From 4b07abf077727764ba16fe0fe28ce5561e66b809 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Fri, 8 Sep 2017 15:00:19 +0200 Subject: Strip out minimal decoder --- Android.bp | 8 - Makefile.am | 116 +- Makefile.vc | 115 +- aac-enc.c | 240 --- configure.ac | 9 - documentation/aacEncoder.pdf | Bin 381677 -> 0 bytes fdk-aac.sym | 7 - libAACdec/src/aac_ram.h | 3 - libAACdec/src/aac_rom.cpp | 185 --- libAACdec/src/aac_rom.h | 18 +- libAACdec/src/aacdec_drc.cpp | 24 +- libAACdec/src/aacdec_drc.h | 6 +- libAACdec/src/aacdec_hcr.cpp | 1591 -------------------- libAACdec/src/aacdec_hcr.h | 126 -- libAACdec/src/aacdec_hcr_bit.cpp | 165 -- libAACdec/src/aacdec_hcr_bit.h | 106 -- libAACdec/src/aacdec_hcr_types.h | 366 ----- libAACdec/src/aacdec_hcrs.cpp | 1409 ----------------- libAACdec/src/aacdec_hcrs.h | 153 -- libAACdec/src/aacdec_pns.cpp | 119 +- libAACdec/src/aacdec_tns.cpp | 185 +-- libAACdec/src/aacdecoder.cpp | 171 +-- libAACdec/src/aacdecoder.h | 14 - libAACdec/src/aacdecoder_lib.cpp | 188 +-- libAACdec/src/block.cpp | 50 +- libAACdec/src/channel.cpp | 22 +- libAACdec/src/channelinfo.h | 24 - libAACdec/src/conceal.cpp | 1866 ----------------------- libAACdec/src/conceal.h | 148 -- libAACdec/src/conceal_types.h | 178 --- libAACdec/src/rvlc.cpp | 1215 --------------- libAACdec/src/rvlc.h | 134 -- libAACdec/src/rvlc_info.h | 176 --- libAACdec/src/rvlcbit.cpp | 131 -- libAACdec/src/rvlcbit.h | 103 -- libAACdec/src/rvlcconceal.cpp | 697 --------- libAACdec/src/rvlcconceal.h | 112 -- libAACenc/include/aacenc_lib.h | 1239 --------------- libAACenc/src/aacEnc_ram.cpp | 194 --- libAACenc/src/aacEnc_ram.h | 226 --- libAACenc/src/aacEnc_rom.cpp | 1232 --------------- libAACenc/src/aacEnc_rom.h | 203 --- libAACenc/src/aacenc.cpp | 1062 ------------- libAACenc/src/aacenc.h | 360 ----- libAACenc/src/aacenc_lib.cpp | 2139 -------------------------- libAACenc/src/aacenc_pns.cpp | 591 -------- libAACenc/src/aacenc_pns.h | 113 -- libAACenc/src/aacenc_tns.cpp | 1341 ----------------- libAACenc/src/aacenc_tns.h | 202 --- libAACenc/src/adj_thr.cpp | 2666 --------------------------------- libAACenc/src/adj_thr.h | 149 -- libAACenc/src/adj_thr_data.h | 151 -- libAACenc/src/band_nrg.cpp | 359 ----- libAACenc/src/band_nrg.h | 149 -- libAACenc/src/bandwidth.cpp | 379 ----- libAACenc/src/bandwidth.h | 106 -- libAACenc/src/bit_cnt.cpp | 1122 -------------- libAACenc/src/bit_cnt.h | 187 --- libAACenc/src/bitenc.cpp | 1474 ------------------ libAACenc/src/bitenc.h | 183 --- libAACenc/src/block_switch.cpp | 545 ------- libAACenc/src/block_switch.h | 146 -- libAACenc/src/channel_map.cpp | 566 ------- libAACenc/src/channel_map.h | 132 -- libAACenc/src/chaosmeasure.cpp | 161 -- libAACenc/src/chaosmeasure.h | 103 -- libAACenc/src/dyn_bits.cpp | 805 ---------- libAACenc/src/dyn_bits.h | 167 --- libAACenc/src/grp_data.cpp | 272 ---- libAACenc/src/grp_data.h | 115 -- libAACenc/src/intensity.cpp | 760 ---------- libAACenc/src/intensity.h | 122 -- libAACenc/src/interface.h | 169 --- libAACenc/src/line_pe.cpp | 209 --- libAACenc/src/line_pe.h | 139 -- libAACenc/src/metadata_compressor.cpp | 1038 ------------- libAACenc/src/metadata_compressor.h | 252 ---- libAACenc/src/metadata_main.cpp | 869 ----------- libAACenc/src/metadata_main.h | 224 --- libAACenc/src/ms_stereo.cpp | 251 ---- libAACenc/src/ms_stereo.h | 107 -- libAACenc/src/noisedet.cpp | 228 --- libAACenc/src/noisedet.h | 108 -- libAACenc/src/pns_func.h | 150 -- libAACenc/src/pnsparam.cpp | 311 ---- libAACenc/src/pnsparam.h | 141 -- libAACenc/src/pre_echo_control.cpp | 170 --- libAACenc/src/pre_echo_control.h | 114 -- libAACenc/src/psy_configuration.cpp | 657 -------- libAACenc/src/psy_configuration.h | 165 -- libAACenc/src/psy_const.h | 160 -- libAACenc/src/psy_data.h | 152 -- libAACenc/src/psy_main.cpp | 1387 ----------------- libAACenc/src/psy_main.h | 174 --- libAACenc/src/qc_data.h | 280 ---- libAACenc/src/qc_main.cpp | 1641 -------------------- libAACenc/src/qc_main.h | 170 --- libAACenc/src/quantize.cpp | 405 ----- libAACenc/src/quantize.h | 121 -- libAACenc/src/sf_estim.cpp | 1330 ---------------- libAACenc/src/sf_estim.h | 118 -- libAACenc/src/spreading.cpp | 114 -- libAACenc/src/spreading.h | 102 -- libAACenc/src/tns_func.h | 145 -- libAACenc/src/tonality.cpp | 204 --- libAACenc/src/tonality.h | 108 -- libAACenc/src/transform.cpp | 264 ---- libAACenc/src/transform.h | 123 -- libMpegTPDec/include/tp_data.h | 37 +- libMpegTPDec/include/tpdec_lib.h | 9 - libMpegTPDec/src/tpdec_asc.cpp | 122 -- libMpegTPDec/src/tpdec_lib.cpp | 10 - libMpegTPEnc/include/mpegFileWrite.h | 140 -- libMpegTPEnc/include/tp_data.h | 350 ----- libMpegTPEnc/include/tpenc_lib.h | 296 ---- libMpegTPEnc/src/tpenc_adif.cpp | 182 --- libMpegTPEnc/src/tpenc_adif.h | 135 -- libMpegTPEnc/src/tpenc_adts.cpp | 315 ---- libMpegTPEnc/src/tpenc_adts.h | 218 --- libMpegTPEnc/src/tpenc_asc.cpp | 576 ------- libMpegTPEnc/src/tpenc_asc.h | 142 -- libMpegTPEnc/src/tpenc_latm.cpp | 881 ----------- libMpegTPEnc/src/tpenc_latm.h | 264 ---- libMpegTPEnc/src/tpenc_lib.cpp | 642 -------- libMpegTPEnc/src/version | 13 - libSBRdec/include/sbrdecoder.h | 347 ----- libSBRdec/src/arm/env_calc_arm.cpp | 148 -- libSBRdec/src/arm/lpp_tran_arm.cpp | 154 -- libSBRdec/src/env_calc.cpp | 2317 ---------------------------- libSBRdec/src/env_calc.h | 165 -- libSBRdec/src/env_dec.cpp | 852 ----------- libSBRdec/src/env_dec.h | 101 -- libSBRdec/src/env_extr.cpp | 1398 ----------------- libSBRdec/src/env_extr.h | 324 ---- libSBRdec/src/huff_dec.cpp | 120 -- libSBRdec/src/huff_dec.h | 100 -- libSBRdec/src/lpp_tran.cpp | 986 ------------ libSBRdec/src/lpp_tran.h | 242 --- libSBRdec/src/psbitdec.cpp | 593 -------- libSBRdec/src/psbitdec.h | 103 -- libSBRdec/src/psdec.cpp | 1414 ----------------- libSBRdec/src/psdec.h | 352 ----- libSBRdec/src/psdec_hybrid.cpp | 652 -------- libSBRdec/src/psdec_hybrid.h | 165 -- libSBRdec/src/sbr_crc.cpp | 183 --- libSBRdec/src/sbr_crc.h | 123 -- libSBRdec/src/sbr_deb.cpp | 90 -- libSBRdec/src/sbr_deb.h | 94 -- libSBRdec/src/sbr_dec.cpp | 1102 -------------- libSBRdec/src/sbr_dec.h | 214 --- libSBRdec/src/sbr_ram.cpp | 194 --- libSBRdec/src/sbr_ram.h | 159 -- libSBRdec/src/sbr_rom.cpp | 1423 ------------------ libSBRdec/src/sbr_rom.h | 235 --- libSBRdec/src/sbr_scale.h | 123 -- libSBRdec/src/sbrdec_drc.cpp | 525 ------- libSBRdec/src/sbrdec_drc.h | 151 -- libSBRdec/src/sbrdec_freq_sca.cpp | 812 ---------- libSBRdec/src/sbrdec_freq_sca.h | 107 -- libSBRdec/src/sbrdecoder.cpp | 1764 ---------------------- libSBRdec/src/transcendent.h | 355 ----- libSBRenc/include/sbr_encoder.h | 430 ------ libSBRenc/src/bit_sbr.cpp | 1057 ------------- libSBRenc/src/bit_sbr.h | 258 ---- libSBRenc/src/cmondata.h | 110 -- libSBRenc/src/code_env.cpp | 641 -------- libSBRenc/src/code_env.h | 153 -- libSBRenc/src/env_bit.cpp | 250 ---- libSBRenc/src/env_bit.h | 126 -- libSBRenc/src/env_est.cpp | 2030 ------------------------- libSBRenc/src/env_est.h | 225 --- libSBRenc/src/fram_gen.cpp | 2065 ------------------------- libSBRenc/src/fram_gen.h | 309 ---- libSBRenc/src/invf_est.cpp | 529 ------- libSBRenc/src/invf_est.h | 175 --- libSBRenc/src/mh_det.cpp | 1471 ------------------ libSBRenc/src/mh_det.h | 196 --- libSBRenc/src/nf_est.cpp | 584 -------- libSBRenc/src/nf_est.h | 147 -- libSBRenc/src/ps_bitenc.cpp | 698 --------- libSBRenc/src/ps_bitenc.h | 177 --- libSBRenc/src/ps_const.h | 148 -- libSBRenc/src/ps_encode.cpp | 1054 ------------- libSBRenc/src/ps_encode.h | 187 --- libSBRenc/src/ps_main.cpp | 618 -------- libSBRenc/src/ps_main.h | 271 ---- libSBRenc/src/resampler.cpp | 507 ------- libSBRenc/src/resampler.h | 151 -- libSBRenc/src/sbr.h | 166 -- libSBRenc/src/sbr_def.h | 275 ---- libSBRenc/src/sbr_encoder.cpp | 2443 ------------------------------ libSBRenc/src/sbr_misc.cpp | 272 ---- libSBRenc/src/sbr_misc.h | 106 -- libSBRenc/src/sbr_ram.cpp | 222 --- libSBRenc/src/sbr_ram.h | 187 --- libSBRenc/src/sbr_rom.cpp | 795 ---------- libSBRenc/src/sbr_rom.h | 127 -- libSBRenc/src/sbrenc_freq_sca.cpp | 691 --------- libSBRenc/src/sbrenc_freq_sca.h | 137 -- libSBRenc/src/ton_corr.cpp | 881 ----------- libSBRenc/src/ton_corr.h | 212 --- libSBRenc/src/tran_det.cpp | 1069 ------------- libSBRenc/src/tran_det.h | 203 --- wavreader.c | 193 --- wavreader.h | 37 - 205 files changed, 44 insertions(+), 87259 deletions(-) delete mode 100644 aac-enc.c delete mode 100644 documentation/aacEncoder.pdf delete mode 100644 libAACdec/src/aacdec_hcr.cpp delete mode 100644 libAACdec/src/aacdec_hcr.h delete mode 100644 libAACdec/src/aacdec_hcr_bit.cpp delete mode 100644 libAACdec/src/aacdec_hcr_bit.h delete mode 100644 libAACdec/src/aacdec_hcr_types.h delete mode 100644 libAACdec/src/aacdec_hcrs.cpp delete mode 100644 libAACdec/src/aacdec_hcrs.h delete mode 100644 libAACdec/src/conceal.cpp delete mode 100644 libAACdec/src/conceal.h delete mode 100644 libAACdec/src/conceal_types.h delete mode 100644 libAACdec/src/rvlc.cpp delete mode 100644 libAACdec/src/rvlc.h delete mode 100644 libAACdec/src/rvlc_info.h delete mode 100644 libAACdec/src/rvlcbit.cpp delete mode 100644 libAACdec/src/rvlcbit.h delete mode 100644 libAACdec/src/rvlcconceal.cpp delete mode 100644 libAACdec/src/rvlcconceal.h delete mode 100644 libAACenc/include/aacenc_lib.h delete mode 100644 libAACenc/src/aacEnc_ram.cpp delete mode 100644 libAACenc/src/aacEnc_ram.h delete mode 100644 libAACenc/src/aacEnc_rom.cpp delete mode 100644 libAACenc/src/aacEnc_rom.h delete mode 100644 libAACenc/src/aacenc.cpp delete mode 100644 libAACenc/src/aacenc.h delete mode 100644 libAACenc/src/aacenc_lib.cpp delete mode 100644 libAACenc/src/aacenc_pns.cpp delete mode 100644 libAACenc/src/aacenc_pns.h delete mode 100644 libAACenc/src/aacenc_tns.cpp delete mode 100644 libAACenc/src/aacenc_tns.h delete mode 100644 libAACenc/src/adj_thr.cpp delete mode 100644 libAACenc/src/adj_thr.h delete mode 100644 libAACenc/src/adj_thr_data.h delete mode 100644 libAACenc/src/band_nrg.cpp delete mode 100644 libAACenc/src/band_nrg.h delete mode 100644 libAACenc/src/bandwidth.cpp delete mode 100644 libAACenc/src/bandwidth.h delete mode 100644 libAACenc/src/bit_cnt.cpp delete mode 100644 libAACenc/src/bit_cnt.h delete mode 100644 libAACenc/src/bitenc.cpp delete mode 100644 libAACenc/src/bitenc.h delete mode 100644 libAACenc/src/block_switch.cpp delete mode 100644 libAACenc/src/block_switch.h delete mode 100644 libAACenc/src/channel_map.cpp delete mode 100644 libAACenc/src/channel_map.h delete mode 100644 libAACenc/src/chaosmeasure.cpp delete mode 100644 libAACenc/src/chaosmeasure.h delete mode 100644 libAACenc/src/dyn_bits.cpp delete mode 100644 libAACenc/src/dyn_bits.h delete mode 100644 libAACenc/src/grp_data.cpp delete mode 100644 libAACenc/src/grp_data.h delete mode 100644 libAACenc/src/intensity.cpp delete mode 100644 libAACenc/src/intensity.h delete mode 100644 libAACenc/src/interface.h delete mode 100644 libAACenc/src/line_pe.cpp delete mode 100644 libAACenc/src/line_pe.h delete mode 100644 libAACenc/src/metadata_compressor.cpp delete mode 100644 libAACenc/src/metadata_compressor.h delete mode 100644 libAACenc/src/metadata_main.cpp delete mode 100644 libAACenc/src/metadata_main.h delete mode 100644 libAACenc/src/ms_stereo.cpp delete mode 100644 libAACenc/src/ms_stereo.h delete mode 100644 libAACenc/src/noisedet.cpp delete mode 100644 libAACenc/src/noisedet.h delete mode 100644 libAACenc/src/pns_func.h delete mode 100644 libAACenc/src/pnsparam.cpp delete mode 100644 libAACenc/src/pnsparam.h delete mode 100644 libAACenc/src/pre_echo_control.cpp delete mode 100644 libAACenc/src/pre_echo_control.h delete mode 100644 libAACenc/src/psy_configuration.cpp delete mode 100644 libAACenc/src/psy_configuration.h delete mode 100644 libAACenc/src/psy_const.h delete mode 100644 libAACenc/src/psy_data.h delete mode 100644 libAACenc/src/psy_main.cpp delete mode 100644 libAACenc/src/psy_main.h delete mode 100644 libAACenc/src/qc_data.h delete mode 100644 libAACenc/src/qc_main.cpp delete mode 100644 libAACenc/src/qc_main.h delete mode 100644 libAACenc/src/quantize.cpp delete mode 100644 libAACenc/src/quantize.h delete mode 100644 libAACenc/src/sf_estim.cpp delete mode 100644 libAACenc/src/sf_estim.h delete mode 100644 libAACenc/src/spreading.cpp delete mode 100644 libAACenc/src/spreading.h delete mode 100644 libAACenc/src/tns_func.h delete mode 100644 libAACenc/src/tonality.cpp delete mode 100644 libAACenc/src/tonality.h delete mode 100644 libAACenc/src/transform.cpp delete mode 100644 libAACenc/src/transform.h delete mode 100644 libMpegTPEnc/include/mpegFileWrite.h delete mode 100644 libMpegTPEnc/include/tp_data.h delete mode 100644 libMpegTPEnc/include/tpenc_lib.h delete mode 100644 libMpegTPEnc/src/tpenc_adif.cpp delete mode 100644 libMpegTPEnc/src/tpenc_adif.h delete mode 100644 libMpegTPEnc/src/tpenc_adts.cpp delete mode 100644 libMpegTPEnc/src/tpenc_adts.h delete mode 100644 libMpegTPEnc/src/tpenc_asc.cpp delete mode 100644 libMpegTPEnc/src/tpenc_asc.h delete mode 100644 libMpegTPEnc/src/tpenc_latm.cpp delete mode 100644 libMpegTPEnc/src/tpenc_latm.h delete mode 100644 libMpegTPEnc/src/tpenc_lib.cpp delete mode 100644 libMpegTPEnc/src/version delete mode 100644 libSBRdec/include/sbrdecoder.h delete mode 100644 libSBRdec/src/arm/env_calc_arm.cpp delete mode 100644 libSBRdec/src/arm/lpp_tran_arm.cpp delete mode 100644 libSBRdec/src/env_calc.cpp delete mode 100644 libSBRdec/src/env_calc.h delete mode 100644 libSBRdec/src/env_dec.cpp delete mode 100644 libSBRdec/src/env_dec.h delete mode 100644 libSBRdec/src/env_extr.cpp delete mode 100644 libSBRdec/src/env_extr.h delete mode 100644 libSBRdec/src/huff_dec.cpp delete mode 100644 libSBRdec/src/huff_dec.h delete mode 100644 libSBRdec/src/lpp_tran.cpp delete mode 100644 libSBRdec/src/lpp_tran.h delete mode 100644 libSBRdec/src/psbitdec.cpp delete mode 100644 libSBRdec/src/psbitdec.h delete mode 100644 libSBRdec/src/psdec.cpp delete mode 100644 libSBRdec/src/psdec.h delete mode 100644 libSBRdec/src/psdec_hybrid.cpp delete mode 100644 libSBRdec/src/psdec_hybrid.h delete mode 100644 libSBRdec/src/sbr_crc.cpp delete mode 100644 libSBRdec/src/sbr_crc.h delete mode 100644 libSBRdec/src/sbr_deb.cpp delete mode 100644 libSBRdec/src/sbr_deb.h delete mode 100644 libSBRdec/src/sbr_dec.cpp delete mode 100644 libSBRdec/src/sbr_dec.h delete mode 100644 libSBRdec/src/sbr_ram.cpp delete mode 100644 libSBRdec/src/sbr_ram.h delete mode 100644 libSBRdec/src/sbr_rom.cpp delete mode 100644 libSBRdec/src/sbr_rom.h delete mode 100644 libSBRdec/src/sbr_scale.h delete mode 100644 libSBRdec/src/sbrdec_drc.cpp delete mode 100644 libSBRdec/src/sbrdec_drc.h delete mode 100644 libSBRdec/src/sbrdec_freq_sca.cpp delete mode 100644 libSBRdec/src/sbrdec_freq_sca.h delete mode 100644 libSBRdec/src/sbrdecoder.cpp delete mode 100644 libSBRdec/src/transcendent.h delete mode 100644 libSBRenc/include/sbr_encoder.h delete mode 100644 libSBRenc/src/bit_sbr.cpp delete mode 100644 libSBRenc/src/bit_sbr.h delete mode 100644 libSBRenc/src/cmondata.h delete mode 100644 libSBRenc/src/code_env.cpp delete mode 100644 libSBRenc/src/code_env.h delete mode 100644 libSBRenc/src/env_bit.cpp delete mode 100644 libSBRenc/src/env_bit.h delete mode 100644 libSBRenc/src/env_est.cpp delete mode 100644 libSBRenc/src/env_est.h delete mode 100644 libSBRenc/src/fram_gen.cpp delete mode 100644 libSBRenc/src/fram_gen.h delete mode 100644 libSBRenc/src/invf_est.cpp delete mode 100644 libSBRenc/src/invf_est.h delete mode 100644 libSBRenc/src/mh_det.cpp delete mode 100644 libSBRenc/src/mh_det.h delete mode 100644 libSBRenc/src/nf_est.cpp delete mode 100644 libSBRenc/src/nf_est.h delete mode 100644 libSBRenc/src/ps_bitenc.cpp delete mode 100644 libSBRenc/src/ps_bitenc.h delete mode 100644 libSBRenc/src/ps_const.h delete mode 100644 libSBRenc/src/ps_encode.cpp delete mode 100644 libSBRenc/src/ps_encode.h delete mode 100644 libSBRenc/src/ps_main.cpp delete mode 100644 libSBRenc/src/ps_main.h delete mode 100644 libSBRenc/src/resampler.cpp delete mode 100644 libSBRenc/src/resampler.h delete mode 100644 libSBRenc/src/sbr.h delete mode 100644 libSBRenc/src/sbr_def.h delete mode 100644 libSBRenc/src/sbr_encoder.cpp delete mode 100644 libSBRenc/src/sbr_misc.cpp delete mode 100644 libSBRenc/src/sbr_misc.h delete mode 100644 libSBRenc/src/sbr_ram.cpp delete mode 100644 libSBRenc/src/sbr_ram.h delete mode 100644 libSBRenc/src/sbr_rom.cpp delete mode 100644 libSBRenc/src/sbr_rom.h delete mode 100644 libSBRenc/src/sbrenc_freq_sca.cpp delete mode 100644 libSBRenc/src/sbrenc_freq_sca.h delete mode 100644 libSBRenc/src/ton_corr.cpp delete mode 100644 libSBRenc/src/ton_corr.h delete mode 100644 libSBRenc/src/tran_det.cpp delete mode 100644 libSBRenc/src/tran_det.h delete mode 100644 wavreader.c delete mode 100644 wavreader.h diff --git a/Android.bp b/Android.bp index 75fe8af..14829e5 100644 --- a/Android.bp +++ b/Android.bp @@ -2,14 +2,10 @@ cc_library_static { name: "libFraunhoferAAC", srcs: [ "libAACdec/src/*.cpp", - "libAACenc/src/*.cpp", "libPCMutils/src/*.cpp", "libFDK/src/*.cpp", "libSYS/src/*.cpp", "libMpegTPDec/src/*.cpp", - "libMpegTPEnc/src/*.cpp", - "libSBRdec/src/*.cpp", - "libSBRenc/src/*.cpp", ], cflags: [ "-Wno-sequence-point", @@ -20,13 +16,9 @@ cc_library_static { ], export_include_dirs: [ "libAACdec/include", - "libAACenc/include", "libPCMutils/include", "libFDK/include", "libSYS/include", "libMpegTPDec/include", - "libMpegTPEnc/include", - "libSBRdec/include", - "libSBRenc/include", ], } diff --git a/Makefile.am b/Makefile.am index 5b2c65b..5187d2a 100644 --- a/Makefile.am +++ b/Makefile.am @@ -3,11 +3,7 @@ AUTOMAKE_OPTIONS = subdir-objects AM_CPPFLAGS = \ -I$(top_srcdir)/libAACdec/include \ - -I$(top_srcdir)/libAACenc/include \ - -I$(top_srcdir)/libSBRdec/include \ - -I$(top_srcdir)/libSBRenc/include \ -I$(top_srcdir)/libMpegTPDec/include \ - -I$(top_srcdir)/libMpegTPEnc/include \ -I$(top_srcdir)/libSYS/include \ -I$(top_srcdir)/libFDK/include \ -I$(top_srcdir)/libPCMutils/include @@ -22,7 +18,6 @@ fdk_aacinclude_HEADERS = \ $(top_srcdir)/libSYS/include/machine_type.h \ $(top_srcdir)/libSYS/include/genericStds.h \ $(top_srcdir)/libSYS/include/FDK_audio.h \ - $(top_srcdir)/libAACenc/include/aacenc_lib.h \ $(top_srcdir)/libAACdec/include/aacdecoder_lib.h pkgconfigdir = $(libdir)/pkgconfig @@ -33,71 +28,21 @@ lib_LTLIBRARIES = libfdk-aac.la libfdk_aac_la_LDFLAGS = -version-info @FDK_AAC_VERSION@ -no-undefined \ -export-symbols $(top_srcdir)/fdk-aac.sym -if EXAMPLE -bin_PROGRAMS = aac-enc$(EXEEXT) - -aac_enc_LDADD = libfdk-aac.la -aac_enc_SOURCES = aac-enc.c wavreader.c - -noinst_HEADERS = wavreader.h -endif - AACDEC_SRC = \ libAACdec/src/aacdec_drc.cpp \ - libAACdec/src/aacdec_hcr.cpp \ libAACdec/src/aacdecoder.cpp \ libAACdec/src/aacdec_pns.cpp \ libAACdec/src/aac_ram.cpp \ libAACdec/src/block.cpp \ libAACdec/src/channelinfo.cpp \ libAACdec/src/ldfiltbank.cpp \ - libAACdec/src/rvlcbit.cpp \ - libAACdec/src/rvlc.cpp \ - libAACdec/src/aacdec_hcr_bit.cpp \ - libAACdec/src/aacdec_hcrs.cpp \ libAACdec/src/aacdecoder_lib.cpp \ libAACdec/src/aacdec_tns.cpp \ libAACdec/src/aac_rom.cpp \ libAACdec/src/channel.cpp \ - libAACdec/src/conceal.cpp \ libAACdec/src/pulsedata.cpp \ - libAACdec/src/rvlcconceal.cpp \ libAACdec/src/stereo.cpp -AACENC_SRC = \ - libAACenc/src/aacenc.cpp \ - libAACenc/src/aacEnc_ram.cpp \ - libAACenc/src/band_nrg.cpp \ - libAACenc/src/block_switch.cpp \ - libAACenc/src/grp_data.cpp \ - libAACenc/src/metadata_main.cpp \ - libAACenc/src/pre_echo_control.cpp \ - libAACenc/src/quantize.cpp \ - libAACenc/src/tonality.cpp \ - libAACenc/src/aacEnc_rom.cpp \ - libAACenc/src/bandwidth.cpp \ - libAACenc/src/channel_map.cpp \ - libAACenc/src/intensity.cpp \ - libAACenc/src/ms_stereo.cpp \ - libAACenc/src/psy_configuration.cpp \ - libAACenc/src/sf_estim.cpp \ - libAACenc/src/transform.cpp \ - libAACenc/src/aacenc_lib.cpp \ - libAACenc/src/aacenc_tns.cpp \ - libAACenc/src/bit_cnt.cpp \ - libAACenc/src/chaosmeasure.cpp \ - libAACenc/src/line_pe.cpp \ - libAACenc/src/noisedet.cpp \ - libAACenc/src/psy_main.cpp \ - libAACenc/src/spreading.cpp \ - libAACenc/src/aacenc_pns.cpp \ - libAACenc/src/adj_thr.cpp \ - libAACenc/src/bitenc.cpp \ - libAACenc/src/dyn_bits.cpp \ - libAACenc/src/metadata_compressor.cpp \ - libAACenc/src/pnsparam.cpp \ - libAACenc/src/qc_main.cpp - FDK_SRC = \ libFDK/src/autocorr2nd.cpp \ libFDK/src/dct.cpp \ @@ -122,56 +67,10 @@ MPEGTPDEC_SRC = \ libMpegTPDec/src/tpdec_latm.cpp \ libMpegTPDec/src/tpdec_lib.cpp -MPEGTPENC_SRC = \ - libMpegTPEnc/src/tpenc_adif.cpp \ - libMpegTPEnc/src/tpenc_adts.cpp \ - libMpegTPEnc/src/tpenc_asc.cpp \ - libMpegTPEnc/src/tpenc_latm.cpp \ - libMpegTPEnc/src/tpenc_lib.cpp - PCMUTILS_SRC = \ libPCMutils/src/limiter.cpp \ libPCMutils/src/pcmutils_lib.cpp -SBRDEC_SRC = \ - libSBRdec/src/env_calc.cpp \ - libSBRdec/src/env_dec.cpp \ - libSBRdec/src/env_extr.cpp \ - libSBRdec/src/huff_dec.cpp \ - libSBRdec/src/lpp_tran.cpp \ - libSBRdec/src/psbitdec.cpp \ - libSBRdec/src/psdec.cpp \ - libSBRdec/src/psdec_hybrid.cpp \ - libSBRdec/src/sbr_crc.cpp \ - libSBRdec/src/sbr_deb.cpp \ - libSBRdec/src/sbr_dec.cpp \ - libSBRdec/src/sbrdec_drc.cpp \ - libSBRdec/src/sbrdec_freq_sca.cpp \ - libSBRdec/src/sbrdecoder.cpp \ - libSBRdec/src/sbr_ram.cpp \ - libSBRdec/src/sbr_rom.cpp - -SBRENC_SRC = \ - libSBRenc/src/bit_sbr.cpp \ - libSBRenc/src/env_bit.cpp \ - libSBRenc/src/fram_gen.cpp \ - libSBRenc/src/mh_det.cpp \ - libSBRenc/src/ps_bitenc.cpp \ - libSBRenc/src/ps_encode.cpp \ - libSBRenc/src/resampler.cpp \ - libSBRenc/src/sbr_encoder.cpp \ - libSBRenc/src/sbr_ram.cpp \ - libSBRenc/src/ton_corr.cpp \ - libSBRenc/src/code_env.cpp \ - libSBRenc/src/env_est.cpp \ - libSBRenc/src/invf_est.cpp \ - libSBRenc/src/nf_est.cpp \ - libSBRenc/src/ps_main.cpp \ - libSBRenc/src/sbrenc_freq_sca.cpp \ - libSBRenc/src/sbr_misc.cpp \ - libSBRenc/src/sbr_rom.cpp \ - libSBRenc/src/tran_det.cpp - SYS_SRC = \ libSYS/src/cmdl_parser.cpp \ libSYS/src/conv_string.cpp \ @@ -179,10 +78,8 @@ SYS_SRC = \ libSYS/src/wav_file.cpp libfdk_aac_la_SOURCES = \ - $(AACDEC_SRC) $(AACENC_SRC) \ - $(MPEGTPDEC_SRC) $(MPEGTPENC_SRC) \ - $(SBRDEC_SRC) $(SBRENC_SRC) \ - $(PCMUTILS_SRC) $(FDK_SRC) $(SYS_SRC) + $(AACDEC_SRC) $(FDK_SRC) $(SYS_SRC) \ + $(MPEGTPDEC_SRC) $(PCMUTILS_SRC) EXTRA_DIST = \ $(top_srcdir)/autogen.sh \ @@ -194,19 +91,10 @@ EXTRA_DIST = \ $(top_srcdir)/documentation/*.pdf \ $(top_srcdir)/libAACdec/src/*.h \ $(top_srcdir)/libAACdec/src/arm/*.cpp \ - $(top_srcdir)/libAACenc/src/*.h \ - $(top_srcdir)/libSBRenc/src/*.h \ - $(top_srcdir)/libSBRenc/include/*.h \ - $(top_srcdir)/libSBRdec/src/*.h \ - $(top_srcdir)/libSBRdec/src/arm/*.cpp \ - $(top_srcdir)/libSBRdec/include/*.h \ $(top_srcdir)/libSYS/include/*.h \ $(top_srcdir)/libSYS/src/linux/*.cpp \ $(top_srcdir)/libSYS/src/mips/*.cpp \ $(top_srcdir)/libPCMutils/include/*.h \ - $(top_srcdir)/libMpegTPEnc/include/*.h \ - $(top_srcdir)/libMpegTPEnc/src/*.h \ - $(top_srcdir)/libMpegTPEnc/src/version \ $(top_srcdir)/libMpegTPDec/include/*.h \ $(top_srcdir)/libMpegTPDec/src/*.h \ $(top_srcdir)/libMpegTPDec/src/version \ diff --git a/Makefile.vc b/Makefile.vc index 6fccb9c..f715967 100644 --- a/Makefile.vc +++ b/Makefile.vc @@ -20,71 +20,26 @@ MKDIR_FLAGS = -p AM_CPPFLAGS = \ -Iwin32 \ -IlibAACdec/include \ - -IlibAACenc/include \ - -IlibSBRdec/include \ - -IlibSBRenc/include \ -IlibMpegTPDec/include \ - -IlibMpegTPEnc/include \ -IlibSYS/include \ -IlibFDK/include \ -IlibPCMutils/include AACDEC_SRC = \ libAACdec/src/aacdec_drc.cpp \ - libAACdec/src/aacdec_hcr.cpp \ libAACdec/src/aacdecoder.cpp \ libAACdec/src/aacdec_pns.cpp \ libAACdec/src/aac_ram.cpp \ libAACdec/src/block.cpp \ libAACdec/src/channelinfo.cpp \ libAACdec/src/ldfiltbank.cpp \ - libAACdec/src/rvlcbit.cpp \ - libAACdec/src/rvlc.cpp \ - libAACdec/src/aacdec_hcr_bit.cpp \ - libAACdec/src/aacdec_hcrs.cpp \ libAACdec/src/aacdecoder_lib.cpp \ libAACdec/src/aacdec_tns.cpp \ libAACdec/src/aac_rom.cpp \ libAACdec/src/channel.cpp \ - libAACdec/src/conceal.cpp \ libAACdec/src/pulsedata.cpp \ - libAACdec/src/rvlcconceal.cpp \ libAACdec/src/stereo.cpp -AACENC_SRC = \ - libAACenc/src/aacenc.cpp \ - libAACenc/src/aacEnc_ram.cpp \ - libAACenc/src/band_nrg.cpp \ - libAACenc/src/block_switch.cpp \ - libAACenc/src/grp_data.cpp \ - libAACenc/src/metadata_main.cpp \ - libAACenc/src/pre_echo_control.cpp \ - libAACenc/src/quantize.cpp \ - libAACenc/src/tonality.cpp \ - libAACenc/src/aacEnc_rom.cpp \ - libAACenc/src/bandwidth.cpp \ - libAACenc/src/channel_map.cpp \ - libAACenc/src/intensity.cpp \ - libAACenc/src/ms_stereo.cpp \ - libAACenc/src/psy_configuration.cpp \ - libAACenc/src/sf_estim.cpp \ - libAACenc/src/transform.cpp \ - libAACenc/src/aacenc_lib.cpp \ - libAACenc/src/aacenc_tns.cpp \ - libAACenc/src/bit_cnt.cpp \ - libAACenc/src/chaosmeasure.cpp \ - libAACenc/src/line_pe.cpp \ - libAACenc/src/noisedet.cpp \ - libAACenc/src/psy_main.cpp \ - libAACenc/src/spreading.cpp \ - libAACenc/src/aacenc_pns.cpp \ - libAACenc/src/adj_thr.cpp \ - libAACenc/src/bitenc.cpp \ - libAACenc/src/dyn_bits.cpp \ - libAACenc/src/metadata_compressor.cpp \ - libAACenc/src/pnsparam.cpp \ - libAACenc/src/qc_main.cpp - FDK_SRC = \ libFDK/src/autocorr2nd.cpp \ libFDK/src/dct.cpp \ @@ -109,56 +64,10 @@ MPEGTPDEC_SRC = \ libMpegTPDec/src/tpdec_latm.cpp \ libMpegTPDec/src/tpdec_lib.cpp -MPEGTPENC_SRC = \ - libMpegTPEnc/src/tpenc_adif.cpp \ - libMpegTPEnc/src/tpenc_adts.cpp \ - libMpegTPEnc/src/tpenc_asc.cpp \ - libMpegTPEnc/src/tpenc_latm.cpp \ - libMpegTPEnc/src/tpenc_lib.cpp - PCMUTILS_SRC = \ libPCMutils/src/limiter.cpp \ libPCMutils/src/pcmutils_lib.cpp -SBRDEC_SRC = \ - libSBRdec/src/env_calc.cpp \ - libSBRdec/src/env_dec.cpp \ - libSBRdec/src/env_extr.cpp \ - libSBRdec/src/huff_dec.cpp \ - libSBRdec/src/lpp_tran.cpp \ - libSBRdec/src/psbitdec.cpp \ - libSBRdec/src/psdec.cpp \ - libSBRdec/src/psdec_hybrid.cpp \ - libSBRdec/src/sbr_crc.cpp \ - libSBRdec/src/sbr_deb.cpp \ - libSBRdec/src/sbr_dec.cpp \ - libSBRdec/src/sbrdec_drc.cpp \ - libSBRdec/src/sbrdec_freq_sca.cpp \ - libSBRdec/src/sbrdecoder.cpp \ - libSBRdec/src/sbr_ram.cpp \ - libSBRdec/src/sbr_rom.cpp - -SBRENC_SRC = \ - libSBRenc/src/bit_sbr.cpp \ - libSBRenc/src/env_bit.cpp \ - libSBRenc/src/fram_gen.cpp \ - libSBRenc/src/mh_det.cpp \ - libSBRenc/src/ps_bitenc.cpp \ - libSBRenc/src/ps_encode.cpp \ - libSBRenc/src/resampler.cpp \ - libSBRenc/src/sbr_encoder.cpp \ - libSBRenc/src/sbr_ram.cpp \ - libSBRenc/src/ton_corr.cpp \ - libSBRenc/src/code_env.cpp \ - libSBRenc/src/env_est.cpp \ - libSBRenc/src/invf_est.cpp \ - libSBRenc/src/nf_est.cpp \ - libSBRenc/src/ps_main.cpp \ - libSBRenc/src/sbrenc_freq_sca.cpp \ - libSBRenc/src/sbr_misc.cpp \ - libSBRenc/src/sbr_rom.cpp \ - libSBRenc/src/tran_det.cpp - SYS_SRC = \ libSYS/src/cmdl_parser.cpp \ libSYS/src/conv_string.cpp \ @@ -166,14 +75,9 @@ SYS_SRC = \ libSYS/src/wav_file.cpp libfdk_aac_SOURCES = \ - $(AACDEC_SRC) $(AACENC_SRC) \ - $(MPEGTPDEC_SRC) $(MPEGTPENC_SRC) \ - $(SBRDEC_SRC) $(SBRENC_SRC) \ + $(AACDEC_SRC) $(MPEGTPDEC_SRC) \ $(PCMUTILS_SRC) $(FDK_SRC) $(SYS_SRC) - -aac_enc_SOURCES = aac-enc.c wavreader.c - prefix = \usr\local prefix_win = $(prefix:/=\) # In case we are using MSYS or MinGW. @@ -194,38 +98,28 @@ STATIC_LIB = fdk-aac.lib SHARED_LIB = fdk-aac-1.dll IMP_LIB = fdk-aac.dll.lib -AAC_ENC_OBJS = $(aac_enc_SOURCES:.c=.obj) FDK_OBJS = $(libfdk_aac_SOURCES:.cpp=.obj) -PROGS = aac-enc.exe - - -all: $(LIB_DEF) $(STATIC_LIB) $(SHARED_LIB) $(IMP_LIB) $(PROGS) +all: $(LIB_DEF) $(STATIC_LIB) $(SHARED_LIB) $(IMP_LIB) clean: - del /f $(LIB_DEF) $(STATIC_LIB) $(SHARED_LIB) $(IMP_LIB) $(PROGS) libfdk-aac.pc 2>NUL + del /f $(LIB_DEF) $(STATIC_LIB) $(SHARED_LIB) $(IMP_LIB) libfdk-aac.pc 2>NUL del /f *.obj *.exp 2>NUL del /f libAACdec\src\*.obj 2>NUL - del /f libAACenc\src\*.obj 2>NUL del /f libFDK\src\*.obj 2>NUL del /f libMpegTPDec\src\*.obj 2>NUL - del /f libMpegTPEnc\src\*.obj 2>NUL del /f libPCMutils\src\*.obj 2>NUL - del /f libSBRdec\src\*.obj 2>NUL - del /f libSBRenc\src\*.obj 2>NUL del /f libSYS\src\*.obj 2>NUL install: $(INST_DIRS) copy libAACdec\include\aacdecoder_lib.h $(incdir) - copy libAACenc\include\aacenc_lib.h $(incdir) copy libSYS\include\FDK_audio.h $(incdir) copy libSYS\include\genericStds.h $(incdir) copy libSYS\include\machine_type.h $(incdir) copy $(STATIC_LIB) $(libdir) copy $(IMP_LIB) $(libdir) copy $(SHARED_LIB) $(bindir) - copy $(PROGS) $(bindir) copy $(LIB_DEF) $(libdir) $(INST_DIRS): @@ -239,9 +133,6 @@ $(IMP_LIB): $(SHARED_LIB) $(SHARED_LIB): $(FDK_OBJS) $(LD) $(LDFLAGS) -OUT:$@ -DEF:$(LIB_DEF) -implib:$(IMP_LIB) -DLL $(FDK_OBJS) -$(PROGS): $(AAC_ENC_OBJS) - $(LD) $(LDFLAGS) -out:$@ $(AAC_ENC_OBJS) $(STATIC_LIB) - .cpp.obj: $(CXX) $(CXXFLAGS) -c -Fo$@ $< diff --git a/aac-enc.c b/aac-enc.c deleted file mode 100644 index 9365cc9..0000000 --- a/aac-enc.c +++ /dev/null @@ -1,240 +0,0 @@ -/* ------------------------------------------------------------------ - * Copyright (C) 2011 Martin Storsjo - * - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either - * express or implied. - * See the License for the specific language governing permissions - * and limitations under the License. - * ------------------------------------------------------------------- - */ - -#include -#include - -#if defined(_MSC_VER) -#include -#else -#include -#endif - -#include -#include "libAACenc/include/aacenc_lib.h" -#include "wavreader.h" - -void usage(const char* name) { - fprintf(stderr, "%s [-r bitrate] [-t aot] [-a afterburner] [-s sbr] [-v vbr] in.wav out.aac\n", name); - fprintf(stderr, "Supported AOTs:\n"); - fprintf(stderr, "\t2\tAAC-LC\n"); - fprintf(stderr, "\t5\tHE-AAC\n"); - fprintf(stderr, "\t29\tHE-AAC v2\n"); - fprintf(stderr, "\t23\tAAC-LD\n"); - fprintf(stderr, "\t39\tAAC-ELD\n"); -} - -int main(int argc, char *argv[]) { - int bitrate = 64000; - int ch; - const char *infile, *outfile; - FILE *out; - void *wav; - int format, sample_rate, channels, bits_per_sample; - int input_size; - uint8_t* input_buf; - int16_t* convert_buf; - int aot = 2; - int afterburner = 1; - int eld_sbr = 0; - int vbr = 0; - HANDLE_AACENCODER handle; - CHANNEL_MODE mode; - AACENC_InfoStruct info = { 0 }; - while ((ch = getopt(argc, argv, "r:t:a:s:v:")) != -1) { - switch (ch) { - case 'r': - bitrate = atoi(optarg); - break; - case 't': - aot = atoi(optarg); - break; - case 'a': - afterburner = atoi(optarg); - break; - case 's': - eld_sbr = atoi(optarg); - break; - case 'v': - vbr = atoi(optarg); - break; - case '?': - default: - usage(argv[0]); - return 1; - } - } - if (argc - optind < 2) { - usage(argv[0]); - return 1; - } - infile = argv[optind]; - outfile = argv[optind + 1]; - - wav = wav_read_open(infile); - if (!wav) { - fprintf(stderr, "Unable to open wav file %s\n", infile); - return 1; - } - if (!wav_get_header(wav, &format, &channels, &sample_rate, &bits_per_sample, NULL)) { - fprintf(stderr, "Bad wav file %s\n", infile); - return 1; - } - if (format != 1) { - fprintf(stderr, "Unsupported WAV format %d\n", format); - return 1; - } - if (bits_per_sample != 16) { - fprintf(stderr, "Unsupported WAV sample depth %d\n", bits_per_sample); - return 1; - } - switch (channels) { - case 1: mode = MODE_1; break; - case 2: mode = MODE_2; break; - case 3: mode = MODE_1_2; break; - case 4: mode = MODE_1_2_1; break; - case 5: mode = MODE_1_2_2; break; - case 6: mode = MODE_1_2_2_1; break; - default: - fprintf(stderr, "Unsupported WAV channels %d\n", channels); - return 1; - } - if (aacEncOpen(&handle, 0, channels) != AACENC_OK) { - fprintf(stderr, "Unable to open encoder\n"); - return 1; - } - if (aacEncoder_SetParam(handle, AACENC_AOT, aot) != AACENC_OK) { - fprintf(stderr, "Unable to set the AOT\n"); - return 1; - } - if (aot == 39 && eld_sbr) { - if (aacEncoder_SetParam(handle, AACENC_SBR_MODE, 1) != AACENC_OK) { - fprintf(stderr, "Unable to set SBR mode for ELD\n"); - return 1; - } - } - if (aacEncoder_SetParam(handle, AACENC_SAMPLERATE, sample_rate) != AACENC_OK) { - fprintf(stderr, "Unable to set the AOT\n"); - return 1; - } - if (aacEncoder_SetParam(handle, AACENC_CHANNELMODE, mode) != AACENC_OK) { - fprintf(stderr, "Unable to set the channel mode\n"); - return 1; - } - if (aacEncoder_SetParam(handle, AACENC_CHANNELORDER, 1) != AACENC_OK) { - fprintf(stderr, "Unable to set the wav channel order\n"); - return 1; - } - if (vbr) { - if (aacEncoder_SetParam(handle, AACENC_BITRATEMODE, vbr) != AACENC_OK) { - fprintf(stderr, "Unable to set the VBR bitrate mode\n"); - return 1; - } - } else { - if (aacEncoder_SetParam(handle, AACENC_BITRATE, bitrate) != AACENC_OK) { - fprintf(stderr, "Unable to set the bitrate\n"); - return 1; - } - } - if (aacEncoder_SetParam(handle, AACENC_TRANSMUX, 2) != AACENC_OK) { - fprintf(stderr, "Unable to set the ADTS transmux\n"); - return 1; - } - if (aacEncoder_SetParam(handle, AACENC_AFTERBURNER, afterburner) != AACENC_OK) { - fprintf(stderr, "Unable to set the afterburner mode\n"); - return 1; - } - if (aacEncEncode(handle, NULL, NULL, NULL, NULL) != AACENC_OK) { - fprintf(stderr, "Unable to initialize the encoder\n"); - return 1; - } - if (aacEncInfo(handle, &info) != AACENC_OK) { - fprintf(stderr, "Unable to get the encoder info\n"); - return 1; - } - - out = fopen(outfile, "wb"); - if (!out) { - perror(outfile); - return 1; - } - - input_size = channels*2*info.frameLength; - input_buf = (uint8_t*) malloc(input_size); - convert_buf = (int16_t*) malloc(input_size); - - while (1) { - AACENC_BufDesc in_buf = { 0 }, out_buf = { 0 }; - AACENC_InArgs in_args = { 0 }; - AACENC_OutArgs out_args = { 0 }; - int in_identifier = IN_AUDIO_DATA; - int in_size, in_elem_size; - int out_identifier = OUT_BITSTREAM_DATA; - int out_size, out_elem_size; - int read, i; - void *in_ptr, *out_ptr; - uint8_t outbuf[20480]; - AACENC_ERROR err; - - read = wav_read_data(wav, input_buf, input_size); - for (i = 0; i < read/2; i++) { - const uint8_t* in = &input_buf[2*i]; - convert_buf[i] = in[0] | (in[1] << 8); - } - if (read <= 0) { - in_args.numInSamples = -1; - } else { - in_ptr = convert_buf; - in_size = read; - in_elem_size = 2; - - in_args.numInSamples = read/2; - in_buf.numBufs = 1; - in_buf.bufs = &in_ptr; - in_buf.bufferIdentifiers = &in_identifier; - in_buf.bufSizes = &in_size; - in_buf.bufElSizes = &in_elem_size; - } - out_ptr = outbuf; - out_size = sizeof(outbuf); - out_elem_size = 1; - out_buf.numBufs = 1; - out_buf.bufs = &out_ptr; - out_buf.bufferIdentifiers = &out_identifier; - out_buf.bufSizes = &out_size; - out_buf.bufElSizes = &out_elem_size; - - if ((err = aacEncEncode(handle, &in_buf, &out_buf, &in_args, &out_args)) != AACENC_OK) { - if (err == AACENC_ENCODE_EOF) - break; - fprintf(stderr, "Encoding failed\n"); - return 1; - } - if (out_args.numOutBytes == 0) - continue; - fwrite(outbuf, 1, out_args.numOutBytes, out); - } - free(input_buf); - free(convert_buf); - fclose(out); - wav_read_close(wav); - aacEncClose(&handle); - - return 0; -} - diff --git a/configure.ac b/configure.ac index 1485ff7..af0d49f 100644 --- a/configure.ac +++ b/configure.ac @@ -7,15 +7,6 @@ AC_CONFIG_MACRO_DIR([m4]) AM_INIT_AUTOMAKE([tar-ustar foreign]) m4_ifdef([AM_SILENT_RULES], [AM_SILENT_RULES([yes])]) -dnl Various options for configure -AC_ARG_ENABLE([example], - [AS_HELP_STRING([--enable-example], - [enable example encoding program (default is no)])], - [example=$enableval], [example=no]) - -dnl Automake conditionals to set -AM_CONDITIONAL(EXAMPLE, test x$example = xyes) - dnl Checks for programs. AC_PROG_CC AC_PROG_CXX diff --git a/documentation/aacEncoder.pdf b/documentation/aacEncoder.pdf deleted file mode 100644 index efb1858..0000000 Binary files a/documentation/aacEncoder.pdf and /dev/null differ diff --git a/fdk-aac.sym b/fdk-aac.sym index 2a06c41..308fa3d 100644 --- a/fdk-aac.sym +++ b/fdk-aac.sym @@ -9,10 +9,3 @@ aacDecoder_GetLibInfo aacDecoder_GetStreamInfo aacDecoder_Open aacDecoder_SetParam -aacEncClose -aacEncEncode -aacEncGetLibInfo -aacEncInfo -aacEncOpen -aacEncoder_GetParam -aacEncoder_SetParam diff --git a/libAACdec/src/aac_ram.h b/libAACdec/src/aac_ram.h index 4527e27..09437a8 100644 --- a/libAACdec/src/aac_ram.h +++ b/libAACdec/src/aac_ram.h @@ -97,9 +97,6 @@ amm-info@iis.fraunhofer.de #include "channel.h" -#include "aacdec_hcr_types.h" -#include "aacdec_hcr.h" - /* End of formal fix.h */ #define MAX_SYNCHS 10 diff --git a/libAACdec/src/aac_rom.cpp b/libAACdec/src/aac_rom.cpp index f3c9b5a..a0b3651 100644 --- a/libAACdec/src/aac_rom.cpp +++ b/libAACdec/src/aac_rom.cpp @@ -1265,120 +1265,6 @@ const SCHAR *aQuantTable[] = {aValTab41, /* 0 - */ /* us aValTab24, /* 30 6 */ aValTab24}; /* 31 6 */ -/* arrays for HCR_TABLE_INFO structures */ -/* maximum length of codeword in each codebook */ -/* codebook: 0,1, 2,3, 4, 5, 6, 7, 8, 9, 10,11,12,13,14,15,16,17,18,19,20,21,22,23,24,25,26,27,28,29,30,31 */ -const UCHAR aMaxCwLen[MAX_CB]={0,11,9,20,16,13,11,14,12,17,14,49,0, 0, 0, 0, 14,17,21,21,25,25,29,29,29,29,33,33,33,37,37,41}; - -/* 11 13 15 17 19 21 23 25 27 39 31 */ -/* CB: 0 1 2 3 4 5 6 7 8 9 10 12 14 16 18 20 22 24 26 28 30 */ -const UCHAR aDimCb[MAX_CB] = {2,4,4,4,4,2,2,2,2,2,2,2,1,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2}; /* codebook dimension - zero cb got a dimension of 2 */ - -/* 11 13 15 17 19 21 23 25 27 39 31 */ -/* CB: 0 1 2 3 4 5 6 7 8 9 10 12 14 16 18 20 22 24 26 28 30 */ -const UCHAR aDimCbShift[MAX_CB]={1,2,2,2,2,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1}; /* codebook dimension */ - -/* 1 -> decode sign bits */ -/* 0 -> decode no sign bits 11 13 15 17 19 21 23 25 27 39 31 */ -/* CB: 0 1 2 3 4 5 6 7 8 9 10 12 14 16 18 20 22 24 26 28 30 */ -const UCHAR aSignCb[MAX_CB]={0,0,0,1,1,0,0,1,1,1,1,1,0,0,0,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1}; - -/* arrays for HCR_CB_PAIRS structures */ -const UCHAR aMinOfCbPair[MAX_CB_PAIRS]={0,1,3,5,7, 9,16,17,18,19,20,21,22,23,24,25,26,27,28,29,30,31,11}; -const UCHAR aMaxOfCbPair[MAX_CB_PAIRS]={0,2,4,6,8,10,16,17,18,19,20,21,22,23,24,25,26,27,28,29,30,31,11}; - -/* priorities of codebooks */ -const UCHAR aCbPriority[MAX_CB]={0,1,1,2,2,3,3,4,4,5,5,22,0,0,0,0,6,7,8,9,10,11,12,13,14,15,16,17,18,19,20,21}; - -const SCHAR aCodebook2StartInt[] = {STOP_THIS_STATE , /* cb 0 */ - BODY_ONLY , /* cb 1 */ - BODY_ONLY , /* cb 2 */ - BODY_SIGN__BODY , /* cb 3 */ - BODY_SIGN__BODY , /* cb 4 */ - BODY_ONLY , /* cb 5 */ - BODY_ONLY , /* cb 6 */ - BODY_SIGN__BODY , /* cb 7 */ - BODY_SIGN__BODY , /* cb 8 */ - BODY_SIGN__BODY , /* cb 9 */ - BODY_SIGN__BODY , /* cb 10 */ - BODY_SIGN_ESC__BODY, /* cb 11 */ - STOP_THIS_STATE , /* cb 12 */ - STOP_THIS_STATE , /* cb 13 */ - STOP_THIS_STATE , /* cb 14 */ - STOP_THIS_STATE , /* cb 15 */ - BODY_SIGN_ESC__BODY, /* cb 16 */ - BODY_SIGN_ESC__BODY, /* cb 17 */ - BODY_SIGN_ESC__BODY, /* cb 18 */ - BODY_SIGN_ESC__BODY, /* cb 19 */ - BODY_SIGN_ESC__BODY, /* cb 20 */ - BODY_SIGN_ESC__BODY, /* cb 21 */ - BODY_SIGN_ESC__BODY, /* cb 22 */ - BODY_SIGN_ESC__BODY, /* cb 23 */ - BODY_SIGN_ESC__BODY, /* cb 24 */ - BODY_SIGN_ESC__BODY, /* cb 25 */ - BODY_SIGN_ESC__BODY, /* cb 26 */ - BODY_SIGN_ESC__BODY, /* cb 27 */ - BODY_SIGN_ESC__BODY, /* cb 28 */ - BODY_SIGN_ESC__BODY, /* cb 29 */ - BODY_SIGN_ESC__BODY, /* cb 30 */ - BODY_SIGN_ESC__BODY}; /* cb 31 */ - -const STATEFUNC aStateConstant2State[] = {NULL , /* 0 = STOP_THIS_STATE */ - Hcr_State_BODY_ONLY , /* 1 = BODY_ONLY */ - Hcr_State_BODY_SIGN__BODY , /* 2 = BODY_SIGN__BODY */ - Hcr_State_BODY_SIGN__SIGN , /* 3 = BODY_SIGN__SIGN */ - Hcr_State_BODY_SIGN_ESC__BODY , /* 4 = BODY_SIGN_ESC__BODY */ - Hcr_State_BODY_SIGN_ESC__SIGN , /* 5 = BODY_SIGN_ESC__SIGN */ - Hcr_State_BODY_SIGN_ESC__ESC_PREFIX, /* 6 = BODY_SIGN_ESC__ESC_PREFIX */ - Hcr_State_BODY_SIGN_ESC__ESC_WORD }; /* 7 = BODY_SIGN_ESC__ESC_WORD */ - -/* CB: 0 1 2 3 4 5 6 7 8 9 10 12 14 16 18 20 22 24 26 28 30 */ -const USHORT aLargestAbsoluteValue[MAX_CB]={0,1,1,2,2,4,4,7,7,12,12,8191, 0, 0, 0, 0,15,31,47,63,95,127,159,191,223,255,319,383,511,767,1023,2047}; /* lav */ -/* CB: 11 13 15 17 19 21 23 25 27 39 31 */ - - -/* ------------------------------------------------------------------------------------------ - description: The table 'HuffTreeRvlcEscape' contains the decode tree for the rvlc - escape sequences. - bit 23 and 11 not used - bit 22 and 10 determine end value --> if set codeword is decoded - bit 21-12 and 9-0 (offset to next node) or (index value) - The escape sequence is the index value. - - input: codeword - output: index ------------------------------------------------------------------------------------------- */ -const UINT aHuffTreeRvlcEscape[53] = { 0x002001,0x400003,0x401004,0x402005,0x403007,0x404006,0x00a405,0x009008, - 0x00b406,0x00c407,0x00d408,0x00e409,0x40b40a,0x40c00f,0x40d010,0x40e011, - 0x40f012,0x410013,0x411014,0x412015,0x016413,0x414415,0x017416,0x417018, - 0x419019,0x01a418,0x01b41a,0x01c023,0x03201d,0x01e020,0x43501f,0x41b41c, - 0x021022,0x41d41e,0x41f420,0x02402b,0x025028,0x026027,0x421422,0x423424, - 0x02902a,0x425426,0x427428,0x02c02f,0x02d02e,0x42942a,0x42b42c,0x030031, - 0x42d42e,0x42f430,0x033034,0x431432,0x433434 }; - -/* ------------------------------------------------------------------------------------------ - description: The table 'HuffTreeRvlc' contains the huffman decoding tree for the RVLC - scale factors. The table contains 15 allowed, symmetric codewords and 8 - forbidden codewords, which are used for error detection. - - usage of bits: bit 23 and 11 not used - bit 22 and 10 determine end value --> if set codeword is decoded - bit 21-12 and 9-0 (offset to next node within the table) or (index+7). - The decoded (index+7) is in the range from 0,1,..,22. If the (index+7) - is in the range 15,16,..,22, then a forbidden codeword is decoded. - - input: A single bit from a RVLC scalefactor codeword - output: [if codeword is not completely decoded:] offset to next node within table or - [if codeword is decoded:] A dpcm value i.e. (index+7) in range from 0,1,..,22. - The differential scalefactor (DPCM value) named 'index' is calculated by - subtracting 7 from the decoded value (index+7). ------------------------------------------------------------------------------------------- */ -const UINT aHuffTreeRvlCodewds[22] = { 0x407001,0x002009,0x003406,0x004405,0x005404,0x006403,0x007400,0x008402, - 0x411401,0x00a408,0x00c00b,0x00e409,0x01000d,0x40f40a,0x41400f,0x01340b, - 0x011015,0x410012,0x41240c,0x416014,0x41540d,0x41340e }; - - - const FIXP_WTB LowDelaySynthesis512[1536] = { /* part 0 */ WTC(0xdac984c0), WTC(0xdb100080), WTC(0xdb56cd00), WTC(0xdb9dec40), WTC(0xdbe55fc0), WTC(0xdc2d2880), WTC(0xdc754780), WTC(0xdcbdbd80), @@ -1764,77 +1650,6 @@ WTC(0xffea32f4), WTC(0xfffae64c), WTC(0x0005aff3), WTC(0x000a7a44), WTC(0x00092f - -/* - * TNS_MAX_BANDS - * entry for each sampling rate - * 1 long window - * 2 SHORT window -*/ -const UCHAR tns_max_bands_tbl[13][2] = -{ - { 31, 9 }, /* 96000 */ - { 31, 9 }, /* 88200 */ - { 34, 10 }, /* 64000 */ - { 40, 14 }, /* 48000 */ - { 42, 14 }, /* 44100 */ - { 51, 14 }, /* 32000 */ - { 46, 14 }, /* 24000 */ - { 46, 14 }, /* 22050 */ - { 42, 14 }, /* 16000 */ - { 42, 14 }, /* 12000 */ - { 42, 14 }, /* 11025 */ - { 39, 14 }, /* 8000 */ - { 39, 14 }, /* 7350 */ -}; - -/* TNS_MAX_BANDS for low delay. The array index is the sampleRateIndex */ -const UCHAR tns_max_bands_tbl_480[13] = { - 31, /* 96000 */ - 31, /* 88200 */ - 31, /* 64000 */ - 31, /* 48000 */ - 32, /* 44100 */ - 37, /* 32000 */ - 30, /* 24000 */ - 30, /* 22050 */ - 30, /* 16000 */ - 30, /* 12000 */ - 30, /* 11025 */ - 30, /* 8000 */ - 30 /* 7350 */ -}; -const UCHAR tns_max_bands_tbl_512[13] = { - 31, /* 96000 */ - 31, /* 88200 */ - 31, /* 64000 */ - 31, /* 48000 */ - 32, /* 44100 */ - 37, /* 32000 */ - 31, /* 24000 */ - 31, /* 22050 */ - 31, /* 16000 */ - 31, /* 12000 */ - 31, /* 11025 */ - 31, /* 8000 */ - 31 /* 7350 */ -}; - -#define TCC(x) (FIXP_DBL(x)) - -const FIXP_TCC FDKaacDec_tnsCoeff3 [8] = -{ - TCC(0x81f1d1d4), TCC(0x9126146c), TCC(0xadb922c4), TCC(0xd438af1f), - TCC(0x00000000), TCC(0x3789809b), TCC(0x64130dd4), TCC(0x7cca7016) -}; -const FIXP_TCC FDKaacDec_tnsCoeff4 [16] = -{ - TCC(0x808bc842), TCC(0x84e2e58c), TCC(0x8d6b49d1), TCC(0x99da920a), - TCC(0xa9c45713), TCC(0xbc9ddeb9), TCC(0xd1c2d51b), TCC(0xe87ae53d), - TCC(0x00000000), TCC(0x1a9cd9b6), TCC(0x340ff254), TCC(0x4b3c8c29), - TCC(0x5f1f5ebb), TCC(0x6ed9ebba), TCC(0x79bc385f), TCC(0x7f4c7e5b) -}; - /* MPEG like mapping (no change). */ const UCHAR channelMappingTablePassthrough[15][8] = { diff --git a/libAACdec/src/aac_rom.h b/libAACdec/src/aac_rom.h index f314a2d..76c45f8 100644 --- a/libAACdec/src/aac_rom.h +++ b/libAACdec/src/aac_rom.h @@ -93,8 +93,6 @@ amm-info@iis.fraunhofer.de #include "common_fix.h" #include "FDK_audio.h" -#include "aacdec_hcr_types.h" -#include "aacdec_hcrs.h" #define AAC_NF_NO_RANDOM_VAL 512 /*!< Size of random number array for noise floor */ @@ -140,7 +138,7 @@ extern const CodeBookDescription AACcodeBookDescriptionTable[13]; extern const CodeBookDescription AACcodeBookDescriptionSCL; -extern const STATEFUNC aStateConstant2State[]; +#define MAX_CB 32 /* last used CB is cb #31 when VCB11 is used */ extern const SCHAR aCodebook2StartInt[]; @@ -158,20 +156,6 @@ extern const SCHAR *aQuantTable[]; extern const USHORT aLargestAbsoluteValue[]; -extern const UINT aHuffTreeRvlcEscape[]; -extern const UINT aHuffTreeRvlCodewds[]; - - -extern const UCHAR tns_max_bands_tbl[13][2]; - -extern const UCHAR tns_max_bands_tbl_480[13]; -extern const UCHAR tns_max_bands_tbl_512[13]; - -#define FIXP_TCC FIXP_DBL - -extern const FIXP_TCC FDKaacDec_tnsCoeff3[8]; -extern const FIXP_TCC FDKaacDec_tnsCoeff4[16]; - extern const USHORT randomSign[AAC_NF_NO_RANDOM_VAL/16]; extern const FIXP_DBL pow2_div24minus1[47]; diff --git a/libAACdec/src/aacdec_drc.cpp b/libAACdec/src/aacdec_drc.cpp index eb8e410..31d81b3 100644 --- a/libAACdec/src/aacdec_drc.cpp +++ b/libAACdec/src/aacdec_drc.cpp @@ -94,7 +94,6 @@ amm-info@iis.fraunhofer.de #include "channelinfo.h" #include "aac_rom.h" - #include "sbrdecoder.h" /* * Dynamic Range Control @@ -844,13 +843,11 @@ static int aacDecoder_drcExtractAndMap ( void aacDecoder_drcApply ( HANDLE_AAC_DRC self, - void *pSbrDec, CAacDecoderChannelInfo *pAacDecoderChannelInfo, CDrcChannelData *pDrcChData, FIXP_DBL *extGain, int ch, /* needed only for SBR */ - int aacFrameSize, - int bSbrPresent ) + int aacFrameSize ) { int band, top, bin, numBands; int bottom = 0; @@ -881,7 +878,6 @@ void aacDecoder_drcApply ( } if (!self->enable) { - sbrDecoder_drcDisable( (HANDLE_SBRDECODER)pSbrDec, ch ); if (extGain != NULL) { INT gainScale = (INT)*extGain; /* The gain scaling must be passed to the function in the buffer pointed on by extGain. */ @@ -1042,7 +1038,6 @@ void aacDecoder_drcApply ( * short blocks must take care that bands fall on * block boundaries! */ - if (!bSbrPresent) { bottom = 0; @@ -1081,23 +1076,6 @@ void aacDecoder_drcApply ( } } } - else { - HANDLE_SBRDECODER hSbrDecoder = (HANDLE_SBRDECODER)pSbrDec; - UINT numBands = pDrcChData->numBands; - - /* feed factors into SBR decoder for application in QMF domain. */ - sbrDecoder_drcFeedChannel ( - hSbrDecoder, - ch, - numBands, - fact_mantissa, - max_exponent, - pDrcChData->drcInterpolationScheme, - winSeq, - pDrcChData->bandTop - ); - } - return; } diff --git a/libAACdec/src/aacdec_drc.h b/libAACdec/src/aacdec_drc.h index c850aa5..7b61d3d 100644 --- a/libAACdec/src/aacdec_drc.h +++ b/libAACdec/src/aacdec_drc.h @@ -146,24 +146,20 @@ int aacDecoder_drcProlog ( /** * \brief Apply DRC. If SBR is present, DRC data is handed over to the SBR decoder. * \param self AAC decoder instance - * \param pSbrDec pointer to SBR decoder instance * \param pAacDecoderChannelInfo AAC decoder channel instance to be processed * \param pDrcDat DRC channel data * \param extGain Pointer to a FIXP_DBL where a externally applyable gain will be stored into (independently on whether it will be apply internally or not). * At function call the buffer must hold the scale (0 >= scale < DFRACT_BITS) to be applied on the gain value. * \param ch channel index * \param aacFrameSize AAC frame size - * \param bSbrPresent flag indicating that SBR is present, in which case DRC is handed over to the SBR instance pSbrDec */ void aacDecoder_drcApply ( HANDLE_AAC_DRC self, - void *pSbrDec, CAacDecoderChannelInfo *pAacDecoderChannelInfo, CDrcChannelData *pDrcDat, FIXP_DBL *extGain, int ch, - int aacFrameSize, - int bSbrPresent ); + int aacFrameSize ); int aacDecoder_drcEpilog ( HANDLE_AAC_DRC self, diff --git a/libAACdec/src/aacdec_hcr.cpp b/libAACdec/src/aacdec_hcr.cpp deleted file mode 100644 index e314e27..0000000 --- a/libAACdec/src/aacdec_hcr.cpp +++ /dev/null @@ -1,1591 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/***************************** MPEG-4 AAC Decoder *************************** - - Author(s): Robert Weidner (DSP Solutions) - Description: HCR Decoder: HCR initialization, preprocess HCR sideinfo, - decode priority codewords (PCWs) - -*******************************************************************************/ - -#include "aacdec_hcr.h" - - - -#include "aacdec_hcr_types.h" -#include "aacdec_hcr_bit.h" -#include "aacdec_hcrs.h" -#include "aac_ram.h" -#include "aac_rom.h" -#include "channel.h" -#include "block.h" - -#include "aacdecoder.h" /* for ID_CPE, ID_SCE ... */ -#include "FDK_bitstream.h" - -extern int mlFileChCurr; - -static void errDetectorInHcrSideinfoShrt(SCHAR cb, - SHORT numLine, - UINT *errorWord); - -static void errDetectorInHcrLengths(SCHAR lengthOfLongestCodeword, - SHORT lengthOfReorderedSpectralData, - UINT *errorWord); - -static void HcrCalcNumCodeword (H_HCR_INFO pHcr); -static void HcrSortCodebookAndNumCodewordInSection(H_HCR_INFO pHcr); -static void HcrPrepareSegmentationGrid (H_HCR_INFO pHcr); -static void HcrExtendedSectionInfo (H_HCR_INFO pHcr); - -static void DeriveNumberOfExtendedSortedSectionsInSets(UINT numSegment, - USHORT *pNumExtendedSortedCodewordInSection, - int numExtendedSortedCodewordInSectionIdx, - USHORT *pNumExtendedSortedSectionsInSets, - int numExtendedSortedSectionsInSetsIdx); - -static INT DecodeEscapeSequence(HANDLE_FDK_BITSTREAM bs, - INT quantSpecCoef, - USHORT *pLeftStartOfSegment, - SCHAR *pRemainingBitsInSegment, - int *pNumDecodedBits - ); - -static int DecodePCW_Sign(HANDLE_FDK_BITSTREAM bs, - UINT codebookDim, - const SCHAR *pQuantVal, - FIXP_DBL *pQuantSpecCoef, - int *quantSpecCoefIdx, - USHORT *pLeftStartOfSegment, - SCHAR *pRemainingBitsInSegment, - int *pNumDecodedBits - ); - -static const SCHAR *DecodePCW_Body(HANDLE_FDK_BITSTREAM bs, - const UINT *pCurrentTree, - const SCHAR *pQuantValBase, - USHORT *pLeftStartOfSegment, - SCHAR *pRemainingBitsInSegment, - int *pNumDecodedBits - ); - -static void DecodePCWs(HANDLE_FDK_BITSTREAM bs, H_HCR_INFO pHcr); - -static void HcrReorderQuantizedSpectralCoefficients( - H_HCR_INFO pHcr, - CAacDecoderChannelInfo *pAacDecoderChannelInfo, - const SamplingRateInfo *pSamplingRateInfo - ); - - -#if CHECK_SEGMENTATION_IMMEDIATELY -static UCHAR errDetectPcwSegmentation(SCHAR remainingBitsInSegment, - H_HCR_INFO pHcr, - PCW_TYPE kind, - FIXP_DBL *qsc_base_of_cw, - UCHAR dimension); -#endif - -#if CHECK_SEGMENTATION_FINAL -static void errDetectWithinSegmentationFinal(H_HCR_INFO pHcr); -#endif - -/*--------------------------------------------------------------------------------------------- - description: Check if codebook and numSect are within allowed range (short only) --------------------------------------------------------------------------------------------- */ -static void errDetectorInHcrSideinfoShrt(SCHAR cb, SHORT numLine,UINT* errorWord) -{ - - - - if ( cb < ZERO_HCB || cb >= MAX_CB_CHECK || cb == BOOKSCL ) { - *errorWord |= CB_OUT_OF_RANGE_SHORT_BLOCK; - } - if ( numLine < 0 || numLine > 1024 ) { - *errorWord |= LINE_IN_SECT_OUT_OF_RANGE_SHORT_BLOCK; - } -} - -/*--------------------------------------------------------------------------------------------- - description: Check both HCR lengths --------------------------------------------------------------------------------------------- */ -static void errDetectorInHcrLengths(SCHAR lengthOfLongestCodeword, - SHORT lengthOfReorderedSpectralData, - UINT *errorWord) -{ - if ( lengthOfReorderedSpectralData < lengthOfLongestCodeword ) { - *errorWord |= HCR_SI_LENGTHS_FAILURE; - } -} - -/*--------------------------------------------------------------------------------------------- - description: Decode (and adapt if necessary) the two HCR sideinfo components: - 'reordered_spectral_data_length' and 'longest_codeword_length' --------------------------------------------------------------------------------------------- */ - -void CHcr_Read(HANDLE_FDK_BITSTREAM bs, - CAacDecoderChannelInfo *pAacDecoderChannelInfo) -{ - INT globalHcrType = getHcrType(&pAacDecoderChannelInfo->pComData->overlay.aac.erHcrInfo); - SHORT lengOfReorderedSpectralData; - SCHAR lengOfLongestCodeword; - - pAacDecoderChannelInfo->pDynData->specificTo.aac.lenOfReorderedSpectralData = 0; - pAacDecoderChannelInfo->pDynData->specificTo.aac.lenOfLongestCodeword = 0; - - - - /* ------- SI-Value No 1 ------- */ - lengOfReorderedSpectralData = FDKreadBits(bs,14) + ERROR_LORSD; - if ( globalHcrType == ID_CPE ) { - if ((lengOfReorderedSpectralData >= 0) && (lengOfReorderedSpectralData <= CPE_TOP_LENGTH)) { - pAacDecoderChannelInfo->pDynData->specificTo.aac.lenOfReorderedSpectralData = lengOfReorderedSpectralData; /* the decoded value is within range */ - } - else { - if (lengOfReorderedSpectralData > CPE_TOP_LENGTH) { - pAacDecoderChannelInfo->pDynData->specificTo.aac.lenOfReorderedSpectralData = CPE_TOP_LENGTH; /* use valid maximum */ - } - } - } - else if (globalHcrType == ID_SCE || globalHcrType == ID_LFE || globalHcrType == ID_CCE ) { - if ((lengOfReorderedSpectralData >= 0) && (lengOfReorderedSpectralData <= SCE_TOP_LENGTH)) { - pAacDecoderChannelInfo->pDynData->specificTo.aac.lenOfReorderedSpectralData = lengOfReorderedSpectralData; /* the decoded value is within range */ - } - else { - if (lengOfReorderedSpectralData > SCE_TOP_LENGTH) { - pAacDecoderChannelInfo->pDynData->specificTo.aac.lenOfReorderedSpectralData = SCE_TOP_LENGTH; /* use valid maximum */ - } - } - } - - /* ------- SI-Value No 2 ------- */ - lengOfLongestCodeword = FDKreadBits(bs,6) + ERROR_LOLC; - if ((lengOfLongestCodeword >= 0) && (lengOfLongestCodeword <= LEN_OF_LONGEST_CW_TOP_LENGTH)) { - pAacDecoderChannelInfo->pDynData->specificTo.aac.lenOfLongestCodeword = lengOfLongestCodeword; /* the decoded value is within range */ - } - else { - if (lengOfLongestCodeword > LEN_OF_LONGEST_CW_TOP_LENGTH) { - pAacDecoderChannelInfo->pDynData->specificTo.aac.lenOfLongestCodeword = LEN_OF_LONGEST_CW_TOP_LENGTH; /* use valid maximum */ - } - } -} - - -/*--------------------------------------------------------------------------------------------- - description: Sets up HCR ROM-Tables --------------------------------------------------------------------------------------------- */ - -void HcrInitRom(H_HCR_INFO pHcr) -{ - pHcr->cbPairs.pMinOfCbPair = aMinOfCbPair; - pHcr->cbPairs.pMaxOfCbPair = aMaxOfCbPair; - - pHcr->tableInfo.pMaxCwLength = aMaxCwLen; - pHcr->tableInfo.pCbDimension = aDimCb; - pHcr->tableInfo.pCbDimShift = aDimCbShift; - pHcr->tableInfo.pCbSign = aSignCb; - pHcr->tableInfo.pCbPriority = aCbPriority; - pHcr->tableInfo.pLargestAbsVal = aLargestAbsoluteValue; -} - -/*--------------------------------------------------------------------------------------------- - description: Set up HCR - must be called before every call to HcrDecoder(). - For short block a sorting algorithm is applied to get the SI in the order - that HCR could assemble the qsc's as if it is a long block. ------------------------------------------------------------------------------------------------ - return: error log --------------------------------------------------------------------------------------------- */ - -UINT HcrInit(H_HCR_INFO pHcr, - CAacDecoderChannelInfo *pAacDecoderChannelInfo, - const SamplingRateInfo *pSamplingRateInfo, - HANDLE_FDK_BITSTREAM bs) -{ - CIcsInfo *pIcsInfo = &pAacDecoderChannelInfo->icsInfo; - SHORT *pNumLinesInSec; - UCHAR *pCodeBk; - SHORT numSection; - SCHAR cb; - int numLine; - int i; - - pHcr->decInOut.lengthOfReorderedSpectralData = pAacDecoderChannelInfo->pDynData->specificTo.aac.lenOfReorderedSpectralData; - pHcr->decInOut.lengthOfLongestCodeword = pAacDecoderChannelInfo->pDynData->specificTo.aac.lenOfLongestCodeword; - pHcr->decInOut.pQuantizedSpectralCoefficientsBase = pAacDecoderChannelInfo->pSpectralCoefficient; - pHcr->decInOut.quantizedSpectralCoefficientsIdx = 0; - pHcr->decInOut.pCodebook = pAacDecoderChannelInfo->pDynData->specificTo.aac.aCodeBooks4Hcr; - pHcr->decInOut.pNumLineInSect = pAacDecoderChannelInfo->pDynData->specificTo.aac.aNumLineInSec4Hcr; - pHcr->decInOut.numSection = pAacDecoderChannelInfo->pDynData->specificTo.aac.numberSection; - pHcr->decInOut.errorLog = 0; - pHcr->nonPcwSideinfo.pResultBase = SPEC_LONG(pAacDecoderChannelInfo->pSpectralCoefficient); - - FDKsyncCache(bs); - pHcr->decInOut.bitstreamIndex = FDKgetBitCnt(bs); - - if (!IsLongBlock(&pAacDecoderChannelInfo->icsInfo)) /* short block */ - { - SHORT band; - SHORT maxBand; - SCHAR group; - SCHAR winGroupLen; - SCHAR window; - SCHAR numUnitInBand; - SCHAR cntUnitInBand; - SCHAR groupWin; - SCHAR cb_prev; - - UCHAR *pCodeBook; - const SHORT *BandOffsets; - SCHAR numOfGroups; - - - pCodeBook = pAacDecoderChannelInfo->pDynData->aCodeBook; /* in */ - pNumLinesInSec = pHcr->decInOut.pNumLineInSect; /* out */ - pCodeBk = pHcr->decInOut.pCodebook; /* out */ - BandOffsets = GetScaleFactorBandOffsets(pIcsInfo, pSamplingRateInfo); /* aux */ - numOfGroups = GetWindowGroups(pIcsInfo); - - numLine = 0; - numSection = 0; - cb = pCodeBook[0]; - cb_prev = pCodeBook[0]; - - /* convert HCR-sideinfo into a unitwise manner: When the cb changes, a new section starts */ - - *pCodeBk++ = cb_prev; - - maxBand = GetScaleFactorBandsTransmitted(&pAacDecoderChannelInfo->icsInfo); - for (band = 0; band < maxBand; band++) { /* from low to high sfbs i.e. from low to high frequencies */ - numUnitInBand = ((BandOffsets[band+1] - BandOffsets[band]) >> FOUR_LOG_DIV_TWO_LOG); /* get the number of units in current sfb */ - for (cntUnitInBand = numUnitInBand; cntUnitInBand != 0; cntUnitInBand-- ) { /* for every unit in the band */ - for (window = 0, group = 0; group < numOfGroups; group++) { - winGroupLen = GetWindowGroupLength(&pAacDecoderChannelInfo->icsInfo,group); - for (groupWin = winGroupLen; groupWin != 0; groupWin--, window++) { - cb = pCodeBook[group * 16 + band]; - if (cb != cb_prev) { -#if CHECK_VALID_HCR_INPUT /* short-block 1 of 2 */ - errDetectorInHcrSideinfoShrt(cb,numLine,&pHcr->decInOut.errorLog ); - if (pHcr->decInOut.errorLog != 0 ) { - return ( pHcr->decInOut.errorLog ); - } -#endif - *pCodeBk++ = cb; - *pNumLinesInSec++ = numLine; - numSection++; - - cb_prev = cb; - numLine = LINES_PER_UNIT; - } - else { - numLine += LINES_PER_UNIT; - } - } - } - } - } - - numSection++; - -#if CHECK_VALID_HCR_INPUT /* short-block 2 of 2 */ - errDetectorInHcrSideinfoShrt(cb,numLine,&pHcr->decInOut.errorLog ); - if ( numSection <= 0 || numSection > 1024/2 ) { - pHcr->decInOut.errorLog |= NUM_SECT_OUT_OF_RANGE_SHORT_BLOCK; - } - errDetectorInHcrLengths(pHcr->decInOut.lengthOfLongestCodeword, - pHcr->decInOut.lengthOfReorderedSpectralData, - &pHcr->decInOut.errorLog); - if (pHcr->decInOut.errorLog != 0 ) { - return ( pHcr->decInOut.errorLog ); - } -#endif - - *pCodeBk = cb; - *pNumLinesInSec = numLine; - pHcr->decInOut.numSection = numSection; - - } else /* end short block prepare SI */ - { /* long block */ -#if CHECK_VALID_HCR_INPUT /* long-block 1 of 1 */ - errDetectorInHcrLengths(pHcr->decInOut.lengthOfLongestCodeword, - pHcr->decInOut.lengthOfReorderedSpectralData, - &pHcr->decInOut.errorLog); - numSection = pHcr->decInOut.numSection; - pNumLinesInSec = pHcr->decInOut.pNumLineInSect; - pCodeBk = pHcr->decInOut.pCodebook; - if ( numSection <= 0 || numSection > 64 ) { - pHcr->decInOut.errorLog |= NUM_SECT_OUT_OF_RANGE_LONG_BLOCK; - numSection = 0; - } - - for ( i = numSection; i != 0; i-- ) - { - cb = *pCodeBk++; - - if ( cb < ZERO_HCB || cb >= MAX_CB_CHECK || cb == BOOKSCL ) { - pHcr->decInOut.errorLog |= CB_OUT_OF_RANGE_LONG_BLOCK; - } - - numLine = *pNumLinesInSec++; - /* FDK_ASSERT(numLine > 0); */ - - if ( (numLine <= 0) || (numLine > 1024) ) { - pHcr->decInOut.errorLog |= LINE_IN_SECT_OUT_OF_RANGE_LONG_BLOCK; - } - } - if (pHcr->decInOut.errorLog != 0 ) { - return ( pHcr->decInOut.errorLog ); - } -#endif /* CHECK_VALID_HCR_INPUT */ - } - - pCodeBk = pHcr->decInOut.pCodebook; - for ( i = 0; i < numSection; i++ ) { - if ( - (*pCodeBk == NOISE_HCB) || - (*pCodeBk == INTENSITY_HCB2) || - (*pCodeBk == INTENSITY_HCB)) - { - *pCodeBk = 0; - } - pCodeBk++; - } - - /* HCR-sideinfo-input is complete and seems to be valid */ - - - - return ( pHcr->decInOut.errorLog ); -} - - - - -#if USE_HCR_DUMMY - -/*--------------------------------------------------------------------------------------------- - - description: This HCR - dummy - function writes only a dirac-sequence in output buffer - --------------------------------------------------------------------------------------------- */ -UINT HcrDecoder(H_HCR_INFO pHcr, - const CAacDecoderChannelInfo *pAacDecoderChannelInfo, - HANDLE_FDK_BITSTREAM bs) -{ - for (SHORT i=0; i < 1024; i++ ) { - pHcr->decInOut.pQuantizedSpectralCoefficients->Long[i] = FL2FXCONST_DBL(0.0f); - if ( i % 30 == 0) { - pHcr->decInOut.pQuantizedSpectralCoefficients->Long[i] = (FIXP_DBL)HCR_DIRAC; - } - } - return 0; -} - -#else /* USE_HCR_DUMMY */ - -/*--------------------------------------------------------------------------------------------- - description: This function decodes the codewords of the spectral coefficients from the - bitstream according to the HCR algorithm and stores the quantized spectral - coefficients in correct order in the output buffer. --------------------------------------------------------------------------------------------- */ - -UINT HcrDecoder(H_HCR_INFO pHcr, - CAacDecoderChannelInfo *pAacDecoderChannelInfo, - const SamplingRateInfo *pSamplingRateInfo, - HANDLE_FDK_BITSTREAM bs) -{ - int pTmp1, pTmp2, pTmp3, pTmp4; -#if DETECT_TOO_LONG_CW_READS - int pTmp5; -#endif - - INT bitCntOffst; - UINT saveBitCnt = FDKgetBitCnt(bs); /* save bitstream position */ - - HcrCalcNumCodeword(pHcr); - - HcrSortCodebookAndNumCodewordInSection(pHcr); - - HcrPrepareSegmentationGrid(pHcr); - - HcrExtendedSectionInfo(pHcr); - - if (( pHcr->decInOut.errorLog & HCR_FATAL_PCW_ERROR_MASK ) != 0 ) { - return ( pHcr->decInOut.errorLog ); /* sideinfo is massively corrupt, return from HCR without having decoded anything */ - } - - DeriveNumberOfExtendedSortedSectionsInSets(pHcr->segmentInfo.numSegment, - pHcr->sectionInfo.pNumExtendedSortedCodewordInSection, - pHcr->sectionInfo.numExtendedSortedCodewordInSectionIdx, - pHcr->sectionInfo.pNumExtendedSortedSectionsInSets, - pHcr->sectionInfo.numExtendedSortedSectionsInSetsIdx); - - /* store */ - pTmp1 = pHcr->sectionInfo.numExtendedSortedCodewordInSectionIdx; - pTmp2 = pHcr->sectionInfo.extendedSortedCodebookIdx; - pTmp3 = pHcr->sectionInfo.numExtendedSortedSectionsInSetsIdx; - pTmp4 = pHcr->decInOut.quantizedSpectralCoefficientsIdx; -#if DETECT_TOO_LONG_CW_READS - pTmp5 = pHcr->sectionInfo.maxLenOfCbInExtSrtSecIdx; -#endif - - /* ------- decode meaningful PCWs ------ */ - DecodePCWs(bs, pHcr); - - if (( pHcr->decInOut.errorLog & HCR_FATAL_PCW_ERROR_MASK ) == 0 ) { - /* ------ decode the non-PCWs -------- */ - DecodeNonPCWs(bs, pHcr); - } - - -#if CHECK_SEGMENTATION_FINAL - errDetectWithinSegmentationFinal(pHcr); -#endif - - /* restore */ - pHcr->sectionInfo.numExtendedSortedCodewordInSectionIdx = pTmp1; - pHcr->sectionInfo.extendedSortedCodebookIdx = pTmp2; - pHcr->sectionInfo.numExtendedSortedSectionsInSetsIdx = pTmp3; - pHcr->decInOut.quantizedSpectralCoefficientsIdx = pTmp4; -#if DETECT_TOO_LONG_CW_READS - pHcr->sectionInfo.maxLenOfCbInExtSrtSecIdx = pTmp5; -#endif - - HcrReorderQuantizedSpectralCoefficients(pHcr, pAacDecoderChannelInfo, pSamplingRateInfo); - - /* restore bitstream position */ - bitCntOffst = saveBitCnt - FDKgetBitCnt(bs); - if( bitCntOffst ) { - FDKpushBiDirectional(bs, bitCntOffst); - } - - return ( pHcr->decInOut.errorLog ); -} - - -#endif /* USE_HCR_DUMMY */ - - - - -/*--------------------------------------------------------------------------------------------- - description: This function reorders the quantized spectral coefficients sectionwise for - long- and short-blocks and compares to the LAV (Largest Absolute Value of - the current codebook) -- a counter is incremented if there is an error - detected. - Additional for short-blocks a unit-based-deinterleaving is applied. - Moreover (for short blocks) the scaling is derived (compare plain huffman - decoder). --------------------------------------------------------------------------------------------- */ - -static void HcrReorderQuantizedSpectralCoefficients( - H_HCR_INFO pHcr, CAacDecoderChannelInfo *pAacDecoderChannelInfo, - const SamplingRateInfo *pSamplingRateInfo - ) -{ - INT qsc; - UINT abs_qsc; - UINT i,j; - USHORT numSpectralValuesInSection; - FIXP_DBL *pTeVa; - USHORT lavErrorCnt = 0; - - UINT numSection = pHcr->decInOut.numSection; - SPECTRAL_PTR pQuantizedSpectralCoefficientsBase = pHcr->decInOut.pQuantizedSpectralCoefficientsBase; - FIXP_DBL *pQuantizedSpectralCoefficients = SPEC_LONG(pHcr->decInOut.pQuantizedSpectralCoefficientsBase); - const UCHAR *pCbDimShift = pHcr->tableInfo.pCbDimShift; - const USHORT *pLargestAbsVal = pHcr->tableInfo.pLargestAbsVal; - UCHAR *pSortedCodebook = pHcr->sectionInfo.pSortedCodebook; - USHORT *pNumSortedCodewordInSection = pHcr->sectionInfo.pNumSortedCodewordInSection; - USHORT *pReorderOffset = pHcr->sectionInfo.pReorderOffset; - FIXP_DBL *pTempValues = pHcr->segmentInfo.pTempValues; - FIXP_DBL *pBak = pHcr->segmentInfo.pTempValues; - - FDKmemclear(pTempValues,1024*sizeof(FIXP_DBL)); - - /* long and short: check if decoded huffman-values (quantized spectral coefficients) are within range */ - for ( i=numSection; i != 0; i-- ) { - numSpectralValuesInSection = *pNumSortedCodewordInSection++ << pCbDimShift[*pSortedCodebook]; - pTeVa = &pTempValues[*pReorderOffset++]; - for( j = numSpectralValuesInSection; j != 0; j-- ) { - qsc = *pQuantizedSpectralCoefficients++; - abs_qsc = FDKabs(qsc); -#if VALID_LAV_ERROR_TRIGGER - if ( abs_qsc <= pLargestAbsVal[*pSortedCodebook] ) { - *pTeVa++ = (FIXP_DBL)qsc; /* the qsc value is within range */ - } - else { /* line is too high .. */ - if ( abs_qsc == Q_VALUE_INVALID ) { /* .. because of previous marking --> dont set LAV flag (would be confusing), just copy out the already marked value */ - *pTeVa++ = (FIXP_DBL) qsc; - } - else { /* .. because a too high value was decoded for this cb --> set LAV flag */ - *pTeVa++ = (FIXP_DBL) Q_VALUE_INVALID; - lavErrorCnt += 1; - } - } -#else - if ( abs_qsc <= pLargestAbsVal[*pSortedCodebook] ) { - *pTeVa++ = qsc; - } - else { - *pTeVa++ = Q_VALUE_INVALID; - lavErrorCnt += 1; - } -#endif - } - pSortedCodebook++; - } - - if (!IsLongBlock(&pAacDecoderChannelInfo->icsInfo)) - { - FIXP_DBL *pOut; - FIXP_DBL locMax; - FIXP_DBL tmp; - SCHAR groupoffset; - SCHAR group; - SCHAR band; - SCHAR groupwin; - SCHAR window; - SCHAR numWinGroup; - SHORT interm; - SCHAR numSfbTransm; - SCHAR winGroupLen; - SHORT index; - INT msb; - INT lsb; - - SHORT *pScaleFacHcr = pAacDecoderChannelInfo->pDynData->aScaleFactor; - SHORT *pSfbSclHcr = pAacDecoderChannelInfo->pDynData->aSfbScale; - const SHORT *BandOffsets = GetScaleFactorBandOffsets(&pAacDecoderChannelInfo->icsInfo, pSamplingRateInfo); - - pBak = pHcr->segmentInfo.pTempValues; - /* deinterleave unitwise for short blocks */ - for ( window = 0; window < (8); window++ ) { - pOut = SPEC(pQuantizedSpectralCoefficientsBase, window, pAacDecoderChannelInfo->granuleLength); - for ( i=0; i < (LINES_PER_UNIT_GROUP); i++ ) { - pTeVa = pBak + (window << FOUR_LOG_DIV_TWO_LOG) + i * 32; /* distance of lines between unit groups has to be constant for every framelength (32)! */ - for ( j=(LINES_PER_UNIT); j != 0; j-- ) { - *pOut++ = *pTeVa++; - } - } - } - - /* short blocks only */ - /* derive global scaling-value for every sfb and every window (as it is done in plain-huffman-decoder at short blocks) */ - groupoffset = 0; - - numWinGroup = GetWindowGroups(&pAacDecoderChannelInfo->icsInfo); - numSfbTransm = GetScaleFactorBandsTransmitted(&pAacDecoderChannelInfo->icsInfo); - - for (group = 0; group < numWinGroup; group++) { - winGroupLen = GetWindowGroupLength(&pAacDecoderChannelInfo->icsInfo,group); - for (band = 0; band < numSfbTransm; band++) { - interm = group * 16 + band; - msb = pScaleFacHcr[interm] >> 2; - lsb = pScaleFacHcr[interm] & 3; - for (groupwin = 0; groupwin < winGroupLen; groupwin++) { - window = groupoffset + groupwin; - pBak = SPEC(pQuantizedSpectralCoefficientsBase, window, pAacDecoderChannelInfo->granuleLength); - locMax = FL2FXCONST_DBL(0.0f); - for (index = BandOffsets[band]; index < BandOffsets[band+1]; index += LINES_PER_UNIT) { - pTeVa = &pBak[index]; - for ( i = LINES_PER_UNIT; i != 0; i --) { - tmp = (*pTeVa < FL2FXCONST_DBL(0.0f))? -*pTeVa++ : *pTeVa++; - locMax = fixMax(tmp,locMax); - } - } - if ( fixp_abs(locMax) > (FIXP_DBL)MAX_QUANTIZED_VALUE ) { - locMax = (FIXP_DBL)MAX_QUANTIZED_VALUE; - } - pSfbSclHcr[window*16+band] = msb - GetScaleFromValue(locMax, lsb); /* save global scale maxima in this sfb */ - } - } - groupoffset += GetWindowGroupLength(&pAacDecoderChannelInfo->icsInfo,group); - } - } else - { - /* copy straight for long-blocks */ - pQuantizedSpectralCoefficients = SPEC_LONG(pQuantizedSpectralCoefficientsBase); - for ( i = 1024; i != 0; i-- ) { - *pQuantizedSpectralCoefficients++ = *pBak++; - } - } - - if ( lavErrorCnt != 0 ) { - pHcr->decInOut.errorLog |= LAV_VIOLATION; - } -} - - -/*--------------------------------------------------------------------------------------------- - description: This function calculates the number of codewords - for each section (numCodewordInSection) and the number of codewords - for all sections (numCodeword). - For zero and intensity codebooks a entry is also done in the variable - numCodewordInSection. It is assumed that the codebook is a two tuples - codebook. This is needed later for the calculation of the base addresses - for the reordering of the quantize spectral coefficients at the end of the - hcr tool. - The variable numCodeword contain the number of codewords which are really - in the bitstream. Zero or intensity codebooks does not increase the - variable numCodewords. ------------------------------------------------------------------------------------------------ - return: - --------------------------------------------------------------------------------------------- */ - -static void HcrCalcNumCodeword(H_HCR_INFO pHcr) -{ - int hcrSection; - UINT numCodeword; - - UINT numSection = pHcr->decInOut.numSection; - UCHAR *pCodebook = pHcr->decInOut.pCodebook; - SHORT *pNumLineInSection = pHcr->decInOut.pNumLineInSect; - const UCHAR *pCbDimShift = pHcr->tableInfo.pCbDimShift; - USHORT *pNumCodewordInSection = pHcr->sectionInfo.pNumCodewordInSection; - - numCodeword = 0; - for ( hcrSection = numSection; hcrSection != 0; hcrSection-- ) { - *pNumCodewordInSection = *pNumLineInSection++ >> pCbDimShift[*pCodebook]; - if ( *pCodebook != 0 ) { - numCodeword += *pNumCodewordInSection; - } - pNumCodewordInSection++; - pCodebook++; - } - pHcr->sectionInfo.numCodeword = numCodeword; -} - - -/*--------------------------------------------------------------------------------------------- - description: This function calculates the number - of sorted codebooks and sorts the codebooks and the numCodewordInSection - according to the priority. --------------------------------------------------------------------------------------------- */ - -static void HcrSortCodebookAndNumCodewordInSection(H_HCR_INFO pHcr) -{ - - UINT i,j,k; - UCHAR temp; - UINT counter; - UINT startOffset; - UINT numZeroSection; - UCHAR *pDest; - UINT numSectionDec; - - UINT numSection = pHcr->decInOut.numSection; - UCHAR *pCodebook = pHcr->decInOut.pCodebook; - UCHAR *pSortedCodebook = pHcr->sectionInfo.pSortedCodebook; - USHORT *pNumCodewordInSection = pHcr->sectionInfo.pNumCodewordInSection; - USHORT *pNumSortedCodewordInSection = pHcr->sectionInfo.pNumSortedCodewordInSection; - UCHAR *pCodebookSwitch = pHcr->sectionInfo.pCodebookSwitch; - USHORT *pReorderOffset = pHcr->sectionInfo.pReorderOffset; - const UCHAR *pCbPriority = pHcr->tableInfo.pCbPriority; - const UCHAR *pMinOfCbPair = pHcr->cbPairs.pMinOfCbPair; - const UCHAR *pMaxOfCbPair = pHcr->cbPairs.pMaxOfCbPair; - const UCHAR *pCbDimShift = pHcr->tableInfo.pCbDimShift; - - UINT searchStart = 0; - - /* calculate *pNumSortedSection and store the priorities in array pSortedCdebook */ - pDest = pSortedCodebook; - numZeroSection = 0; - for ( i=numSection; i != 0; i-- ) { - if ( pCbPriority[*pCodebook] == 0 ) { - numZeroSection += 1; - } - *pDest++ = pCbPriority[*pCodebook++]; - } - pHcr->sectionInfo.numSortedSection = numSection - numZeroSection; /* numSortedSection contains no zero or intensity section */ - pCodebook = pHcr->decInOut.pCodebook; - - /* sort priorities of the codebooks in array pSortedCdebook[] */ - numSectionDec = numSection - 1; - if ( numSectionDec > 0 ) { - counter = numSectionDec; - for ( j=numSectionDec; j != 0; j-- ) { - for ( i=0; i < counter; i++ ) { - /* swap priorities */ - if ( pSortedCodebook[i+1] > pSortedCodebook[i] ) { - temp = pSortedCodebook[i]; - pSortedCodebook[i] = pSortedCodebook[i+1]; - pSortedCodebook[i+1] = temp; - } - } - counter -= 1; - } - } - - /* clear codebookSwitch array */ - for ( i = numSection; i != 0; i--) { - *pCodebookSwitch++ = 0; - } - pCodebookSwitch = pHcr->sectionInfo.pCodebookSwitch; - - /* sort sectionCodebooks and numCodwordsInSection and calculate pReorderOffst[j] */ - for ( j = 0; j < numSection; j++ ) { - for ( i = searchStart; i < numSection; i++ ) { - if ( pCodebookSwitch[i] == 0 && ( pMinOfCbPair[pSortedCodebook[j]] == pCodebook[i] || pMaxOfCbPair[pSortedCodebook[j]] == pCodebook[i] )) { - pCodebookSwitch[i] = 1; - pSortedCodebook[j] = pCodebook[i]; /* sort codebook */ - pNumSortedCodewordInSection[j] = pNumCodewordInSection[i]; /* sort NumCodewordInSection */ - - startOffset = 0; - for ( k = 0; k < i; k++ ) { /* make entry in pReorderOffst */ - startOffset += pNumCodewordInSection[k] << pCbDimShift[pCodebook[k]]; - } - pReorderOffset[j] = startOffset; /* offset for reordering the codewords */ - - if(i == searchStart) { - UINT k = i; - while(pCodebookSwitch[k++] == 1) searchStart++; - } - break; - } - } - } -} - - -/*--------------------------------------------------------------------------------------------- - description: This function calculates the segmentation, which includes numSegment, - leftStartOfSegment, rightStartOfSegment and remainingBitsInSegment. - The segmentation could be visualized a as kind of 'overlay-grid' for the - bitstream-block holding the HCR-encoded quantized-spectral-coefficients. --------------------------------------------------------------------------------------------- */ - -static void HcrPrepareSegmentationGrid(H_HCR_INFO pHcr) -{ - USHORT i,j; - USHORT numSegment = 0; - USHORT segmentStart = 0; - UCHAR segmentWidth; - UCHAR lastSegmentWidth; - UCHAR sortedCodebook; - UCHAR endFlag = 0; - USHORT intermediateResult; - - SCHAR lengthOfLongestCodeword = pHcr->decInOut.lengthOfLongestCodeword; - SHORT lengthOfReorderedSpectralData = pHcr->decInOut.lengthOfReorderedSpectralData; - UINT numSortedSection = pHcr->sectionInfo.numSortedSection; - UCHAR *pSortedCodebook = pHcr->sectionInfo.pSortedCodebook; - USHORT *pNumSortedCodewordInSection = pHcr->sectionInfo.pNumSortedCodewordInSection; - USHORT *pLeftStartOfSegment = pHcr->segmentInfo.pLeftStartOfSegment; - USHORT *pRightStartOfSegment = pHcr->segmentInfo.pRightStartOfSegment; - SCHAR *pRemainingBitsInSegment = pHcr->segmentInfo.pRemainingBitsInSegment; - USHORT bitstreamIndex = pHcr->decInOut.bitstreamIndex; - const UCHAR *pMaxCwLength = pHcr->tableInfo.pMaxCwLength; - - for ( i=numSortedSection; i != 0; i-- ) { - sortedCodebook = *pSortedCodebook++; - segmentWidth = FDKmin(pMaxCwLength[sortedCodebook],lengthOfLongestCodeword); - - for ( j = *pNumSortedCodewordInSection; j != 0 ; j-- ) { - /* width allows a new segment */ - intermediateResult = bitstreamIndex + segmentStart; - if ( (segmentStart + segmentWidth) <= lengthOfReorderedSpectralData ) { - /* store segment start, segment length and increment the number of segments */ - *pLeftStartOfSegment++ = intermediateResult; - *pRightStartOfSegment++ = intermediateResult + segmentWidth - 1; - *pRemainingBitsInSegment++ = segmentWidth; - segmentStart += segmentWidth; - numSegment += 1; - } - /* width does not allow a new segment */ - else { - /* correct the last segment length */ - pLeftStartOfSegment--; - pRightStartOfSegment--; - pRemainingBitsInSegment--; - segmentStart = *pLeftStartOfSegment - bitstreamIndex; - - lastSegmentWidth = lengthOfReorderedSpectralData - segmentStart; - *pRemainingBitsInSegment = lastSegmentWidth; - *pRightStartOfSegment = bitstreamIndex + segmentStart + lastSegmentWidth - 1; - endFlag = 1; - break; - } - } - pNumSortedCodewordInSection++; - if (endFlag != 0) { - break; - } - } - pHcr->segmentInfo.numSegment = numSegment; - -} - - -/*--------------------------------------------------------------------------------------------- - description: This function adapts the sorted section boundaries to the boundaries of - segmentation. If the section lengths does not fit completely into the - current segment, the section is spitted into two so called 'extended - sections'. The extended-section-info (pNumExtendedSortedCodewordInSectin - and pExtendedSortedCodebook) is updated in this case. - --------------------------------------------------------------------------------------------- */ - -static void HcrExtendedSectionInfo(H_HCR_INFO pHcr) -{ - UINT srtSecCnt = 0; /* counter for sorted sections */ - UINT xSrtScCnt = 0; /* counter for extended sorted sections */ - UINT remainNumCwInSortSec; - UINT inSegmentRemainNumCW; - - UINT numSortedSection = pHcr->sectionInfo.numSortedSection; - UCHAR *pSortedCodebook = pHcr->sectionInfo.pSortedCodebook; - USHORT *pNumSortedCodewordInSection = pHcr->sectionInfo.pNumSortedCodewordInSection; - UCHAR *pExtendedSortedCoBo = pHcr->sectionInfo.pExtendedSortedCodebook; - USHORT *pNumExtSortCwInSect = pHcr->sectionInfo.pNumExtendedSortedCodewordInSection; - UINT numSegment = pHcr->segmentInfo.numSegment; -#if DETECT_TOO_LONG_CW_READS - UCHAR *pMaxLenOfCbInExtSrtSec = pHcr->sectionInfo.pMaxLenOfCbInExtSrtSec; - SCHAR lengthOfLongestCodeword = pHcr->decInOut.lengthOfLongestCodeword; - const UCHAR *pMaxCwLength = pHcr->tableInfo.pMaxCwLength; -#endif - - remainNumCwInSortSec = pNumSortedCodewordInSection[srtSecCnt]; - inSegmentRemainNumCW = numSegment; - - while (srtSecCnt < numSortedSection) { - if (inSegmentRemainNumCW < remainNumCwInSortSec) { - - pNumExtSortCwInSect[xSrtScCnt] = inSegmentRemainNumCW; - pExtendedSortedCoBo[xSrtScCnt] = pSortedCodebook[srtSecCnt]; - - remainNumCwInSortSec -= inSegmentRemainNumCW; - inSegmentRemainNumCW = numSegment; - /* data of a sorted section was not integrated in extended sorted section */ - } - else if (inSegmentRemainNumCW == remainNumCwInSortSec) { - pNumExtSortCwInSect[xSrtScCnt] = inSegmentRemainNumCW; - pExtendedSortedCoBo[xSrtScCnt] = pSortedCodebook[srtSecCnt]; - - srtSecCnt++; - remainNumCwInSortSec = pNumSortedCodewordInSection[srtSecCnt]; - inSegmentRemainNumCW = numSegment; - /* data of a sorted section was integrated in extended sorted section */ - } - else { /* inSegmentRemainNumCW > remainNumCwInSortSec */ - pNumExtSortCwInSect[xSrtScCnt] = remainNumCwInSortSec; - pExtendedSortedCoBo[xSrtScCnt] = pSortedCodebook[srtSecCnt]; - - - inSegmentRemainNumCW -= remainNumCwInSortSec; - srtSecCnt++; - remainNumCwInSortSec = pNumSortedCodewordInSection[srtSecCnt]; - /* data of a sorted section was integrated in extended sorted section */ - } -#if DETECT_TOO_LONG_CW_READS - pMaxLenOfCbInExtSrtSec[xSrtScCnt] = FDKmin(pMaxCwLength[pExtendedSortedCoBo[xSrtScCnt]],lengthOfLongestCodeword); -#endif - - - - xSrtScCnt += 1; - - if ( xSrtScCnt >= (MAX_SFB_HCR + MAX_HCR_SETS) ) { - pHcr->decInOut.errorLog |= EXTENDED_SORTED_COUNTER_OVERFLOW; - return; - } - - } - pNumExtSortCwInSect[xSrtScCnt] = 0; - -} - - -/*--------------------------------------------------------------------------------------------- - description: This function calculates the number of extended sorted sections which - belong to the sets. Each set from set 0 (one and only set for the PCWs) - till to the last set gets a entry in the array to which - 'pNumExtendedSortedSectinsInSets' points to. - - Calculation: The entrys in pNumExtendedSortedCodewordInSectin are added - untill the value numSegment is reached. Then the sum_variable is cleared - and the calculation starts from the beginning. As much extended sorted - Sections are summed up to reach the value numSegment, as much is the - current entry in *pNumExtendedSortedCodewordInSectin. --------------------------------------------------------------------------------------------- */ -static void DeriveNumberOfExtendedSortedSectionsInSets(UINT numSegment, - USHORT *pNumExtendedSortedCodewordInSection, - int numExtendedSortedCodewordInSectionIdx, - USHORT *pNumExtendedSortedSectionsInSets, - int numExtendedSortedSectionsInSetsIdx) -{ - USHORT counter = 0; - UINT cwSum = 0; - USHORT *pNumExSortCwInSec = pNumExtendedSortedCodewordInSection; - USHORT *pNumExSortSecInSets = pNumExtendedSortedSectionsInSets; - - while (pNumExSortCwInSec[numExtendedSortedCodewordInSectionIdx] != 0) - { - cwSum += pNumExSortCwInSec[numExtendedSortedCodewordInSectionIdx]; - numExtendedSortedCodewordInSectionIdx++; - if (numExtendedSortedCodewordInSectionIdx >= (MAX_SFB_HCR+MAX_HCR_SETS)) { - return; - } - if (cwSum > numSegment) { - return; - } - counter++; - if (counter > 1024/4) { - return; - } - if ( cwSum == numSegment ) { - pNumExSortSecInSets[numExtendedSortedSectionsInSetsIdx] = counter; - numExtendedSortedSectionsInSetsIdx++; - if (numExtendedSortedSectionsInSetsIdx >= MAX_HCR_SETS) { - return; - } - counter = 0; - cwSum = 0; - } - } - pNumExSortSecInSets[numExtendedSortedSectionsInSetsIdx] = counter; /* save last entry for the last - probably shorter - set */ -} - - -/*--------------------------------------------------------------------------------------------- - description: This function decodes all priority codewords (PCWs) in a spectrum (within - set 0). The calculation of the PCWs is managed in two loops. The - loopcounter of the outer loop is set to the first value pointer - pNumExtendedSortedSectionsInSets points to. This value represents the - number of extended sorted sections within set 0. - The loopcounter of the inner loop is set to the first value pointer - pNumExtendedSortedCodewordInSectin points to. The value represents the - number of extended sorted codewords in sections (the original sections have - been splitted to go along with the borders of the sets). - Each time the number of the extended sorted codewords in sections are de- - coded, the pointer 'pNumExtendedSortedCodewordInSectin' is incremented by - one. --------------------------------------------------------------------------------------------- */ -static void DecodePCWs(HANDLE_FDK_BITSTREAM bs, H_HCR_INFO pHcr) -{ - UINT i; - USHORT extSortSec; - USHORT curExtSortCwInSec; - UCHAR codebook; - UCHAR dimension; - const UINT *pCurrentTree; - const SCHAR *pQuantValBase; - const SCHAR *pQuantVal; - - USHORT *pNumExtendedSortedCodewordInSection = pHcr->sectionInfo.pNumExtendedSortedCodewordInSection; - int numExtendedSortedCodewordInSectionIdx = pHcr->sectionInfo.numExtendedSortedCodewordInSectionIdx; - UCHAR *pExtendedSortedCodebook = pHcr->sectionInfo.pExtendedSortedCodebook; - int extendedSortedCodebookIdx = pHcr->sectionInfo.extendedSortedCodebookIdx; - USHORT *pNumExtendedSortedSectionsInSets = pHcr->sectionInfo.pNumExtendedSortedSectionsInSets; - int numExtendedSortedSectionsInSetsIdx = pHcr->sectionInfo.numExtendedSortedSectionsInSetsIdx; - FIXP_DBL *pQuantizedSpectralCoefficients = SPEC_LONG(pHcr->decInOut.pQuantizedSpectralCoefficientsBase); - int quantizedSpectralCoefficientsIdx = pHcr->decInOut.quantizedSpectralCoefficientsIdx; - USHORT *pLeftStartOfSegment = pHcr->segmentInfo.pLeftStartOfSegment; - SCHAR *pRemainingBitsInSegment = pHcr->segmentInfo.pRemainingBitsInSegment; -#if DETECT_TOO_LONG_CW_READS - UCHAR *pMaxLenOfCbInExtSrtSec = pHcr->sectionInfo.pMaxLenOfCbInExtSrtSec; - int maxLenOfCbInExtSrtSecIdx = pHcr->sectionInfo.maxLenOfCbInExtSrtSecIdx; - UCHAR maxAllowedCwLen; - int numDecodedBits; -#endif - const UCHAR *pCbDimension = pHcr->tableInfo.pCbDimension; - const UCHAR *pCbSign = pHcr->tableInfo.pCbSign; - - /* clear result array */ - //pQSC = &pQuantizedSpectralCoefficients[quantizedSpectralCoefficientsIdx]; - //pQSC = *pQuantizedSpectralCoefficients; - - FDKmemclear(pQuantizedSpectralCoefficients+quantizedSpectralCoefficientsIdx,1024*sizeof(FIXP_DBL)); - - /* decode all PCWs in the extended sorted section(s) belonging to set 0 */ - for ( extSortSec = pNumExtendedSortedSectionsInSets[numExtendedSortedSectionsInSetsIdx]; extSortSec != 0; extSortSec-- ) { - - codebook = pExtendedSortedCodebook[extendedSortedCodebookIdx]; /* get codebook for this extended sorted section and increment ptr to cb of next ext. sort sec */ - extendedSortedCodebookIdx++; - if (extendedSortedCodebookIdx >= (MAX_SFB_HCR+MAX_HCR_SETS)) { - return; - } - dimension = pCbDimension[codebook]; /* get dimension of codebook of this extended sort. sec. */ - pCurrentTree = aHuffTable [codebook]; /* convert codebook to pointer to QSCs */ - pQuantValBase = aQuantTable [codebook]; /* convert codebook to index to table of QSCs */ -#if DETECT_TOO_LONG_CW_READS - maxAllowedCwLen = pMaxLenOfCbInExtSrtSec[maxLenOfCbInExtSrtSecIdx]; - maxLenOfCbInExtSrtSecIdx++; - if (maxLenOfCbInExtSrtSecIdx >= (MAX_SFB_HCR+MAX_HCR_SETS)) { - return; - } -#endif - - /* switch for decoding with different codebooks: */ - if ( pCbSign[codebook] == 0 ) { /* no sign bits follow after the codeword-body */ - /* PCW_BodyONLY */ - /*==============*/ - - for ( curExtSortCwInSec = pNumExtendedSortedCodewordInSection[numExtendedSortedCodewordInSectionIdx] ; curExtSortCwInSec != 0; curExtSortCwInSec--) { - numDecodedBits = 0; - - /* decode PCW_BODY */ - pQuantVal = DecodePCW_Body(bs, - pCurrentTree, - pQuantValBase, - pLeftStartOfSegment, - pRemainingBitsInSegment, - &numDecodedBits - ); - - /* result is written out here because NO sign bits follow the body */ - for( i=dimension; i != 0 ; i-- ) { - pQuantizedSpectralCoefficients[quantizedSpectralCoefficientsIdx] = (FIXP_DBL) *pQuantVal++; /* write quant. spec. coef. into spectrum; sign is already valid */ - quantizedSpectralCoefficientsIdx++; - if (quantizedSpectralCoefficientsIdx >= 1024) { - return; - } - } - - /* one more PCW should be decoded */ - -#if DETECT_TOO_LONG_CW_READS - if ( maxAllowedCwLen < (numDecodedBits + ERROR_PCW_BODY_ONLY_TOO_LONG) ) { - pHcr->decInOut.errorLog |= TOO_MANY_PCW_BODY_BITS_DECODED; - } -#endif - -#if CHECK_SEGMENTATION_IMMEDIATELY - if (1 == errDetectPcwSegmentation(*pRemainingBitsInSegment-ERROR_PCW_BODY,pHcr,PCW_BODY,pQuantizedSpectralCoefficients+quantizedSpectralCoefficientsIdx-dimension,dimension)) { - return; - } -#endif - pLeftStartOfSegment++; /* update pointer for decoding the next PCW */ - pRemainingBitsInSegment++; /* update pointer for decoding the next PCW */ - } - } - else if (( pCbSign[codebook] == 1 ) && ( codebook < 11 )) { /* possibly there follow 1,2,3 or 4 sign bits after the codeword-body */ - /* PCW_Body and PCW_Sign */ - /*=======================*/ - - for ( curExtSortCwInSec = pNumExtendedSortedCodewordInSection[numExtendedSortedCodewordInSectionIdx] ; curExtSortCwInSec != 0; curExtSortCwInSec--) - { - int err; - numDecodedBits = 0; - - pQuantVal = DecodePCW_Body(bs, - pCurrentTree, - pQuantValBase, - pLeftStartOfSegment, - pRemainingBitsInSegment, - &numDecodedBits - ); - - err = DecodePCW_Sign( bs, - dimension, - pQuantVal, - pQuantizedSpectralCoefficients, - &quantizedSpectralCoefficientsIdx, - pLeftStartOfSegment, - pRemainingBitsInSegment, - &numDecodedBits - ); - if (err != 0) { - return; - } - /* one more PCW should be decoded */ - -#if DETECT_TOO_LONG_CW_READS - if ( maxAllowedCwLen < (numDecodedBits + ERROR_PCW_BODY_SIGN_TOO_LONG) ) { - pHcr->decInOut.errorLog |= TOO_MANY_PCW_BODY_SIGN_BITS_DECODED; - } -#endif - -#if CHECK_SEGMENTATION_IMMEDIATELY - if (1 == errDetectPcwSegmentation(*pRemainingBitsInSegment-ERROR_PCW_BODY_SIGN,pHcr,PCW_BODY_SIGN, pQuantizedSpectralCoefficients+quantizedSpectralCoefficientsIdx-dimension,dimension)) { - return; - } -#endif - pLeftStartOfSegment++; - pRemainingBitsInSegment++; - } - } - else if (( pCbSign[codebook] == 1 ) && ( codebook >= 11 )) { /* possibly there follow some sign bits and maybe one or two escape sequences after the cw-body */ - /* PCW_Body, PCW_Sign and maybe PCW_Escape */ - /*=========================================*/ - - for ( curExtSortCwInSec = pNumExtendedSortedCodewordInSection[numExtendedSortedCodewordInSectionIdx] ; curExtSortCwInSec != 0; curExtSortCwInSec--) - { - int err; - numDecodedBits = 0; - - /* decode PCW_BODY */ - pQuantVal = DecodePCW_Body(bs, - pCurrentTree, - pQuantValBase, - pLeftStartOfSegment, - pRemainingBitsInSegment, - &numDecodedBits - ); - - err = DecodePCW_Sign( bs, - dimension, - pQuantVal, - pQuantizedSpectralCoefficients, - &quantizedSpectralCoefficientsIdx, - pLeftStartOfSegment, - pRemainingBitsInSegment, - &numDecodedBits - ); - if (err != 0) { - return; - } - - /* decode PCW_ESCAPE if present */ - quantizedSpectralCoefficientsIdx -= DIMENSION_OF_ESCAPE_CODEBOOK; - - if ( fixp_abs(pQuantizedSpectralCoefficients[quantizedSpectralCoefficientsIdx]) == (FIXP_DBL)ESCAPE_VALUE ) { - pQuantizedSpectralCoefficients[quantizedSpectralCoefficientsIdx] = (FIXP_DBL) DecodeEscapeSequence( bs, - pQuantizedSpectralCoefficients[quantizedSpectralCoefficientsIdx], - pLeftStartOfSegment, - pRemainingBitsInSegment, - &numDecodedBits - ); - } - quantizedSpectralCoefficientsIdx++; - if (quantizedSpectralCoefficientsIdx >= 1024) { - return; - } - - if ( fixp_abs(pQuantizedSpectralCoefficients[quantizedSpectralCoefficientsIdx]) == (FIXP_DBL)ESCAPE_VALUE ) { - pQuantizedSpectralCoefficients[quantizedSpectralCoefficientsIdx] = (FIXP_DBL) DecodeEscapeSequence( bs, - pQuantizedSpectralCoefficients[quantizedSpectralCoefficientsIdx], - pLeftStartOfSegment, - pRemainingBitsInSegment, - &numDecodedBits - ); - } - quantizedSpectralCoefficientsIdx++; - if (quantizedSpectralCoefficientsIdx >= 1024) { - return; - } - - /* one more PCW should be decoded */ - -#if DETECT_TOO_LONG_CW_READS - if ( maxAllowedCwLen < (numDecodedBits + ERROR_PCW_BODY_SIGN_ESC_TOO_LONG) ) { - pHcr->decInOut.errorLog |= TOO_MANY_PCW_BODY_SIGN_ESC_BITS_DECODED; - } -#endif - -#if CHECK_SEGMENTATION_IMMEDIATELY - if (1 == errDetectPcwSegmentation(*pRemainingBitsInSegment-ERROR_PCW_BODY_SIGN_ESC,pHcr,PCW_BODY_SIGN_ESC,pQuantizedSpectralCoefficients+quantizedSpectralCoefficientsIdx-DIMENSION_OF_ESCAPE_CODEBOOK,DIMENSION_OF_ESCAPE_CODEBOOK)) { - return; - } -#endif - pLeftStartOfSegment++; - pRemainingBitsInSegment++; - } - } - - /* all PCWs belonging to this extended section should be decoded */ - numExtendedSortedCodewordInSectionIdx++; - if (numExtendedSortedCodewordInSectionIdx >= MAX_SFB_HCR+MAX_HCR_SETS) { - return; - } - } - /* all PCWs should be decoded */ - - numExtendedSortedSectionsInSetsIdx++; - if (numExtendedSortedSectionsInSetsIdx >= MAX_HCR_SETS) { - return; - } - - /* Write back indexes into structure */ - pHcr->sectionInfo.numExtendedSortedCodewordInSectionIdx = numExtendedSortedCodewordInSectionIdx; - pHcr->sectionInfo.extendedSortedCodebookIdx = extendedSortedCodebookIdx; - pHcr->sectionInfo.numExtendedSortedSectionsInSetsIdx = numExtendedSortedSectionsInSetsIdx; - pHcr->decInOut.quantizedSpectralCoefficientsIdx = quantizedSpectralCoefficientsIdx; - pHcr->sectionInfo.maxLenOfCbInExtSrtSecIdx = maxLenOfCbInExtSrtSecIdx; -} - -#if CHECK_SEGMENTATION_IMMEDIATELY -/*--------------------------------------------------------------------------------------------- - description: This function checks immediately after every decoded PCW, whether out of - the current segment too many bits have been read or not. If an error occurrs, - probably the sideinfo or the HCR-bitstream block holding the huffman - encoded quantized spectral coefficients is distorted. In this case the two - or four quantized spectral coefficients belonging to the current codeword - are marked (for being detected by concealment later). --------------------------------------------------------------------------------------------- */ -static UCHAR errDetectPcwSegmentation(SCHAR remainingBitsInSegment, - H_HCR_INFO pHcr, - PCW_TYPE kind, - FIXP_DBL *qsc_base_of_cw, - UCHAR dimension) -{ - SCHAR i; - if ( remainingBitsInSegment < 0 ) { - /* log the error */ - switch (kind) { - case PCW_BODY: - pHcr->decInOut.errorLog |= SEGMENT_OVERRIDE_ERR_PCW_BODY; - break; - case PCW_BODY_SIGN: - pHcr->decInOut.errorLog |= SEGMENT_OVERRIDE_ERR_PCW_BODY_SIGN; - break; - case PCW_BODY_SIGN_ESC: - pHcr->decInOut.errorLog |= SEGMENT_OVERRIDE_ERR_PCW_BODY_SIGN_ESC; - break; - } - /* mark the erred lines */ - for ( i = dimension; i != 0; i-- ) { - *qsc_base_of_cw++ = (FIXP_DBL) Q_VALUE_INVALID; - } - return 1; - } - return 0; -} -#endif - -#if CHECK_SEGMENTATION_FINAL -/*--------------------------------------------------------------------------------------------- - description: This function checks if all segments are empty after decoding. There - are _no lines markded_ as invalid because it could not be traced back - where from the remaining bits are. --------------------------------------------------------------------------------------------- */ -static void errDetectWithinSegmentationFinal(H_HCR_INFO pHcr) -{ - UCHAR segmentationErrorFlag = 0; - USHORT i; - SCHAR *pRemainingBitsInSegment = pHcr->segmentInfo.pRemainingBitsInSegment; - UINT numSegment = pHcr->segmentInfo.numSegment; - - for ( i=numSegment; i != 0 ; i--) { - if (*pRemainingBitsInSegment++ != 0) { - segmentationErrorFlag = 1; - } - } - if (segmentationErrorFlag == 1) { - pHcr->decInOut.errorLog |= BIT_IN_SEGMENTATION_ERROR; - } -} -#endif - -/*--------------------------------------------------------------------------------------------- - description: This function walks one step within the decoding tree. Which branch is - taken depends on the decoded carryBit input parameter. --------------------------------------------------------------------------------------------- */ -void CarryBitToBranchValue(UCHAR carryBit, - UINT treeNode, - UINT *branchValue, - UINT *branchNode) -{ - if (carryBit == 0) { - *branchNode = (treeNode & MASK_LEFT) >> LEFT_OFFSET; /* MASK_LEFT: 00FFF000 */ - } - else { - *branchNode = treeNode & MASK_RIGHT; /* MASK_RIGHT: 00000FFF */ - } - - *branchValue = *branchNode & CLR_BIT_10; /* clear bit 10 (if set) */ -} - - -/*--------------------------------------------------------------------------------------------- - description: Decodes the body of a priority codeword (PCW) ------------------------------------------------------------------------------------------------ - return: - return value is pointer to first of two or four quantized spectral - coefficients --------------------------------------------------------------------------------------------- */ -static const SCHAR *DecodePCW_Body(HANDLE_FDK_BITSTREAM bs, - const UINT *pCurrentTree, - const SCHAR *pQuantValBase, - USHORT *pLeftStartOfSegment, - SCHAR *pRemainingBitsInSegment, - int *pNumDecodedBits - ) -{ - UCHAR carryBit; - UINT branchNode; - UINT treeNode; - UINT branchValue; - const SCHAR *pQuantVal; - - /* decode PCW_BODY */ - treeNode = *pCurrentTree; /* get first node of current tree belonging to current codebook */ - - /* decode whole PCW-codeword-body */ - while (1) { - - carryBit = HcrGetABitFromBitstream(bs, - pLeftStartOfSegment, - pLeftStartOfSegment, /* dummy */ - FROM_LEFT_TO_RIGHT); - *pRemainingBitsInSegment -= 1; - *pNumDecodedBits += 1; - - CarryBitToBranchValue(carryBit, - treeNode, - &branchValue, - &branchNode); - - if ((branchNode & TEST_BIT_10) == TEST_BIT_10) { /* test bit 10 ; if set --> codeword-body is complete */ - break; /* end of branch in tree reached i.e. a whole PCW-Body is decoded */ - } - else { - treeNode = *(pCurrentTree + branchValue); /* update treeNode for further step in decoding tree */ - } - - } - - pQuantVal = pQuantValBase + branchValue; /* update pointer to valid first of 2 or 4 quantized values */ - - return pQuantVal; -} - - -/*--------------------------------------------------------------------------------------------- - description: This function decodes one escape sequence. In case of a escape codebook - and in case of the absolute value of the quantized spectral value == 16, - a escapeSequence is decoded in two steps: - 1. escape prefix - 2. escape word --------------------------------------------------------------------------------------------- */ - -static INT DecodeEscapeSequence(HANDLE_FDK_BITSTREAM bs, - INT quantSpecCoef, - USHORT *pLeftStartOfSegment, - SCHAR *pRemainingBitsInSegment, - int *pNumDecodedBits - ) -{ - UINT i; - INT sign; - UINT escapeOnesCounter = 0; - UINT carryBit; - INT escape_word = 0; - - /* decode escape prefix */ - while (1) { - carryBit = HcrGetABitFromBitstream(bs, - pLeftStartOfSegment, - pLeftStartOfSegment, /* dummy */ - FROM_LEFT_TO_RIGHT); - *pRemainingBitsInSegment -= 1; - *pNumDecodedBits += 1; - - if (carryBit != 0) { - escapeOnesCounter += 1; - } - else { - escapeOnesCounter += 4; - break; - } - } - - /* decode escape word */ - for( i=escapeOnesCounter; i != 0 ; i-- ) { - carryBit = HcrGetABitFromBitstream(bs, - pLeftStartOfSegment, - pLeftStartOfSegment, /* dummy */ - FROM_LEFT_TO_RIGHT); - *pRemainingBitsInSegment -= 1; - *pNumDecodedBits += 1; - - escape_word <<= 1; - escape_word = escape_word | carryBit; - } - - sign = (quantSpecCoef >= 0) ? 1 : -1; - - quantSpecCoef = sign * (((INT ) 1 << escapeOnesCounter) + escape_word); - - return quantSpecCoef; -} - - -/*--------------------------------------------------------------------------------------------- - description: Decodes the Signbits of a priority codeword (PCW) and writes out the - resulting quantized spectral values into unsorted sections ------------------------------------------------------------------------------------------------ - output: - two or four lines at position in corresponding section (which are not - located at the desired position, i.e. they must be reordered in the last - of eight function of HCR) ------------------------------------------------------------------------------------------------ - return: - updated pQuantSpecCoef pointer (to next empty storage for a line) --------------------------------------------------------------------------------------------- */ -static int DecodePCW_Sign(HANDLE_FDK_BITSTREAM bs, - UINT codebookDim, - const SCHAR *pQuantVal, - FIXP_DBL *pQuantSpecCoef, - int *quantSpecCoefIdx, - USHORT *pLeftStartOfSegment, - SCHAR *pRemainingBitsInSegment, - int *pNumDecodedBits - ) -{ - UINT i; - UINT carryBit; - INT quantSpecCoef; - - for( i=codebookDim; i != 0 ; i-- ) { - quantSpecCoef = *pQuantVal++; - if (quantSpecCoef != 0) { - carryBit = HcrGetABitFromBitstream(bs, - pLeftStartOfSegment, - pLeftStartOfSegment, /* dummy */ - FROM_LEFT_TO_RIGHT); - *pRemainingBitsInSegment -= 1; - *pNumDecodedBits += 1; - if (*pRemainingBitsInSegment < 0 || *pNumDecodedBits >= (1024>>1)) { - return -1; - } - - /* adapt sign of values according to the decoded sign bit */ - if (carryBit != 0) { - pQuantSpecCoef[*quantSpecCoefIdx] = -(FIXP_DBL)quantSpecCoef; - } - else { - pQuantSpecCoef[*quantSpecCoefIdx] = (FIXP_DBL)quantSpecCoef; - } - } - else { - pQuantSpecCoef[*quantSpecCoefIdx] = FL2FXCONST_DBL(0.0f); - } - *quantSpecCoefIdx += 1 ; - if (*quantSpecCoefIdx >= 1024) { - return -1; - } - } - return 0; -} - - -/*--------------------------------------------------------------------------------------------- - description: Mutes spectral lines which have been marked as erroneous (Q_VALUE_INVALID) --------------------------------------------------------------------------------------------- */ -void HcrMuteErroneousLines(H_HCR_INFO hHcr) -{ - int c; - FIXP_DBL *RESTRICT pLong = SPEC_LONG(hHcr->decInOut.pQuantizedSpectralCoefficientsBase); - - /* if there is a line with value Q_VALUE_INVALID mute it */ - for (c = 0; c < 1024; c++) { - if (pLong[c] == (FIXP_DBL)Q_VALUE_INVALID) { -#if HCR_LISTEN_TO_MUTED_LINES - pLong[c] = (FIXP_DBL)HCR_DIRAC; /* marking */ -#else - pLong[c] = FL2FXCONST_DBL(0.0f); /* muting */ -#endif - } - } -} - - -/*--------------------------------------------------------------------------------------------- - description: Sets global HCR type --------------------------------------------------------------------------------------------- */ -void setHcrType(H_HCR_INFO hHcr, MP4_ELEMENT_ID type) -{ - switch (type) { - case ID_SCE: - hHcr->globalHcrType = 0; - break; - case ID_CPE: - hHcr->globalHcrType = 1; - break; - default: - break; - } -} - - -/*--------------------------------------------------------------------------------------------- - description: Gets HCR type from the HCR data structure ------------------------------------------------------------------------------------------------ - return: - global HCR type --------------------------------------------------------------------------------------------- */ -INT getHcrType(H_HCR_INFO hHcr) -{ - return hHcr->globalHcrType; -} - - - - diff --git a/libAACdec/src/aacdec_hcr.h b/libAACdec/src/aacdec_hcr.h deleted file mode 100644 index 6fc527b..0000000 --- a/libAACdec/src/aacdec_hcr.h +++ /dev/null @@ -1,126 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/***************************** MPEG-4 AAC Decoder *************************** - - Author(s): Robert Weidner (DSP Solutions) - Description: HCR Decoder: Interface function declaration; common defines - and structures; defines for switching error-generator, - -detector, and -concealment - -*******************************************************************************/ - -#ifndef _AACDEC_HCR_H_ -#define _AACDEC_HCR_H_ - - - -#include "channelinfo.h" -#include "FDK_bitstream.h" - -void HcrInitRom (H_HCR_INFO hHcr); -UINT HcrInit(H_HCR_INFO pHcr, - CAacDecoderChannelInfo *pAacDecoderChannelInfo, - const SamplingRateInfo *pSamplingRateInfo, - HANDLE_FDK_BITSTREAM bs); -UINT HcrDecoder (H_HCR_INFO hHcr, - CAacDecoderChannelInfo *pAacDecoderChannelInfo, - const SamplingRateInfo *pSamplingRateInfo, - HANDLE_FDK_BITSTREAM bs); -void CarryBitToBranchValue( - UCHAR carryBit, - UINT treeNode, - UINT *branchValue, - UINT *branchNode - ); - -void CHcr_Read (HANDLE_FDK_BITSTREAM bs, - CAacDecoderChannelInfo *pAacDecoderChannelInfo); -void HcrMuteErroneousLines(H_HCR_INFO hHcr); - -void setHcrType(H_HCR_INFO hHcr, MP4_ELEMENT_ID type); -INT getHcrType(H_HCR_INFO hHcr); - - - -#endif /* _AACDEC_HCR_H_ */ diff --git a/libAACdec/src/aacdec_hcr_bit.cpp b/libAACdec/src/aacdec_hcr_bit.cpp deleted file mode 100644 index df2685b..0000000 --- a/libAACdec/src/aacdec_hcr_bit.cpp +++ /dev/null @@ -1,165 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/***************************** MPEG-4 AAC Decoder *************************** - - Author(s): Robert Weidner (DSP Solutions) - Description: HCR Decoder: Bitstream reading - -*******************************************************************************/ - -#include "aacdec_hcr_bit.h" - - -/*--------------------------------------------------------------------------------------------- - description: This function toggles the read direction. ------------------------------------------------------------------------------------------------ - input: current read direction ------------------------------------------------------------------------------------------------ - return: new read direction --------------------------------------------------------------------------------------------- */ -UCHAR ToggleReadDirection(UCHAR readDirection) -{ - if ( readDirection == FROM_LEFT_TO_RIGHT ) { - return FROM_RIGHT_TO_LEFT; - } - else { - return FROM_LEFT_TO_RIGHT; - } -} - - -/*--------------------------------------------------------------------------------------------- - description: This function returns a bit from the bitstream according to read direction. - It is called very often, therefore it makes sense to inline it (runtime). ------------------------------------------------------------------------------------------------ - input: - handle to FDK bitstream - - reference value marking start of bitfield - - pLeftStartOfSegment - - pRightStartOfSegment - - readDirection ------------------------------------------------------------------------------------------------ - return: - bit from bitstream --------------------------------------------------------------------------------------------- */ -UINT HcrGetABitFromBitstream(HANDLE_FDK_BITSTREAM bs, - USHORT *pLeftStartOfSegment, - USHORT *pRightStartOfSegment, - UCHAR readDirection) -{ - UINT bit; - INT readBitOffset; - - if (readDirection == FROM_LEFT_TO_RIGHT) { - readBitOffset = *pLeftStartOfSegment-FDKgetBitCnt(bs); - if( readBitOffset ) { - FDKpushBiDirectional(bs, readBitOffset); - } - - bit = FDKreadBits(bs, 1); - - *pLeftStartOfSegment += 1; - } - else { - readBitOffset = *pRightStartOfSegment-FDKgetBitCnt(bs); - if( readBitOffset ) { - FDKpushBiDirectional(bs, readBitOffset); - } - - /* to be replaced with a brother function of FDKreadBits() */ - bit = FDKreadBits(bs, 1); - FDKpushBack(bs, 2); - - *pRightStartOfSegment -= 1; - } - - -#if ERROR_GENERATOR_BIT_STREAM_HCR - static int a; - if ((++a % MODULO_DIVISOR_HCR) == 0) { - bit = (bit == 0) ? 1 : 0; - } -#endif - - return (bit); -} - diff --git a/libAACdec/src/aacdec_hcr_bit.h b/libAACdec/src/aacdec_hcr_bit.h deleted file mode 100644 index 8994ff1..0000000 --- a/libAACdec/src/aacdec_hcr_bit.h +++ /dev/null @@ -1,106 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/***************************** MPEG-4 AAC Decoder *************************** - - Author(s): Robert Weidner (DSP Solutions) - Description: HCR Decoder: Bitstream reading prototypes - -*******************************************************************************/ - -#ifndef _AACDEC_HCR_BIT_H_ -#define _AACDEC_HCR_BIT_H_ - - - -#include "aacdec_hcr.h" - -UCHAR ToggleReadDirection(UCHAR readDirection); - -UINT HcrGetABitFromBitstream(HANDLE_FDK_BITSTREAM bs, - USHORT *pLeftStartOfSegment, - USHORT *pRightStartOfSegment, - UCHAR readDirection); - - -#endif /* _AACDEC_HCR_BIT_H_ */ diff --git a/libAACdec/src/aacdec_hcr_types.h b/libAACdec/src/aacdec_hcr_types.h deleted file mode 100644 index 323ec4e..0000000 --- a/libAACdec/src/aacdec_hcr_types.h +++ /dev/null @@ -1,366 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/***************************** MPEG-4 AAC Decoder *************************** - - Author(s): Robert Weidner (DSP Solutions) - Description: HCR Decoder: Common defines and structures; defines for - switching error-generator, -detector, and -concealment; - -*******************************************************************************/ - -#ifndef _AACDEC_HCR_TYPES_H_ -#define _AACDEC_HCR_TYPES_H_ - - - -#include "FDK_bitstream.h" -#include "overlapadd.h" - -/* ------------------------------------------------ */ -/* ------------------------------------------------ */ - -#define LINES_PER_UNIT 4 - -/* ------------------------------------------------ */ -/* ------------------------------------------------ */ -/* ----------- basic HCR configuration ------------ */ - - - #define MAX_SFB_HCR (((1024/8) / LINES_PER_UNIT) * 8) /* (8 * 16) is not enough because sfbs are split in units for blocktype short */ - #define NUMBER_OF_UNIT_GROUPS (LINES_PER_UNIT * 8) - #define LINES_PER_UNIT_GROUP (1024 / NUMBER_OF_UNIT_GROUPS) /* 15 16 30 32 */ - - -/* ------------------------------------------------ */ -/* ------------------------------------------------ */ -/* ------------------------------------------------ */ - -#define FROM_LEFT_TO_RIGHT 0 -#define FROM_RIGHT_TO_LEFT 1 - -#define MAX_CB_PAIRS 23 -#define MAX_HCR_SETS 14 - -#define ESCAPE_VALUE 16 -#define POSITION_OF_FLAG_A 21 -#define POSITION_OF_FLAG_B 20 - -#define MAX_CB 32 /* last used CB is cb #31 when VCB11 is used */ - -#define MAX_CB_CHECK 32 /* support for VCB11 available -- is more general, could therefore used in both cases */ - -#define NUMBER_OF_BIT_IN_WORD 32 - -/* log */ -#define THIRTYTWO_LOG_DIV_TWO_LOG 5 -#define EIGHT_LOG_DIV_TWO_LOG 3 -#define FOUR_LOG_DIV_TWO_LOG 2 - -/* borders */ -#define CPE_TOP_LENGTH 12288 -#define SCE_TOP_LENGTH 6144 -#define LEN_OF_LONGEST_CW_TOP_LENGTH 49 - -/* qsc's of high level */ -#define Q_VALUE_INVALID 8192 /* mark a invalid line with this value (to be concealed later on) */ -#define HCR_DIRAC 500 /* a line of high level */ - -/* masks */ -#define MASK_LEFT 0xFFF000 -#define MASK_RIGHT 0xFFF -#define CLR_BIT_10 0x3FF -#define TEST_BIT_10 0x400 - -#define LEFT_OFFSET 12 - -/* when set HCR is replaced by a dummy-module which just fills the outputbuffer with a dirac sequence */ -/* use this if HCR is suspected to write in other modules -- if error is stell there, HCR is innocent */ -#define USE_HCR_DUMMY 0 - - -/* ------------------------------ */ -/* - insert HCR errors - */ -/* ------------------------------ */ - - /* modify input lengths -- high protected */ -#define ERROR_LORSD 0 /* offset: error if different from zero */ -#define ERROR_LOLC 0 /* offset: error if different from zero */ - - /* segments are earlier empty as expected when decoding PCWs */ -#define ERROR_PCW_BODY 0 /* set a positive values to trigger the error (make segments earlyer appear to be empty) */ -#define ERROR_PCW_BODY_SIGN 0 /* set a positive values to trigger the error (make segments earlyer appear to be empty) */ -#define ERROR_PCW_BODY_SIGN_ESC 0 /* set a positive values to trigger the error (make segments earlyer appear to be empty) */ - - /* pretend there are too many bits decoded (enlarge length of codeword) at PCWs -- use a positive value */ -#define ERROR_PCW_BODY_ONLY_TOO_LONG 0 /* set a positive values to trigger the error */ -#define ERROR_PCW_BODY_SIGN_TOO_LONG 0 /* set a positive values to trigger the error */ -#define ERROR_PCW_BODY_SIGN_ESC_TOO_LONG 0 /* set a positive values to trigger the error */ - - /* modify HCR bitstream block */ -#define ERROR_GENERATOR_BIT_STREAM_HCR 0 /* modify every -bit when reading from bitstream */ /* !!! BEWARE!!! if RVLC is active, also RVLC data at ESC2 will be modified !!! */ -#define MODULO_DIVISOR_HCR 30 - - -/* ------------------------------ */ -/* - detect HCR errors - */ -/* ------------------------------ */ - /* check input data */ -#define CHECK_VALID_HCR_INPUT 1 /* it is highly recommended to check input data */ - - /* during decoding */ -#define CHECK_SEGMENTATION_IMMEDIATELY 1 /* the 2 or 4 lines of a detected PCW-decoding-error is marked */ - -#define CHECK_SEGMENTATION_FINAL 1 /* all the segments are checked -- therefore -- if this check passes, its a kind of evidence that the - decoded PCWs and non-PCWs are fine */ - -#define DETECT_TOO_LONG_CW_READS 1 /* if a codeword is decoded there exists a border for the number of bits, which are allowed to read for this - codeword. This border is the minimum of the length of the longest codeword (for the currently used - codebook) and the separately transmitted 'lengthOfLongestCodeword' in this frame and channel. The number - of decoded bits is counted (for PCWs only -- there it makes really sense in my opinion). If this number - exceeds the border (derived as minimum -- see above), a error is detected. */ - -#define STATE_MACHINE_ERROR_CHECK 1 /* test if the number of remaining bits in a segment is _below_ zero. If there are no errors the lowest - allowed value for remainingBitsInSegment is zero. This check also could be set to zero (save runtime) */ - /* other */ -#define VALID_LAV_ERROR_TRIGGER 1 /* when set to '1', avoid setting the LAV-Flag in errorLog due to a previous-line-marking (at PCW decoder). A little - more runtime is needed then when writing values out into output-buffer. */ - -#define HCR_LISTEN_TO_MUTED_LINES 0 /* listen to the "error-concealment" for testing */ - -/* ------------------------------ */ -/* - conceal HCR errors - */ -/* ------------------------------ */ - -#define HCR_ERROR_CONCEALMENT 1 /* if set to '1', HCR _mutes_ the erred quantized spectral coefficients */ - - -// ------------------------------------------------------------------------------------------------------------------ -// errorLog: A word of 32 bits used for logging possible errors within HCR -// in case of distorted bitstreams. Table of all known errors: -// ------------------------------------------------------------------------------------------------------------------------ - // bit fatal location meaning - // ----+-----+-----------+-------------------------------------- -#define SEGMENT_OVERRIDE_ERR_PCW_BODY 0x80000000 // 31 no PCW-Dec During PCW decoding it is checked after every PCW if there are too many bits decoded (immediate check). -#define SEGMENT_OVERRIDE_ERR_PCW_BODY_SIGN 0x40000000 // 30 no PCW-Dec During PCW decoding it is checked after every PCW if there are too many bits decoded (immediate check). -#define SEGMENT_OVERRIDE_ERR_PCW_BODY_SIGN_ESC 0x20000000 // 29 no PCW-Dec During PCW decoding it is checked after every PCW if there are too many bits decoded (immediate check). -#define EXTENDED_SORTED_COUNTER_OVERFLOW 0x10000000 // 28 yes Init-Dec Error during extending sideinfo (neither a PCW nor a nonPCW was decoded so far) - // 0x08000000 // 27 reserved - // 0x04000000 // 26 reserved - // 0x02000000 // 25 reserved - // 0x01000000 // 24 reserved - // 0x00800000 // 23 reserved - // 0x00400000 // 22 reserved - // 0x00200000 // 21 reserved - // 0x00100000 // 20 reserved - - /* special errors */ -#define TOO_MANY_PCW_BODY_BITS_DECODED 0x00080000 // 19 yes PCW-Dec During PCW-body-decoding too many bits have been read from bitstream -- advice: skip non-PCW decoding -#define TOO_MANY_PCW_BODY_SIGN_BITS_DECODED 0x00040000 // 18 yes PCW-Dec During PCW-body-sign-decoding too many bits have been read from bitstream -- advice: skip non-PCW decoding -#define TOO_MANY_PCW_BODY_SIGN_ESC_BITS_DECODED 0x00020000 // 17 yes PCW-Dec During PCW-body-sign-esc-decoding too many bits have been read from bitstream -- advice: skip non-PCW decoding - - - // 0x00010000 // 16 reserved -#define STATE_ERROR_BODY_ONLY 0x00008000 // 15 no NonPCW-Dec State machine returned with error -#define STATE_ERROR_BODY_SIGN__BODY 0x00004000 // 14 no NonPCW-Dec State machine returned with error -#define STATE_ERROR_BODY_SIGN__SIGN 0x00002000 // 13 no NonPCW-Dec State machine returned with error -#define STATE_ERROR_BODY_SIGN_ESC__BODY 0x00001000 // 12 no NonPCW-Dec State machine returned with error -#define STATE_ERROR_BODY_SIGN_ESC__SIGN 0x00000800 // 11 no NonPCW-Dec State machine returned with error -#define STATE_ERROR_BODY_SIGN_ESC__ESC_PREFIX 0x00000400 // 10 no NonPCW-Dec State machine returned with error -#define STATE_ERROR_BODY_SIGN_ESC__ESC_WORD 0x00000200 // 9 no NonPCW-Dec State machine returned with error -#define HCR_SI_LENGTHS_FAILURE 0x00000100 // 8 yes Init-Dec LengthOfLongestCodeword must not be less than lenghtOfReorderedSpectralData -#define NUM_SECT_OUT_OF_RANGE_SHORT_BLOCK 0x00000080 // 7 yes Init-Dec The number of sections is not within the allowed range (short block) -#define NUM_SECT_OUT_OF_RANGE_LONG_BLOCK 0x00000040 // 6 yes Init-Dec The number of sections is not within the allowed range (long block) -#define LINE_IN_SECT_OUT_OF_RANGE_SHORT_BLOCK 0x00000020 // 5 yes Init-Dec The number of lines per section is not within the allowed range (short block) -#define CB_OUT_OF_RANGE_SHORT_BLOCK 0x00000010 // 4 yes Init-Dec The codebook is not within the allowed range (short block) -#define LINE_IN_SECT_OUT_OF_RANGE_LONG_BLOCK 0x00000008 // 3 yes Init-Dec The number of lines per section is not within the allowed range (long block) -#define CB_OUT_OF_RANGE_LONG_BLOCK 0x00000004 // 2 yes Init-Dec The codebook is not within the allowed range (long block) -#define LAV_VIOLATION 0x00000002 // 1 no Final The absolute value of at least one decoded line was too high for the according codebook. -#define BIT_IN_SEGMENTATION_ERROR 0x00000001 // 0 no Final After PCW and non-PWC-decoding at least one segment is not zero (global check). - - /*----------*/ -#define HCR_FATAL_PCW_ERROR_MASK 0x100E01FC - - -typedef enum { - PCW_BODY, - PCW_BODY_SIGN, - PCW_BODY_SIGN_ESC -} PCW_TYPE; - - -/* interface Decoder <---> HCR */ -typedef struct { - UINT errorLog; - SPECTRAL_PTR pQuantizedSpectralCoefficientsBase; - int quantizedSpectralCoefficientsIdx; - SHORT lengthOfReorderedSpectralData; - SHORT numSection; - SHORT *pNumLineInSect; - USHORT bitstreamIndex; - SCHAR lengthOfLongestCodeword; - UCHAR *pCodebook; -} HCR_INPUT_OUTPUT; - -typedef struct { - const UCHAR *pMinOfCbPair; - const UCHAR *pMaxOfCbPair; -} HCR_CB_PAIRS; - -typedef struct{ - const USHORT *pLargestAbsVal; - const UCHAR *pMaxCwLength; - const UCHAR *pCbDimension; - const UCHAR *pCbDimShift; - const UCHAR *pCbSign; - const UCHAR *pCbPriority; -} HCR_TABLE_INFO; - -typedef struct{ - UINT numSegment; - UINT pSegmentBitfield[((1024>>1)/NUMBER_OF_BIT_IN_WORD+1)]; - UINT pCodewordBitfield[((1024>>1)/NUMBER_OF_BIT_IN_WORD+1)]; - UINT segmentOffset; - FIXP_DBL pTempValues[1024]; - USHORT pLeftStartOfSegment[1024>>1]; - USHORT pRightStartOfSegment[1024>>1]; - SCHAR pRemainingBitsInSegment[1024>>1]; - UCHAR readDirection; - UCHAR numWordForBitfield; - USHORT pNumBitValidInLastWord; -} HCR_SEGMENT_INFO; - -typedef struct{ - - UINT numCodeword; - UINT numSortedSection; - USHORT pNumCodewordInSection[MAX_SFB_HCR]; - USHORT pNumSortedCodewordInSection[MAX_SFB_HCR]; - USHORT pNumExtendedSortedCodewordInSection[MAX_SFB_HCR+MAX_HCR_SETS]; - int numExtendedSortedCodewordInSectionIdx; - USHORT pNumExtendedSortedSectionsInSets[MAX_HCR_SETS]; - int numExtendedSortedSectionsInSetsIdx; - USHORT pReorderOffset[MAX_SFB_HCR]; - UCHAR pSortedCodebook[MAX_SFB_HCR]; - - UCHAR pExtendedSortedCodebook[MAX_SFB_HCR+MAX_HCR_SETS]; - int extendedSortedCodebookIdx; -#if DETECT_TOO_LONG_CW_READS - UCHAR pMaxLenOfCbInExtSrtSec[MAX_SFB_HCR+MAX_HCR_SETS]; - int maxLenOfCbInExtSrtSecIdx; -#endif - UCHAR pCodebookSwitch[MAX_SFB_HCR]; -} HCR_SECTION_INFO; - -typedef UINT (*STATEFUNC)(HANDLE_FDK_BITSTREAM, void*); - -typedef struct{ - /* worst-case and 1024/4 non-PCWs exist in worst-case */ - FIXP_DBL *pResultBase; /* Base address for spectral data output target buffer */ - UINT iNode[1024>>2]; /* Helper indices for code books */ - USHORT iResultPointer[1024>>2]; /* Helper indices for accessing pResultBase */ - UINT pEscapeSequenceInfo[1024>>2]; - UINT codewordOffset; - STATEFUNC pState; - UCHAR pCodebook[1024>>2]; - UCHAR pCntSign[1024>>2]; - /* this array holds the states coded as integer values within the range [0,1,..,7] */ - SCHAR pSta[1024>>2]; -} HCR_NON_PCW_SIDEINFO; - -typedef struct{ - HCR_INPUT_OUTPUT decInOut; - HCR_CB_PAIRS cbPairs; - HCR_TABLE_INFO tableInfo; - HCR_SEGMENT_INFO segmentInfo; - HCR_SECTION_INFO sectionInfo; - HCR_NON_PCW_SIDEINFO nonPcwSideinfo; - - INT globalHcrType; -} CErHcrInfo; - - -typedef CErHcrInfo *H_HCR_INFO; - - -#endif /* _AACDEC_HCR_TYPES_H_ */ diff --git a/libAACdec/src/aacdec_hcrs.cpp b/libAACdec/src/aacdec_hcrs.cpp deleted file mode 100644 index c0b2173..0000000 --- a/libAACdec/src/aacdec_hcrs.cpp +++ /dev/null @@ -1,1409 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/***************************** MPEG-4 AAC Decoder *************************** - - Author(s): Robert Weidner (DSP Solutions) - Description: HCR Decoder: Prepare decoding of non-PCWs, segmentation- and - bitfield-handling, HCR-Statemachine - -*******************************************************************************/ - -#include "aacdec_hcrs.h" - - -#include "aacdec_hcr.h" - -#include "aacdec_hcr_bit.h" -#include "aac_rom.h" -#include "aac_ram.h" - - -static UINT InitSegmentBitfield(UINT *pNumSegment, - SCHAR *pRemainingBitsInSegment, - UINT *pSegmentBitfield, - UCHAR *pNumWordForBitfield, - USHORT *pNumBitValidInLastWord); - -static void InitNonPCWSideInformationForCurrentSet(H_HCR_INFO pHcr); - -static INT ModuloValue(INT input, INT bufferlength); - -static void ClearBitFromBitfield(STATEFUNC *ptrState, - UINT offset, - UINT *pBitfield); - - -/*--------------------------------------------------------------------------------------------- - description: This function decodes all non-priority codewords (non-PCWs) by using a - state-machine. --------------------------------------------------------------------------------------------- */ -void DecodeNonPCWs(HANDLE_FDK_BITSTREAM bs, H_HCR_INFO pHcr) -{ - UINT numValidSegment; - INT segmentOffset; - INT codewordOffsetBase; - INT codewordOffset; - UINT trial; - - UINT *pNumSegment; - SCHAR *pRemainingBitsInSegment; - UINT *pSegmentBitfield; - UCHAR *pNumWordForBitfield; - USHORT *pNumBitValidInLastWord; - UINT *pCodewordBitfield; - INT bitfieldWord; - INT bitInWord; - UINT tempWord; - UINT interMediateWord; - INT tempBit; - INT carry; - - UINT numCodeword; - UCHAR numSet; - UCHAR currentSet; - UINT codewordInSet; - UINT remainingCodewordsInSet; - SCHAR *pSta; - UINT ret; - - pNumSegment = &(pHcr->segmentInfo.numSegment); - pRemainingBitsInSegment = pHcr->segmentInfo.pRemainingBitsInSegment; - pSegmentBitfield = pHcr->segmentInfo.pSegmentBitfield; - pNumWordForBitfield = &(pHcr->segmentInfo.numWordForBitfield); - pNumBitValidInLastWord = &(pHcr->segmentInfo.pNumBitValidInLastWord); - pSta = pHcr->nonPcwSideinfo.pSta; - - numValidSegment = InitSegmentBitfield(pNumSegment, - pRemainingBitsInSegment, - pSegmentBitfield, - pNumWordForBitfield, - pNumBitValidInLastWord); - - if ( numValidSegment != 0 ) { - numCodeword = pHcr->sectionInfo.numCodeword; - numSet = ((numCodeword - 1) / *pNumSegment) + 1; - - - pHcr->segmentInfo.readDirection = FROM_RIGHT_TO_LEFT; - - /* Process sets subsequently */ - for ( currentSet = 1; currentSet < numSet ; currentSet++ ) { - - - - /* step 1 */ - numCodeword -= *pNumSegment; /* number of remaining non PCWs [for all sets] */ - if ( numCodeword < *pNumSegment ) { - codewordInSet = numCodeword; /* for last set */ - } - else { - codewordInSet = *pNumSegment; /* for all sets except last set */ - } - - /* step 2 */ - /* prepare array 'CodewordBitfield'; as much ones are written from left in all words, as much decodedCodewordInSetCounter nonPCWs exist in this set */ - tempWord = 0xFFFFFFFF; - pCodewordBitfield = pHcr->segmentInfo.pCodewordBitfield; - - for ( bitfieldWord = *pNumWordForBitfield; bitfieldWord !=0; bitfieldWord-- ) { /* loop over all used words */ - if ( codewordInSet > NUMBER_OF_BIT_IN_WORD ) { /* more codewords than number of bits => fill ones */ - /* fill a whole word with ones */ - *pCodewordBitfield++ = tempWord; - codewordInSet -= NUMBER_OF_BIT_IN_WORD; /* subtract number of bits */ - } - else { - /* prepare last tempWord */ - for (remainingCodewordsInSet = codewordInSet; remainingCodewordsInSet < NUMBER_OF_BIT_IN_WORD ; remainingCodewordsInSet++ ) { - tempWord = tempWord & ~(1 << (NUMBER_OF_BIT_IN_WORD-1-remainingCodewordsInSet)); /* set a zero at bit number (NUMBER_OF_BIT_IN_WORD-1-i) in tempWord */ - } - *pCodewordBitfield++ = tempWord; - tempWord = 0x00000000; - } - } - pCodewordBitfield = pHcr->segmentInfo.pCodewordBitfield; - - /* step 3 */ - /* build non-PCW sideinfo for each non-PCW of the current set */ - InitNonPCWSideInformationForCurrentSet(pHcr); - - /* step 4 */ - /* decode all non-PCWs belonging to this set */ - - /* loop over trials */ - codewordOffsetBase = 0; - for ( trial = *pNumSegment; trial > 0; trial-- ) { - - /* loop over number of words in bitfields */ - segmentOffset = 0; /* start at zero in every segment */ - pHcr->segmentInfo.segmentOffset = segmentOffset; /* store in structure for states */ - codewordOffset = codewordOffsetBase; - pHcr->nonPcwSideinfo.codewordOffset = codewordOffset; /* store in structure for states */ - - for ( bitfieldWord=0; bitfieldWord < *pNumWordForBitfield; bitfieldWord++ ) { - - /* derive tempWord with bitwise and */ - tempWord = pSegmentBitfield[bitfieldWord] & pCodewordBitfield[bitfieldWord]; - - /* if tempWord is not zero, decode something */ - if ( tempWord != 0 ) { - - - /* loop over all bits in tempWord; start state machine if & is true */ - for ( bitInWord = NUMBER_OF_BIT_IN_WORD; bitInWord > 0; bitInWord-- ) { - - interMediateWord = ((UINT)1 << (bitInWord-1) ); - if ( ( tempWord & interMediateWord ) == interMediateWord ) { - - /* get state and start state machine */ - pHcr->nonPcwSideinfo.pState = aStateConstant2State[pSta[codewordOffset]]; - - while(pHcr->nonPcwSideinfo.pState) { - ret = ((STATEFUNC) pHcr->nonPcwSideinfo.pState)(bs, pHcr); -#if STATE_MACHINE_ERROR_CHECK - if ( ret != 0 ) { - return; - } -#endif - } - } - - /* update both offsets */ - segmentOffset += 1; /* add NUMBER_OF_BIT_IN_WORD times one */ - pHcr->segmentInfo.segmentOffset = segmentOffset; - codewordOffset += 1; /* add NUMBER_OF_BIT_IN_WORD times one */ - codewordOffset = ModuloValue(codewordOffset,*pNumSegment); /* index of the current codeword lies within modulo range */ - pHcr->nonPcwSideinfo.codewordOffset = codewordOffset; - } - } - else { - segmentOffset += NUMBER_OF_BIT_IN_WORD; /* add NUMBER_OF_BIT_IN_WORD at once */ - pHcr->segmentInfo.segmentOffset = segmentOffset; - codewordOffset += NUMBER_OF_BIT_IN_WORD; /* add NUMBER_OF_BIT_IN_WORD at once */ - codewordOffset = ModuloValue(codewordOffset,*pNumSegment); /* index of the current codeword lies within modulo range */ - pHcr->nonPcwSideinfo.codewordOffset = codewordOffset; - } - } /* end of bitfield word loop */ - - /* decrement codeword - pointer */ - codewordOffsetBase -= 1; - codewordOffsetBase = ModuloValue(codewordOffsetBase,*pNumSegment); /* index of the current codeword base lies within modulo range */ - - /* rotate numSegment bits in codewordBitfield */ - /* rotation of *numSegment bits in bitfield of codewords (circle-rotation) */ - /* get last valid bit */ - tempBit = pCodewordBitfield[*pNumWordForBitfield-1] & (1 << (NUMBER_OF_BIT_IN_WORD - *pNumBitValidInLastWord)); - tempBit = tempBit >> (NUMBER_OF_BIT_IN_WORD - *pNumBitValidInLastWord); - - /* write zero into place where tempBit was fetched from */ - pCodewordBitfield[*pNumWordForBitfield-1] = pCodewordBitfield[*pNumWordForBitfield-1] & ~(1 << (NUMBER_OF_BIT_IN_WORD - *pNumBitValidInLastWord)); - - /* rotate last valid word */ - pCodewordBitfield[*pNumWordForBitfield-1] = pCodewordBitfield[*pNumWordForBitfield-1] >> 1; - - /* transfare carry bit 0 from current word into bitposition 31 from next word and rotate current word */ - for ( bitfieldWord = *pNumWordForBitfield-2; bitfieldWord > -1 ; bitfieldWord-- ) { - /* get carry (=bit at position 0) from current word */ - carry = pCodewordBitfield[bitfieldWord] & 1; - - /* put the carry bit at position 31 into word right from current word */ - pCodewordBitfield[bitfieldWord+1] = pCodewordBitfield[bitfieldWord+1] | (carry << (NUMBER_OF_BIT_IN_WORD-1)); - - /* shift current word */ - pCodewordBitfield[bitfieldWord] = pCodewordBitfield[bitfieldWord] >> 1; - } - - /* put tempBit into free bit-position 31 from first word */ - pCodewordBitfield[0] = pCodewordBitfield[0] | (tempBit << (NUMBER_OF_BIT_IN_WORD-1)); - - } /* end of trial loop */ - - /* toggle read direction */ - pHcr->segmentInfo.readDirection = ToggleReadDirection(pHcr->segmentInfo.readDirection); - - } - /* end of set loop */ - - /* all non-PCWs of this spectrum are decoded */ - } - - /* all PCWs and all non PCWs are decoded. They are unbacksorted in output buffer. Here is the Interface with comparing QSCs to asm decoding */ -} - - -/*--------------------------------------------------------------------------------------------- - description: This function prepares the bitfield used for the - segments. The list is set up once to be used in all following sets. If a - segment is decoded empty, the according bit from the Bitfield is removed. ------------------------------------------------------------------------------------------------ - return: numValidSegment = the number of valid segments --------------------------------------------------------------------------------------------- */ -static UINT InitSegmentBitfield(UINT *pNumSegment, - SCHAR *pRemainingBitsInSegment, - UINT *pSegmentBitfield, - UCHAR *pNumWordForBitfield, - USHORT *pNumBitValidInLastWord) -{ - SHORT i; - USHORT r; - UCHAR bitfieldWord; - UINT tempWord; - USHORT numValidSegment; - - *pNumWordForBitfield = ((*pNumSegment-1) >> THIRTYTWO_LOG_DIV_TWO_LOG) + 1; - - /* loop over all words, which are completely used or only partial */ - /* bit in pSegmentBitfield is zero if segment is empty; bit in pSegmentBitfield is one if segment is not empty */ - numValidSegment = 0; - *pNumBitValidInLastWord = *pNumSegment; - - /* loop over words */ - for ( bitfieldWord=0; bitfieldWord < *pNumWordForBitfield - 1; bitfieldWord++ ) { - tempWord = 0xFFFFFFFF; /* set ones */ - r = bitfieldWord << THIRTYTWO_LOG_DIV_TWO_LOG; - for ( i=0; i < NUMBER_OF_BIT_IN_WORD; i++) { - if ( pRemainingBitsInSegment[r + i] == 0 ) { - tempWord = tempWord & ~(1 << (NUMBER_OF_BIT_IN_WORD-1-i)); /* set a zero at bit number (NUMBER_OF_BIT_IN_WORD-1-i) in tempWord */ - } - else { - numValidSegment += 1; /* count segments which are not empty */ - } - } - pSegmentBitfield[bitfieldWord] = tempWord; /* store result */ - *pNumBitValidInLastWord -= NUMBER_OF_BIT_IN_WORD; /* calculate number of zeros on LSB side in the last word */ - } - - - /* calculate last word: prepare special tempWord */ - tempWord = 0xFFFFFFFF; - for ( i=0; i < ( NUMBER_OF_BIT_IN_WORD - *pNumBitValidInLastWord ); i++ ) { - tempWord = tempWord & ~(1 << i); /* clear bit i in tempWord */ - } - - /* calculate last word */ - r = bitfieldWord << THIRTYTWO_LOG_DIV_TWO_LOG; - for ( i=0; i<*pNumBitValidInLastWord; i++) { - if ( pRemainingBitsInSegment[r + i] == 0 ) { - tempWord = tempWord & ~(1 << (NUMBER_OF_BIT_IN_WORD-1-i)); /* set a zero at bit number (NUMBER_OF_BIT_IN_WORD-1-i) in tempWord */ - } - else { - numValidSegment += 1; /* count segments which are not empty */ - } - } - pSegmentBitfield[bitfieldWord] = tempWord; /* store result */ - - - - return numValidSegment; -} - - -/*--------------------------------------------------------------------------------------------- - description: This function sets up sideinfo for the non-PCW decoder (for the current set). ----------------------------------------------------------------------------------------------*/ -static void InitNonPCWSideInformationForCurrentSet(H_HCR_INFO pHcr) -{ - USHORT i,k; - UCHAR codebookDim; - UINT startNode; - - UCHAR *pCodebook = pHcr->nonPcwSideinfo.pCodebook; - UINT *iNode = pHcr->nonPcwSideinfo.iNode; - UCHAR *pCntSign = pHcr->nonPcwSideinfo.pCntSign; - USHORT *iResultPointer = pHcr->nonPcwSideinfo.iResultPointer; - UINT *pEscapeSequenceInfo = pHcr->nonPcwSideinfo.pEscapeSequenceInfo; - SCHAR *pSta = pHcr->nonPcwSideinfo.pSta; - USHORT *pNumExtendedSortedCodewordInSection = pHcr->sectionInfo.pNumExtendedSortedCodewordInSection; - int numExtendedSortedCodewordInSectionIdx = pHcr->sectionInfo.numExtendedSortedCodewordInSectionIdx; - UCHAR *pExtendedSortedCodebook = pHcr->sectionInfo.pExtendedSortedCodebook; - int extendedSortedCodebookIdx = pHcr->sectionInfo.extendedSortedCodebookIdx; - USHORT *pNumExtendedSortedSectionsInSets = pHcr->sectionInfo.pNumExtendedSortedSectionsInSets; - int numExtendedSortedSectionsInSetsIdx = pHcr->sectionInfo.numExtendedSortedSectionsInSetsIdx; - FIXP_DBL *pQuantizedSpectralCoefficients = SPEC_LONG(pHcr->decInOut.pQuantizedSpectralCoefficientsBase); - int quantizedSpectralCoefficientsIdx = pHcr->decInOut.quantizedSpectralCoefficientsIdx; - const UCHAR *pCbDimension = pHcr->tableInfo.pCbDimension; - int iterationCounter = 0; - - /* loop over number of extended sorted sections in the current set so all codewords sideinfo variables within this set can be prepared for decoding */ - for ( i=pNumExtendedSortedSectionsInSets[numExtendedSortedSectionsInSetsIdx]; i != 0; i-- ) { - - codebookDim = pCbDimension[pExtendedSortedCodebook[extendedSortedCodebookIdx]]; - startNode = *aHuffTable[pExtendedSortedCodebook[extendedSortedCodebookIdx]]; - - for ( k = pNumExtendedSortedCodewordInSection[numExtendedSortedCodewordInSectionIdx]; k != 0; k-- ) { - iterationCounter++; - if (iterationCounter > (1024>>2)) { - return; - } - *pSta++ = aCodebook2StartInt[pExtendedSortedCodebook[extendedSortedCodebookIdx]]; - *pCodebook++ = pExtendedSortedCodebook[extendedSortedCodebookIdx]; - *iNode++ = startNode; - *pCntSign++ = 0; - *iResultPointer++ = quantizedSpectralCoefficientsIdx; - *pEscapeSequenceInfo++ = 0; - quantizedSpectralCoefficientsIdx += codebookDim; /* update pointer by codebookDim --> point to next starting value for writing out */ - if (quantizedSpectralCoefficientsIdx >= 1024) { - return; - } - } - numExtendedSortedCodewordInSectionIdx++; /* inc ptr for next ext sort sec in current set */ - extendedSortedCodebookIdx++; /* inc ptr for next ext sort sec in current set */ - if (numExtendedSortedCodewordInSectionIdx >= (MAX_SFB_HCR+MAX_HCR_SETS) || extendedSortedCodebookIdx >= (MAX_SFB_HCR+MAX_HCR_SETS)) { - return; - } - } - numExtendedSortedSectionsInSetsIdx++; /* inc ptr for next set of non-PCWs */ - if (numExtendedSortedCodewordInSectionIdx >= (MAX_SFB_HCR+MAX_HCR_SETS)) { - return; - } - - /* Write back indexes */ - pHcr->sectionInfo.numExtendedSortedCodewordInSectionIdx = numExtendedSortedCodewordInSectionIdx; - pHcr->sectionInfo.extendedSortedCodebookIdx = extendedSortedCodebookIdx; - pHcr->sectionInfo.numExtendedSortedSectionsInSetsIdx = numExtendedSortedSectionsInSetsIdx; - pHcr->sectionInfo.numExtendedSortedCodewordInSectionIdx = numExtendedSortedCodewordInSectionIdx; - pHcr->decInOut.quantizedSpectralCoefficientsIdx = quantizedSpectralCoefficientsIdx; -} - - -/*--------------------------------------------------------------------------------------------- - description: This function returns the input value if the value is in the - range of bufferlength. If is smaller, one bufferlength is added, - if is bigger one bufferlength is subtracted. ------------------------------------------------------------------------------------------------ - return: modulo result --------------------------------------------------------------------------------------------- */ -static INT ModuloValue(INT input, INT bufferlength) -{ - if ( input > (bufferlength - 1) ) { - return (input - bufferlength); - } - if ( input < 0 ) { - return (input + bufferlength); - } - return input; -} - - -/*--------------------------------------------------------------------------------------------- - description: This function clears a bit from current bitfield and - switches off the statemachine. - - A bit is cleared in two cases: - a) a codeword is decoded, then a bit is cleared in codeword bitfield - b) a segment is decoded empty, then a bit is cleared in segment bitfield --------------------------------------------------------------------------------------------- */ -static void ClearBitFromBitfield(STATEFUNC *ptrState, - UINT offset, - UINT *pBitfield) -{ - UINT numBitfieldWord; - UINT numBitfieldBit; - - /* get both values needed for clearing the bit */ - numBitfieldWord = offset >> THIRTYTWO_LOG_DIV_TWO_LOG; /* int = wordNr */ - numBitfieldBit = offset - (numBitfieldWord << THIRTYTWO_LOG_DIV_TWO_LOG); /* fract = bitNr */ - - /* clear a bit in bitfield */ - pBitfield[numBitfieldWord] = pBitfield[numBitfieldWord] & ~(1 << (NUMBER_OF_BIT_IN_WORD-1 - numBitfieldBit)); - - /* switch off state machine because codeword is decoded and/or because segment is empty */ - *ptrState = NULL; -} - - - -/* ========================================================================================= - the states of the statemachine - ========================================================================================= */ - - -/*--------------------------------------------------------------------------------------------- - description: Decodes the body of a codeword. This State is used for codebooks 1,2,5 and 6. - No sign bits are decoded, because the table of the quantized spectral values - has got a valid sign at the quantized spectral lines. ------------------------------------------------------------------------------------------------ - output: Two or four quantizes spectral values written at position where pResultPointr - points to ------------------------------------------------------------------------------------------------ - return: 0 --------------------------------------------------------------------------------------------- */ -UINT Hcr_State_BODY_ONLY(HANDLE_FDK_BITSTREAM bs, void *ptr) -{ - H_HCR_INFO pHcr = (H_HCR_INFO)ptr; - UINT *pSegmentBitfield; - UINT *pCodewordBitfield; - UINT segmentOffset; - FIXP_DBL *pResultBase; - UINT *iNode; - USHORT *iResultPointer; - UINT codewordOffset; - UINT branchNode; - UINT branchValue; - UINT iQSC; - UINT treeNode; - UCHAR carryBit; - USHORT *pLeftStartOfSegment; - USHORT *pRightStartOfSegment; - SCHAR *pRemainingBitsInSegment; - UCHAR readDirection; - UCHAR *pCodebook; - UCHAR dimCntr; - const UINT *pCurrentTree; - const UCHAR *pCbDimension; - const SCHAR *pQuantVal; - const SCHAR *pQuantValBase; - - pRemainingBitsInSegment = pHcr->segmentInfo.pRemainingBitsInSegment; - pLeftStartOfSegment = pHcr->segmentInfo.pLeftStartOfSegment; - pRightStartOfSegment = pHcr->segmentInfo.pRightStartOfSegment; - readDirection = pHcr->segmentInfo.readDirection; - pSegmentBitfield = pHcr->segmentInfo.pSegmentBitfield; - pCodewordBitfield = pHcr->segmentInfo.pCodewordBitfield; - segmentOffset = pHcr->segmentInfo.segmentOffset; - - pCodebook = pHcr->nonPcwSideinfo.pCodebook; - iNode = pHcr->nonPcwSideinfo.iNode; - pResultBase = pHcr->nonPcwSideinfo.pResultBase; - iResultPointer = pHcr->nonPcwSideinfo.iResultPointer; - codewordOffset = pHcr->nonPcwSideinfo.codewordOffset; - - pCbDimension = pHcr->tableInfo.pCbDimension; - - treeNode = iNode[codewordOffset]; - pCurrentTree = aHuffTable[pCodebook[codewordOffset]]; - - - for ( ; pRemainingBitsInSegment[segmentOffset] > 0 ; pRemainingBitsInSegment[segmentOffset] -= 1 ) { - - carryBit = HcrGetABitFromBitstream( bs, - &pLeftStartOfSegment[segmentOffset], - &pRightStartOfSegment[segmentOffset], - readDirection); - - CarryBitToBranchValue(carryBit, /* make a step in decoding tree */ - treeNode, - &branchValue, - &branchNode); - - /* if end of branch reached write out lines and count bits needed for sign, otherwise store node in codeword sideinfo */ - if ((branchNode & TEST_BIT_10) == TEST_BIT_10) { /* test bit 10 ; ==> body is complete */ - pQuantValBase = aQuantTable[pCodebook[codewordOffset]]; /* get base address of quantized values belonging to current codebook */ - pQuantVal = pQuantValBase + branchValue; /* set pointer to first valid line [of 2 or 4 quantized values] */ - - iQSC = iResultPointer[codewordOffset]; /* get position of first line for writing out result */ - - for ( dimCntr = pCbDimension[pCodebook[codewordOffset]]; dimCntr != 0; dimCntr-- ) { - pResultBase[iQSC++] = (FIXP_DBL)*pQuantVal++; /* write out 2 or 4 lines into spectrum; no Sign bits available in this state */ - } - - ClearBitFromBitfield(&(pHcr->nonPcwSideinfo.pState), - segmentOffset, - pCodewordBitfield); /* clear a bit in bitfield and switch off statemachine */ - pRemainingBitsInSegment[segmentOffset] -= 1; /* last reinitialzation of for loop counter (see above) is done here */ - break; /* end of branch in tree reached i.e. a whole nonPCW-Body is decoded */ - } - else { /* body is not decoded completely: */ - treeNode = *(pCurrentTree + branchValue); /* update treeNode for further step in decoding tree */ - } - } - iNode[codewordOffset] = treeNode; /* store updated treeNode because maybe decoding of codeword body not finished yet */ - - if ( pRemainingBitsInSegment[segmentOffset] <= 0 ) { - ClearBitFromBitfield(&(pHcr->nonPcwSideinfo.pState), - segmentOffset, - pSegmentBitfield); /* clear a bit in bitfield and switch off statemachine */ - -#if STATE_MACHINE_ERROR_CHECK - if ( pRemainingBitsInSegment[segmentOffset] < 0 ) { - pHcr->decInOut.errorLog |= STATE_ERROR_BODY_ONLY; - return BODY_ONLY; - } -#endif - } - - return STOP_THIS_STATE; -} - - -/*--------------------------------------------------------------------------------------------- - description: Decodes the codeword body, writes out result and counts the number of quantized - spectral values, which are different form zero. For those values sign bits are - needed. - - If sign bit counter cntSign is different from zero, switch to next state to - decode sign Bits there. - If sign bit counter cntSign is zero, no sign bits are needed and codeword is - decoded. ------------------------------------------------------------------------------------------------ - output: Two or four written quantizes spectral values written at position where - pResultPointr points to. The signs of those lines may be wrong. If the signs - [on just one signle sign] is wrong, the next state will correct it. ------------------------------------------------------------------------------------------------ - return: 0 --------------------------------------------------------------------------------------------- */ -UINT Hcr_State_BODY_SIGN__BODY(HANDLE_FDK_BITSTREAM bs, void *ptr) -{ - H_HCR_INFO pHcr = (H_HCR_INFO)ptr; - SCHAR *pRemainingBitsInSegment; - USHORT *pLeftStartOfSegment; - USHORT *pRightStartOfSegment; - UCHAR readDirection; - UINT *pSegmentBitfield; - UINT *pCodewordBitfield; - UINT segmentOffset; - - UCHAR *pCodebook; - UINT *iNode; - UCHAR *pCntSign; - FIXP_DBL *pResultBase; - USHORT *iResultPointer; - UINT codewordOffset; - - UINT iQSC; - UINT cntSign; - UCHAR dimCntr; - UCHAR carryBit; - SCHAR *pSta; - UINT treeNode; - UINT branchValue; - UINT branchNode; - const UCHAR *pCbDimension; - const UINT *pCurrentTree; - const SCHAR *pQuantValBase; - const SCHAR *pQuantVal; - - pRemainingBitsInSegment = pHcr->segmentInfo.pRemainingBitsInSegment; - pLeftStartOfSegment = pHcr->segmentInfo.pLeftStartOfSegment; - pRightStartOfSegment = pHcr->segmentInfo.pRightStartOfSegment; - readDirection = pHcr->segmentInfo.readDirection; - pSegmentBitfield = pHcr->segmentInfo.pSegmentBitfield; - pCodewordBitfield = pHcr->segmentInfo.pCodewordBitfield; - segmentOffset = pHcr->segmentInfo.segmentOffset; - - pCodebook = pHcr->nonPcwSideinfo.pCodebook; - iNode = pHcr->nonPcwSideinfo.iNode; - pCntSign = pHcr->nonPcwSideinfo.pCntSign; - pResultBase = pHcr->nonPcwSideinfo.pResultBase; - iResultPointer = pHcr->nonPcwSideinfo.iResultPointer; - codewordOffset = pHcr->nonPcwSideinfo.codewordOffset; - pSta = pHcr->nonPcwSideinfo.pSta; - - pCbDimension = pHcr->tableInfo.pCbDimension; - - treeNode = iNode[codewordOffset]; - pCurrentTree = aHuffTable[pCodebook[codewordOffset]]; - - - for ( ; pRemainingBitsInSegment[segmentOffset] > 0 ; pRemainingBitsInSegment[segmentOffset] -= 1 ) { - - carryBit = HcrGetABitFromBitstream( bs, - &pLeftStartOfSegment[segmentOffset], - &pRightStartOfSegment[segmentOffset], - readDirection); - - CarryBitToBranchValue(carryBit, /* make a step in decoding tree */ - treeNode, - &branchValue, - &branchNode); - - /* if end of branch reached write out lines and count bits needed for sign, otherwise store node in codeword sideinfo */ - if ((branchNode & TEST_BIT_10) == TEST_BIT_10) { /* test bit 10 ; if set body complete */ - /* body completely decoded; branchValue is valid, set pQuantVal to first (of two or four) quantized spectral coefficients */ - pQuantValBase = aQuantTable[pCodebook[codewordOffset]]; /* get base address of quantized values belonging to current codebook */ - pQuantVal = pQuantValBase + branchValue; /* set pointer to first valid line [of 2 or 4 quantized values] */ - - iQSC = iResultPointer[codewordOffset]; /* get position of first line for writing result */ - - /* codeword decoding result is written out here: Write out 2 or 4 quantized spectral values with probably */ - /* wrong sign and count number of values which are different from zero for sign bit decoding [which happens in next state] */ - cntSign = 0; - for ( dimCntr = pCbDimension[pCodebook[codewordOffset]]; dimCntr != 0; dimCntr-- ) { - pResultBase[iQSC++] = (FIXP_DBL)*pQuantVal; /* write quant. spec. coef. into spectrum */ - if ( *pQuantVal++ != 0 ) { - cntSign += 1; - } - } - - if ( cntSign == 0 ) { - ClearBitFromBitfield(&(pHcr->nonPcwSideinfo.pState), - segmentOffset, - pCodewordBitfield); /* clear a bit in bitfield and switch off statemachine */ - } - else { - pCntSign[codewordOffset] = cntSign; /* write sign count result into codewordsideinfo of current codeword */ - pSta[codewordOffset] = BODY_SIGN__SIGN; /* change state */ - pHcr->nonPcwSideinfo.pState = aStateConstant2State[pSta[codewordOffset]]; /* get state from separate array of cw-sideinfo */ - } - pRemainingBitsInSegment[segmentOffset] -= 1; /* last reinitialzation of for loop counter (see above) is done here */ - break; /* end of branch in tree reached i.e. a whole nonPCW-Body is decoded */ - } - else {/* body is not decoded completely: */ - treeNode = *(pCurrentTree + branchValue); /* update treeNode for further step in decoding tree */ - } - } - iNode[codewordOffset] = treeNode; /* store updated treeNode because maybe decoding of codeword body not finished yet */ - - if ( pRemainingBitsInSegment[segmentOffset] <= 0 ) { - ClearBitFromBitfield(&(pHcr->nonPcwSideinfo.pState), - segmentOffset, - pSegmentBitfield); /* clear a bit in bitfield and switch off statemachine */ - -#if STATE_MACHINE_ERROR_CHECK - if ( pRemainingBitsInSegment[segmentOffset] < 0 ) { - pHcr->decInOut.errorLog |= STATE_ERROR_BODY_SIGN__BODY; - return BODY_SIGN__BODY; - } -#endif - } - - return STOP_THIS_STATE; -} - - -/*--------------------------------------------------------------------------------------------- - description: This state decodes the sign bits belonging to a codeword. The state is called - as often in different "trials" until pCntSgn[codewordOffset] is zero. ------------------------------------------------------------------------------------------------ - output: The two or four quantizes spectral values (written in previous state) have - now the correct sign. ------------------------------------------------------------------------------------------------ - return: 0 --------------------------------------------------------------------------------------------- */ -UINT Hcr_State_BODY_SIGN__SIGN(HANDLE_FDK_BITSTREAM bs, void *ptr) -{ - H_HCR_INFO pHcr = (H_HCR_INFO)ptr; - SCHAR *pRemainingBitsInSegment; - USHORT *pLeftStartOfSegment; - USHORT *pRightStartOfSegment; - UCHAR readDirection; - UINT *pSegmentBitfield; - UINT *pCodewordBitfield; - UINT segmentOffset; - - UCHAR *pCntSign; - FIXP_DBL *pResultBase; - USHORT *iResultPointer; - UINT codewordOffset; - UCHAR carryBit; - UINT iQSC; - UCHAR cntSign; - - pRemainingBitsInSegment = pHcr->segmentInfo.pRemainingBitsInSegment; - pLeftStartOfSegment = pHcr->segmentInfo.pLeftStartOfSegment; - pRightStartOfSegment = pHcr->segmentInfo.pRightStartOfSegment; - readDirection = pHcr->segmentInfo.readDirection; - pSegmentBitfield = pHcr->segmentInfo.pSegmentBitfield; - pCodewordBitfield = pHcr->segmentInfo.pCodewordBitfield; - segmentOffset = pHcr->segmentInfo.segmentOffset; - - pCntSign = pHcr->nonPcwSideinfo.pCntSign; - pResultBase = pHcr->nonPcwSideinfo.pResultBase; - iResultPointer = pHcr->nonPcwSideinfo.iResultPointer; - codewordOffset = pHcr->nonPcwSideinfo.codewordOffset; - iQSC = iResultPointer[codewordOffset]; - cntSign = pCntSign[codewordOffset]; - - - - /* loop for sign bit decoding */ - for ( ; pRemainingBitsInSegment[segmentOffset] > 0 ; pRemainingBitsInSegment[segmentOffset] -= 1 ) { - - carryBit = HcrGetABitFromBitstream( bs, - &pLeftStartOfSegment[segmentOffset], - &pRightStartOfSegment[segmentOffset], - readDirection); - cntSign -= 1; /* decrement sign counter because one sign bit has been read */ - - /* search for a line (which was decoded in previous state) which is not zero. [This value will get a sign] */ - while ( pResultBase[iQSC] == (FIXP_DBL)0 ) { - iQSC++; /* points to current value different from zero */ - if (iQSC >= 1024) { - return BODY_SIGN__SIGN; - } - } - - /* put sign together with line; if carryBit is zero, the sign is ok already; no write operation necessary in this case */ - if ( carryBit != 0 ) { - pResultBase[iQSC] = -pResultBase[iQSC]; /* carryBit = 1 --> minus */ - } - - iQSC++; /* update pointer to next (maybe valid) value */ - - if ( cntSign == 0 ) { /* if (cntSign==0) ==> set state CODEWORD_DECODED */ - ClearBitFromBitfield(&(pHcr->nonPcwSideinfo.pState), - segmentOffset, - pCodewordBitfield); /* clear a bit in bitfield and switch off statemachine */ - pRemainingBitsInSegment[segmentOffset] -= 1; /* last reinitialzation of for loop counter (see above) is done here */ - break; /* whole nonPCW-Body and according sign bits are decoded */ - } - } - pCntSign[codewordOffset] = cntSign; - iResultPointer[codewordOffset] = iQSC; /* store updated pResultPointer */ - - if ( pRemainingBitsInSegment[segmentOffset] <= 0 ) { - ClearBitFromBitfield(&(pHcr->nonPcwSideinfo.pState), - segmentOffset, - pSegmentBitfield); /* clear a bit in bitfield and switch off statemachine */ - -#if STATE_MACHINE_ERROR_CHECK - if ( pRemainingBitsInSegment[segmentOffset] < 0 ) { - pHcr->decInOut.errorLog |= STATE_ERROR_BODY_SIGN__SIGN; - return BODY_SIGN__SIGN; - } -#endif - } - - return STOP_THIS_STATE; -} - - -/*--------------------------------------------------------------------------------------------- - description: Decodes the codeword body in case of codebook is 11. Writes out resulting - two or four lines [with probably wrong sign] and counts the number of - lines, which are different form zero. This information is needed in next - state where sign bits will be decoded, if necessary. - If sign bit counter cntSign is zero, no sign bits are needed and codeword is - decoded completely. ------------------------------------------------------------------------------------------------ - output: Two lines (quantizes spectral coefficients) which are probably wrong. The - sign may be wrong and if one or two values is/are 16, the following states - will decode the escape sequence to correct the values which are wirtten here. ------------------------------------------------------------------------------------------------ - return: 0 --------------------------------------------------------------------------------------------- */ -UINT Hcr_State_BODY_SIGN_ESC__BODY(HANDLE_FDK_BITSTREAM bs, void *ptr) -{ - H_HCR_INFO pHcr = (H_HCR_INFO)ptr; - SCHAR *pRemainingBitsInSegment; - USHORT *pLeftStartOfSegment; - USHORT *pRightStartOfSegment; - UCHAR readDirection; - UINT *pSegmentBitfield; - UINT *pCodewordBitfield; - UINT segmentOffset; - - UINT *iNode; - UCHAR *pCntSign; - FIXP_DBL *pResultBase; - USHORT *iResultPointer; - UINT codewordOffset; - - UCHAR carryBit; - UINT iQSC; - UINT cntSign; - UINT dimCntr; - UINT treeNode; - SCHAR *pSta; - UINT branchNode; - UINT branchValue; - const UINT *pCurrentTree; - const SCHAR *pQuantValBase; - const SCHAR *pQuantVal; - - pRemainingBitsInSegment = pHcr->segmentInfo.pRemainingBitsInSegment; - pLeftStartOfSegment = pHcr->segmentInfo.pLeftStartOfSegment; - pRightStartOfSegment = pHcr->segmentInfo.pRightStartOfSegment; - readDirection = pHcr->segmentInfo.readDirection; - pSegmentBitfield = pHcr->segmentInfo.pSegmentBitfield; - pCodewordBitfield = pHcr->segmentInfo.pCodewordBitfield; - segmentOffset = pHcr->segmentInfo.segmentOffset; - - iNode = pHcr->nonPcwSideinfo.iNode; - pCntSign = pHcr->nonPcwSideinfo.pCntSign; - pResultBase = pHcr->nonPcwSideinfo.pResultBase; - iResultPointer = pHcr->nonPcwSideinfo.iResultPointer; - codewordOffset = pHcr->nonPcwSideinfo.codewordOffset; - pSta = pHcr->nonPcwSideinfo.pSta; - - treeNode = iNode[codewordOffset]; - pCurrentTree = aHuffTable[ESCAPE_CODEBOOK]; - - - for ( ; pRemainingBitsInSegment[segmentOffset] > 0 ; pRemainingBitsInSegment[segmentOffset] -= 1 ) { - - carryBit = HcrGetABitFromBitstream( bs, - &pLeftStartOfSegment[segmentOffset], - &pRightStartOfSegment[segmentOffset], - readDirection); - - /* make a step in tree */ - CarryBitToBranchValue(carryBit, - treeNode, - &branchValue, - &branchNode); - - /* if end of branch reached write out lines and count bits needed for sign, otherwise store node in codeword sideinfo */ - if ((branchNode & TEST_BIT_10) == TEST_BIT_10) { /* test bit 10 ; if set body complete */ - - /* body completely decoded; branchValue is valid */ - /* set pQuantVol to first (of two or four) quantized spectral coefficients */ - pQuantValBase = aQuantTable[ESCAPE_CODEBOOK]; /* get base address of quantized values belonging to current codebook */ - pQuantVal = pQuantValBase + branchValue; /* set pointer to first valid line [of 2 or 4 quantized values] */ - - /* make backup from original resultPointer in node storage for state BODY_SIGN_ESC__SIGN */ - iNode[codewordOffset] = iResultPointer[codewordOffset]; - - /* get position of first line for writing result */ - iQSC = iResultPointer[codewordOffset]; - - /* codeword decoding result is written out here: Write out 2 or 4 quantized spectral values with probably */ - /* wrong sign and count number of values which are different from zero for sign bit decoding [which happens in next state] */ - cntSign = 0; - - for ( dimCntr = DIMENSION_OF_ESCAPE_CODEBOOK; dimCntr != 0; dimCntr-- ) { - pResultBase[iQSC++] = (FIXP_DBL)*pQuantVal; /* write quant. spec. coef. into spectrum */ - if ( *pQuantVal++ != 0 ) { - cntSign += 1; - } - } - - if ( cntSign == 0 ) { - ClearBitFromBitfield(&(pHcr->nonPcwSideinfo.pState), - segmentOffset, - pCodewordBitfield); /* clear a bit in bitfield and switch off statemachine */ - /* codeword decoded */ - } - else { - /* write sign count result into codewordsideinfo of current codeword */ - pCntSign[codewordOffset] = cntSign; - pSta[codewordOffset] = BODY_SIGN_ESC__SIGN; /* change state */ - pHcr->nonPcwSideinfo.pState = aStateConstant2State[pSta[codewordOffset]]; /* get state from separate array of cw-sideinfo */ - } - pRemainingBitsInSegment[segmentOffset] -= 1; /* the last reinitialzation of for loop counter (see above) is done here */ - break; /* end of branch in tree reached i.e. a whole nonPCW-Body is decoded */ - } - else { /* body is not decoded completely: */ - /* update treeNode for further step in decoding tree and store updated treeNode because maybe no more bits left in segment */ - treeNode = *(pCurrentTree + branchValue); - iNode[codewordOffset] = treeNode; - } - } - - if ( pRemainingBitsInSegment[segmentOffset] <= 0 ) { - ClearBitFromBitfield(&(pHcr->nonPcwSideinfo.pState), - segmentOffset, - pSegmentBitfield); /* clear a bit in bitfield and switch off statemachine */ - -#if STATE_MACHINE_ERROR_CHECK - if ( pRemainingBitsInSegment[segmentOffset] < 0 ) { - pHcr->decInOut.errorLog |= STATE_ERROR_BODY_SIGN_ESC__BODY; - return BODY_SIGN_ESC__BODY; - } -#endif - } - - return STOP_THIS_STATE; -} - - -/*--------------------------------------------------------------------------------------------- - description: This state decodes the sign bits, if a codeword of codebook 11 needs some. - A flag named 'flagB' in codeword sideinfo is set, if the second line of - quantized spectral values is 16. The 'flagB' is used in case of decoding - of a escape sequence is necessary as far as the second line is concerned. - - If only the first line needs an escape sequence, the flagB is cleared. - If only the second line needs an escape sequence, the flagB is not used. - - For storing sideinfo in case of escape sequence decoding one single word - can be used for both escape sequences because they are decoded not at the - same time: - - - bit 23 22 21 20 19 18 17 16 15 14 13 12 11 10 9 8 7 6 5 4 3 2 1 0 - ===== == == =========== =========== =================================== - ^ ^ ^ ^ ^ ^ - | | | | | | - res. flagA flagB escapePrefixUp escapePrefixDown escapeWord - ------------------------------------------------------------------------------------------------ - output: Two lines with correct sign. If one or two values is/are 16, the lines are - not valid, otherwise they are. ------------------------------------------------------------------------------------------------ - return: 0 --------------------------------------------------------------------------------------------- */ -UINT Hcr_State_BODY_SIGN_ESC__SIGN(HANDLE_FDK_BITSTREAM bs, void *ptr) -{ - H_HCR_INFO pHcr = (H_HCR_INFO)ptr; - SCHAR *pRemainingBitsInSegment; - USHORT *pLeftStartOfSegment; - USHORT *pRightStartOfSegment; - UCHAR readDirection; - UINT *pSegmentBitfield; - UINT *pCodewordBitfield; - UINT segmentOffset; - - UINT *iNode; - UCHAR *pCntSign; - FIXP_DBL *pResultBase; - USHORT *iResultPointer; - UINT *pEscapeSequenceInfo; - UINT codewordOffset; - - UINT iQSC; - UCHAR cntSign; - UINT flagA; - UINT flagB; - UINT flags; - UCHAR carryBit; - SCHAR *pSta; - - pRemainingBitsInSegment = pHcr->segmentInfo.pRemainingBitsInSegment; - pLeftStartOfSegment = pHcr->segmentInfo.pLeftStartOfSegment; - pRightStartOfSegment = pHcr->segmentInfo.pRightStartOfSegment; - readDirection = pHcr->segmentInfo.readDirection; - pSegmentBitfield = pHcr->segmentInfo.pSegmentBitfield; - pCodewordBitfield = pHcr->segmentInfo.pCodewordBitfield; - segmentOffset = pHcr->segmentInfo.segmentOffset; - - iNode = pHcr->nonPcwSideinfo.iNode; - pCntSign = pHcr->nonPcwSideinfo.pCntSign; - pResultBase = pHcr->nonPcwSideinfo.pResultBase; - iResultPointer = pHcr->nonPcwSideinfo.iResultPointer; - pEscapeSequenceInfo = pHcr->nonPcwSideinfo.pEscapeSequenceInfo; - codewordOffset = pHcr->nonPcwSideinfo.codewordOffset; - pSta = pHcr->nonPcwSideinfo.pSta; - - iQSC = iResultPointer[codewordOffset]; - cntSign = pCntSign[codewordOffset]; - - - /* loop for sign bit decoding */ - for ( ; pRemainingBitsInSegment[segmentOffset] > 0 ; pRemainingBitsInSegment[segmentOffset] -= 1 ) { - - carryBit = HcrGetABitFromBitstream( bs, - &pLeftStartOfSegment[segmentOffset], - &pRightStartOfSegment[segmentOffset], - readDirection); - - /* decrement sign counter because one sign bit has been read */ - cntSign -= 1; - pCntSign[codewordOffset] = cntSign; - - /* get a quantized spectral value (which was decoded in previous state) which is not zero. [This value will get a sign] */ - while ( pResultBase[iQSC] == (FIXP_DBL)0 ) { - iQSC++; - } - iResultPointer[codewordOffset] = iQSC; - - /* put negative sign together with quantized spectral value; if carryBit is zero, the sign is ok already; no write operation necessary in this case */ - if ( carryBit != 0 ) { - pResultBase[iQSC] = - pResultBase[iQSC]; /* carryBit = 1 --> minus */ - } - iQSC++; /* update index to next (maybe valid) value */ - iResultPointer[codewordOffset] = iQSC; - - if ( cntSign == 0 ) { - /* all sign bits are decoded now */ - pRemainingBitsInSegment[segmentOffset] -= 1; /* last reinitialzation of for loop counter (see above) is done here */ - - /* check decoded values if codeword is decoded: Check if one or two escape sequences 16 follow */ - - /* step 0 */ - /* restore pointer to first decoded quantized value [ = original pResultPointr] from index iNode prepared in State_BODY_SIGN_ESC__BODY */ - iQSC = iNode[codewordOffset]; - - /* step 1 */ - /* test first value if escape sequence follows */ - flagA = 0; /* for first possible escape sequence */ - if ( fixp_abs(pResultBase[iQSC++]) == (FIXP_DBL)ESCAPE_VALUE ) { - flagA = 1; - } - - /* step 2 */ - /* test second value if escape sequence follows */ - flagB = 0; /* for second possible escape sequence */ - if ( fixp_abs(pResultBase[iQSC]) == (FIXP_DBL)ESCAPE_VALUE ) { - flagB = 1; - } - - - /* step 3 */ - /* evaluate flag result and go on if necessary */ - if ( !flagA && !flagB ) { - ClearBitFromBitfield(&(pHcr->nonPcwSideinfo.pState), - segmentOffset, - pCodewordBitfield); /* clear a bit in bitfield and switch off statemachine */ - } - else { - /* at least one of two lines is 16 */ - /* store both flags at correct positions in non PCW codeword sideinfo pEscapeSequenceInfo[codewordOffset] */ - flags = 0; - flags = flagA << POSITION_OF_FLAG_A; - flags |= (flagB << POSITION_OF_FLAG_B); - pEscapeSequenceInfo[codewordOffset] = flags; - - - /* set next state */ - pSta[codewordOffset] = BODY_SIGN_ESC__ESC_PREFIX; - pHcr->nonPcwSideinfo.pState = aStateConstant2State[pSta[codewordOffset]]; /* get state from separate array of cw-sideinfo */ - - /* set result pointer to the first line of the two decoded lines */ - iResultPointer[codewordOffset] = iNode[codewordOffset]; - - if ( !flagA && flagB ) { - /* update pResultPointr ==> state Stat_BODY_SIGN_ESC__ESC_WORD writes to correct position. Second value is the one and only escape value */ - iQSC = iResultPointer[codewordOffset]; - iQSC++; - iResultPointer[codewordOffset] = iQSC; - } - - } /* at least one of two lines is 16 */ - break; /* nonPCW-Body at cb 11 and according sign bits are decoded */ - - } /* if ( cntSign == 0 ) */ - } /* loop over remaining Bits in segment */ - - if ( pRemainingBitsInSegment[segmentOffset] <= 0 ) { - ClearBitFromBitfield(&(pHcr->nonPcwSideinfo.pState), - segmentOffset, - pSegmentBitfield); /* clear a bit in bitfield and switch off statemachine */ - -#if STATE_MACHINE_ERROR_CHECK - if ( pRemainingBitsInSegment[segmentOffset] < 0 ) { - pHcr->decInOut.errorLog |= STATE_ERROR_BODY_SIGN_ESC__SIGN; - return BODY_SIGN_ESC__SIGN; - } -#endif - - } - return STOP_THIS_STATE; -} - - -/*--------------------------------------------------------------------------------------------- - description: Decode escape prefix of first or second escape sequence. The escape prefix - consists of ones. The following zero is also decoded here. ------------------------------------------------------------------------------------------------ - output: If the single separator-zero which follows the escape-prefix-ones is not yet decoded: - The value 'escapePrefixUp' in word pEscapeSequenceInfo[codewordOffset] is updated. - - If the single separator-zero which follows the escape-prefix-ones is decoded: - Two updated values 'escapePrefixUp' and 'escapePrefixDown' in word - pEscapeSequenceInfo[codewordOffset]. This State is finished. Switch to next state. ------------------------------------------------------------------------------------------------ - return: 0 --------------------------------------------------------------------------------------------- */ -UINT Hcr_State_BODY_SIGN_ESC__ESC_PREFIX(HANDLE_FDK_BITSTREAM bs, void *ptr) -{ - H_HCR_INFO pHcr = (H_HCR_INFO)ptr; - SCHAR *pRemainingBitsInSegment; - USHORT *pLeftStartOfSegment; - USHORT *pRightStartOfSegment; - UCHAR readDirection; - UINT *pSegmentBitfield; - UINT segmentOffset; - UINT *pEscapeSequenceInfo; - UINT codewordOffset; - UCHAR carryBit; - UINT escapePrefixUp; - SCHAR *pSta; - - pRemainingBitsInSegment = pHcr->segmentInfo.pRemainingBitsInSegment; - pLeftStartOfSegment = pHcr->segmentInfo.pLeftStartOfSegment; - pRightStartOfSegment = pHcr->segmentInfo.pRightStartOfSegment; - readDirection = pHcr->segmentInfo.readDirection; - pSegmentBitfield = pHcr->segmentInfo.pSegmentBitfield; - segmentOffset = pHcr->segmentInfo.segmentOffset; - pEscapeSequenceInfo = pHcr->nonPcwSideinfo.pEscapeSequenceInfo; - codewordOffset = pHcr->nonPcwSideinfo.codewordOffset; - pSta = pHcr->nonPcwSideinfo.pSta; - - escapePrefixUp = (pEscapeSequenceInfo[codewordOffset] & MASK_ESCAPE_PREFIX_UP) >> LSB_ESCAPE_PREFIX_UP; - - - /* decode escape prefix */ - for ( ; pRemainingBitsInSegment[segmentOffset] > 0 ; pRemainingBitsInSegment[segmentOffset] -= 1 ) { - - carryBit = HcrGetABitFromBitstream( bs, - &pLeftStartOfSegment[segmentOffset], - &pRightStartOfSegment[segmentOffset], - readDirection); - - /* count ones and store sum in escapePrefixUp */ - if ( carryBit == 1 ) { - escapePrefixUp += 1; /* update conter for ones */ - - /* store updated counter in sideinfo of current codeword */ - pEscapeSequenceInfo[codewordOffset] &= ~MASK_ESCAPE_PREFIX_UP; /* delete old escapePrefixUp */ - escapePrefixUp <<= LSB_ESCAPE_PREFIX_UP; /* shift to correct position */ - pEscapeSequenceInfo[codewordOffset] |= escapePrefixUp; /* insert new escapePrefixUp */ - escapePrefixUp >>= LSB_ESCAPE_PREFIX_UP; /* shift back down */ - } - else { /* separator [zero] reached */ - pRemainingBitsInSegment[segmentOffset] -= 1; /* last reinitialzation of for loop counter (see above) is done here */ - escapePrefixUp += 4; /* if escape_separator '0' appears, add 4 and ==> break */ - - /* store escapePrefixUp in pEscapeSequenceInfo[codewordOffset] at bit position escapePrefixUp */ - pEscapeSequenceInfo[codewordOffset] &= ~MASK_ESCAPE_PREFIX_UP; /* delete old escapePrefixUp */ - escapePrefixUp <<= LSB_ESCAPE_PREFIX_UP; /* shift to correct position */ - pEscapeSequenceInfo[codewordOffset] |= escapePrefixUp; /* insert new escapePrefixUp */ - escapePrefixUp >>= LSB_ESCAPE_PREFIX_UP; /* shift back down */ - - /* store escapePrefixUp in pEscapeSequenceInfo[codewordOffset] at bit position escapePrefixDown */ - pEscapeSequenceInfo[codewordOffset] &= ~MASK_ESCAPE_PREFIX_DOWN; /* delete old escapePrefixDown */ - escapePrefixUp <<= LSB_ESCAPE_PREFIX_DOWN; /* shift to correct position */ - pEscapeSequenceInfo[codewordOffset] |= escapePrefixUp; /* insert new escapePrefixDown */ - escapePrefixUp >>= LSB_ESCAPE_PREFIX_DOWN; /* shift back down */ - - pSta[codewordOffset] = BODY_SIGN_ESC__ESC_WORD; /* set next state */ - pHcr->nonPcwSideinfo.pState = aStateConstant2State[pSta[codewordOffset]]; /* get state from separate array of cw-sideinfo */ - break; - } - } - - if ( pRemainingBitsInSegment[segmentOffset] <= 0 ) { - ClearBitFromBitfield(&(pHcr->nonPcwSideinfo.pState), - segmentOffset, - pSegmentBitfield); /* clear a bit in bitfield and switch off statemachine */ - -#if STATE_MACHINE_ERROR_CHECK - if ( pRemainingBitsInSegment[segmentOffset] < 0 ) { - pHcr->decInOut.errorLog |= STATE_ERROR_BODY_SIGN_ESC__ESC_PREFIX; - return BODY_SIGN_ESC__ESC_PREFIX; - } -#endif - } - - return STOP_THIS_STATE; -} - - -/*--------------------------------------------------------------------------------------------- - description: Decode escapeWord of escape sequence. If the escape sequence is decoded - completely, assemble quantized-spectral-escape-coefficient and replace the - previous decoded 16 by the new value. - Test flagB. If flagB is set, the second escape sequence must be decoded. If - flagB is not set, the codeword is decoded and the state machine is switched - off. ------------------------------------------------------------------------------------------------ - output: Two lines with valid sign. At least one of both lines has got the correct - value. ------------------------------------------------------------------------------------------------ - return: 0 --------------------------------------------------------------------------------------------- */ -UINT Hcr_State_BODY_SIGN_ESC__ESC_WORD(HANDLE_FDK_BITSTREAM bs, void *ptr) -{ - H_HCR_INFO pHcr = (H_HCR_INFO)ptr; - SCHAR *pRemainingBitsInSegment; - USHORT *pLeftStartOfSegment; - USHORT *pRightStartOfSegment; - UCHAR readDirection; - UINT *pSegmentBitfield; - UINT *pCodewordBitfield; - UINT segmentOffset; - - FIXP_DBL *pResultBase; - USHORT *iResultPointer; - UINT *pEscapeSequenceInfo; - UINT codewordOffset; - - UINT escapeWord; - UINT escapePrefixDown; - UINT escapePrefixUp; - UCHAR carryBit; - UINT iQSC; - INT sign; - UINT flagA; - UINT flagB; - SCHAR *pSta; - - pRemainingBitsInSegment = pHcr->segmentInfo.pRemainingBitsInSegment; - pLeftStartOfSegment = pHcr->segmentInfo.pLeftStartOfSegment; - pRightStartOfSegment = pHcr->segmentInfo.pRightStartOfSegment; - readDirection = pHcr->segmentInfo.readDirection; - pSegmentBitfield = pHcr->segmentInfo.pSegmentBitfield; - pCodewordBitfield = pHcr->segmentInfo.pCodewordBitfield; - segmentOffset = pHcr->segmentInfo.segmentOffset; - - pResultBase = pHcr->nonPcwSideinfo.pResultBase; - iResultPointer = pHcr->nonPcwSideinfo.iResultPointer; - pEscapeSequenceInfo = pHcr->nonPcwSideinfo.pEscapeSequenceInfo; - codewordOffset = pHcr->nonPcwSideinfo.codewordOffset; - pSta = pHcr->nonPcwSideinfo.pSta; - - escapeWord = pEscapeSequenceInfo[codewordOffset] & MASK_ESCAPE_WORD; - escapePrefixDown = (pEscapeSequenceInfo[codewordOffset] & MASK_ESCAPE_PREFIX_DOWN) >> LSB_ESCAPE_PREFIX_DOWN; - - - /* decode escape word */ - for ( ; pRemainingBitsInSegment[segmentOffset] > 0 ; pRemainingBitsInSegment[segmentOffset] -= 1 ) { - - carryBit = HcrGetABitFromBitstream( bs, - &pLeftStartOfSegment[segmentOffset], - &pRightStartOfSegment[segmentOffset], - readDirection); - - /* build escape word */ - escapeWord <<= 1; /* left shift previous decoded part of escapeWord by on bit */ - escapeWord = escapeWord | carryBit; /* assemble escape word by bitwise or */ - - /* decrement counter for length of escape word because one more bit was decoded */ - escapePrefixDown -= 1; - - /* store updated escapePrefixDown */ - pEscapeSequenceInfo[codewordOffset] &= ~MASK_ESCAPE_PREFIX_DOWN; /* delete old escapePrefixDown */ - escapePrefixDown <<= LSB_ESCAPE_PREFIX_DOWN; /* shift to correct position */ - pEscapeSequenceInfo[codewordOffset] |= escapePrefixDown; /* insert new escapePrefixDown */ - escapePrefixDown >>= LSB_ESCAPE_PREFIX_DOWN; /* shift back */ - - - /* store updated escapeWord */ - pEscapeSequenceInfo[codewordOffset] &= ~MASK_ESCAPE_WORD; /* delete old escapeWord */ - pEscapeSequenceInfo[codewordOffset] |= escapeWord; /* insert new escapeWord */ - - - if ( escapePrefixDown == 0 ) { - pRemainingBitsInSegment[segmentOffset] -= 1; /* last reinitialzation of for loop counter (see above) is done here */ - - /* escape sequence decoded. Assemble escape-line and replace original line */ - - /* step 0 */ - /* derive sign */ - iQSC = iResultPointer[codewordOffset]; - sign = (pResultBase[iQSC] >= (FIXP_DBL)0) ? 1 : -1; /* get sign of escape value 16 */ - - /* step 1 */ - /* get escapePrefixUp */ - escapePrefixUp = (pEscapeSequenceInfo[codewordOffset] & MASK_ESCAPE_PREFIX_UP) >> LSB_ESCAPE_PREFIX_UP; - - /* step 2 */ - /* calculate escape value */ - pResultBase[iQSC] = (FIXP_DBL)(sign * (((INT) 1 << escapePrefixUp) + escapeWord)); - - /* get both flags from sideinfo (flags are not shifted to the lsb-position) */ - flagA = pEscapeSequenceInfo[codewordOffset] & MASK_FLAG_A; - flagB = pEscapeSequenceInfo[codewordOffset] & MASK_FLAG_B; - - /* step 3 */ - /* clear the whole escape sideinfo word */ - pEscapeSequenceInfo[codewordOffset] = 0; - - /* change state in dependence of flag flagB */ - if ( flagA != 0 ) { - /* first escape sequence decoded; previous decoded 16 has been replaced by valid line */ - - /* clear flagA in sideinfo word because this escape sequence has already beed decoded */ - pEscapeSequenceInfo[codewordOffset] &= ~MASK_FLAG_A; - - if ( flagB == 0 ) { - ClearBitFromBitfield(&(pHcr->nonPcwSideinfo.pState), - segmentOffset, - pCodewordBitfield); /* clear a bit in bitfield and switch off statemachine */ - } - else { - /* updated pointer to next and last 16 */ - iQSC++; - iResultPointer[codewordOffset] = iQSC; - - /* change state */ - pSta[codewordOffset] = BODY_SIGN_ESC__ESC_PREFIX; - pHcr->nonPcwSideinfo.pState = aStateConstant2State[pSta[codewordOffset]]; /* get state from separate array of cw-sideinfo */ - } - } - else { - ClearBitFromBitfield(&(pHcr->nonPcwSideinfo.pState), - segmentOffset, - pCodewordBitfield); /* clear a bit in bitfield and switch off statemachine */ - } - break; - } - } - - if ( pRemainingBitsInSegment[segmentOffset] <= 0 ) { - ClearBitFromBitfield(&(pHcr->nonPcwSideinfo.pState), - segmentOffset, - pSegmentBitfield); /* clear a bit in bitfield and switch off statemachine */ - -#if STATE_MACHINE_ERROR_CHECK - if ( pRemainingBitsInSegment[segmentOffset] < 0 ) { - pHcr->decInOut.errorLog |= STATE_ERROR_BODY_SIGN_ESC__ESC_WORD; - return BODY_SIGN_ESC__ESC_WORD; - } -#endif - } - - return STOP_THIS_STATE; -} - diff --git a/libAACdec/src/aacdec_hcrs.h b/libAACdec/src/aacdec_hcrs.h deleted file mode 100644 index 678ba26..0000000 --- a/libAACdec/src/aacdec_hcrs.h +++ /dev/null @@ -1,153 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/***************************** MPEG-4 AAC Decoder *************************** - - Author(s): Robert Weidner (DSP Solutions) - Description: HCR Decoder: Defines of state-constants, masks and - state-prototypes - -*******************************************************************************/ - -#ifndef _AACDEC_HCRS_H_ -#define _AACDEC_HCRS_H_ - - - -#include "FDK_bitstream.h" -#include "aacdec_hcr_types.h" - /* The four different kinds of types of states are: */ -/* different states are defined as constants */ /* start middle=self next stop */ -#define STOP_THIS_STATE 0 /* */ -#define BODY_ONLY 1 /* X X X */ -#define BODY_SIGN__BODY 2 /* X X X X [stop if no sign] */ -#define BODY_SIGN__SIGN 3 /* X X [stop if sign bits decoded] */ -#define BODY_SIGN_ESC__BODY 4 /* X X X X [stop if no sign] */ -#define BODY_SIGN_ESC__SIGN 5 /* X X X [stop if no escape sequence] */ -#define BODY_SIGN_ESC__ESC_PREFIX 6 /* X X */ -#define BODY_SIGN_ESC__ESC_WORD 7 /* X X X [stop if abs(second qsc) != 16] */ - -/* examples: */ - -/* BODY_ONLY means only the codeword body will be decoded; no sign bits will follow and no escapesequence will follow */ - -/* BODY_SIGN__BODY means that the codeword consists of two parts; body and sign part. The part '__BODY' after the two underscores shows */ -/* that the bits which are currently decoded belong to the '__BODY' of the codeword and not to the sign part. */ - -/* BODY_SIGN_ESC__ESC_PB means that the codeword consists of three parts; body, sign and (here: two) escape sequences; */ -/* P = Prefix = ones */ -/* W = Escape Word */ -/* A = first possible (of two) Escape sequeces */ -/* B = second possible (of two) Escape sequeces */ -/* The part after the two underscores shows that the current bits which are decoded belong to the '__ESC_PB' - part of the */ -/* codeword. That means the body and the sign bits are decoded completely and the bits which are decoded now belong to */ -/* the escape sequence [P = prefix; B=second possible escape sequence] */ - - -#define MSB_31_MASK 0x80000000 /* masks MSB (= Bit 31) in a 32 bit word */ -#define DIMENSION_OF_ESCAPE_CODEBOOK 2 /* for cb >= 11 is dimension 2 */ -#define ESCAPE_CODEBOOK 11 - -#define MASK_ESCAPE_PREFIX_UP 0x000F0000 -#define LSB_ESCAPE_PREFIX_UP 16 - -#define MASK_ESCAPE_PREFIX_DOWN 0x0000F000 -#define LSB_ESCAPE_PREFIX_DOWN 12 - -#define MASK_ESCAPE_WORD 0x00000FFF -#define MASK_FLAG_A 0x00200000 -#define MASK_FLAG_B 0x00100000 - - -extern void DecodeNonPCWs(HANDLE_FDK_BITSTREAM bs, H_HCR_INFO hHcr); - -UINT Hcr_State_BODY_ONLY (HANDLE_FDK_BITSTREAM, void*); -UINT Hcr_State_BODY_SIGN__BODY (HANDLE_FDK_BITSTREAM, void*); -UINT Hcr_State_BODY_SIGN__SIGN (HANDLE_FDK_BITSTREAM, void*); -UINT Hcr_State_BODY_SIGN_ESC__BODY (HANDLE_FDK_BITSTREAM, void*); -UINT Hcr_State_BODY_SIGN_ESC__SIGN (HANDLE_FDK_BITSTREAM, void*); -UINT Hcr_State_BODY_SIGN_ESC__ESC_PREFIX (HANDLE_FDK_BITSTREAM, void*); -UINT Hcr_State_BODY_SIGN_ESC__ESC_WORD (HANDLE_FDK_BITSTREAM, void*); - -#endif /* _AACDEC_HCRS_H_ */ - diff --git a/libAACdec/src/aacdec_pns.cpp b/libAACdec/src/aacdec_pns.cpp index 541ef07..47449ba 100644 --- a/libAACdec/src/aacdec_pns.cpp +++ b/libAACdec/src/aacdec_pns.cpp @@ -246,82 +246,6 @@ void CPns_Read (CPnsData *pPnsData, } -/** - * \brief Generate a vector of noise of given length. The noise values are - * scaled in order to yield a noise energy of 1.0 - * \param spec pointer to were the noise values will be written to. - * \param size amount of noise values to be generated. - * \param pRandomState pointer to the state of the random generator being used. - * \return exponent of generated noise vector. - */ -static int GenerateRandomVector (FIXP_DBL *RESTRICT spec, - int size, - int *pRandomState) -{ - int i, invNrg_e = 0, nrg_e = 0; - FIXP_DBL invNrg_m, nrg_m = FL2FXCONST_DBL(0.0f) ; - FIXP_DBL *RESTRICT ptr = spec; - int randomState = *pRandomState; - -#define GEN_NOISE_NRG_SCALE 7 - - /* Generate noise and calculate energy. */ - for (i=0; i>GEN_NOISE_NRG_SCALE); - *ptr++ = (FIXP_DBL)randomState; - } - nrg_e = GEN_NOISE_NRG_SCALE*2 + 1; - - /* weight noise with = 1 / sqrt_nrg; */ - invNrg_m = invSqrtNorm2(nrg_m<<1, &invNrg_e); - invNrg_e += -((nrg_e-1)>>1); - - for (i=size; i--; ) - { - spec[i] = fMult(spec[i], invNrg_m); - } - - /* Store random state */ - *pRandomState = randomState; - - return invNrg_e; -} - -static void ScaleBand (FIXP_DBL *RESTRICT spec, int size, int scaleFactor, int specScale, int noise_e, int out_of_phase) -{ - int i, shift, sfExponent; - FIXP_DBL sfMatissa; - - /* Get gain from scale factor value = 2^(scaleFactor * 0.25) */ - sfMatissa = MantissaTable[scaleFactor & 0x03][0]; - /* sfExponent = (scaleFactor >> 2) + ExponentTable[scaleFactor & 0x03][0]; */ - /* Note: ExponentTable[scaleFactor & 0x03][0] is always 1. */ - sfExponent = (scaleFactor >> 2) + 1; - - if (out_of_phase != 0) { - sfMatissa = -sfMatissa; - } - - /* +1 because of fMultDiv2 below. */ - shift = sfExponent - specScale + 1 + noise_e; - - /* Apply gain to noise values */ - if (shift>=0) { - shift = fixMin( shift, DFRACT_BITS-1 ); - for (i = size ; i-- != 0; ) { - spec [i] = fMultDiv2 (spec [i], sfMatissa) << shift; - } - } else { - shift = fixMin( -shift, DFRACT_BITS-1 ); - for (i = size ; i-- != 0; ) { - spec [i] = fMultDiv2 (spec [i], sfMatissa) >> shift; - } - } -} - - /*! \brief Apply PNS @@ -338,45 +262,10 @@ void CPns_Apply (const CPnsData *pPnsData, const INT granuleLength, const int channel) { +#if 0 if (pPnsData->PnsActive) { - const short *BandOffsets = GetScaleFactorBandOffsets(pIcsInfo, pSamplingRateInfo); - - int ScaleFactorBandsTransmitted = GetScaleFactorBandsTransmitted(pIcsInfo); - - for (int window = 0, group = 0; group < GetWindowGroups(pIcsInfo); group++) { - for (int groupwin = 0; groupwin < GetWindowGroupLength(pIcsInfo, group); groupwin++, window++) { - FIXP_DBL *spectrum = SPEC(pSpectrum, window, granuleLength); - - for (int band = 0 ; band < ScaleFactorBandsTransmitted; band++) { - if (CPns_IsPnsUsed (pPnsData, group, band)) { - UINT pns_band = group*16+band; - - int bandWidth = BandOffsets [band + 1] - BandOffsets [band] ; - int noise_e; - - FDK_ASSERT(bandWidth >= 0); - - if (channel > 0 && CPns_IsCorrelated(pPnsData, group, band)) - { - noise_e = GenerateRandomVector (spectrum + BandOffsets [band], bandWidth, - &pPnsData->randomSeed [pns_band]) ; - } - else - { - pPnsData->randomSeed [pns_band] = *pPnsData->currentSeed ; - - noise_e = GenerateRandomVector (spectrum + BandOffsets [band], bandWidth, - pPnsData->currentSeed) ; - } - - int outOfPhase = CPns_IsOutOfPhase (pPnsData, group, band); - - ScaleBand (spectrum + BandOffsets [band], bandWidth, - pScaleFactor[pns_band], - pSpecScale[window], noise_e, outOfPhase) ; - } - } - } - } + /** disabled */ + FDKprintf("PNS not implemented\n"); } +#endif } diff --git a/libAACdec/src/aacdec_tns.cpp b/libAACdec/src/aacdec_tns.cpp index 352f04a..bbbba7d 100644 --- a/libAACdec/src/aacdec_tns.cpp +++ b/libAACdec/src/aacdec_tns.cpp @@ -213,116 +213,6 @@ AAC_DECODER_ERROR CTns_Read(HANDLE_FDK_BITSTREAM bs, } -static void CTns_Filter (FIXP_DBL *spec, int size, int inc, FIXP_TCC coeff [], int order) -{ - // - Simple all-pole filter of order "order" defined by - // y(n) = x(n) - a(2)*y(n-1) - ... - a(order+1)*y(n-order) - // - // - The state variables of the filter are initialized to zero every time - // - // - The output data is written over the input data ("in-place operation") - // - // - An input vector of "size" samples is processed and the index increment - // to the next data sample is given by "inc" - - int i,j,N; - FIXP_DBL *pSpec; - FIXP_DBL maxVal=FL2FXCONST_DBL(0.0); - INT s; - - FDK_ASSERT(order <= TNS_MAXIMUM_ORDER); - C_ALLOC_SCRATCH_START(state, FIXP_DBL, TNS_MAXIMUM_ORDER); - FDKmemclear(state, order*sizeof(FIXP_DBL)); - - for (i=0; i FL2FXCONST_DBL(0.03125*0.70710678118) ) - s = fixMax(CntLeadingZeros(maxVal)-6,0); - else - s = fixMax(CntLeadingZeros(maxVal)-5,0); - - s = fixMin(s,2); - s = s-1; - - if (inc == -1) - pSpec = &spec[size - 1]; - else - pSpec = &spec[0]; - - FIXP_TCC *pCoeff; - -#define FIRST_PART_FLTR \ - FIXP_DBL x, *pState = state; \ - pCoeff = coeff; \ - \ - if (s < 0) \ - x = (pSpec [0]>>1) + fMultDiv2 (*pCoeff++, pState [0]) ; \ - else \ - x = (pSpec [0]<> s; \ - *pState =(-x) << 1; \ - pSpec += inc ; - - - if (order>8) - { - N = (order-1)&7; - - for (i = size ; i != 0 ; i--) - { - FIRST_PART_FLTR - - for (j = N; j > 0 ; j--) { INNER_FLTR_INLINE } - - INNER_FLTR_INLINE INNER_FLTR_INLINE INNER_FLTR_INLINE INNER_FLTR_INLINE - INNER_FLTR_INLINE INNER_FLTR_INLINE INNER_FLTR_INLINE INNER_FLTR_INLINE - - LAST_PART_FLTR - } - - } else if (order>4) { - - N = (order-1)&3; - - for (i = size ; i != 0 ; i--) - { - FIRST_PART_FLTR - for (j = N; j > 0 ; j--) { INNER_FLTR_INLINE } - - INNER_FLTR_INLINE INNER_FLTR_INLINE INNER_FLTR_INLINE INNER_FLTR_INLINE - - LAST_PART_FLTR - } - - } else { - - N = order-1; - - for (i = size ; i != 0 ; i--) - { - FIRST_PART_FLTR - - for (j = N; j > 0 ; j--) { INNER_FLTR_INLINE } - - LAST_PART_FLTR - } - } - - C_ALLOC_SCRATCH_END(state, FIXP_DBL, TNS_MAXIMUM_ORDER); -} - /*! \brief Apply tns to spectral lines @@ -338,75 +228,10 @@ void CTns_Apply ( const INT granuleLength ) { - int window,index,start,stop,size; - - - if (pTnsData->Active) - { - C_AALLOC_SCRATCH_START(coeff, FIXP_TCC, TNS_MAXIMUM_ORDER); - - for (window=0; window < GetWindowsPerFrame(pIcsInfo); window++) - { - FIXP_DBL *pSpectrum = SPEC(pSpectralCoefficient, window, granuleLength); - - for (index=0; index < pTnsData->NumberOfFilters[window]; index++) - { - CFilter *RESTRICT filter = &pTnsData->Filter[window][index]; - - if (filter->Order > 0) - { - FIXP_TCC *pCoeff; - int tns_max_bands; - - pCoeff = &coeff[filter->Order-1]; - if (filter->Resolution == 3) - { - int i; - for (i=0; i < filter->Order; i++) - *pCoeff-- = FDKaacDec_tnsCoeff3[filter->Coeff[i]+4]; - } - else - { - int i; - for (i=0; i < filter->Order; i++) - *pCoeff-- = FDKaacDec_tnsCoeff4[filter->Coeff[i]+8]; - } - - switch (granuleLength) { - case 480: - tns_max_bands = tns_max_bands_tbl_480[pSamplingRateInfo->samplingRateIndex]; - break; - case 512: - tns_max_bands = tns_max_bands_tbl_512[pSamplingRateInfo->samplingRateIndex]; - break; - default: - tns_max_bands = GetMaximumTnsBands(pIcsInfo, pSamplingRateInfo->samplingRateIndex); - break; - } - - start = fixMin( fixMin(filter->StartBand, tns_max_bands), - GetScaleFactorBandsTransmitted(pIcsInfo) ); - - start = GetScaleFactorBandOffsets(pIcsInfo, pSamplingRateInfo)[start]; - - stop = fixMin( fixMin(filter->StopBand, tns_max_bands), - GetScaleFactorBandsTransmitted(pIcsInfo) ); - - stop = GetScaleFactorBandOffsets(pIcsInfo, pSamplingRateInfo)[stop]; - - size = stop - start; - - if (size > 0) { - CTns_Filter(&pSpectrum[start], - size, - filter->Direction, - coeff, - filter->Order ); - } - } - } - } - C_AALLOC_SCRATCH_END(coeff, FIXP_TCC, TNS_MAXIMUM_ORDER); +#if 0 + if (pTnsData->Active) { + /** disabled */ + FDKprintf("TNS disabled\n"); } - +#endif } diff --git a/libAACdec/src/aacdecoder.cpp b/libAACdec/src/aacdecoder.cpp index 579e470..27dc727 100644 --- a/libAACdec/src/aacdecoder.cpp +++ b/libAACdec/src/aacdecoder.cpp @@ -144,18 +144,10 @@ amm-info@iis.fraunhofer.de #include "aacdec_pns.h" - #include "sbrdecoder.h" - - - - - #include "aacdec_hcr.h" - #include "rvlc.h" #include "tpdec_lib.h" -#include "conceal.h" #include "FDK_crc.h" @@ -181,7 +173,6 @@ void CAacDecoder_SyncQmfMode(HANDLE_AACDECODER self) /* Set SBR to current QMF mode. Error does not matter. */ - sbrDecoder_SetParam(self->hSbrDecoder, SBR_QMF_MODE, (self->qmfModeCurr == MODE_LP)); self->psPossible = ((CAN_DO_PS(self->streamInfo.aot) && self->streamInfo.aacNumChannels == 1 && ! (self->flags & AC_MPS_PRESENT))) && self->qmfModeCurr == MODE_HQ ; FDK_ASSERT( ! ( (self->flags & AC_MPS_PRESENT) && self->psPossible ) ); } @@ -514,39 +505,8 @@ AAC_DECODER_ERROR CAacDecoder_ExtPayloadParse (HANDLE_AACDECODER self, crcFlag = 1; case EXT_SBR_DATA: if (IS_CHANNEL_ELEMENT(previous_element)) { - SBR_ERROR sbrError; - CAacDecoder_SyncQmfMode(self); - sbrError = sbrDecoder_InitElement( - self->hSbrDecoder, - self->streamInfo.aacSampleRate, - self->streamInfo.extSamplingRate, - self->streamInfo.aacSamplesPerFrame, - self->streamInfo.aot, - previous_element, - elIndex - ); - - if (sbrError == SBRDEC_OK) { - sbrError = sbrDecoder_Parse ( - self->hSbrDecoder, - hBs, - count, - *count, - crcFlag, - previous_element, - elIndex, - self->flags & AC_INDEP ); - /* Enable SBR for implicit SBR signalling but only if no severe error happend. */ - if ( (sbrError == SBRDEC_OK) - || (sbrError == SBRDEC_PARSE_ERROR) ) { - self->sbrEnabled = 1; - } - } else { - /* Do not try to apply SBR because initializing the element failed. */ - self->sbrEnabled = 0; - } /* Citation from ISO/IEC 14496-3 chapter 4.5.2.1.5.2 Fill elements containing an extension_payload() with an extension_type of EXT_SBR_DATA or EXT_SBR_DATA_CRC shall not contain any other extension_payload of any other extension_type. @@ -556,9 +516,7 @@ AAC_DECODER_ERROR CAacDecoder_ExtPayloadParse (HANDLE_AACDECODER self, *count = 0; } else { /* If this is not a fill element with a known length, we are screwed and further parsing makes no sense. */ - if (sbrError != SBRDEC_OK) { - self->frameOK = 0; - } + self->frameOK = 0; } } else { error = AAC_DEC_PARSE_ERROR; @@ -744,9 +702,6 @@ LINKSPEC_CPP HANDLE_AACDECODER CAacDecoder_Open(TRANSPORT_TYPE bsFormat) /*!< /* initialize stream info */ CStreamInfoInit(&self->streamInfo); - /* initialize error concealment common data */ - CConcealment_InitCommonData(&self->concealCommonData); - self->hDrcInfo = GetDrcInfo(); if (self->hDrcInfo == NULL) { goto bail; @@ -757,7 +712,7 @@ LINKSPEC_CPP HANDLE_AACDECODER CAacDecoder_Open(TRANSPORT_TYPE bsFormat) /*!< aacDecoder_drcSetParam ( self->hDrcInfo, DRC_BS_DELAY, - CConcealment_GetDelay(&self->concealCommonData) + 0 ); @@ -835,18 +790,15 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_Init(HANDLE_AACDECODER self, const CS case AOT_AAC_LC: self->streamInfo.profile = 1; - case AOT_ER_AAC_SCAL: - if (asc->m_sc.m_gaSpecificConfig.m_layer > 0) { - /* aac_scalable_extension_element() currently not supported. */ - return AAC_DEC_UNSUPPORTED_FORMAT; - } + case AOT_PS: + break; + case AOT_DRM_AAC: case AOT_SBR: - case AOT_PS: + case AOT_ER_AAC_SCAL: case AOT_ER_AAC_LD: case AOT_ER_AAC_ELD: - case AOT_DRM_AAC: - break; + return AAC_DEC_UNSUPPORTED_FORMAT; default: return AAC_DEC_UNSUPPORTED_AOT; @@ -952,7 +904,6 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_Init(HANDLE_AACDECODER self, const CS self->streamInfo.extSamplingRate = asc->m_extensionSamplingFrequency; self->flags |= (asc->m_sbrPresentFlag) ? AC_SBR_PRESENT : 0; self->flags |= (asc->m_psPresentFlag) ? AC_PS_PRESENT : 0; - self->sbrEnabled = 0; /* --------- vcb11 ------------ */ self->flags |= (asc->m_vcb11Flag) ? AC_ER_VCB11 : 0; @@ -964,11 +915,7 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_Init(HANDLE_AACDECODER self, const CS self->flags |= (asc->m_hcrFlag) ? AC_ER_HCR : 0; if (asc->m_aot == AOT_ER_AAC_ELD) { - self->flags |= AC_ELD; - self->flags |= (asc->m_sbrPresentFlag) ? AC_SBR_PRESENT : 0; /* Need to set the SBR flag for backward-compatibility - reasons. Even if SBR is not supported. */ - self->flags |= (asc->m_sc.m_eldSpecificConfig.m_sbrCrcFlag) ? AC_SBRCRC : 0; - self->flags |= (asc->m_sc.m_eldSpecificConfig.m_useLdQmfTimeAlign) ? AC_LD_MPS : 0; + return AAC_DEC_UNSUPPORTED_ER_FORMAT; } self->flags |= (asc->m_aot == AOT_ER_AAC_LD) ? AC_LD : 0; self->flags |= (asc->m_epConfig >= 0) ? AC_ER : 0; @@ -981,10 +928,6 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_Init(HANDLE_AACDECODER self, const CS } - if (asc->m_sbrPresentFlag) { - self->sbrEnabled = 1; - self->sbrEnabledPrev = 1; - } if (asc->m_psPresentFlag) { self->flags |= AC_PS_PRESENT; } @@ -1059,9 +1002,6 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_Init(HANDLE_AACDECODER self, const CS /* Make allocated channel count persistent in decoder context. */ self->aacChannels = ascChannels; } - - HcrInitRom(&self->aacCommonData.overlay.aac.erHcrInfo); - setHcrType(&self->aacCommonData.overlay.aac.erHcrInfo, ID_SCE); } /* Make amount of signalled channels persistent in decoder context. */ @@ -1091,12 +1031,6 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_Init(HANDLE_AACDECODER self, const CS /* Reset DRC control data for this channel */ aacDecoder_drcInitChannelData ( &self->pAacDecoderStaticChannelInfo[ch]->drcData ); - - /* Reset concealment only if ASC changed. Otherwise it will be done with any config callback. - E.g. every time the LATM SMC is present. */ - CConcealment_InitChannelData(&self->pAacDecoderStaticChannelInfo[ch]->concealmentInfo, - &self->concealCommonData, - self->streamInfo.aacSamplesPerFrame ); } } @@ -1191,10 +1125,6 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame( int ch; /* Clear history */ for (ch = 0; ch < self->aacChannels; ch++) { - /* Reset concealment */ - CConcealment_InitChannelData(&self->pAacDecoderStaticChannelInfo[ch]->concealmentInfo, - &self->concealCommonData, - self->streamInfo.aacSamplesPerFrame ); /* Clear overlap-add buffers to avoid clicks. */ FDKmemclear(self->pAacDecoderStaticChannelInfo[ch]->pOverlapBuffer, OverlapBufferSize*sizeof(FIXP_DBL)); } @@ -1222,9 +1152,6 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame( else type = self->elements[element_count]; - setHcrType(&self->aacCommonData.overlay.aac.erHcrInfo, type); - - if ((INT)FDKgetValidBits(bs) < 0) self->frameOK = 0; @@ -1306,27 +1233,6 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame( else { self->frameOK = 0; } - /* Create SBR element for SBR for upsampling for LFE elements, - and if SBR was explicitly signaled, because the first frame(s) - may not contain SBR payload (broken encoder, bit errors). */ - if ( (self->flags & AC_SBR_PRESENT) || (self->sbrEnabled == 1) ) - { - SBR_ERROR sbrError; - - sbrError = sbrDecoder_InitElement( - self->hSbrDecoder, - self->streamInfo.aacSampleRate, - self->streamInfo.extSamplingRate, - self->streamInfo.aacSamplesPerFrame, - self->streamInfo.aot, - type, - previous_element_index - ); - if (sbrError != SBRDEC_OK) { - /* Do not try to apply SBR because initializing the element failed. */ - self->sbrEnabled = 0; - } - } } el_cnt[type]++; @@ -1497,42 +1403,12 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame( if ( (bitCnt > 0) && (self->flags & AC_SBR_PRESENT) && (self->flags & (AC_USAC|AC_RSVD50|AC_ELD|AC_DRM)) ) { - SBR_ERROR err = SBRDEC_OK; - int elIdx, numChElements = el_cnt[ID_SCE] + el_cnt[ID_CPE]; - - for (elIdx = 0; elIdx < numChElements; elIdx += 1) - { - err = sbrDecoder_Parse ( - self->hSbrDecoder, - bs, - &bitCnt, - -1, - self->flags & AC_SBRCRC, - self->elements[elIdx], - elIdx, - self->flags & AC_INDEP ); - - if (err != SBRDEC_OK) { - break; - } - } - switch (err) { - case SBRDEC_PARSE_ERROR: - /* Can not go on parsing because we do not - know the length of the SBR extension data. */ - FDKpushFor(bs, bitCnt); - bitCnt = 0; - break; - case SBRDEC_OK: - self->sbrEnabled = 1; - break; - default: - self->frameOK = 0; - break; - } + /* Can not go on parsing because we do not + know the length of the SBR extension data. */ + FDKpushFor(bs, bitCnt); + bitCnt = 0; } - if (self->flags & AC_DRM) { if ((bitCnt = (INT)FDKgetValidBits(bs)) != 0) { @@ -1630,13 +1506,11 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame( self->aacChannelsPrev = aacChannels; /* store */ FDKmemcpy(self->channelTypePrev, self->channelType, (8)*sizeof(AUDIO_CHANNEL_TYPE)); /* store */ FDKmemcpy(self->channelIndicesPrev, self->channelIndices, (8)*sizeof(UCHAR)); /* store */ - self->sbrEnabledPrev = self->sbrEnabled; } else { if (self->aacChannels > 0) { aacChannels = self->aacChannelsPrev; /* restore */ FDKmemcpy(self->channelType, self->channelTypePrev, (8)*sizeof(AUDIO_CHANNEL_TYPE)); /* restore */ FDKmemcpy(self->channelIndices, self->channelIndicesPrev, (8)*sizeof(UCHAR)); /* restore */ - self->sbrEnabled = self->sbrEnabledPrev; } } @@ -1720,20 +1594,6 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame( FDKmemclear(pAacDecoderChannelInfo->pSpectralCoefficient, sizeof(FIXP_DBL)*self->streamInfo.aacSamplesPerFrame); } - /* - Conceal defective spectral data - */ - CConcealment_Apply(&self->pAacDecoderStaticChannelInfo[c]->concealmentInfo, - pAacDecoderChannelInfo, - self->pAacDecoderStaticChannelInfo[c], - &self->samplingRateInfo, - self->streamInfo.aacSamplesPerFrame, - 0, - (self->frameOK && !(flags&AACDEC_CONCEAL)), - self->flags - ); - - if (flags & (AACDEC_INTR|AACDEC_CLRHIST)) { /* Reset DRC control data for this channel */ aacDecoder_drcInitChannelData ( &self->pAacDecoderStaticChannelInfo[c]->drcData ); @@ -1743,13 +1603,11 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame( /* DRC processing */ aacDecoder_drcApply ( self->hDrcInfo, - self->hSbrDecoder, pAacDecoderChannelInfo, &self->pAacDecoderStaticChannelInfo[c]->drcData, self->extGain, c, - self->streamInfo.aacSamplesPerFrame, - self->sbrEnabled + self->streamInfo.aacSamplesPerFrame ); switch (pAacDecoderChannelInfo->renderMode) @@ -1798,9 +1656,6 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame( ); } - /* Add additional concealment delay */ - self->streamInfo.outputDelay += CConcealment_GetDelay(&self->concealCommonData) * self->streamInfo.aacSamplesPerFrame; - /* Map DRC data to StreamInfo structure */ aacDecoder_drcGetInfo ( self->hDrcInfo, diff --git a/libAACdec/src/aacdecoder.h b/libAACdec/src/aacdecoder.h index 25bc35d..d38d6e2 100644 --- a/libAACdec/src/aacdecoder.h +++ b/libAACdec/src/aacdecoder.h @@ -105,9 +105,6 @@ amm-info@iis.fraunhofer.de #include "genericStds.h" -#include "sbrdecoder.h" - - #include "aacdec_drc.h" #include "pcmutils_lib.h" @@ -154,11 +151,6 @@ typedef enum { MODE_LP = 1 } QMF_MODE; -typedef struct { - int bsDelay; -} SBR_PARAMS; - - /* AAC decoder (opaque toward userland) struct declaration */ struct AAC_DECODER_INSTANCE { INT aacChannels; /*!< Amount of AAC decoder channels allocated. */ @@ -197,18 +189,12 @@ struct AAC_DECODER_INSTANCE { CAacDecoderCommonData aacCommonData; /*!< Temporal shared data for all channels hooked into pAacDecoderChannelInfo */ - CConcealParams concealCommonData; - INT aacChannelsPrev; /*!< The amount of AAC core channels of the last successful decode call. */ AUDIO_CHANNEL_TYPE channelTypePrev[(8)]; /*!< Array holding the channelType values of the last successful decode call. */ UCHAR channelIndicesPrev[(8)]; /*!< Array holding the channelIndices values of the last successful decode call. */ - HANDLE_SBRDECODER hSbrDecoder; /*!< SBR decoder handle. */ - UCHAR sbrEnabled; /*!< flag to store if SBR has been detected */ - UCHAR sbrEnabledPrev; /*!< flag to store if SBR has been detected from previous frame */ UCHAR psPossible; /*!< flag to store if PS is possible */ - SBR_PARAMS sbrParams; /*!< struct to store all sbr parameters */ QMF_MODE qmfModeCurr; /*!< The current QMF mode */ QMF_MODE qmfModeUser; /*!< The QMF mode requested by the library user */ diff --git a/libAACdec/src/aacdecoder_lib.cpp b/libAACdec/src/aacdecoder_lib.cpp index 50efb0f..cb985fb 100644 --- a/libAACdec/src/aacdecoder_lib.cpp +++ b/libAACdec/src/aacdecoder_lib.cpp @@ -96,13 +96,10 @@ amm-info@iis.fraunhofer.de #include "FDK_core.h" /* FDK_tools version info */ - #include "sbrdecoder.h" -#include "conceal.h" - #include "aacdec_drc.h" @@ -207,17 +204,6 @@ static INT aacDecoder_ConfigCallback(void *handle, const CSAudioSpecificConfig * } } if (err == AAC_DEC_OK) { - if ( self->flags & (AC_USAC|AC_RSVD50|AC_LD|AC_ELD) - && CConcealment_GetDelay(&self->concealCommonData) > 0 ) - { - /* Revert to error concealment method Noise Substitution. - Because interpolation is not implemented for USAC/RSVD50 or - the additional delay is unwanted for low delay codecs. */ - setConcealMethod(self, 1); -#ifdef DEBUG - FDKprintf(" Concealment method was reverted to 1 !\n"); -#endif - } errTp = TRANSPORTDEC_OK; } else { if (IS_INIT_ERROR(err)) { @@ -263,75 +249,20 @@ setConcealMethod ( const HANDLE_AACDECODER self, /*!< Handle of the decoder i const INT method ) { AAC_DECODER_ERROR errorStatus = AAC_DEC_OK; - CConcealParams *pConcealData = NULL; - HANDLE_SBRDECODER hSbrDec = NULL; HANDLE_AAC_DRC hDrcInfo = NULL; HANDLE_PCM_DOWNMIX hPcmDmx = NULL; - CConcealmentMethod backupMethod = ConcealMethodNone; - int backupDelay = 0; - int bsDelay = 0; /* check decoder handle */ if (self != NULL) { - pConcealData = &self->concealCommonData; - hSbrDec = self->hSbrDecoder; hDrcInfo = self->hDrcInfo; hPcmDmx = self->hPcmUtils; } - - /* Get current method/delay */ - backupMethod = CConcealment_GetMethod(pConcealData); - backupDelay = CConcealment_GetDelay(pConcealData); - - /* Be sure to set AAC and SBR concealment method simultaneously! */ - errorStatus = - CConcealment_SetParams( - pConcealData, - (int)method, // concealMethod - AACDEC_CONCEAL_PARAM_NOT_SPECIFIED, // concealFadeOutSlope - AACDEC_CONCEAL_PARAM_NOT_SPECIFIED, // concealFadeInSlope - AACDEC_CONCEAL_PARAM_NOT_SPECIFIED, // concealMuteRelease - AACDEC_CONCEAL_PARAM_NOT_SPECIFIED // concealComfNoiseLevel - ); - if ( (errorStatus != AAC_DEC_OK) - && (errorStatus != AAC_DEC_INVALID_HANDLE) ) { - goto bail; - } - - /* Get new delay */ - bsDelay = CConcealment_GetDelay(pConcealData); - - { - SBR_ERROR sbrErr = SBRDEC_OK; - - /* set SBR bitstream delay */ - sbrErr = sbrDecoder_SetParam ( - hSbrDec, - SBR_SYSTEM_BITSTREAM_DELAY, - bsDelay - ); - - switch (sbrErr) { - case SBRDEC_OK: - case SBRDEC_NOT_INITIALIZED: - if (self != NULL) { - /* save the param value and set later - (when SBR has been initialized) */ - self->sbrParams.bsDelay = bsDelay; - } - break; - default: - errorStatus = AAC_DEC_SET_PARAM_FAIL; - goto bail; - } - } - errorStatus = aacDecoder_drcSetParam ( hDrcInfo, DRC_BS_DELAY, - bsDelay + 0 ); if ( (errorStatus != AAC_DEC_OK) && (errorStatus != AAC_DEC_INVALID_HANDLE) ) { @@ -343,7 +274,7 @@ setConcealMethod ( const HANDLE_AACDECODER self, /*!< Handle of the decoder i pcmDmx_SetParam ( hPcmDmx, DMX_BS_DATA_DELAY, - bsDelay + 0 ); switch (err) { case PCMDMX_INVALID_HANDLE: @@ -361,32 +292,17 @@ bail: if ( (errorStatus != AAC_DEC_OK) && (errorStatus != AAC_DEC_INVALID_HANDLE) ) { - /* Revert to the initial state */ - CConcealment_SetParams ( - pConcealData, - (int)backupMethod, - AACDEC_CONCEAL_PARAM_NOT_SPECIFIED, - AACDEC_CONCEAL_PARAM_NOT_SPECIFIED, - AACDEC_CONCEAL_PARAM_NOT_SPECIFIED, - AACDEC_CONCEAL_PARAM_NOT_SPECIFIED - ); - /* Revert SBR bitstream delay */ - sbrDecoder_SetParam ( - hSbrDec, - SBR_SYSTEM_BITSTREAM_DELAY, - backupDelay - ); /* Revert DRC bitstream delay */ aacDecoder_drcSetParam ( hDrcInfo, DRC_BS_DELAY, - backupDelay + 0 ); /* Revert PCM mixdown bitstream delay */ pcmDmx_SetParam ( hPcmDmx, DMX_BS_DATA_DELAY, - backupDelay + 0 ); } @@ -400,14 +316,12 @@ aacDecoder_SetParam ( const HANDLE_AACDECODER self, /*!< Handle of the decode const INT value) /*!< Parameter valued */ { AAC_DECODER_ERROR errorStatus = AAC_DEC_OK; - CConcealParams *pConcealData = NULL; HANDLE_AAC_DRC hDrcInfo = NULL; HANDLE_PCM_DOWNMIX hPcmDmx = NULL; TDLimiterPtr hPcmTdl = NULL; /* check decoder handle */ if (self != NULL) { - pConcealData = &self->concealCommonData; hDrcInfo = self->hDrcInfo; hPcmDmx = self->hPcmUtils; hPcmTdl = self->hLimiter; @@ -666,14 +580,7 @@ LINKSPEC_CPP HANDLE_AACDECODER aacDecoder_Open(TRANSPORT_TYPE transportFmt, UINT /* Register Config Update callback. */ transportDec_RegisterAscCallback(pIn, aacDecoder_ConfigCallback, (void*)aacDec); - /* open SBR decoder */ - if ( SBRDEC_OK != sbrDecoder_Open ( &aacDec->hSbrDecoder )) { - err = -1; - goto bail; - } aacDec->qmfModeUser = NOT_DEFINED; - transportDec_RegisterSbrCallback(aacDec->hInput, (cbSbr_t)sbrDecoder_Header, (void*)aacDec->hSbrDecoder); - pcmDmx_Open( &aacDec->hPcmUtils ); if (aacDec->hPcmUtils == NULL) { @@ -692,7 +599,7 @@ LINKSPEC_CPP HANDLE_AACDECODER aacDecoder_Open(TRANSPORT_TYPE transportFmt, UINT /* Assure that all modules have same delay */ - if ( setConcealMethod(aacDec, CConcealment_GetMethod(&aacDec->concealCommonData)) ) { + if ( setConcealMethod(aacDec, 0) ) { err = -1; goto bail; } @@ -736,10 +643,6 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_Fill( static void aacDecoder_SignalInterruption(HANDLE_AACDECODER self) { CAacDecoder_SignalInterruption(self); - - if ( self->hSbrDecoder != NULL ) { - sbrDecoder_SetParam(self->hSbrDecoder, SBR_BS_INTERRUPTION, 0); - } } static void aacDecoder_UpdateBitStreamCounters(CStreamInfo *pSi, HANDLE_FDK_BITSTREAM hBs, int nBits, AAC_DECODER_ERROR ErrorStatus) @@ -846,7 +749,6 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame( /* Signal bit stream interruption to other modules if required. */ if ( fTpInterruption || (flags & (AACDEC_INTR|AACDEC_CLRHIST)) ) { - sbrDecoder_SetParam(self->hSbrDecoder, SBR_CLEAR_HISTORY, (flags&AACDEC_CLRHIST)); aacDecoder_SignalInterruption(self); if ( ! (flags & AACDEC_INTR) ) { ErrorStatus = AAC_DEC_TRANSPORT_SYNC_ERROR; @@ -909,84 +811,11 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame( /* sbr decoder */ - if (ErrorStatus || (flags & AACDEC_CONCEAL) || self->pAacDecoderStaticChannelInfo[0]->concealmentInfo.concealState > ConcealState_FadeIn) + if (ErrorStatus || (flags & AACDEC_CONCEAL)) { self->frameOK = 0; /* if an error has occured do concealment in the SBR decoder too */ } - if (self->sbrEnabled) - { - SBR_ERROR sbrError = SBRDEC_OK; - int chIdx, numCoreChannel = self->streamInfo.numChannels; - int chOutMapIdx = ((self->chMapIndex==0) && (numCoreChannel<7)) ? numCoreChannel : self->chMapIndex; - - /* set params */ - sbrDecoder_SetParam ( self->hSbrDecoder, - SBR_SYSTEM_BITSTREAM_DELAY, - self->sbrParams.bsDelay); - sbrDecoder_SetParam ( self->hSbrDecoder, - SBR_FLUSH_DATA, - (flags & AACDEC_FLUSH) ); - - if ( self->streamInfo.aot == AOT_ER_AAC_ELD ) { - /* Configure QMF */ - sbrDecoder_SetParam ( self->hSbrDecoder, - SBR_LD_QMF_TIME_ALIGN, - (self->flags & AC_LD_MPS) ? 1 : 0 ); - } - - { - PCMDMX_ERROR dmxErr; - INT maxOutCh = 0; - - dmxErr = pcmDmx_GetParam(self->hPcmUtils, MAX_NUMBER_OF_OUTPUT_CHANNELS, &maxOutCh); - if ( (dmxErr == PCMDMX_OK) && (maxOutCh == 1) ) { - /* Disable PS processing if we have to create a mono output signal. */ - self->psPossible = 0; - } - } - - - /* apply SBR processing */ - sbrError = sbrDecoder_Apply ( self->hSbrDecoder, - pTimeData, - &self->streamInfo.numChannels, - &self->streamInfo.sampleRate, - self->channelOutputMapping[chOutMapIdx], - interleaved, - self->frameOK, - &self->psPossible); - - - if (sbrError == SBRDEC_OK) { - #define UPS_SCALE 2 /* Maximum upsampling factor is 4 (CELP+SBR) */ - FIXP_DBL upsampleFactor = FL2FXCONST_DBL(1.0f/(1<flags |= AC_SBR_PRESENT; - if (self->streamInfo.aacSampleRate != self->streamInfo.sampleRate) { - if (self->streamInfo.frameSize == 768) { - upsampleFactor = FL2FXCONST_DBL(8.0f/(3<streamInfo.frameSize = (INT)fMult((FIXP_DBL)self->streamInfo.aacSamplesPerFrame<streamInfo.outputDelay = (UINT)(INT)fMult((FIXP_DBL)self->streamInfo.outputDelay<streamInfo.outputDelay += sbrDecoder_GetDelay( self->hSbrDecoder ); - - if (self->psPossible) { - self->flags |= AC_PS_PRESENT; - } - for (chIdx = numCoreChannel; chIdx < self->streamInfo.numChannels; chIdx+=1) { - self->channelType[chIdx] = ACT_FRONT; - self->channelIndices[chIdx] = chIdx; - } - } - } - - { INT pcmLimiterScale = 0; PCMDMX_ERROR dmxErr = PCMDMX_OK; @@ -1084,10 +913,6 @@ LINKSPEC_CPP void aacDecoder_Close ( HANDLE_AACDECODER self ) - if (self->hSbrDecoder != NULL) { - sbrDecoder_Close(&self->hSbrDecoder); - } - if (self->hInput != NULL) { transportDec_Close(&self->hInput); } @@ -1109,7 +934,6 @@ LINKSPEC_CPP INT aacDecoder_GetLibInfo ( LIB_INFO *info ) return -1; } - sbrDecoder_GetLibInfo( info ); transportDec_GetLibInfo( info ); FDK_toolsGetLibInfo( info ); pcmDmx_GetLibInfo( info ); diff --git a/libAACdec/src/block.cpp b/libAACdec/src/block.cpp index bda565c..0ab3d5c 100644 --- a/libAACdec/src/block.cpp +++ b/libAACdec/src/block.cpp @@ -97,8 +97,6 @@ amm-info@iis.fraunhofer.de -#include "aacdec_hcr.h" -#include "rvlc.h" #if defined(__arm__) @@ -280,9 +278,7 @@ AAC_DECODER_ERROR CBlock_ReadSectionData(HANDLE_FDK_BITSTREAM bs, UCHAR sect_cb; UCHAR *pCodeBook = pAacDecoderChannelInfo->pDynData->aCodeBook; /* HCR input (long) */ - SHORT *pNumLinesInSec = pAacDecoderChannelInfo->pDynData->specificTo.aac.aNumLineInSec4Hcr; int numLinesInSecIdx = 0; - UCHAR *pHcrCodeBook = pAacDecoderChannelInfo->pDynData->specificTo.aac.aCodeBooks4Hcr; const SHORT *BandOffsets = GetScaleFactorBandOffsets(&pAacDecoderChannelInfo->icsInfo, pSamplingRateInfo); pAacDecoderChannelInfo->pDynData->specificTo.aac.numberSection = 0; AAC_DECODER_ERROR ErrorStatus = AAC_DEC_OK; @@ -326,19 +322,8 @@ AAC_DECODER_ERROR CBlock_ReadSectionData(HANDLE_FDK_BITSTREAM bs, top = band + sect_len; if (flags & AC_ER_HCR) { - /* HCR input (long) -- collecting sideinfo (for HCR-_long_ only) */ - if (numLinesInSecIdx >= MAX_SFB_HCR) { - return AAC_DEC_PARSE_ERROR; - } - pNumLinesInSec[numLinesInSecIdx] = BandOffsets[top] - BandOffsets[band]; - numLinesInSecIdx++; - if (sect_cb == BOOKSCL) - { - return AAC_DEC_INVALID_CODE_BOOK; - } else { - *pHcrCodeBook++ = sect_cb; - } - pAacDecoderChannelInfo->pDynData->specificTo.aac.numberSection++; + /* HCR disabled */ + return AAC_DEC_PARSE_ERROR; } /* Check spectral line limits */ @@ -586,38 +571,11 @@ AAC_DECODER_ERROR CBlock_ReadSpectralData(HANDLE_FDK_BITSTREAM bs, } /* plain huffman decoding (short) finished */ } - /* HCR - Huffman Codeword Reordering short */ else /* if ( flags & AC_ER_HCR ) */ { - H_HCR_INFO hHcr = &pAacDecoderChannelInfo->pComData->overlay.aac.erHcrInfo; - int hcrStatus = 0; - - /* advanced Huffman decoding starts here (HCR decoding :) */ - if ( pAacDecoderChannelInfo->pDynData->specificTo.aac.lenOfReorderedSpectralData != 0 ) { - - /* HCR initialization short */ - hcrStatus = HcrInit(hHcr, pAacDecoderChannelInfo, pSamplingRateInfo, bs); - - if (hcrStatus != 0) { - return AAC_DEC_DECODE_FRAME_ERROR; - } - - /* HCR decoding short */ - hcrStatus = HcrDecoder(hHcr, pAacDecoderChannelInfo, pSamplingRateInfo, bs); - - if (hcrStatus != 0) { -#if HCR_ERROR_CONCEALMENT - HcrMuteErroneousLines(hHcr); -#else - return AAC_DEC_DECODE_FRAME_ERROR; -#endif /* HCR_ERROR_CONCEALMENT */ - } - - FDKpushFor (bs, pAacDecoderChannelInfo->pDynData->specificTo.aac.lenOfReorderedSpectralData); - } + /* HCR - Huffman Codeword Reordering disabled */ + return AAC_DEC_DECODE_FRAME_ERROR; } - /* HCR - Huffman Codeword Reordering short finished */ - if ( IsLongBlock(&pAacDecoderChannelInfo->icsInfo) && !(flags & (AC_ELD|AC_SCALABLE)) ) diff --git a/libAACdec/src/channel.cpp b/libAACdec/src/channel.cpp index 4b182e0..f0117ee 100644 --- a/libAACdec/src/channel.cpp +++ b/libAACdec/src/channel.cpp @@ -91,15 +91,10 @@ amm-info@iis.fraunhofer.de #include "channel.h" #include "aacdecoder.h" #include "block.h" -#include "aacdec_tns.h" #include "FDK_bitstream.h" #include "FDK_tools_rom.h" -#include "conceal.h" -#include "rvlc.h" - -#include "aacdec_hcr.h" static @@ -186,13 +181,6 @@ void CChannelElement_Decode( CAacDecoderChannelInfo *pAacDecoderChannelInfo[2], } } - - CRvlc_ElementCheck( - pAacDecoderChannelInfo, - pAacDecoderStaticChannelInfo, - flags, - el_channels - ); } void CChannel_CodebookTableInit(CAacDecoderChannelInfo *pAacDecoderChannelInfo) @@ -331,7 +319,7 @@ AAC_DECODER_ERROR CChannelElement_Read(HANDLE_FDK_BITSTREAM hBs, case scale_factor_data: if (flags & AC_ER_RVLC) { /* read RVLC data from bitstream (error sens. cat. 1) */ - CRvlc_Read(pAacDecoderChannelInfo[ch], hBs); + error = AAC_DEC_DECODE_FRAME_ERROR; } else { @@ -369,17 +357,13 @@ AAC_DECODER_ERROR CChannelElement_Read(HANDLE_FDK_BITSTREAM hBs, case esc2_rvlc: if (flags & AC_ER_RVLC) { - CRvlc_Decode( - pAacDecoderChannelInfo[ch], - pAacDecoderStaticChannelInfo[ch], - hBs - ); + error = AAC_DEC_UNSUPPORTED_FORMAT; } break; case esc1_hcr: if (flags & AC_ER_HCR) { - CHcr_Read(hBs, pAacDecoderChannelInfo[ch] ); + error = AAC_DEC_UNSUPPORTED_FORMAT; } break; diff --git a/libAACdec/src/channelinfo.h b/libAACdec/src/channelinfo.h index e092ab3..54234a3 100644 --- a/libAACdec/src/channelinfo.h +++ b/libAACdec/src/channelinfo.h @@ -105,11 +105,8 @@ amm-info@iis.fraunhofer.de #include "aacdec_pns.h" -#include "aacdec_hcr_types.h" -#include "rvlc_info.h" -#include "conceal_types.h" #include "aacdec_drc_types.h" @@ -193,7 +190,6 @@ typedef struct CDrcChannelData drcData; - CConcealmentInfo concealmentInfo; } CAacDecoderStaticChannelInfo; @@ -214,8 +210,6 @@ typedef struct { struct { CPulseData PulseData; - SHORT aNumLineInSec4Hcr[MAX_SFB_HCR]; /* needed once for all channels except for Drm syntax */ - UCHAR aCodeBooks4Hcr[MAX_SFB_HCR]; /* needed once for all channels except for Drm syntax. Same as "aCodeBook" ? */ SHORT lenOfReorderedSpectralData; SCHAR lenOfLongestCodeword; SCHAR numberSection; @@ -246,18 +240,6 @@ typedef struct { INT pnsRandomSeed[(8*16)]; CJointStereoData jointStereoData; /* One for one element */ - - shouldBeUnion { - struct { - CErHcrInfo erHcrInfo; - CErRvlcInfo erRvlcInfo; - SHORT aRvlcScfEsc[RVLC_MAX_SFB]; /* needed once for all channels */ - SHORT aRvlcScfFwd[RVLC_MAX_SFB]; /* needed once for all channels */ - SHORT aRvlcScfBwd[RVLC_MAX_SFB]; /* needed once for all channels */ - } aac; - - } overlay; - } CAacDecoderCommonData; @@ -440,11 +422,5 @@ inline UCHAR GetScaleFactorBandsTotal(const CIcsInfo *pIcsInfo) return pIcsInfo->TotalSfBands; } -/* Note: This function applies to AAC-LC only ! */ -inline UCHAR GetMaximumTnsBands(const CIcsInfo *pIcsInfo, const int samplingRateIndex) -{ - return tns_max_bands_tbl[samplingRateIndex][!IsLongBlock(pIcsInfo)]; -} - #endif /* #ifndef CHANNELINFO_H */ diff --git a/libAACdec/src/conceal.cpp b/libAACdec/src/conceal.cpp deleted file mode 100644 index 1c313ef..0000000 --- a/libAACdec/src/conceal.cpp +++ /dev/null @@ -1,1866 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/***************************** MPEG-4 AAC Decoder ************************** - - Author(s): Josef Hoepfl - Description: independent channel concealment - -******************************************************************************/ - -/*! - \page concealment AAC core concealment - - This AAC core implementation includes a concealment function, which can be enabled - using the several defines during compilation. - - There are various tests inside the core, starting with simple CRC tests and ending in - a variety of plausibility checks. If such a check indicates an invalid bitstream, then - concealment is applied. - - Concealment is also applied when the calling main program indicates a distorted or missing - data frame using the frameOK flag. This is used for error detection on the transport layer. - (See below) - - There are three concealment-modes: - - 1) Muting: The spectral data is simply set to zero in case of an detected error. - - 2) Noise substitution: In case of an detected error, concealment copies the last frame and adds - attenuates the spectral data. For this mode you have to set the #CONCEAL_NOISE define. - Noise substitution adds no additional delay. - - 3) Interpolation: The interpolation routine swaps the spectral data from the previous and the - current frame just before the final frequency to time conversion. In case a single frame is - corrupted, concealmant interpolates between the last good and the first good frame to create - the spectral data for the missing frame. If multiple frames are corrupted, concealment - implements first a fade out based on slightly modified spectral values from the last good - frame. As soon as good frames are available, concealmant fades in the new spectral data. - For this mode you have to set the #CONCEAL_INTER define. Note that in this case, you also - need to set #SBR_BS_DELAY_ENABLE, which basically adds approriate delay in the SBR decoder. - Note that the Interpolating-Concealment increases the delay of your decoder by one frame - and that it does require additional resources such as memory and computational complexity. - -

How concealment can be used with errors on the transport layer

- - Many errors can or have to be detected on the transport layer. For example in IP based systems - packet loss can occur. The transport protocol used should indicate such packet loss by inserting - an empty frame with frameOK=0. -*/ - -#include "conceal.h" - -#include "aac_rom.h" -#include "genericStds.h" - - -/* PNS (of block) */ -#include "aacdec_pns.h" -#include "block.h" - -#include "FDK_tools_rom.h" - -#define CONCEAL_DFLT_COMF_NOISE_LEVEL ( 46 ) /* ~= -70 dB */ - - -/* default settings */ -#define CONCEAL_DFLT_FADEOUT_FRAMES ( 5 ) -#define CONCEAL_DFLT_FADEIN_FRAMES ( 5 ) -#define CONCEAL_DFLT_MUTE_RELEASE_FRAMES ( 3 ) - -#define CONCEAL_DFLT_FADE_FACTOR ( 0.707106781186548f ) /* 1/sqrt(2) */ - -/* some often used constants: */ -#define FIXP_ZERO FL2FXCONST_DBL(0.0f) -#define FIXP_ONE FL2FXCONST_DBL(1.0f) -#define FIXP_FL_CORRECTION FL2FXCONST_DBL(0.53333333333333333f) - -/* For parameter conversion */ -#define CONCEAL_PARAMETER_BITS ( 8 ) -#define CONCEAL_MAX_QUANT_FACTOR ( (1<method = ConcealMethodInter; - - pConcealCommonData->numFadeOutFrames = CONCEAL_DFLT_FADEOUT_FRAMES; - pConcealCommonData->numFadeInFrames = CONCEAL_DFLT_FADEIN_FRAMES; - pConcealCommonData->numMuteReleaseFrames = CONCEAL_DFLT_MUTE_RELEASE_FRAMES; - - pConcealCommonData->comfortNoiseLevel = CONCEAL_DFLT_COMF_NOISE_LEVEL; - - /* Init fade factors (symetric) */ - pConcealCommonData->fadeOutFactor[0] = FL2FXCONST_SGL( CONCEAL_DFLT_FADE_FACTOR ); - pConcealCommonData->fadeInFactor[0] = pConcealCommonData->fadeOutFactor[0]; - - for (i = 1; i < CONCEAL_MAX_NUM_FADE_FACTORS; i++) { - pConcealCommonData->fadeOutFactor[i] = FX_DBL2FX_SGL(fMult(pConcealCommonData->fadeOutFactor[i-1],FL2FXCONST_SGL(CONCEAL_DFLT_FADE_FACTOR))); - pConcealCommonData->fadeInFactor[i] = pConcealCommonData->fadeOutFactor[i]; - } - } -} - - - -/*! - \brief Get current concealment method. - - \pConcealCommonData Pointer to common concealment data (for all channels) - - \return Concealment method. -*/ -CConcealmentMethod - CConcealment_GetMethod( CConcealParams *pConcealCommonData ) -{ - CConcealmentMethod method = ConcealMethodNone; - - if (pConcealCommonData != NULL) { - method = pConcealCommonData->method; - } - - return (method); -} - - -/*! - \brief Init concealment information for each channel - - The function initializes the concealment information. Two methods can be chosen: - 0 = interpolation method (adds delay) - 1 = noise substitution (no delay, low complexity) - - \return none -*/ -void - CConcealment_InitChannelData ( - CConcealmentInfo *pConcealChannelInfo, - CConcealParams *pConcealCommonData, - int samplesPerFrame ) -{ - int i; - - pConcealChannelInfo->pConcealParams = pConcealCommonData; - - FDKmemclear(pConcealChannelInfo->spectralCoefficient, 1024 * sizeof(FIXP_CNCL)); - - for (i = 0; i < 8; i++) { - pConcealChannelInfo->specScale[i] = 0; - } - - pConcealChannelInfo->iRandomPhase = 0; - - pConcealChannelInfo->windowSequence = 0; - pConcealChannelInfo->windowShape = 0; - - pConcealChannelInfo->prevFrameOk[0] = 1; - pConcealChannelInfo->prevFrameOk[1] = 1; - - pConcealChannelInfo->cntFadeFrames = 0; - pConcealChannelInfo->cntValidFrames = 0; - - pConcealChannelInfo->concealState = ConcealState_Ok; - -} - - -/*! - \brief Set error concealment parameters - - \concealParams - \method - \fadeOutSlope - \fadeInSlope - \muteRelease - \comfNoiseLevel - - \return none -*/ -AAC_DECODER_ERROR - CConcealment_SetParams ( - CConcealParams *concealParams, - int method, - int fadeOutSlope, - int fadeInSlope, - int muteRelease, - int comfNoiseLevel ) -{ - /* set concealment technique */ - if (method != AACDEC_CONCEAL_PARAM_NOT_SPECIFIED) { - switch ((CConcealmentMethod)method) - { - case ConcealMethodMute: - case ConcealMethodNoise: - case ConcealMethodInter: - /* Be sure to enable delay adjustment of SBR decoder! */ - if (concealParams == NULL) { - return AAC_DEC_INVALID_HANDLE; - } else { - /* set param */ - concealParams->method = (CConcealmentMethod)method; - } - break; - - default: - return AAC_DEC_SET_PARAM_FAIL; - } - } - - /* set number of frames for fade-out slope */ - if (fadeOutSlope != AACDEC_CONCEAL_PARAM_NOT_SPECIFIED) { - if ( (fadeOutSlope < CONCEAL_MAX_NUM_FADE_FACTORS) - && (fadeOutSlope >= 0) ) - { - if (concealParams == NULL) { - return AAC_DEC_INVALID_HANDLE; - } else { - /* set param */ - concealParams->numFadeOutFrames = fadeOutSlope; - } - } else { - return AAC_DEC_SET_PARAM_FAIL; - } - } - - /* set number of frames for fade-in slope */ - if (fadeInSlope != AACDEC_CONCEAL_PARAM_NOT_SPECIFIED) { - if ( (fadeInSlope < CONCEAL_MAX_NUM_FADE_FACTORS) - && (fadeInSlope >= 1) ) - { - if (concealParams == NULL) { - return AAC_DEC_INVALID_HANDLE; - } else { - /* set param */ - concealParams->numFadeInFrames = fadeInSlope; - } - } else { - return AAC_DEC_SET_PARAM_FAIL; - } - } - - /* set number of error-free frames after which the muting will be released */ - if (muteRelease != AACDEC_CONCEAL_PARAM_NOT_SPECIFIED) { - if ( (muteRelease < (CONCEAL_MAX_NUM_FADE_FACTORS<<1)) - && (muteRelease >= 0) ) - { - if (concealParams == NULL) { - return AAC_DEC_INVALID_HANDLE; - } else { - /* set param */ - concealParams->numMuteReleaseFrames = muteRelease; - } - } else { - return AAC_DEC_SET_PARAM_FAIL; - } - } - - /* set confort noise level which will be inserted while in state 'muting' */ - if (comfNoiseLevel != AACDEC_CONCEAL_PARAM_NOT_SPECIFIED) { - if ( (comfNoiseLevel < -1) - || (comfNoiseLevel > 127) ) { - return AAC_DEC_SET_PARAM_FAIL; - } - if (concealParams == NULL) { - return AAC_DEC_INVALID_HANDLE; - } else { - concealParams->comfortNoiseLevel = comfNoiseLevel; - } - } - - return (AAC_DEC_OK); -} - - -/*! - \brief Set fade-out/in attenuation factor vectors - - \concealParams - \fadeOutAttenuationVector - \fadeInAttenuationVector - - \return 0 if OK all other values indicate errors -*/ -AAC_DECODER_ERROR - CConcealment_SetAttenuation ( - CConcealParams *concealParams, - SHORT *fadeOutAttenuationVector, - SHORT *fadeInAttenuationVector ) -{ - if ( (fadeOutAttenuationVector == NULL) - && (fadeInAttenuationVector == NULL) ) { - return AAC_DEC_SET_PARAM_FAIL; - } - - /* Fade-out factors */ - if (fadeOutAttenuationVector != NULL) - { - int i; - - /* check quantized factors first */ - for (i = 0; i < CONCEAL_MAX_NUM_FADE_FACTORS; i++) { - if ((fadeOutAttenuationVector[i] < 0) || (fadeOutAttenuationVector[i] > CONCEAL_MAX_QUANT_FACTOR)) { - return AAC_DEC_SET_PARAM_FAIL; - } - } - if (concealParams == NULL) { - return AAC_DEC_INVALID_HANDLE; - } - - /* now dequantize factors */ - for (i = 0; i < CONCEAL_MAX_NUM_FADE_FACTORS; i++) - { - concealParams->fadeOutFactor[i] = - FX_DBL2FX_SGL( fLdPow( CONCEAL_MIN_ATTENUATION_FACTOR_025_LD, - 0, - (FIXP_DBL)((INT)(FL2FXCONST_DBL(1.0/2.0)>>(CONCEAL_PARAMETER_BITS-1)) * (INT)fadeOutAttenuationVector[i]), - CONCEAL_PARAMETER_BITS - ) - ); - } - } - - /* Fade-in factors */ - if (fadeInAttenuationVector != NULL) - { - int i; - - /* check quantized factors first */ - for (i = 0; i < CONCEAL_MAX_NUM_FADE_FACTORS; i++) { - if ((fadeInAttenuationVector[i] < 0) || (fadeInAttenuationVector[i] > CONCEAL_MAX_QUANT_FACTOR)) { - return AAC_DEC_SET_PARAM_FAIL; - } - } - if (concealParams == NULL) { - return AAC_DEC_INVALID_HANDLE; - } - - /* now dequantize factors */ - for (i = 0; i < CONCEAL_MAX_NUM_FADE_FACTORS; i++) - { - concealParams->fadeInFactor[i] = - FX_DBL2FX_SGL( fLdPow( CONCEAL_MIN_ATTENUATION_FACTOR_025_LD, - 0, - (FIXP_DBL)((INT)(FIXP_ONE>>CONCEAL_PARAMETER_BITS) * (INT)fadeInAttenuationVector[i]), - CONCEAL_PARAMETER_BITS - ) - ); - } - } - - return (AAC_DEC_OK); -} - - -/*! - \brief Get state of concealment module. - - \pConcealChannelInfo - - \return Concealment state. -*/ -CConcealmentState - CConcealment_GetState ( - CConcealmentInfo *pConcealChannelInfo - ) -{ - CConcealmentState state = ConcealState_Ok; - - if (pConcealChannelInfo != NULL) { - state = pConcealChannelInfo->concealState; - } - - return (state); -} - - -static void CConcealment_fakePnsData ( - CPnsData *pPnsData, - CIcsInfo *pIcsInfo, - const SamplingRateInfo *pSamplingRateInfo, - SHORT *pSpecScale, - SHORT *pScaleFactor, - const int level ) -{ - CPnsInterChannelData *pInterChannelData = pPnsData->pPnsInterChannelData; - - int pnsBand, band, group, win; - //int delta = 0; - int windowsPerFrame = GetWindowsPerFrame(pIcsInfo); - int refLevel = (windowsPerFrame > 1) ? 82 : 91; - - FDK_ASSERT(level >= 0 && level <= 127); - - for (win = 0; win < windowsPerFrame; win++) { - pSpecScale[win] = 31; - } - - /* fake ICS info if necessary */ - if (!IsValid(pIcsInfo)) { - pIcsInfo->WindowGroups = 1; - if (IsLongBlock(pIcsInfo)) { - pIcsInfo->TotalSfBands = pSamplingRateInfo->NumberOfScaleFactorBands_Long; - pIcsInfo->WindowGroupLength[0] = 1; - } - else { - pIcsInfo->TotalSfBands = pSamplingRateInfo->NumberOfScaleFactorBands_Short; - pIcsInfo->WindowGroupLength[0] = 8; - } - pIcsInfo->MaxSfBands = pIcsInfo->TotalSfBands; - } - - /* global activate PNS */ - pPnsData->PnsActive = 1; - /* set energy level */ - pPnsData->CurrentEnergy = fixMax( 0, refLevel - level ); - - /* - value: | Avg. RMS power | Avg. RMS power | - | specScale = 22 | specScale = 31 | - -------+----------------+----------------+ - 5 | | -99.0 dB - 15 | | -90.0 dB - 25 | | -89.7 dB - 35 | | -85.3 dB - ... | ... | ... - 45 | -69.9 dB | -70.0 dB - 50 | -62.2 dB | - 55 | -55.6 dB | -54.6 dB - 60 | -47.0 dB | - 65 | -39.5 dB | -39.5 dB - 70 | -31.9 dB | - 75 | -24.4 dB | -24.4 dB - 80 | -16.9 dB | - 85 | -9.4 dB (c) | -9.4 dB - 90 | -3.9 dB (c) | - 95 | | -2.1 dB - 100 | | -1.6 dB - 105 | | -1.4 dB - */ - - for (group=0; group < GetWindowGroups(pIcsInfo); group++) - { - for (band=0; band < GetScaleFactorBandsTransmitted(pIcsInfo); band++) - { - pnsBand = group * 16 + band; - - if (pnsBand >= NO_OFBANDS) { - return; - } - //pPnsData->CurrentEnergy += delta ; - pScaleFactor[pnsBand] = pPnsData->CurrentEnergy; - pInterChannelData->correlated[pnsBand] = 0; - pPnsData->pnsUsed[pnsBand] = 1; - } - } -} - - -/*! - \brief Store data for concealment techniques applied later - - Interface function to store data for different concealment strategies - - \return none - */ -void - CConcealment_Store ( - CConcealmentInfo *hConcealmentInfo, - CAacDecoderChannelInfo *pAacDecoderChannelInfo, - CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo ) -{ - if ( !(pAacDecoderChannelInfo->renderMode == AACDEC_RENDER_LPD - ) ) - { - FIXP_DBL *pSpectralCoefficient = SPEC_LONG(pAacDecoderChannelInfo->pSpectralCoefficient); - SHORT *pSpecScale = pAacDecoderChannelInfo->specScale; - CIcsInfo *pIcsInfo = &pAacDecoderChannelInfo->icsInfo; - - SHORT tSpecScale[8]; - UCHAR tWindowShape, tWindowSequence; - - /* store old window infos for swapping */ - tWindowSequence = hConcealmentInfo->windowSequence; - tWindowShape = hConcealmentInfo->windowShape; - - /* store old scale factors for swapping */ - FDKmemcpy(tSpecScale, hConcealmentInfo->specScale, 8*sizeof(SHORT)); - - /* store new window infos */ - hConcealmentInfo->windowSequence = GetWindowSequence(pIcsInfo); - hConcealmentInfo->windowShape = GetWindowShape(pIcsInfo); - hConcealmentInfo->lastWinGrpLen = *(GetWindowGroupLengthTable(pIcsInfo)+GetWindowGroups(pIcsInfo)-1); - - /* store new scale factors */ - FDKmemcpy(hConcealmentInfo->specScale, pSpecScale, 8*sizeof(SHORT)); - - if (CConcealment_GetDelay(hConcealmentInfo->pConcealParams) == 0) - { - /* store new spectral bins */ -#if (CNCL_FRACT_BITS == DFRACT_BITS) - FDKmemcpy(hConcealmentInfo->spectralCoefficient, pSpectralCoefficient, 1024 * sizeof(FIXP_CNCL)); -#else - FIXP_CNCL *RESTRICT pCncl = &hConcealmentInfo->spectralCoefficient[1024-1]; - FIXP_DBL *RESTRICT pSpec = &pSpectralCoefficient[1024-1]; - int i; - - for (i = 1024; i != 0; i--) { - *pCncl-- = FX_DBL2FX_CNCL(*pSpec--); - } -#endif - } - else - { - FIXP_CNCL *RESTRICT pCncl = &hConcealmentInfo->spectralCoefficient[1024-1]; - FIXP_DBL *RESTRICT pSpec = &pSpectralCoefficient[1024-1]; - int i; - - /* swap spectral data */ - for (i = 1024; i != 0; i--) { - FIXP_DBL tSpec = *pSpec; - *pSpec-- = FX_CNCL2FX_DBL(*pCncl); - *pCncl-- = FX_DBL2FX_CNCL( tSpec); - } - - /* complete swapping of window infos */ - pIcsInfo->WindowSequence = tWindowSequence; - pIcsInfo->WindowShape = tWindowShape; - - /* complete swapping of scale factors */ - FDKmemcpy(pSpecScale, tSpecScale, 8*sizeof(SHORT)); - } - } - -} - - -/*! - \brief Apply concealment - - Interface function to different concealment strategies - - \return none - */ -int - CConcealment_Apply ( - CConcealmentInfo *hConcealmentInfo, - CAacDecoderChannelInfo *pAacDecoderChannelInfo, - CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, - const SamplingRateInfo *pSamplingRateInfo, - const int samplesPerFrame, - const UCHAR lastLpdMode, - const int frameOk, - const UINT flags) -{ - int appliedProcessing = 0; - - if ( (frameOk == 0) - && (pAacDecoderChannelInfo->renderMode != (AACDEC_RENDER_MODE)hConcealmentInfo->lastRenderMode) ) { - /* restore the last render mode to stay in the same domain which allows to do a proper concealment */ - pAacDecoderChannelInfo->renderMode = (AACDEC_RENDER_MODE)hConcealmentInfo->lastRenderMode; - } else { - /* otherwise store the current mode */ - hConcealmentInfo->lastRenderMode = (SCHAR)pAacDecoderChannelInfo->renderMode; - } - - if ( frameOk ) - { - /* Rescue current data for concealment in future frames */ - CConcealment_Store ( hConcealmentInfo, - pAacDecoderChannelInfo, - pAacDecoderStaticChannelInfo ); - /* Reset index to random sign vector to make sign calculation frame agnostic - (only depends on number of subsequently concealed spectral blocks) */ - hConcealmentInfo->iRandomPhase = 0; - } - - /* hand current frame status to the state machine */ - CConcealment_UpdateState( hConcealmentInfo, - frameOk ); - - { - /* Create data for signal rendering according to the selected concealment method and decoder operating mode. */ - - - if ( !(pAacDecoderChannelInfo->renderMode == AACDEC_RENDER_LPD - ) - ) - { - switch (hConcealmentInfo->pConcealParams->method) - { - default: - case ConcealMethodMute: - if (!frameOk) { - /* Mute spectral data in case of errors */ - FDKmemclear(pAacDecoderChannelInfo->pSpectralCoefficient, samplesPerFrame * sizeof(FIXP_DBL)); - /* Set last window shape */ - pAacDecoderChannelInfo->icsInfo.WindowShape = hConcealmentInfo->windowShape; - appliedProcessing = 1; - } - break; - - case ConcealMethodNoise: - /* Noise substitution error concealment technique */ - appliedProcessing = - CConcealment_ApplyNoise (hConcealmentInfo, - pAacDecoderChannelInfo, - pAacDecoderStaticChannelInfo, - pSamplingRateInfo, - samplesPerFrame, - flags); - break; - - case ConcealMethodInter: - /* Energy interpolation concealment based on 3GPP */ - appliedProcessing = - CConcealment_ApplyInter (hConcealmentInfo, - pAacDecoderChannelInfo, - pSamplingRateInfo, - samplesPerFrame, - 0, /* don't use tonal improvement */ - frameOk); - break; - - } - } - } - /* update history */ - hConcealmentInfo->prevFrameOk[0] = hConcealmentInfo->prevFrameOk[1]; - hConcealmentInfo->prevFrameOk[1] = frameOk; - - return appliedProcessing; -} - -/*! -\brief Apply concealment noise substitution - - In case of frame lost this function produces a noisy frame with respect to the - energies values of past frame. - -\return none - */ -static int - CConcealment_ApplyNoise (CConcealmentInfo *pConcealmentInfo, - CAacDecoderChannelInfo *pAacDecoderChannelInfo, - CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, - const SamplingRateInfo *pSamplingRateInfo, - const int samplesPerFrame, - const UINT flags) -{ - CConcealParams *pConcealCommonData = pConcealmentInfo->pConcealParams; - - FIXP_DBL *pSpectralCoefficient = SPEC_LONG(pAacDecoderChannelInfo->pSpectralCoefficient); - SHORT *pSpecScale = pAacDecoderChannelInfo->specScale; - CIcsInfo *pIcsInfo = &pAacDecoderChannelInfo->icsInfo; - - int appliedProcessing = 0; - - FDK_ASSERT((samplesPerFrame>=480) && (samplesPerFrame<=1024)); - FDK_ASSERT((samplesPerFrame&0x1F) == 0); - - switch (pConcealmentInfo->concealState) - { - case ConcealState_Ok: - /* Nothing to do here! */ - break; - - case ConcealState_Single: - case ConcealState_FadeOut: - { - /* restore frequency coefficients from buffer with a specific muting */ - FIXP_SGL fac; - int win, numWindows = 1; - int windowLen = samplesPerFrame; - int tFadeFrames, lastWindow = 0; - int win_idx_stride = 1; - - FDK_ASSERT(pConcealmentInfo != NULL); - FDK_ASSERT(pConcealmentInfo->cntFadeFrames >= 0); - FDK_ASSERT(pConcealmentInfo->cntFadeFrames < CONCEAL_MAX_NUM_FADE_FACTORS); - FDK_ASSERT(pConcealmentInfo->cntFadeFrames <= pConcealCommonData->numFadeOutFrames); - - /* get attenuation factor */ - tFadeFrames = pConcealmentInfo->cntFadeFrames; - fac = pConcealCommonData->fadeOutFactor[tFadeFrames]; - - /* set old window parameters */ - { - pIcsInfo->WindowShape = pConcealmentInfo->windowShape; - pIcsInfo->WindowSequence = pConcealmentInfo->windowSequence; - - if (pConcealmentInfo->windowSequence == 2) { - /* short block handling */ - numWindows = 8; - windowLen = samplesPerFrame >> 3; - lastWindow = numWindows - pConcealmentInfo->lastWinGrpLen; - } - } - - for (win = 0; win < numWindows; win++) { - FIXP_CNCL *pCncl = pConcealmentInfo->spectralCoefficient + (lastWindow * windowLen); - FIXP_DBL *pOut = pSpectralCoefficient + (win * windowLen); - int i; - - FDK_ASSERT((lastWindow * windowLen + windowLen) <= samplesPerFrame); - - /* restore frequency coefficients from buffer with a specific attenuation */ - for (i = 0; i < windowLen; i++) { - pOut[i] = fMult(pCncl[i], fac); - } - - /* apply random change of sign for spectral coefficients */ - CConcealment_ApplyRandomSign(pConcealmentInfo->iRandomPhase, - pOut, - windowLen ); - - /* Increment random phase index to avoid repetition artifacts. */ - pConcealmentInfo->iRandomPhase = (pConcealmentInfo->iRandomPhase + 1) & (AAC_NF_NO_RANDOM_VAL - 1); - - /* set old scale factors */ - pSpecScale[win*win_idx_stride] = pConcealmentInfo->specScale[win_idx_stride*lastWindow++]; - - if ( (lastWindow >= numWindows) - && (numWindows > 1) ) - { - /* end of sequence -> rewind */ - lastWindow = numWindows - pConcealmentInfo->lastWinGrpLen; - /* update the attenuation factor to get a faster fade-out */ - tFadeFrames += 1; - if (tFadeFrames < pConcealCommonData->numFadeOutFrames) { - fac = pConcealCommonData->fadeOutFactor[tFadeFrames]; - } else { - fac = (FIXP_SGL)0; - } - } - } - - /* store temp vars */ - pConcealmentInfo->cntFadeFrames = tFadeFrames; - appliedProcessing = 1; - } - break; - - case ConcealState_Mute: - { - /* set dummy window parameters */ - pIcsInfo->Valid = 0; /* Trigger the generation of a consitent IcsInfo */ - pIcsInfo->WindowShape = pConcealmentInfo->windowShape; /* Prevent an invalid WindowShape (required for F/T transform) */ - pIcsInfo->WindowSequence = CConcealment_GetWinSeq(pConcealmentInfo->windowSequence); - pConcealmentInfo->windowSequence = pIcsInfo->WindowSequence; /* Store for next frame (spectrum in concealment buffer can't be used at all) */ - - /* mute spectral data */ - FDKmemclear(pSpectralCoefficient, samplesPerFrame * sizeof(FIXP_DBL)); - - if ( !(flags & (AC_USAC|AC_RSVD50)) - && pConcealCommonData->comfortNoiseLevel >= 0 - && pConcealCommonData->comfortNoiseLevel <= 61 /* -90dB */) - { - /* insert comfort noise using PNS */ - CConcealment_fakePnsData ( - &pAacDecoderChannelInfo->data.aac.PnsData, - pIcsInfo, - pSamplingRateInfo, - pAacDecoderChannelInfo->pDynData->aSfbScale, - pAacDecoderChannelInfo->pDynData->aScaleFactor, - pConcealCommonData->comfortNoiseLevel - ); - - CPns_Apply ( - &pAacDecoderChannelInfo->data.aac.PnsData, - pIcsInfo, - pAacDecoderChannelInfo->pSpectralCoefficient, - pAacDecoderChannelInfo->specScale, - pAacDecoderChannelInfo->pDynData->aScaleFactor, - pSamplingRateInfo, - pAacDecoderChannelInfo->granuleLength, - 0 /* always apply to first channel */ - ); - } - appliedProcessing = 1; - } - break; - - case ConcealState_FadeIn: - { - FDK_ASSERT(pConcealmentInfo->cntFadeFrames >= 0); - FDK_ASSERT(pConcealmentInfo->cntFadeFrames < CONCEAL_MAX_NUM_FADE_FACTORS); - FDK_ASSERT(pConcealmentInfo->cntFadeFrames < pConcealCommonData->numFadeInFrames); - - /* attenuate signal to get a smooth fade-in */ - FIXP_DBL *RESTRICT pOut = &pSpectralCoefficient[samplesPerFrame-1]; - FIXP_SGL fac = pConcealCommonData->fadeInFactor[pConcealmentInfo->cntFadeFrames]; - int i; - - for (i = samplesPerFrame; i != 0; i--) { - *pOut = fMult(*pOut, fac); - pOut--; - } - appliedProcessing = 1; - } - break; - - default: - /* we shouldn't come here anyway */ - FDK_ASSERT(0); - break; - } - - return appliedProcessing; -} - - -/*! - \brief Apply concealment interpolation - - The function swaps the data from the current and the previous frame. If an - error has occured, frame interpolation is performed to restore the missing - frame. In case of multiple faulty frames, fade-in and fade-out is applied. - - \return none -*/ -static int - CConcealment_ApplyInter ( - CConcealmentInfo *pConcealmentInfo, - CAacDecoderChannelInfo *pAacDecoderChannelInfo, - const SamplingRateInfo *pSamplingRateInfo, - const int samplesPerFrame, - const int improveTonal, - const int frameOk ) -{ - CConcealParams *pConcealCommonData = pConcealmentInfo->pConcealParams; - - FIXP_DBL *pSpectralCoefficient = SPEC_LONG(pAacDecoderChannelInfo->pSpectralCoefficient); - CIcsInfo *pIcsInfo = &pAacDecoderChannelInfo->icsInfo; - SHORT *pSpecScale = pAacDecoderChannelInfo->specScale; - - - int sfbEnergyPrev[64]; - int sfbEnergyAct [64]; - - int i, appliedProcessing = 0; - - /* clear/init */ - FDKmemclear(sfbEnergyPrev, 64 * sizeof(int)); - FDKmemclear(sfbEnergyAct, 64 * sizeof(int)); - - - if (!frameOk) - { - /* Restore last frame from concealment buffer */ - pIcsInfo->WindowShape = pConcealmentInfo->windowShape; - pIcsInfo->WindowSequence = pConcealmentInfo->windowSequence; - - /* Restore spectral data */ - for (i = 0; i < samplesPerFrame; i++) { - pSpectralCoefficient[i] = FX_CNCL2FX_DBL(pConcealmentInfo->spectralCoefficient[i]); - } - - /* Restore scale factors */ - FDKmemcpy(pSpecScale, pConcealmentInfo->specScale, 8*sizeof(SHORT)); - } - - /* if previous frame was not ok */ - if (!pConcealmentInfo->prevFrameOk[1]) { - - /* if current frame (f_n) is ok and the last but one frame (f_(n-2)) - was ok, too, then interpolate both frames in order to generate - the current output frame (f_(n-1)). Otherwise, use the last stored - frame (f_(n-2) or f_(n-3) or ...). */ - if (frameOk && pConcealmentInfo->prevFrameOk[0]) - { - appliedProcessing = 1; - - - /* Interpolate both frames in order to generate the current output frame (f_(n-1)). */ - if (pIcsInfo->WindowSequence == EightShortSequence) { - /* f_(n-2) == EightShortSequence */ - /* short--??????--short, short--??????--long interpolation */ - /* short--short---short, short---long---long interpolation */ - - int wnd; - - if (pConcealmentInfo->windowSequence == EightShortSequence) { /* f_n == EightShortSequence */ - /* short--short---short interpolation */ - - int scaleFactorBandsTotal = pSamplingRateInfo->NumberOfScaleFactorBands_Short; - const SHORT *pSfbOffset = pSamplingRateInfo->ScaleFactorBands_Short; - pIcsInfo->WindowShape = 1; - pIcsInfo->WindowSequence = EightShortSequence; - - for (wnd = 0; wnd < 8; wnd++) - { - CConcealment_CalcBandEnergy( - &pSpectralCoefficient[wnd * (samplesPerFrame / 8)], /* spec_(n-2) */ - pSamplingRateInfo, - EightShortSequence, - CConcealment_NoExpand, - sfbEnergyPrev); - - CConcealment_CalcBandEnergy( - &pConcealmentInfo->spectralCoefficient[wnd * (samplesPerFrame / 8)], /* spec_n */ - pSamplingRateInfo, - EightShortSequence, - CConcealment_NoExpand, - sfbEnergyAct); - - CConcealment_InterpolateBuffer( - &pSpectralCoefficient[wnd * (samplesPerFrame / 8)], /* spec_(n-1) */ - &pSpecScale[wnd], - &pConcealmentInfo->specScale[wnd], - &pSpecScale[wnd], - sfbEnergyPrev, - sfbEnergyAct, - scaleFactorBandsTotal, - pSfbOffset); - - } - } else { /* f_n != EightShortSequence */ - /* short---long---long interpolation */ - - int scaleFactorBandsTotal = pSamplingRateInfo->NumberOfScaleFactorBands_Long; - const SHORT *pSfbOffset = pSamplingRateInfo->ScaleFactorBands_Long; - SHORT specScaleOut; - - CConcealment_CalcBandEnergy(&pSpectralCoefficient[samplesPerFrame - (samplesPerFrame / 8)], /* [wnd] spec_(n-2) */ - pSamplingRateInfo, - EightShortSequence, - CConcealment_Expand, - sfbEnergyAct); - - CConcealment_CalcBandEnergy(pConcealmentInfo->spectralCoefficient, /* spec_n */ - pSamplingRateInfo, - OnlyLongSequence, - CConcealment_NoExpand, - sfbEnergyPrev); - - pIcsInfo->WindowShape = 0; - pIcsInfo->WindowSequence = LongStopSequence; - - for (i = 0; i < samplesPerFrame ; i++) { - pSpectralCoefficient[i] = pConcealmentInfo->spectralCoefficient[i]; /* spec_n */ - } - - for (i = 0; i < 8; i++) { /* search for max(specScale) */ - if (pSpecScale[i] > pSpecScale[0]) { - pSpecScale[0] = pSpecScale[i]; - } - } - - CConcealment_InterpolateBuffer( - pSpectralCoefficient, /* spec_(n-1) */ - &pConcealmentInfo->specScale[0], - &pSpecScale[0], - &specScaleOut, - sfbEnergyPrev, - sfbEnergyAct, - scaleFactorBandsTotal, - pSfbOffset); - - pSpecScale[0] = specScaleOut; - } - } else { - /* long--??????--short, long--??????--long interpolation */ - /* long---long---short, long---long---long interpolation */ - - int scaleFactorBandsTotal = pSamplingRateInfo->NumberOfScaleFactorBands_Long; - const SHORT *pSfbOffset = pSamplingRateInfo->ScaleFactorBands_Long; - SHORT specScaleAct = pConcealmentInfo->specScale[0]; - - CConcealment_CalcBandEnergy(pSpectralCoefficient, /* spec_(n-2) */ - pSamplingRateInfo, - OnlyLongSequence, - CConcealment_NoExpand, - sfbEnergyPrev); - - if (pConcealmentInfo->windowSequence == EightShortSequence) { /* f_n == EightShortSequence */ - /* long---long---short interpolation */ - - pIcsInfo->WindowShape = 1; - pIcsInfo->WindowSequence = LongStartSequence; - - for (i = 1; i < 8; i++) { /* search for max(specScale) */ - if (pConcealmentInfo->specScale[i] > specScaleAct) { - specScaleAct = pConcealmentInfo->specScale[i]; - } - } - - /* Expand first short spectrum */ - CConcealment_CalcBandEnergy(pConcealmentInfo->spectralCoefficient, /* spec_n */ - pSamplingRateInfo, - EightShortSequence, - CConcealment_Expand, /* !!! */ - sfbEnergyAct); - } else { - /* long---long---long interpolation */ - - pIcsInfo->WindowShape = 0; - pIcsInfo->WindowSequence = OnlyLongSequence; - - CConcealment_CalcBandEnergy(pConcealmentInfo->spectralCoefficient, /* spec_n */ - pSamplingRateInfo, - OnlyLongSequence, - CConcealment_NoExpand, - sfbEnergyAct); - } - - CConcealment_InterpolateBuffer( - pSpectralCoefficient, /* spec_(n-1) */ - &pSpecScale[0], - &specScaleAct, - &pSpecScale[0], - sfbEnergyPrev, - sfbEnergyAct, - scaleFactorBandsTotal, - pSfbOffset); - - } - } - - /* Noise substitution of sign of the output spectral coefficients */ - CConcealment_ApplyRandomSign (pConcealmentInfo->iRandomPhase, - pSpectralCoefficient, - samplesPerFrame); - /* Increment random phase index to avoid repetition artifacts. */ - pConcealmentInfo->iRandomPhase = (pConcealmentInfo->iRandomPhase + 1) & (AAC_NF_NO_RANDOM_VAL - 1); - } - - /* scale spectrum according to concealment state */ - switch (pConcealmentInfo->concealState) - { - case ConcealState_Single: - appliedProcessing = 1; - break; - - case ConcealState_FadeOut: - { - FDK_ASSERT(pConcealmentInfo->cntFadeFrames >= 0); - FDK_ASSERT(pConcealmentInfo->cntFadeFrames < CONCEAL_MAX_NUM_FADE_FACTORS); - FDK_ASSERT(pConcealmentInfo->cntFadeFrames < pConcealCommonData->numFadeOutFrames); - - /* restore frequency coefficients from buffer with a specific muting */ - FIXP_DBL *RESTRICT pOut = &pSpectralCoefficient[samplesPerFrame-1]; - FIXP_SGL fac = pConcealCommonData->fadeOutFactor[pConcealmentInfo->cntFadeFrames]; - - for (i = samplesPerFrame; i != 0; i--) { - *pOut = fMult(*pOut, fac); - pOut--; - } - appliedProcessing = 1; - } - break; - - case ConcealState_FadeIn: - { - FDK_ASSERT(pConcealmentInfo->cntFadeFrames >= 0); - FDK_ASSERT(pConcealmentInfo->cntFadeFrames < CONCEAL_MAX_NUM_FADE_FACTORS); - FDK_ASSERT(pConcealmentInfo->cntFadeFrames < pConcealCommonData->numFadeInFrames); - - /* attenuate signal to get a smooth fade-in */ - FIXP_DBL *RESTRICT pOut = &pSpectralCoefficient[samplesPerFrame-1]; - FIXP_SGL fac = pConcealCommonData->fadeInFactor[pConcealmentInfo->cntFadeFrames]; - - for (i = samplesPerFrame; i != 0; i--) { - *pOut = fMult(*pOut, fac); - pOut--; - } - appliedProcessing = 1; - } - break; - - case ConcealState_Mute: - { - int fac = pConcealCommonData->comfortNoiseLevel; - - /* set dummy window parameters */ - pIcsInfo->Valid = 0; /* Trigger the generation of a consitent IcsInfo */ - pIcsInfo->WindowShape = pConcealmentInfo->windowShape; /* Prevent an invalid WindowShape (required for F/T transform) */ - pIcsInfo->WindowSequence = CConcealment_GetWinSeq(pConcealmentInfo->windowSequence); - pConcealmentInfo->windowSequence = pIcsInfo->WindowSequence; /* Store for next frame (spectrum in concealment buffer can't be used at all) */ - - /* mute spectral data */ - FDKmemclear(pSpectralCoefficient, samplesPerFrame * sizeof(FIXP_DBL)); - - if (fac >= 0 && fac <= 61) { - /* insert comfort noise using PNS */ - CConcealment_fakePnsData ( - &pAacDecoderChannelInfo->data.aac.PnsData, - pIcsInfo, - pSamplingRateInfo, - pAacDecoderChannelInfo->specScale, - pAacDecoderChannelInfo->pDynData->aScaleFactor, - fac - ); - - CPns_Apply ( - &pAacDecoderChannelInfo->data.aac.PnsData, - pIcsInfo, - pAacDecoderChannelInfo->pSpectralCoefficient, - pAacDecoderChannelInfo->specScale, - pAacDecoderChannelInfo->pDynData->aScaleFactor, - pSamplingRateInfo, - pAacDecoderChannelInfo->granuleLength, - 0 /* always apply to first channel */ - ); - } - appliedProcessing = 1; - } - break; - - default: - /* nothing to do here */ - break; - } - - return appliedProcessing; -} - - -/*! - \brief Calculate the spectral energy - - The function calculates band-wise the spectral energy. This is used for - frame interpolation. - - \return none -*/ -static void - CConcealment_CalcBandEnergy ( - FIXP_DBL *spectrum, - const SamplingRateInfo *pSamplingRateInfo, - const int blockType, - CConcealmentExpandType expandType, - int *sfbEnergy ) -{ - const SHORT *pSfbOffset; - int line, sfb, scaleFactorBandsTotal = 0; - - /* In the following calculations, enAccu is initialized with LSB-value in order to avoid zero energy-level */ - - line = 0; - - switch(blockType) { - - case OnlyLongSequence: - case LongStartSequence: - case LongStopSequence: - - if (expandType == CConcealment_NoExpand) { - /* standard long calculation */ - scaleFactorBandsTotal = pSamplingRateInfo->NumberOfScaleFactorBands_Long; - pSfbOffset = pSamplingRateInfo->ScaleFactorBands_Long; - - for (sfb = 0; sfb < scaleFactorBandsTotal; sfb++) { - FIXP_DBL enAccu = (FIXP_DBL)(LONG)1; - int sfbScale = (sizeof(LONG)<<3) - CntLeadingZeros(pSfbOffset[sfb+1] - pSfbOffset[sfb]) - 1; - /* scaling depends on sfb width. */ - for ( ; line < pSfbOffset[sfb+1]; line++) { - enAccu += fPow2Div2(*(spectrum + line)) >> sfbScale; - } - *(sfbEnergy + sfb) = CntLeadingZeros(enAccu) - 1; - } - } - else { - /* compress long to short */ - scaleFactorBandsTotal = pSamplingRateInfo->NumberOfScaleFactorBands_Short; - pSfbOffset = pSamplingRateInfo->ScaleFactorBands_Short; - - for (sfb = 0; sfb < scaleFactorBandsTotal; sfb++) { - FIXP_DBL enAccu = (FIXP_DBL)(LONG)1; - int sfbScale = (sizeof(LONG)<<3) - CntLeadingZeros(pSfbOffset[sfb+1] - pSfbOffset[sfb]) - 1; - /* scaling depends on sfb width. */ - for (; line < pSfbOffset[sfb+1] << 3; line++) { - enAccu += (enAccu + (fPow2Div2(*(spectrum + line)) >> sfbScale)) >> 3; - } - *(sfbEnergy + sfb) = CntLeadingZeros(enAccu) - 1; - } - } - break; - - case EightShortSequence: - - if (expandType == CConcealment_NoExpand) { - /* standard short calculation */ - scaleFactorBandsTotal = pSamplingRateInfo->NumberOfScaleFactorBands_Short; - pSfbOffset = pSamplingRateInfo->ScaleFactorBands_Short; - - for (sfb = 0; sfb < scaleFactorBandsTotal; sfb++) { - FIXP_DBL enAccu = (FIXP_DBL)(LONG)1; - int sfbScale = (sizeof(LONG)<<3) - CntLeadingZeros(pSfbOffset[sfb+1] - pSfbOffset[sfb]) - 1; - /* scaling depends on sfb width. */ - for ( ; line < pSfbOffset[sfb+1]; line++) { - enAccu += fPow2Div2(*(spectrum + line)) >> sfbScale; - } - *(sfbEnergy + sfb) = CntLeadingZeros(enAccu) - 1; - } - } - else { - /* expand short to long spectrum */ - scaleFactorBandsTotal = pSamplingRateInfo->NumberOfScaleFactorBands_Long; - pSfbOffset = pSamplingRateInfo->ScaleFactorBands_Long; - - for (sfb = 0; sfb < scaleFactorBandsTotal; sfb++) { - FIXP_DBL enAccu = (FIXP_DBL)(LONG)1; - int sfbScale = (sizeof(LONG)<<3) - CntLeadingZeros(pSfbOffset[sfb+1] - pSfbOffset[sfb]) - 1; - /* scaling depends on sfb width. */ - for ( ; line < pSfbOffset[sfb+1]; line++) { - enAccu += fPow2Div2(*(spectrum + (line >> 3))) >> sfbScale; - } - *(sfbEnergy + sfb) = CntLeadingZeros(enAccu) - 1; - } - } - break; - } -} - - -/*! - \brief Interpolate buffer - - The function creates the interpolated spectral data according to the - energy of the last good frame and the current (good) frame. - - \return none -*/ -static void - CConcealment_InterpolateBuffer ( - FIXP_DBL *spectrum, - SHORT *pSpecScalePrv, - SHORT *pSpecScaleAct, - SHORT *pSpecScaleOut, - int *enPrv, - int *enAct, - int sfbCnt, - const SHORT *pSfbOffset ) -{ - int sfb, line = 0; - int fac_shift; - int fac_mod; - FIXP_DBL accu; - - for (sfb = 0; sfb < sfbCnt; sfb++) { - - fac_shift = enPrv[sfb] - enAct[sfb] + ((*pSpecScaleAct - *pSpecScalePrv) << 1); - fac_mod = fac_shift & 3; - fac_shift = (fac_shift >> 2) + 1; - fac_shift += *pSpecScalePrv - fixMax(*pSpecScalePrv, *pSpecScaleAct); - - for (; line < pSfbOffset[sfb+1]; line++) { - accu = fMult(*(spectrum+line), facMod4Table[fac_mod]); - if (fac_shift < 0) { - accu >>= -fac_shift; - } else { - accu <<= fac_shift; - } - *(spectrum+line) = accu; - } - } - *pSpecScaleOut = fixMax(*pSpecScalePrv, *pSpecScaleAct); -} - - - - -static INT findEquiFadeFrame ( - CConcealParams *pConcealCommonData, - INT actFadeIndex, - int direction ) -{ - FIXP_SGL *pFactor; - FIXP_SGL referenceVal; - FIXP_SGL minDiff = (FIXP_SGL)MAXVAL_SGL; - - INT numFrames = 0; - INT nextFadeIndex = 0; - - int i; - - /* init depending on direction */ - if (direction == 0) { /* FADE-OUT => FADE-IN */ - numFrames = pConcealCommonData->numFadeInFrames; - referenceVal = pConcealCommonData->fadeOutFactor[actFadeIndex] >> 1; - pFactor = pConcealCommonData->fadeInFactor; - } - else { /* FADE-IN => FADE-OUT */ - numFrames = pConcealCommonData->numFadeOutFrames; - referenceVal = pConcealCommonData->fadeInFactor[actFadeIndex] >> 1; - pFactor = pConcealCommonData->fadeOutFactor; - } - - /* search for minimum difference */ - for (i = 0; i < numFrames; i++) { - FIXP_SGL diff = fixp_abs((pFactor[i]>>1) - referenceVal); - if (diff < minDiff) { - minDiff = diff; - nextFadeIndex = i; - } - } - - /* check and adjust depending on direction */ - if (direction == 0) { /* FADE-OUT => FADE-IN */ - if (((pFactor[nextFadeIndex]>>1) <= referenceVal) && (nextFadeIndex > 0)) { - nextFadeIndex -= 1; - } - } - else { /* FADE-IN => FADE-OUT */ - if (((pFactor[nextFadeIndex]>>1) >= referenceVal) && (nextFadeIndex < numFrames-1)) { - nextFadeIndex += 1; - } - } - - return (nextFadeIndex); -} - - -/*! - \brief Update the concealment state - - The function updates the state of the concealment state-machine. The - states are: mute, fade-in, fade-out, interpolate and frame-ok. - - \return none -*/ -static void - CConcealment_UpdateState ( - CConcealmentInfo *pConcealmentInfo, - int frameOk ) -{ - CConcealParams *pConcealCommonData = pConcealmentInfo->pConcealParams; - - switch (pConcealCommonData->method) - { - case ConcealMethodNoise: - { - if (pConcealmentInfo->concealState != ConcealState_Ok) { - /* count the valid frames during concealment process */ - if (frameOk) { - pConcealmentInfo->cntValidFrames += 1; - } else { - pConcealmentInfo->cntValidFrames = 0; - } - } - - /* -- STATE MACHINE for Noise Substitution -- */ - switch (pConcealmentInfo->concealState) - { - case ConcealState_Ok: - if (!frameOk) { - if (pConcealCommonData->numFadeOutFrames > 0) { - /* change to state SINGLE-FRAME-LOSS */ - pConcealmentInfo->concealState = ConcealState_Single; - } else { - /* change to state MUTE */ - pConcealmentInfo->concealState = ConcealState_Mute; - } - pConcealmentInfo->cntFadeFrames = 0; - pConcealmentInfo->cntValidFrames = 0; - } - break; - - case ConcealState_Single: /* Just a pre-stage before fade-out begins. Stay here only one frame! */ - pConcealmentInfo->cntFadeFrames += 1; - if (frameOk) { - if (pConcealmentInfo->cntValidFrames > pConcealCommonData->numMuteReleaseFrames) { - /* change to state FADE-IN */ - pConcealmentInfo->concealState = ConcealState_FadeIn; - pConcealmentInfo->cntFadeFrames = findEquiFadeFrame( pConcealCommonData, - pConcealmentInfo->cntFadeFrames-1, - 0 /* FadeOut -> FadeIn */); - } else { - /* change to state OK */ - pConcealmentInfo->concealState = ConcealState_Ok; - } - } else { - if (pConcealmentInfo->cntFadeFrames >= pConcealCommonData->numFadeOutFrames) { - /* change to state MUTE */ - pConcealmentInfo->concealState = ConcealState_Mute; - } else { - /* change to state FADE-OUT */ - pConcealmentInfo->concealState = ConcealState_FadeOut; - } - } - break; - - case ConcealState_FadeOut: - pConcealmentInfo->cntFadeFrames += 1; /* used to address the fade-out factors */ - if (pConcealmentInfo->cntValidFrames > pConcealCommonData->numMuteReleaseFrames) { - if (pConcealCommonData->numFadeInFrames > 0) { - /* change to state FADE-IN */ - pConcealmentInfo->concealState = ConcealState_FadeIn; - pConcealmentInfo->cntFadeFrames = findEquiFadeFrame( pConcealCommonData, - pConcealmentInfo->cntFadeFrames-1, - 0 /* FadeOut -> FadeIn */); - } else { - /* change to state OK */ - pConcealmentInfo->concealState = ConcealState_Ok; - } - } else { - if (pConcealmentInfo->cntFadeFrames >= pConcealCommonData->numFadeOutFrames) { - /* change to state MUTE */ - pConcealmentInfo->concealState = ConcealState_Mute; - } - } - break; - - case ConcealState_Mute: - if (pConcealmentInfo->cntValidFrames > pConcealCommonData->numMuteReleaseFrames) { - if (pConcealCommonData->numFadeInFrames > 0) { - /* change to state FADE-IN */ - pConcealmentInfo->concealState = ConcealState_FadeIn; - pConcealmentInfo->cntFadeFrames = pConcealCommonData->numFadeInFrames - 1; - } else { - /* change to state OK */ - pConcealmentInfo->concealState = ConcealState_Ok; - } - } - break; - - case ConcealState_FadeIn: - pConcealmentInfo->cntFadeFrames -= 1; /* used to address the fade-in factors */ - if (frameOk) { - if (pConcealmentInfo->cntFadeFrames < 0) { - /* change to state OK */ - pConcealmentInfo->concealState = ConcealState_Ok; - } - } else { - if (pConcealCommonData->numFadeOutFrames > 0) { - /* change to state FADE-OUT */ - pConcealmentInfo->concealState = ConcealState_FadeOut; - pConcealmentInfo->cntFadeFrames = findEquiFadeFrame( pConcealCommonData, - pConcealmentInfo->cntFadeFrames+1, - 1 /* FadeIn -> FadeOut */); - } else { - /* change to state MUTE */ - pConcealmentInfo->concealState = ConcealState_Mute; - } - } - break; - - default: - FDK_ASSERT(0); - break; - } - } - break; - - case ConcealMethodInter: - case ConcealMethodTonal: - { - if (pConcealmentInfo->concealState != ConcealState_Ok) { - /* count the valid frames during concealment process */ - if ( pConcealmentInfo->prevFrameOk[1] || - (pConcealmentInfo->prevFrameOk[0] && !pConcealmentInfo->prevFrameOk[1] && frameOk) ) { - /* The frame is OK even if it can be estimated by the energy interpolation algorithm */ - pConcealmentInfo->cntValidFrames += 1; - } else { - pConcealmentInfo->cntValidFrames = 0; - } - } - - /* -- STATE MACHINE for energy interpolation -- */ - switch (pConcealmentInfo->concealState) - { - case ConcealState_Ok: - if (!(pConcealmentInfo->prevFrameOk[1] || - (pConcealmentInfo->prevFrameOk[0] && !pConcealmentInfo->prevFrameOk[1] && frameOk))) { - if (pConcealCommonData->numFadeOutFrames > 0) { - /* Fade out only if the energy interpolation algorithm can not be applied! */ - pConcealmentInfo->concealState = ConcealState_FadeOut; - } else { - /* change to state MUTE */ - pConcealmentInfo->concealState = ConcealState_Mute; - } - pConcealmentInfo->cntFadeFrames = 0; - pConcealmentInfo->cntValidFrames = 0; - } - break; - - case ConcealState_Single: - pConcealmentInfo->concealState = ConcealState_Ok; - break; - - case ConcealState_FadeOut: - pConcealmentInfo->cntFadeFrames += 1; - - if (pConcealmentInfo->cntValidFrames > pConcealCommonData->numMuteReleaseFrames) { - if (pConcealCommonData->numFadeInFrames > 0) { - /* change to state FADE-IN */ - pConcealmentInfo->concealState = ConcealState_FadeIn; - pConcealmentInfo->cntFadeFrames = findEquiFadeFrame( pConcealCommonData, - pConcealmentInfo->cntFadeFrames-1, - 0 /* FadeOut -> FadeIn */); - } else { - /* change to state OK */ - pConcealmentInfo->concealState = ConcealState_Ok; - } - } else { - if (pConcealmentInfo->cntFadeFrames >= pConcealCommonData->numFadeOutFrames) { - /* change to state MUTE */ - pConcealmentInfo->concealState = ConcealState_Mute; - } - } - break; - - case ConcealState_Mute: - if (pConcealmentInfo->cntValidFrames > pConcealCommonData->numMuteReleaseFrames) { - if (pConcealCommonData->numFadeInFrames > 0) { - /* change to state FADE-IN */ - pConcealmentInfo->concealState = ConcealState_FadeIn; - pConcealmentInfo->cntFadeFrames = pConcealCommonData->numFadeInFrames - 1; - } else { - /* change to state OK */ - pConcealmentInfo->concealState = ConcealState_Ok; - } - } - break; - - case ConcealState_FadeIn: - pConcealmentInfo->cntFadeFrames -= 1; /* used to address the fade-in factors */ - - if (frameOk || pConcealmentInfo->prevFrameOk[1]) { - if (pConcealmentInfo->cntFadeFrames < 0) { - /* change to state OK */ - pConcealmentInfo->concealState = ConcealState_Ok; - } - } else { - if (pConcealCommonData->numFadeOutFrames > 0) { - /* change to state FADE-OUT */ - pConcealmentInfo->concealState = ConcealState_FadeOut; - pConcealmentInfo->cntFadeFrames = findEquiFadeFrame( pConcealCommonData, - pConcealmentInfo->cntFadeFrames+1, - 1 /* FadeIn -> FadeOut */); - } else { - /* change to state MUTE */ - pConcealmentInfo->concealState = ConcealState_Mute; - } - } - break; - } /* End switch(pConcealmentInfo->concealState) */ - } - break; - - default: - /* Don't need a state machine for other concealment methods. */ - break; - } - -} - - -/*! -\brief Randomizes the sign of the spectral data - - The function toggles the sign of the spectral data randomly. This is - useful to ensure the quality of the concealed frames. - -\return none - */ -static -void CConcealment_ApplyRandomSign (int randomPhase, - FIXP_DBL *spec, - int samplesPerFrame - ) -{ - int i; - USHORT packedSign=0; - - /* random table 512x16bit has been reduced to 512 packed sign bits = 32x16 bit */ - - /* read current packed sign word */ - packedSign = randomSign[randomPhase>>4]; - packedSign >>= (randomPhase&0xf); - - for (i = 0; i < samplesPerFrame ; i++) { - if ((randomPhase & 0xf) == 0) { - packedSign = randomSign[randomPhase>>4]; - } - - if (packedSign & 0x1) { - spec[i] = -spec[i]; - } - packedSign >>= 1; - - randomPhase = (randomPhase + 1) & (AAC_NF_NO_RANDOM_VAL - 1); - } -} - - -/*! - \brief Get fadeing factor for current concealment state. - - The function returns the factor used for fading that belongs to the current internal state. - - \return Fade factor - */ -FIXP_DBL - CConcealment_GetFadeFactor ( - CConcealmentInfo *hConcealmentInfo, - const int fPreviousFactor - ) -{ - FIXP_DBL fac = (FIXP_DBL)0; - - CConcealParams *pConcealCommonData = hConcealmentInfo->pConcealParams; - - if (hConcealmentInfo->pConcealParams->method > ConcealMethodMute) { - switch (hConcealmentInfo->concealState) { - default: - case ConcealState_Mute: - /* Nothing to do here */ - break; - case ConcealState_Ok: - fac = (FIXP_DBL)MAXVAL_DBL; - break; - case ConcealState_Single: - case ConcealState_FadeOut: - { - int idx = hConcealmentInfo->cntFadeFrames - ((fPreviousFactor != 0) ? 1 : 0); - fac = (idx < 0) ? (FIXP_DBL)MAXVAL_DBL : FX_SGL2FX_DBL(pConcealCommonData->fadeOutFactor[idx]); - } - break; - case ConcealState_FadeIn: - { - int idx = hConcealmentInfo->cntFadeFrames + ((fPreviousFactor != 0) ? 1 : 0); - fac = (idx >= hConcealmentInfo->pConcealParams->numFadeInFrames) ? (FIXP_DBL)0 : FX_SGL2FX_DBL(pConcealCommonData->fadeInFactor[idx]); - } - break; - } - } - - return (fac); -} - - -/*! - \brief Get fadeing factor for current concealment state. - - The function returns the state (ok or not) of the previous frame. - If called before the function CConcealment_Apply() set the fBeforeApply - flag to get the correct value. - - \return Frame OK flag of previous frame. - */ -int - CConcealment_GetLastFrameOk ( - CConcealmentInfo *hConcealmentInfo, - const int fBeforeApply - ) -{ - int prevFrameOk = 1; - - if (hConcealmentInfo != NULL) { - prevFrameOk = hConcealmentInfo->prevFrameOk[fBeforeApply & 0x1]; - } - - return prevFrameOk; -} - -/*! - \brief Get the number of delay frames introduced by concealment technique. - - \return Number of delay frames. - */ -UINT - CConcealment_GetDelay ( - CConcealParams *pConcealCommonData - ) -{ - UINT frameDelay = 0; - - if (pConcealCommonData != NULL) { - switch (pConcealCommonData->method) { - case ConcealMethodTonal: - case ConcealMethodInter: - frameDelay = 1; - break; - default: - break; - } - } - - return frameDelay; -} - diff --git a/libAACdec/src/conceal.h b/libAACdec/src/conceal.h deleted file mode 100644 index 20e674f..0000000 --- a/libAACdec/src/conceal.h +++ /dev/null @@ -1,148 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/***************************** MPEG-4 AAC Decoder ************************** - - Author(s): Josef Hoepfl - Description: independent channel concealment - -******************************************************************************/ - -#ifndef _CONCEAL_H_ -#define _CONCEAL_H_ - -#include "aacdecoder_lib.h" - -#include "channelinfo.h" - -#define AACDEC_CONCEAL_PARAM_NOT_SPECIFIED ( 0xFFFE ) - -void CConcealment_InitCommonData (CConcealParams *pConcealCommonData); - -void CConcealment_InitChannelData (CConcealmentInfo *hConcealmentInfo, - CConcealParams *pConcealCommonData, - int samplesPerFrame); - -CConcealmentMethod - CConcealment_GetMethod (CConcealParams *pConcealCommonData); - -UINT - CConcealment_GetDelay (CConcealParams *pConcealCommonData); - -AAC_DECODER_ERROR - CConcealment_SetParams (CConcealParams *concealParams, - int method, - int fadeOutSlope, - int fadeInSlope, - int muteRelease, - int comfNoiseLevel); - -CConcealmentState - CConcealment_GetState (CConcealmentInfo *hConcealmentInfo); - -AAC_DECODER_ERROR - CConcealment_SetAttenuation (CConcealParams *concealParams, - SHORT *fadeOutAttenuationVector, - SHORT *fadeInAttenuationVector); - -void CConcealment_Store (CConcealmentInfo *hConcealmentInfo, - CAacDecoderChannelInfo *pAacDecoderChannelInfo, - CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo ); - -int CConcealment_Apply (CConcealmentInfo *hConcealmentInfo, - CAacDecoderChannelInfo *pAacDecoderChannelInfo, - CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, - const SamplingRateInfo *pSamplingRateInfo, - const int samplesPerFrame, - const UCHAR lastLpdMode, - const int FrameOk, - const UINT flags); - -FIXP_DBL - CConcealment_GetFadeFactor (CConcealmentInfo *hConcealmentInfo, - const int fPreviousFactor); - -int CConcealment_GetLastFrameOk (CConcealmentInfo *hConcealmentInfo, - const int fBeforeApply); - -#endif /* #ifndef _CONCEAL_H_ */ diff --git a/libAACdec/src/conceal_types.h b/libAACdec/src/conceal_types.h deleted file mode 100644 index 31bc645..0000000 --- a/libAACdec/src/conceal_types.h +++ /dev/null @@ -1,178 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/***************************** MPEG-4 AAC Decoder ************************** - - Author(s): Christian Griebel - Description: Error concealment structs and types - -******************************************************************************/ - -#ifndef CONCEAL_TYPES_H -#define CONCEAL_TYPES_H - - - -#include "machine_type.h" -#include "common_fix.h" - -#include "rvlc_info.h" - - -#define CONCEAL_MAX_NUM_FADE_FACTORS ( 16 ) - - #define FIXP_CNCL FIXP_DBL - #define FL2FXCONST_CNCL FL2FXCONST_DBL - #define FX_DBL2FX_CNCL - #define FX_CNCL2FX_DBL - #define CNCL_FRACT_BITS DFRACT_BITS - -/* Warning: Do not ever change these values. */ -typedef enum -{ - ConcealMethodNone = -1, - ConcealMethodMute = 0, - ConcealMethodNoise = 1, - ConcealMethodInter = 2, - ConcealMethodTonal = 3 - -} CConcealmentMethod; - - -typedef enum -{ - ConcealState_Ok, - ConcealState_Single, - ConcealState_FadeIn, - ConcealState_Mute, - ConcealState_FadeOut - -} CConcealmentState; - - -typedef struct -{ - FIXP_SGL fadeOutFactor[CONCEAL_MAX_NUM_FADE_FACTORS]; - FIXP_SGL fadeInFactor [CONCEAL_MAX_NUM_FADE_FACTORS]; - - CConcealmentMethod method; - - int numFadeOutFrames; - int numFadeInFrames; - int numMuteReleaseFrames; - int comfortNoiseLevel; - -} CConcealParams; - - - -typedef struct -{ - CConcealParams *pConcealParams; - - FIXP_CNCL spectralCoefficient[1024]; - SHORT specScale[8]; - - INT iRandomPhase; - INT prevFrameOk[2]; - INT cntFadeFrames; - INT cntValidFrames; - - SHORT aRvlcPreviousScaleFactor[RVLC_MAX_SFB]; /* needed once per channel */ - UCHAR aRvlcPreviousCodebook[RVLC_MAX_SFB]; /* needed once per channel */ - SCHAR rvlcPreviousScaleFactorOK; - SCHAR rvlcPreviousBlockType; - - - SCHAR lastRenderMode; - - UCHAR windowShape; - UCHAR windowSequence; - UCHAR lastWinGrpLen; - - CConcealmentState concealState; - -} CConcealmentInfo; - - -#endif /* #ifndef CONCEAL_TYPES_H */ diff --git a/libAACdec/src/rvlc.cpp b/libAACdec/src/rvlc.cpp deleted file mode 100644 index 16f0bf5..0000000 --- a/libAACdec/src/rvlc.cpp +++ /dev/null @@ -1,1215 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief RVLC Decoder - \author Robert Weidner -*/ - -#include "rvlc.h" - - -#include "block.h" - -#include "aac_rom.h" -#include "rvlcbit.h" -#include "rvlcconceal.h" -#include "aacdec_hcr.h" - -/*--------------------------------------------------------------------------------------------- - function: rvlcInit - - description: init RVLC by data from channelinfo, which was decoded previously and - set up pointers ------------------------------------------------------------------------------------------------ - input: - pointer rvlc structure - - pointer channel info structure - - pointer bitstream structure ------------------------------------------------------------------------------------------------ - return: - --------------------------------------------------------------------------------------------- */ - -static -void rvlcInit (CErRvlcInfo *pRvlc, - CAacDecoderChannelInfo *pAacDecoderChannelInfo, - HANDLE_FDK_BITSTREAM bs) -{ - /* RVLC common initialization part 2 of 2 */ - SHORT *pScfEsc = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfEsc; - SHORT *pScfFwd = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd; - SHORT *pScfBwd = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd; - SHORT *pScaleFactor = pAacDecoderChannelInfo->pDynData->aScaleFactor; - int bnds; - - pAacDecoderChannelInfo->pDynData->specificTo.aac.rvlcIntensityUsed = 0; - - pRvlc->numDecodedEscapeWordsEsc = 0; - pRvlc->numDecodedEscapeWordsFwd = 0; - pRvlc->numDecodedEscapeWordsBwd = 0; - - pRvlc->intensity_used = 0; - pRvlc->errorLogRvlc = 0; - - pRvlc->conceal_max = CONCEAL_MAX_INIT; - pRvlc->conceal_min = CONCEAL_MIN_INIT; - - pRvlc->conceal_max_esc = CONCEAL_MAX_INIT; - pRvlc->conceal_min_esc = CONCEAL_MIN_INIT; - - pRvlc->pHuffTreeRvlcEscape = aHuffTreeRvlcEscape; - pRvlc->pHuffTreeRvlCodewds = aHuffTreeRvlCodewds; - - /* init scf arrays (for savety (in case of there are only zero codebooks)) */ - for (bnds = 0; bnds < RVLC_MAX_SFB; bnds++) { - pScfFwd[bnds] = 0; - pScfBwd[bnds] = 0; - pScfEsc[bnds] = 0; - pScaleFactor[bnds] = 0; - } - - /* set base bitstream ptr to the RVL-coded part (start of RVLC data (ESC 2)) */ - FDKsyncCache (bs); - - pRvlc->bitstreamIndexRvlFwd = FDKgetBitCnt(bs); /* first bit within RVL coded block as start address for forward decoding */ - pRvlc->bitstreamIndexRvlBwd = FDKgetBitCnt(bs) + pRvlc->length_of_rvlc_sf - 1; /* last bit within RVL coded block as start address for backward decoding */ - - /* skip RVLC-bitstream-part -- pointing now to escapes (if present) or to TNS data (if present) */ - FDKpushFor (bs, pRvlc->length_of_rvlc_sf); - - if ( pRvlc->sf_escapes_present != 0 ) { - - /* locate internal bitstream ptr at escapes (which is the second part) */ - FDKsyncCache (bs); - pRvlc->bitstreamIndexEsc = FDKgetBitCnt(bs); - - /* skip escapeRVLC-bitstream-part -- pointing to TNS data (if present) to make decoder continue */ - /* decoding of RVLC should work despite this second pushFor during initialization because */ - /* bitstream initialization is valid for both ESC2 data parts (RVL-coded values and ESC-coded values) */ - FDKpushFor (bs, pRvlc->length_of_rvlc_escapes); - } - -#if VERBOSE_RVLC_INIT - DebugOutputInit(pRvlc,pAacDecoderChannelInfo); -#endif -} - - -/*--------------------------------------------------------------------------------------------- - function: rvlcCheckIntensityCb - - description: Check if a intensity codebook is used in the current channel. ------------------------------------------------------------------------------------------------ - input: - pointer rvlc structure - - pointer channel info structure ------------------------------------------------------------------------------------------------ - output: - intensity_used: 0 no intensity codebook is used - 1 intensity codebook is used ------------------------------------------------------------------------------------------------ - return: - --------------------------------------------------------------------------------------------- */ - -static -void rvlcCheckIntensityCb (CErRvlcInfo *pRvlc, - CAacDecoderChannelInfo *pAacDecoderChannelInfo) -{ - int group, band, bnds; - - pRvlc->intensity_used = 0; - - for (group=0; group < pRvlc->numWindowGroups; group++) { - for (band=0; band < pRvlc->maxSfbTransmitted; band++) { - bnds = 16*group+band; - if ( (pAacDecoderChannelInfo->pDynData->aCodeBook[bnds] == INTENSITY_HCB) || (pAacDecoderChannelInfo->pDynData->aCodeBook[bnds] == INTENSITY_HCB2) ) { - pRvlc->intensity_used = 1; - break; - } - } - } -} - - -/*--------------------------------------------------------------------------------------------- - function: rvlcDecodeEscapeWord - - description: Decode a huffman coded RVLC Escape-word. This value is part of a DPCM coded - scalefactor. ------------------------------------------------------------------------------------------------ - input: - pointer rvlc structure ------------------------------------------------------------------------------------------------ - return: - a single RVLC-Escape value which had to be applied to a DPCM value (which - has a absolute value of 7) --------------------------------------------------------------------------------------------- */ - -static -SCHAR rvlcDecodeEscapeWord (CErRvlcInfo *pRvlc, - HANDLE_FDK_BITSTREAM bs) -{ - int i; - SCHAR value; - UCHAR carryBit; - UINT treeNode; - UINT branchValue; - UINT branchNode; - - USHORT* pBitstreamIndexEsc; - const UINT* pEscTree; - - pEscTree = pRvlc->pHuffTreeRvlcEscape; - pBitstreamIndexEsc = &(pRvlc->bitstreamIndexEsc); - treeNode = *pEscTree; /* init at starting node */ - - for (i=MAX_LEN_RVLC_ESCAPE_WORD-1; i >= 0; i--) { - carryBit = rvlcReadBitFromBitstream(bs, /* get next bit */ - pBitstreamIndexEsc, - FWD); - - CarryBitToBranchValue(carryBit, /* huffman decoding, do a single step in huffman decoding tree */ - treeNode, - &branchValue, - &branchNode); - - if ((branchNode & TEST_BIT_10) == TEST_BIT_10) { /* test bit 10 ; if set --> a RVLC-escape-word is completely decoded */ - value = (SCHAR) branchNode & CLR_BIT_10; - pRvlc->length_of_rvlc_escapes -= (MAX_LEN_RVLC_ESCAPE_WORD - i); - - if (pRvlc->length_of_rvlc_escapes < 0) { - pRvlc->errorLogRvlc |= RVLC_ERROR_ALL_ESCAPE_WORDS_INVALID; - value = -1; - } - - return value; - } - else { - treeNode = *(pEscTree + branchValue); /* update treeNode for further step in decoding tree */ - } - } - - pRvlc->errorLogRvlc |= RVLC_ERROR_ALL_ESCAPE_WORDS_INVALID; - - return -1; /* should not be reached */ -} - - -/*--------------------------------------------------------------------------------------------- - function: rvlcDecodeEscapes - - description: Decodes all huffman coded RVLC Escape Words. - Here a difference to the pseudo-code-implementation from standard can be - found. A while loop (and not two nested for loops) is used for two reasons: - - 1. The plain huffman encoded escapes are decoded before the RVL-coded - scalefactors. Therefore the escapes are present in the second step - when decoding the RVL-coded-scalefactor values in forward and - backward direction. - - When the RVL-coded scalefactors are decoded and there a escape is - needed, then it is just taken out of the array in ascending order. - - 2. It's faster. ------------------------------------------------------------------------------------------------ - input: - pointer rvlc structure - - handle to FDK bitstream ------------------------------------------------------------------------------------------------ - return: - 0 ok the decoded escapes seem to be valid - - 1 error there was a error detected during decoding escapes - --> all escapes are invalid --------------------------------------------------------------------------------------------- */ - -static -void rvlcDecodeEscapes (CErRvlcInfo *pRvlc, - SHORT *pEsc, - HANDLE_FDK_BITSTREAM bs) -{ - SCHAR escWord; - SCHAR escCnt=0; - SHORT* pEscBitCntSum; - - pEscBitCntSum = &(pRvlc->length_of_rvlc_escapes); - - /* Decode all RVLC-Escape words with a plain Huffman-Decoder */ - while ( *pEscBitCntSum > 0 ) { - escWord = rvlcDecodeEscapeWord(pRvlc, bs); - - if (escWord >= 0) { - - pEsc[escCnt] = escWord; - escCnt++; - } - else { - pRvlc->errorLogRvlc |= RVLC_ERROR_ALL_ESCAPE_WORDS_INVALID; - pRvlc->numDecodedEscapeWordsEsc = escCnt; - - return; - } - } /* all RVLC escapes decoded */ - - pRvlc->numDecodedEscapeWordsEsc = escCnt; -} - - -/*--------------------------------------------------------------------------------------------- - function: decodeRVLCodeword - - description: Decodes a RVL-coded dpcm-word (-part). ------------------------------------------------------------------------------------------------ - input: - FDK bitstream handle - - pointer rvlc structure ------------------------------------------------------------------------------------------------ - return: - a dpcm value which is within range [0,1,..,14] in case of no errors. - The offset of 7 must be subtracted to get a valid dpcm scalefactor value. - In case of errors a forbidden codeword is detected --> returning -1 --------------------------------------------------------------------------------------------- */ - -SCHAR decodeRVLCodeword (HANDLE_FDK_BITSTREAM bs, CErRvlcInfo *pRvlc) -{ - int i; - SCHAR value; - UCHAR carryBit; - UINT branchValue; - UINT branchNode; - - const UINT *pRvlCodeTree = pRvlc->pHuffTreeRvlCodewds; - UCHAR direction = pRvlc->direction; - USHORT *pBitstrIndxRvl = pRvlc->pBitstrIndxRvl_RVL; - UINT treeNode = *pRvlCodeTree; - - for (i=MAX_LEN_RVLC_CODE_WORD-1; i >= 0; i--) { - carryBit = rvlcReadBitFromBitstream(bs, /* get next bit */ - pBitstrIndxRvl, - direction); - - CarryBitToBranchValue(carryBit, /* huffman decoding, do a single step in huffman decoding tree */ - treeNode, - &branchValue, - &branchNode); - - if ((branchNode & TEST_BIT_10) == TEST_BIT_10) { /* test bit 10 ; if set --> a RVLC-codeword is completely decoded */ - value = (SCHAR) (branchNode & CLR_BIT_10); - *pRvlc->pRvlBitCnt_RVL -= (MAX_LEN_RVLC_CODE_WORD - i); - - /* check available bits for decoding */ - if (*pRvlc->pRvlBitCnt_RVL < 0) { - if (direction == FWD) { - pRvlc->errorLogRvlc |= RVLC_ERROR_RVL_SUM_BIT_COUNTER_BELOW_ZERO_FWD; } - else { - pRvlc->errorLogRvlc |= RVLC_ERROR_RVL_SUM_BIT_COUNTER_BELOW_ZERO_BWD; } - value = -1; /* signalize an error in return value, because too many bits was decoded */ - } - - /* check max value of dpcm value */ - if (value > MAX_ALLOWED_DPCM_INDEX) { - if (direction == FWD) { - pRvlc->errorLogRvlc |= RVLC_ERROR_FORBIDDEN_CW_DETECTED_FWD; - } - else { - pRvlc->errorLogRvlc |= RVLC_ERROR_FORBIDDEN_CW_DETECTED_BWD; - } - value = -1; /* signalize an error in return value, because a forbidden cw was detected*/ - } - - return value; /* return a dpcm value with offset +7 or an error status */ - } - else { - treeNode = *(pRvlCodeTree + branchValue); /* update treeNode for further step in decoding tree */ - } - } - - return -1; -} - - -/*--------------------------------------------------------------------------------------------- - function: rvlcDecodeForward - - description: Decode RVL-coded codewords in forward direction. ------------------------------------------------------------------------------------------------ - input: - pointer rvlc structure - - pointer channel info structure - - handle to FDK bitstream ------------------------------------------------------------------------------------------------ - return: - --------------------------------------------------------------------------------------------- */ - -static -void rvlcDecodeForward (CErRvlcInfo *pRvlc, - CAacDecoderChannelInfo *pAacDecoderChannelInfo, - HANDLE_FDK_BITSTREAM bs) -{ - int band = 0; - int group = 0; - int bnds = 0; - - SHORT dpcm; - - SHORT factor = pAacDecoderChannelInfo->pDynData->RawDataInfo.GlobalGain - SF_OFFSET; - SHORT position = - SF_OFFSET; - SHORT noisenrg = pAacDecoderChannelInfo->pDynData->RawDataInfo.GlobalGain - SF_OFFSET - 90 - 256; - - SHORT* pScfFwd = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd; - SHORT* pScfEsc = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfEsc; - UCHAR* pEscFwdCnt = &(pRvlc->numDecodedEscapeWordsFwd); - - pRvlc->pRvlBitCnt_RVL = &(pRvlc->length_of_rvlc_sf_fwd); - pRvlc->pBitstrIndxRvl_RVL = &(pRvlc->bitstreamIndexRvlFwd); - - *pEscFwdCnt = 0; - pRvlc->direction = FWD; - pRvlc->noise_used = 0; - pRvlc->sf_used = 0; - pRvlc->lastScf = 0; - pRvlc->lastNrg = 0; - pRvlc->lastIs = 0; - - rvlcCheckIntensityCb(pRvlc,pAacDecoderChannelInfo); - - /* main loop fwd long */ - for (group=0; group < pRvlc->numWindowGroups; group++) { - for (band=0; band < pRvlc->maxSfbTransmitted; band++) { - bnds = 16*group+band; - - switch (pAacDecoderChannelInfo->pDynData->aCodeBook[bnds]) { - - case ZERO_HCB : - pScfFwd[bnds] = 0; - break; - - case INTENSITY_HCB2 : - case INTENSITY_HCB : - /* store dpcm_is_position */ - dpcm = decodeRVLCodeword(bs, pRvlc); - if ( dpcm < 0 ) { - pRvlc->conceal_max = bnds; - return; - } - dpcm -= TABLE_OFFSET; - if ((dpcm == MIN_RVL) || (dpcm == MAX_RVL)) { - if (pRvlc->length_of_rvlc_escapes) { - pRvlc->conceal_max = bnds; - return; - } - else { - if (dpcm == MIN_RVL) { - dpcm -= *pScfEsc++; - } - else { - dpcm += *pScfEsc++; - } - (*pEscFwdCnt)++; - if (pRvlc->conceal_max_esc == CONCEAL_MAX_INIT) { - pRvlc->conceal_max_esc = bnds; - } - } - } - position += dpcm; - pScfFwd[bnds] = position; - pRvlc->lastIs = position; - break; - - case NOISE_HCB : - if (pRvlc->noise_used == 0) { - pRvlc->noise_used = 1; - pRvlc->first_noise_band = bnds; - noisenrg += pRvlc->dpcm_noise_nrg; - pScfFwd[bnds] = 100 + noisenrg; - pRvlc->lastNrg = noisenrg; - } - else { - dpcm = decodeRVLCodeword(bs, pRvlc); - if ( dpcm < 0 ) { - pRvlc->conceal_max = bnds; - return; - } - dpcm -= TABLE_OFFSET; - if ((dpcm == MIN_RVL) || (dpcm == MAX_RVL)) { - if (pRvlc->length_of_rvlc_escapes) { - pRvlc->conceal_max = bnds; - return; - } - else { - if (dpcm == MIN_RVL) { - dpcm -= *pScfEsc++; - } - else { - dpcm += *pScfEsc++; - } - (*pEscFwdCnt)++; - if (pRvlc->conceal_max_esc == CONCEAL_MAX_INIT) { - pRvlc->conceal_max_esc = bnds; - } - } - } - noisenrg += dpcm; - pScfFwd[bnds] = 100 + noisenrg; - pRvlc->lastNrg = noisenrg; - } - pAacDecoderChannelInfo->data.aac.PnsData.pnsUsed[bnds] = 1; - break ; - - default : - pRvlc->sf_used = 1; - dpcm = decodeRVLCodeword(bs, pRvlc); - if ( dpcm < 0 ) { - pRvlc->conceal_max = bnds; - return; - } - dpcm -= TABLE_OFFSET; - if ((dpcm == MIN_RVL) || (dpcm == MAX_RVL)) { - if (pRvlc->length_of_rvlc_escapes) { - pRvlc->conceal_max = bnds; - return; - } - else { - if (dpcm == MIN_RVL) { - dpcm -= *pScfEsc++; } - else { - dpcm += *pScfEsc++; - } - (*pEscFwdCnt)++; - if (pRvlc->conceal_max_esc == CONCEAL_MAX_INIT) { - pRvlc->conceal_max_esc = bnds; - } - } - } - factor += dpcm; - pScfFwd[bnds] = factor; - pRvlc->lastScf = factor; - break; - } - } - } - - /* postfetch fwd long */ - if (pRvlc->intensity_used) { - dpcm = decodeRVLCodeword(bs, pRvlc); /* dpcm_is_last_position */ - if ( dpcm < 0 ) { - pRvlc->conceal_max = bnds; - return; - } - dpcm -= TABLE_OFFSET; - if ((dpcm == MIN_RVL) || (dpcm == MAX_RVL)) { - if (pRvlc->length_of_rvlc_escapes) { - pRvlc->conceal_max = bnds; - return; - } - else { - if (dpcm == MIN_RVL) { - dpcm -= *pScfEsc++; - } - else { - dpcm += *pScfEsc++; - } - (*pEscFwdCnt)++; - if (pRvlc->conceal_max_esc == CONCEAL_MAX_INIT) { - pRvlc->conceal_max_esc = bnds; - } - } - } - pRvlc->dpcm_is_last_position = dpcm; - } -} - - -/*--------------------------------------------------------------------------------------------- - function: rvlcDecodeBackward - - description: Decode RVL-coded codewords in backward direction. ------------------------------------------------------------------------------------------------ - input: - pointer rvlc structure - - pointer channel info structure - - handle FDK bitstream ------------------------------------------------------------------------------------------------ - return: - --------------------------------------------------------------------------------------------- */ - -static -void rvlcDecodeBackward (CErRvlcInfo *pRvlc, - CAacDecoderChannelInfo *pAacDecoderChannelInfo, - HANDLE_FDK_BITSTREAM bs) -{ - SHORT band, group, dpcm, offset; - SHORT bnds = pRvlc->maxSfbTransmitted-1; - - SHORT factor = pRvlc->rev_global_gain - SF_OFFSET; - SHORT position = pRvlc->dpcm_is_last_position - SF_OFFSET; - SHORT noisenrg = pRvlc->rev_global_gain + pRvlc->dpcm_noise_last_position - SF_OFFSET - 90 - 256; - - SHORT *pScfBwd = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd; - SHORT *pScfEsc = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfEsc; - UCHAR *pEscEscCnt = &(pRvlc->numDecodedEscapeWordsEsc); - UCHAR *pEscBwdCnt = &(pRvlc->numDecodedEscapeWordsBwd); - - pRvlc->pRvlBitCnt_RVL = &(pRvlc->length_of_rvlc_sf_bwd); - pRvlc->pBitstrIndxRvl_RVL = &(pRvlc->bitstreamIndexRvlBwd); - - *pEscBwdCnt = 0; - pRvlc->direction = BWD; - pScfEsc += *pEscEscCnt - 1; /* set pScfEsc to last entry */ - pRvlc->firstScf = 0; - pRvlc->firstNrg = 0; - pRvlc->firstIs = 0; - - /* prefetch long BWD */ - if (pRvlc->intensity_used) { - dpcm = decodeRVLCodeword(bs, pRvlc); /* dpcm_is_last_position */ - if ( dpcm < 0 ) { - pRvlc->dpcm_is_last_position = 0; - pRvlc->conceal_min = bnds; - return; - } - dpcm -= TABLE_OFFSET; - if ((dpcm == MIN_RVL) || (dpcm == MAX_RVL)) { - if (pRvlc->length_of_rvlc_escapes) { - pRvlc->conceal_min = bnds; - return; - } - else { - if (dpcm == MIN_RVL) { - dpcm -= *pScfEsc--; - } - else { - dpcm += *pScfEsc--; - } - (*pEscBwdCnt)++; - if (pRvlc->conceal_min_esc == CONCEAL_MIN_INIT) { - pRvlc->conceal_min_esc = bnds; - } - } - } - pRvlc->dpcm_is_last_position = dpcm; - } - - /* main loop long BWD */ - for (group=pRvlc->numWindowGroups-1; group >= 0; group--) { - for (band=pRvlc->maxSfbTransmitted-1; band >= 0; band--) { - bnds = 16*group+band; - if ((band == 0) && (pRvlc->numWindowGroups != 1)) - offset = 16 - pRvlc->maxSfbTransmitted + 1; - else - offset = 1; - - switch (pAacDecoderChannelInfo->pDynData->aCodeBook[bnds]) { - - case ZERO_HCB : - pScfBwd[bnds] = 0; - break; - - case INTENSITY_HCB2 : - case INTENSITY_HCB : - /* store dpcm_is_position */ - dpcm = decodeRVLCodeword(bs, pRvlc); - if ( dpcm < 0 ) { - pScfBwd[bnds] = position; - pRvlc->conceal_min = FDKmax(0,bnds-offset); - return; - } - dpcm -= TABLE_OFFSET; - if ((dpcm == MIN_RVL) || (dpcm == MAX_RVL)) { - if (pRvlc->length_of_rvlc_escapes) { - pScfBwd[bnds] = position; - pRvlc->conceal_min = FDKmax(0,bnds-offset); - return; - } - else { - if (dpcm == MIN_RVL) { - dpcm -= *pScfEsc--; - } - else { - dpcm += *pScfEsc--; - } - (*pEscBwdCnt)++; - if (pRvlc->conceal_min_esc == CONCEAL_MIN_INIT) { - pRvlc->conceal_min_esc = FDKmax(0,bnds-offset); - } - } - } - pScfBwd[bnds] = position; - position -= dpcm; - pRvlc->firstIs = position; - break; - - case NOISE_HCB : - if ( bnds == pRvlc->first_noise_band ) { - pScfBwd[bnds] = pRvlc->dpcm_noise_nrg + pAacDecoderChannelInfo->pDynData->RawDataInfo.GlobalGain - SF_OFFSET - 90 - 256; - pRvlc->firstNrg = pScfBwd[bnds]; - } - else { - dpcm = decodeRVLCodeword(bs, pRvlc); - if ( dpcm < 0 ) { - pScfBwd[bnds] = noisenrg; - pRvlc->conceal_min = FDKmax(0,bnds-offset); - return; - } - dpcm -= TABLE_OFFSET; - if ((dpcm == MIN_RVL) || (dpcm == MAX_RVL)) { - if (pRvlc->length_of_rvlc_escapes) { - pScfBwd[bnds] = noisenrg; - pRvlc->conceal_min = FDKmax(0,bnds-offset); - return; - } - else { - if (dpcm == MIN_RVL) { - dpcm -= *pScfEsc--; - } - else { - dpcm += *pScfEsc--; - } - (*pEscBwdCnt)++; - if (pRvlc->conceal_min_esc == CONCEAL_MIN_INIT) { - pRvlc->conceal_min_esc = FDKmax(0,bnds-offset); - } - } - } - pScfBwd[bnds] = noisenrg; - noisenrg -= dpcm; - pRvlc->firstNrg = noisenrg; - } - break ; - - default : - dpcm = decodeRVLCodeword(bs, pRvlc); - if ( dpcm < 0 ) { - pScfBwd[bnds] = factor; - pRvlc->conceal_min = FDKmax(0,bnds-offset); - return; - } - dpcm -= TABLE_OFFSET; - if ((dpcm == MIN_RVL) || (dpcm == MAX_RVL)) { - if (pRvlc->length_of_rvlc_escapes) { - pScfBwd[bnds] = factor; - pRvlc->conceal_min = FDKmax(0,bnds-offset); - return; - } - else { - if (dpcm == MIN_RVL) { - dpcm -= *pScfEsc--; - } - else { - dpcm += *pScfEsc--; - } - (*pEscBwdCnt)++; - if (pRvlc->conceal_min_esc == CONCEAL_MIN_INIT) { - pRvlc->conceal_min_esc = FDKmax(0,bnds-offset); - } - } - } - pScfBwd[bnds] = factor; - factor -= dpcm; - pRvlc->firstScf = factor; - break; - } - } - } -} - - -/*--------------------------------------------------------------------------------------------- - function: rvlcFinalErrorDetection - - description: Call RVLC concealment if error was detected in decoding process ------------------------------------------------------------------------------------------------ - input: - pointer rvlc structure - - pointer channel info structure ------------------------------------------------------------------------------------------------ - return: - --------------------------------------------------------------------------------------------- */ - -static -void rvlcFinalErrorDetection (CAacDecoderChannelInfo *pAacDecoderChannelInfo, - CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo) -{ - CErRvlcInfo *pRvlc = &pAacDecoderChannelInfo->pComData->overlay.aac.erRvlcInfo; - UCHAR ErrorStatusComplete = 0; - UCHAR ErrorStatusLengthFwd = 0; - UCHAR ErrorStatusLengthBwd = 0; - UCHAR ErrorStatusLengthEscapes = 0; - UCHAR ErrorStatusFirstScf = 0; - UCHAR ErrorStatusLastScf = 0; - UCHAR ErrorStatusFirstNrg = 0; - UCHAR ErrorStatusLastNrg = 0; - UCHAR ErrorStatusFirstIs = 0; - UCHAR ErrorStatusLastIs = 0; - UCHAR ErrorStatusForbiddenCwFwd = 0; - UCHAR ErrorStatusForbiddenCwBwd = 0; - UCHAR ErrorStatusNumEscapesFwd = 0; - UCHAR ErrorStatusNumEscapesBwd = 0; - UCHAR ConcealStatus = 1; - UCHAR currentBlockType; /* short: 0, not short: 1*/ - -#if VERBOSE_RVLC_OUTPUT - CHAR Strategy[60]="No"; - SHORT conceal_max; - SHORT conceal_min; -#endif - - pAacDecoderChannelInfo->pDynData->specificTo.aac.rvlcCurrentScaleFactorOK = 1; - - /* invalid escape words, bit counter unequal zero, forbidden codeword detected */ - if (pRvlc->errorLogRvlc & RVLC_ERROR_FORBIDDEN_CW_DETECTED_FWD) - ErrorStatusForbiddenCwFwd = 1; - - if (pRvlc->errorLogRvlc & RVLC_ERROR_FORBIDDEN_CW_DETECTED_BWD) - ErrorStatusForbiddenCwBwd = 1; - - /* bit counter forward unequal zero */ - if (pRvlc->length_of_rvlc_sf_fwd) - ErrorStatusLengthFwd = 1; - - /* bit counter backward unequal zero */ - if (pRvlc->length_of_rvlc_sf_bwd) - ErrorStatusLengthBwd = 1; - - /* bit counter escape sequences unequal zero */ - if (pRvlc->sf_escapes_present) - if (pRvlc->length_of_rvlc_escapes) - ErrorStatusLengthEscapes = 1; - - if (pRvlc->sf_used) { - /* first decoded scf does not match to global gain in backward direction */ - if (pRvlc->firstScf != (pAacDecoderChannelInfo->pDynData->RawDataInfo.GlobalGain - SF_OFFSET) ) - ErrorStatusFirstScf = 1; - - /* last decoded scf does not match to rev global gain in forward direction */ - if (pRvlc->lastScf != (pRvlc->rev_global_gain - SF_OFFSET) ) - ErrorStatusLastScf = 1; - } - - if (pRvlc->noise_used) { - /* first decoded nrg does not match to dpcm_noise_nrg in backward direction */ - if (pRvlc->firstNrg != (pAacDecoderChannelInfo->pDynData->RawDataInfo.GlobalGain + pRvlc->dpcm_noise_nrg - SF_OFFSET -90 - 256) ) - ErrorStatusFirstNrg = 1; - - /* last decoded nrg does not match to dpcm_noise_last_position in forward direction */ - if (pRvlc->lastNrg != (pRvlc->rev_global_gain + pRvlc->dpcm_noise_last_position - SF_OFFSET - 90 - 256) ) - ErrorStatusLastNrg = 1; - } - - if (pRvlc->intensity_used) { - /* first decoded is position does not match in backward direction */ - if (pRvlc->firstIs != (-SF_OFFSET) ) - ErrorStatusFirstIs = 1; - - /* last decoded is position does not match in forward direction */ - if (pRvlc->lastIs != (pRvlc->dpcm_is_last_position - SF_OFFSET) ) - ErrorStatusLastIs = 1; - } - - /* decoded escapes and used escapes in forward direction do not fit */ - if ((pRvlc->numDecodedEscapeWordsFwd != pRvlc->numDecodedEscapeWordsEsc) && (pRvlc->conceal_max == CONCEAL_MAX_INIT)) { - ErrorStatusNumEscapesFwd = 1; - } - - /* decoded escapes and used escapes in backward direction do not fit */ - if ((pRvlc->numDecodedEscapeWordsBwd != pRvlc->numDecodedEscapeWordsEsc) && (pRvlc->conceal_min == CONCEAL_MIN_INIT)) { - ErrorStatusNumEscapesBwd = 1; - } - -#if VERBOSE_RVLC_OUTPUT - conceal_max = pRvlc->conceal_max; - conceal_min = pRvlc->conceal_min; -#endif - - if ( ErrorStatusLengthEscapes - || ( - ( (pRvlc->conceal_max == CONCEAL_MAX_INIT) - && (pRvlc->numDecodedEscapeWordsFwd != pRvlc->numDecodedEscapeWordsEsc) - && (ErrorStatusLastScf || ErrorStatusLastNrg || ErrorStatusLastIs) ) - - && - - ( (pRvlc->conceal_min == CONCEAL_MIN_INIT) - && (pRvlc->numDecodedEscapeWordsBwd != pRvlc->numDecodedEscapeWordsEsc) - && (ErrorStatusFirstScf || ErrorStatusFirstNrg || ErrorStatusFirstIs) ) - ) - || ( (pRvlc->conceal_max == CONCEAL_MAX_INIT) - && ((pRvlc->rev_global_gain - SF_OFFSET - pRvlc->lastScf) < -15) - ) - || ( (pRvlc->conceal_min == CONCEAL_MIN_INIT) - && ((pAacDecoderChannelInfo->pDynData->RawDataInfo.GlobalGain - SF_OFFSET - pRvlc->firstScf) < -15) - ) - ) { - if ((pRvlc->conceal_max == CONCEAL_MAX_INIT) || (pRvlc->conceal_min == CONCEAL_MIN_INIT)) { - pRvlc->conceal_max = 0; - pRvlc->conceal_min = FDKmax(0, (pRvlc->numWindowGroups-1)*16+pRvlc->maxSfbTransmitted-1); - } - else { - pRvlc->conceal_max = FDKmin(pRvlc->conceal_max,pRvlc->conceal_max_esc); - pRvlc->conceal_min = FDKmax(pRvlc->conceal_min,pRvlc->conceal_min_esc); - } - } - - ErrorStatusComplete = ErrorStatusLastScf || ErrorStatusFirstScf || ErrorStatusLastNrg || ErrorStatusFirstNrg - || ErrorStatusLastIs || ErrorStatusFirstIs || ErrorStatusForbiddenCwFwd || ErrorStatusForbiddenCwBwd - || ErrorStatusLengthFwd || ErrorStatusLengthBwd || ErrorStatusLengthEscapes || ErrorStatusNumEscapesFwd - || ErrorStatusNumEscapesBwd; - - currentBlockType = (GetWindowSequence(&pAacDecoderChannelInfo->icsInfo) == EightShortSequence) ? 0 : 1; - - - if (!ErrorStatusComplete) { - int band; - int group; - int bnds; - int lastSfbIndex; - - lastSfbIndex = (pRvlc->numWindowGroups > 1) ? 16 : 64; - - for (group=0; group < pRvlc->numWindowGroups; group++) { - for (band=0; bandmaxSfbTransmitted; band++) { - bnds = 16*group+band; - pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = pAacDecoderStaticChannelInfo->concealmentInfo.aRvlcPreviousScaleFactor[bnds] = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds]; - } - } - - for (group=0; group < pRvlc->numWindowGroups; group++) - { - for (band=0; bandmaxSfbTransmitted; band++) { - bnds = 16*group+band; - pAacDecoderStaticChannelInfo->concealmentInfo.aRvlcPreviousCodebook[bnds] = pAacDecoderChannelInfo->pDynData->aCodeBook[bnds]; - } - for (; band = 0 && bnds < RVLC_MAX_SFB); - pAacDecoderStaticChannelInfo->concealmentInfo.aRvlcPreviousCodebook[bnds] = ZERO_HCB; - } - } - } - else { - int band; - int group; - - /* A single bit error was detected in decoding of dpcm values. It also could be an error with more bits in decoding - of escapes and dpcm values whereby an illegal codeword followed not directly after the corrupted bits but just - after decoding some more (wrong) scalefactors. Use the smaller scalefactor from forward decoding, backward decoding - and previous frame. */ - if ( ((pRvlc->conceal_min != CONCEAL_MIN_INIT) || (pRvlc->conceal_max != CONCEAL_MAX_INIT)) && (pRvlc->conceal_min <= pRvlc->conceal_max) - && (pAacDecoderStaticChannelInfo->concealmentInfo.rvlcPreviousBlockType == currentBlockType) && pAacDecoderStaticChannelInfo->concealmentInfo.rvlcPreviousScaleFactorOK - && pRvlc->sf_concealment && ConcealStatus ) - { - BidirectionalEstimation_UseScfOfPrevFrameAsReference (pAacDecoderChannelInfo, pAacDecoderStaticChannelInfo); - ConcealStatus=0; -#if VERBOSE_RVLC_OUTPUT - FDKstrcpy(Strategy,"Yes (BidirectionalEstimation_UseScfOfPrevFrameAsReference)"); -#endif - } - - /* A single bit error was detected in decoding of dpcm values. It also could be an error with more bits in decoding - of escapes and dpcm values whereby an illegal codeword followed not directly after the corrupted bits but just - after decoding some more (wrong) scalefactors. Use the smaller scalefactor from forward and backward decoding. */ - if ( (pRvlc->conceal_min <= pRvlc->conceal_max) && ((pRvlc->conceal_min != CONCEAL_MIN_INIT) || (pRvlc->conceal_max != CONCEAL_MAX_INIT)) - && !(pAacDecoderStaticChannelInfo->concealmentInfo.rvlcPreviousScaleFactorOK && pRvlc->sf_concealment && (pAacDecoderStaticChannelInfo->concealmentInfo.rvlcPreviousBlockType == currentBlockType)) - && ConcealStatus ) - { - BidirectionalEstimation_UseLowerScfOfCurrentFrame (pAacDecoderChannelInfo); - ConcealStatus=0; -#if VERBOSE_RVLC_OUTPUT - FDKstrcpy(Strategy,"Yes (BidirectionalEstimation_UseLowerScfOfCurrentFrame)"); -#endif - } - - /* No errors were detected in decoding of escapes and dpcm values however the first and last value - of a group (is,nrg,sf) is incorrect */ - if ( (pRvlc->conceal_min <= pRvlc->conceal_max) && ((ErrorStatusLastScf && ErrorStatusFirstScf) - || (ErrorStatusLastNrg && ErrorStatusFirstNrg) || (ErrorStatusLastIs && ErrorStatusFirstIs)) - && !(ErrorStatusForbiddenCwFwd || ErrorStatusForbiddenCwBwd || ErrorStatusLengthEscapes ) && ConcealStatus) - { - StatisticalEstimation (pAacDecoderChannelInfo); - ConcealStatus=0; -#if VERBOSE_RVLC_OUTPUT - FDKstrcpy(Strategy,"Yes (StatisticalEstimation)"); -#endif - } - - /* A error with more bits in decoding of escapes and dpcm values was detected. Use the smaller scalefactor from forward - decoding, backward decoding and previous frame. */ - if ( (pRvlc->conceal_min <= pRvlc->conceal_max) && pAacDecoderStaticChannelInfo->concealmentInfo.rvlcPreviousScaleFactorOK && pRvlc->sf_concealment - && (pAacDecoderStaticChannelInfo->concealmentInfo.rvlcPreviousBlockType == currentBlockType) && ConcealStatus ) - { - PredictiveInterpolation(pAacDecoderChannelInfo, pAacDecoderStaticChannelInfo); - ConcealStatus=0; -#if VERBOSE_RVLC_OUTPUT - FDKstrcpy(Strategy,"Yes (PredictiveInterpolation)"); -#endif - } - - /* Call frame concealment, because no better strategy was found. Setting the scalefactors to zero is done for debugging - purposes */ - if (ConcealStatus) { - for (group=0; group < pRvlc->numWindowGroups; group++) { - for (band=0; bandmaxSfbTransmitted; band++) { - pAacDecoderChannelInfo->pDynData->aScaleFactor[16*group+band] = 0; - } - } - pAacDecoderChannelInfo->pDynData->specificTo.aac.rvlcCurrentScaleFactorOK = 0; -#if VERBOSE_RVLC_OUTPUT - FDKstrcpy(Strategy,"Yes (FrameConcealment)"); -#endif - } - } - -#if VERBOSE_RVLC_OUTPUT - DebugOutputDistortedBitstreams(pRvlc,pAacDecoderChannelInfo,ErrorStatusLengthFwd,ErrorStatusLengthBwd, - ErrorStatusLengthEscapes,ErrorStatusFirstScf,ErrorStatusLastScf, - ErrorStatusFirstNrg,ErrorStatusLastNrg,ErrorStatusFirstIs,ErrorStatusLastIs, - ErrorStatusForbiddenCwFwd,ErrorStatusForbiddenCwBwd,ErrorStatusNumEscapesFwd, - ErrorStatusNumEscapesBwd,conceal_max,conceal_min,Strategy); -#endif -} - - -/*--------------------------------------------------------------------------------------------- - function: CRvlc_Read - - description: Read RVLC ESC1 data (side info) from bitstream. ------------------------------------------------------------------------------------------------ - input: - pointer rvlc structure - - pointer channel info structure - - pointer bitstream structure ------------------------------------------------------------------------------------------------ - return: - --------------------------------------------------------------------------------------------- */ - -void CRvlc_Read ( - CAacDecoderChannelInfo *pAacDecoderChannelInfo, - HANDLE_FDK_BITSTREAM bs) -{ - CErRvlcInfo *pRvlc = &pAacDecoderChannelInfo->pComData->overlay.aac.erRvlcInfo; - - int group,band; - - /* RVLC long specific initialization Init part 1 of 2 */ - pRvlc->numWindowGroups = GetWindowGroups(&pAacDecoderChannelInfo->icsInfo); - pRvlc->maxSfbTransmitted = GetScaleFactorBandsTransmitted(&pAacDecoderChannelInfo->icsInfo); - pRvlc->noise_used = 0; /* noise detection */ - pRvlc->dpcm_noise_nrg = 0; /* only for debugging */ - pRvlc->dpcm_noise_last_position = 0; /* only for debugging */ - pRvlc->length_of_rvlc_escapes = -1; /* default value is used for error detection and concealment */ - - /* read only error sensitivity class 1 data (ESC 1 - data) */ - pRvlc->sf_concealment = FDKreadBits(bs,1); /* #1 */ - pRvlc->rev_global_gain = FDKreadBits(bs,8); /* #2 */ - - if (GetWindowSequence(&pAacDecoderChannelInfo->icsInfo) == EightShortSequence) { - pRvlc->length_of_rvlc_sf = FDKreadBits(bs,11); /* #3 */ - } - else { - pRvlc->length_of_rvlc_sf = FDKreadBits(bs,9); /* #3 */ - } - - /* check if noise codebook is used */ - for (group = 0; group < pRvlc->numWindowGroups; group++) { - for (band=0; band < pRvlc->maxSfbTransmitted; band++) { - if (pAacDecoderChannelInfo->pDynData->aCodeBook[16*group+band] == NOISE_HCB) { - pRvlc->noise_used = 1; - break; - } - } - } - - if (pRvlc->noise_used) - pRvlc->dpcm_noise_nrg = FDKreadBits(bs, 9); /* #4 PNS */ - - pRvlc->sf_escapes_present = FDKreadBits(bs, 1); /* #5 */ - - if ( pRvlc->sf_escapes_present) { - pRvlc->length_of_rvlc_escapes = FDKreadBits(bs, 8); /* #6 */ - } - - if (pRvlc->noise_used) { - pRvlc->dpcm_noise_last_position = FDKreadBits(bs, 9); /* #7 PNS */ - pRvlc->length_of_rvlc_sf -= 9; - } - - pRvlc->length_of_rvlc_sf_fwd = pRvlc->length_of_rvlc_sf; - pRvlc->length_of_rvlc_sf_bwd = pRvlc->length_of_rvlc_sf; -} - - -/*--------------------------------------------------------------------------------------------- - function: CRvlc_Decode - - description: Decode rvlc data - The function reads both the escape sequences and the scalefactors in forward - and backward direction. If an error occured during decoding process which can - not be concealed with the rvlc concealment frame concealment will be initiated. - Then the element "rvlcCurrentScaleFactorOK" in the decoder channel info is set - to 0 otherwise it is set to 1. ------------------------------------------------------------------------------------------------ - input: - pointer rvlc structure - - pointer channel info structure - - pointer to persistent channel info structure - - pointer bitstream structure ------------------------------------------------------------------------------------------------ - return: ErrorStatus = AAC_DEC_OK --------------------------------------------------------------------------------------------- */ - -void CRvlc_Decode ( - CAacDecoderChannelInfo *pAacDecoderChannelInfo, - CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, - HANDLE_FDK_BITSTREAM bs - ) -{ - CErRvlcInfo *pRvlc = &pAacDecoderChannelInfo->pComData->overlay.aac.erRvlcInfo; - INT bitCntOffst; - UINT saveBitCnt; - - rvlcInit(pRvlc,pAacDecoderChannelInfo,bs); - - /* save bitstream position */ - saveBitCnt = FDKgetBitCnt(bs); - -#if RVLC_ADVANCED_BITSTREAM_ERROR_GENERATOR_SF - GenerateSingleBitError(pRvlc, - &(pRvlc->bitstreamIndexRvlFwd), - pRvlc->length_of_rvlc_sf, - 0); -#endif - -#if RVLC_ADVANCED_BITSTREAM_ERROR_GENERATOR_ESC - if (pRvlc->sf_escapes_present) - GenerateSingleBitError(pRvlc, - &(pRvlc->bitstreamIndexEsc), - pRvlc->length_of_rvlc_escapes, - 1); -#endif - - if ( pRvlc->sf_escapes_present) - rvlcDecodeEscapes(pRvlc, pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfEsc, bs); - - rvlcDecodeForward(pRvlc,pAacDecoderChannelInfo, bs); - rvlcDecodeBackward(pRvlc,pAacDecoderChannelInfo, bs); - rvlcFinalErrorDetection(pAacDecoderChannelInfo, pAacDecoderStaticChannelInfo); - - pAacDecoderChannelInfo->pDynData->specificTo.aac.rvlcIntensityUsed = pRvlc->intensity_used; - pAacDecoderChannelInfo->data.aac.PnsData.PnsActive = pRvlc->noise_used; - - /* restore bitstream position */ - bitCntOffst = saveBitCnt - FDKgetBitCnt(bs); - if( bitCntOffst ) { - FDKpushBiDirectional(bs, bitCntOffst); - } -} - -void CRvlc_ElementCheck ( - CAacDecoderChannelInfo *pAacDecoderChannelInfo[], - CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo[], - const UINT flags, - const INT elChannels - ) -{ - int ch; - - /* Required for MPS residuals. */ - if (pAacDecoderStaticChannelInfo == NULL) { - return; - } - - /* RVLC specific sanity checks */ - if ( (flags & AC_ER_RVLC) && (elChannels == 2)) { /* to be reviewed */ - if ( ( (pAacDecoderChannelInfo[0]->pDynData->specificTo.aac.rvlcCurrentScaleFactorOK == 0) || - (pAacDecoderChannelInfo[1]->pDynData->specificTo.aac.rvlcCurrentScaleFactorOK == 0) ) - && pAacDecoderChannelInfo[0]->pComData->jointStereoData.MsMaskPresent ) { - pAacDecoderChannelInfo[0]->pDynData->specificTo.aac.rvlcCurrentScaleFactorOK = 0; - pAacDecoderChannelInfo[1]->pDynData->specificTo.aac.rvlcCurrentScaleFactorOK = 0; - } - - if ( (pAacDecoderChannelInfo[0]->pDynData->specificTo.aac.rvlcCurrentScaleFactorOK == 0) - && (pAacDecoderChannelInfo[1]->pDynData->specificTo.aac.rvlcCurrentScaleFactorOK == 1) - && (pAacDecoderChannelInfo[1]->pDynData->specificTo.aac.rvlcIntensityUsed == 1) ){ - pAacDecoderChannelInfo[1]->pDynData->specificTo.aac.rvlcCurrentScaleFactorOK = 0; - } - } - - for (ch = 0; ch < elChannels; ch ++) - { - pAacDecoderStaticChannelInfo[ch]->concealmentInfo.rvlcPreviousBlockType = (GetWindowSequence(&pAacDecoderChannelInfo[ch]->icsInfo) == EightShortSequence) ? 0 : 1; - if (flags & AC_ER_RVLC) { - pAacDecoderStaticChannelInfo[ch]->concealmentInfo.rvlcPreviousScaleFactorOK = pAacDecoderChannelInfo[ch]->pDynData->specificTo.aac.rvlcCurrentScaleFactorOK; - } - else { - pAacDecoderStaticChannelInfo[ch]->concealmentInfo.rvlcPreviousScaleFactorOK = 0; - } - } -} - - diff --git a/libAACdec/src/rvlc.h b/libAACdec/src/rvlc.h deleted file mode 100644 index 18d5fa3..0000000 --- a/libAACdec/src/rvlc.h +++ /dev/null @@ -1,134 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief Defines structures and prototypes for RVLC - \author Robert Weidner -*/ - -#ifndef RVLC_H -#define RVLC_H - - - -#include "aacdecoder.h" -#include "channel.h" -#include "rvlc_info.h" - -/* ------------------------------------------------------------------- */ -/* errorLogRvlc: A word of 32 bits used for logging possible errors */ -/* within RVLC in case of distorted bitstreams. */ -/* ------------------------------------------------------------------- */ -#define RVLC_ERROR_ALL_ESCAPE_WORDS_INVALID 0x80000000 /* ESC-Dec During RVLC-Escape-decoding there have been more bits decoded as there are available */ -#define RVLC_ERROR_RVL_SUM_BIT_COUNTER_BELOW_ZERO_FWD 0x40000000 /* RVL-Dec negative sum-bitcounter during RVL-fwd-decoding (long+shrt) */ -#define RVLC_ERROR_RVL_SUM_BIT_COUNTER_BELOW_ZERO_BWD 0x20000000 /* RVL-Dec negative sum-bitcounter during RVL-fwd-decoding (long+shrt) */ -#define RVLC_ERROR_FORBIDDEN_CW_DETECTED_FWD 0x08000000 /* RVL-Dec forbidden codeword detected fwd (long+shrt) */ -#define RVLC_ERROR_FORBIDDEN_CW_DETECTED_BWD 0x04000000 /* RVL-Dec forbidden codeword detected bwd (long+shrt) */ - - - -void CRvlc_Read (CAacDecoderChannelInfo *pAacDecoderChannelInfo, - HANDLE_FDK_BITSTREAM bs); - -void CRvlc_Decode (CAacDecoderChannelInfo *pAacDecoderChannelInfo, - CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, - HANDLE_FDK_BITSTREAM bs); - -/** - * \brief performe sanity checks to the channel data corresponding to one channel element. - * \param pAacDecoderChannelInfo - * \param pAacDecoderStaticChannelInfo - * \param elChannels amount of channels of the channel element. - */ -void CRvlc_ElementCheck ( - CAacDecoderChannelInfo *pAacDecoderChannelInfo[], - CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo[], - const UINT flags, - const INT elChannels - ); - - - - -#endif /* RVLC_H */ diff --git a/libAACdec/src/rvlc_info.h b/libAACdec/src/rvlc_info.h deleted file mode 100644 index 63934af..0000000 --- a/libAACdec/src/rvlc_info.h +++ /dev/null @@ -1,176 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief Defines structures for RVLC - \author Robert Weidner -*/ -#ifndef RVLC_INFO_H -#define RVLC_INFO_H - - - -#define FWD 0 /* bitstream decoding direction forward (RVL coded part) */ -#define BWD 1 /* bitstream decoding direction backward (RVL coded part) */ - -#define MAX_RVL 7 /* positive RVLC escape */ -#define MIN_RVL -7 /* negative RVLC escape */ -#define MAX_ALLOWED_DPCM_INDEX 14 /* the maximum allowed index of a decoded dpcm value (offset 'TABLE_OFFSET' incl --> must be subtracted) */ -#define TABLE_OFFSET 7 /* dpcm offset of valid output values of rvl table decoding, the rvl table ouly returns positive values, therefore the offset */ -#define MAX_LEN_RVLC_CODE_WORD 9 /* max length of a RVL codeword in bits */ -#define MAX_LEN_RVLC_ESCAPE_WORD 20 /* max length of huffman coded RVLC escape word in bits */ - -#define DPCM_NOISE_NRG_BITS 9 -#define SF_OFFSET 100 /* offset for correcting scf value */ - -#define CONCEAL_MAX_INIT 1311 /* arbitrary value */ -#define CONCEAL_MIN_INIT -1311 /* arbitrary value */ - -#define RVLC_MAX_SFB ((8) * (16)) - -/* sideinfo of RVLC */ -typedef struct -{ - /* ------- ESC 1 Data: --------- */ /* order of RVLC-bitstream components in bitstream (RVLC-initialization), every component appears only once in bitstream */ - INT sf_concealment; /* 1 */ - INT rev_global_gain; /* 2 */ - SHORT length_of_rvlc_sf; /* 3 */ /* original value, gets modified (subtract 9) in case of noise (PNS); is kept for later use */ - INT dpcm_noise_nrg; /* 4 optional */ - INT sf_escapes_present; /* 5 */ - SHORT length_of_rvlc_escapes; /* 6 optional */ - INT dpcm_noise_last_position; /* 7 optional */ - - INT dpcm_is_last_position; - - SHORT length_of_rvlc_sf_fwd; /* length_of_rvlc_sf used for forward decoding */ - SHORT length_of_rvlc_sf_bwd; /* length_of_rvlc_sf used for backward decoding */ - - /* for RVL-Codeword decoder to distinguish between fwd and bwd decoding */ - SHORT *pRvlBitCnt_RVL; - USHORT *pBitstrIndxRvl_RVL; - - UCHAR numWindowGroups; - UCHAR maxSfbTransmitted; - UCHAR first_noise_group; - UCHAR first_noise_band; - UCHAR direction; - - /* bitstream indices */ - USHORT bitstreamIndexRvlFwd; /* base address of RVL-coded-scalefactor data (ESC 2) for forward decoding */ - USHORT bitstreamIndexRvlBwd; /* base address of RVL-coded-scalefactor data (ESC 2) for backward decoding */ - USHORT bitstreamIndexEsc; /* base address where RVLC-escapes start (ESC 2) */ - - /* decoding trees */ - const UINT *pHuffTreeRvlCodewds; - const UINT *pHuffTreeRvlcEscape; - - /* escape counters */ - UCHAR numDecodedEscapeWordsFwd; /* when decoding RVL-codes forward */ - UCHAR numDecodedEscapeWordsBwd; /* when decoding RVL-codes backward */ - UCHAR numDecodedEscapeWordsEsc; /* when decoding the escape-Words */ - - SCHAR noise_used; - SCHAR intensity_used; - SCHAR sf_used; - - SHORT firstScf; - SHORT lastScf; - SHORT firstNrg; - SHORT lastNrg; - SHORT firstIs; - SHORT lastIs; - - /* ------ RVLC error detection ------ */ - UINT errorLogRvlc; /* store RVLC errors */ - SHORT conceal_min; /* is set at backward decoding */ - SHORT conceal_max; /* is set at forward decoding */ - SHORT conceal_min_esc; /* is set at backward decoding */ - SHORT conceal_max_esc; /* is set at forward decoding */ -} CErRvlcInfo; - -typedef CErRvlcInfo RVLC_INFO; /* temp */ - - - -#endif /* RVLC_INFO_H */ diff --git a/libAACdec/src/rvlcbit.cpp b/libAACdec/src/rvlcbit.cpp deleted file mode 100644 index 6efbb93..0000000 --- a/libAACdec/src/rvlcbit.cpp +++ /dev/null @@ -1,131 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief RVLC bitstream reading - \author Robert Weidner -*/ - -#include "rvlcbit.h" - - -/*--------------------------------------------------------------------------------------------- - function: rvlcReadBitFromBitstream - - description: This function returns a bit from the bitstream according to read direction. - It is called very often, therefore it makes sense to inline it (runtime). ------------------------------------------------------------------------------------------------ - input: - bitstream - - pPosition - - readDirection ------------------------------------------------------------------------------------------------ - return: - bit from bitstream --------------------------------------------------------------------------------------------- */ - -UCHAR rvlcReadBitFromBitstream (HANDLE_FDK_BITSTREAM bs, - USHORT *pPosition, - UCHAR readDirection) -{ - UINT bit; - INT readBitOffset = *pPosition-FDKgetBitCnt(bs); - - if( readBitOffset ) { - FDKpushBiDirectional(bs, readBitOffset); - } - - if (readDirection == FWD) { - bit = FDKreadBits(bs, 1); - - *pPosition += 1; - } else { - /* to be replaced with a brother function of FDKreadBits() */ - bit = FDKreadBits(bs, 1); - FDKpushBack(bs, 2); - - *pPosition -= 1; - } - - return (bit); -} - diff --git a/libAACdec/src/rvlcbit.h b/libAACdec/src/rvlcbit.h deleted file mode 100644 index 02fba88..0000000 --- a/libAACdec/src/rvlcbit.h +++ /dev/null @@ -1,103 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/***************************** MPEG-4 AAC Decoder *************************** - - Author(s): Robert Weidner (DSP Solutions) - Description: RVLC Decoder: Bitstream reading - -*******************************************************************************/ - -#ifndef RVLCBIT_H -#define RVLCBIT_H - - - -#include "rvlc.h" - -UCHAR rvlcReadBitFromBitstream (HANDLE_FDK_BITSTREAM bs, - USHORT *pPosition, - UCHAR readDirection); - - -#endif /* RVLCBIT_H */ diff --git a/libAACdec/src/rvlcconceal.cpp b/libAACdec/src/rvlcconceal.cpp deleted file mode 100644 index cf33dd5..0000000 --- a/libAACdec/src/rvlcconceal.cpp +++ /dev/null @@ -1,697 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief rvlc concealment - \author Josef Hoepfl -*/ - -#include "rvlcconceal.h" - - -#include "block.h" -#include "rvlc.h" - -/*--------------------------------------------------------------------------------------------- - function: calcRefValFwd - - description: The function determines the scalefactor which is closed to the scalefactorband - conceal_min. The same is done for intensity data and noise energies. ------------------------------------------------------------------------------------------------ - output: - reference value scf - - reference value internsity data - - reference value noise energy ------------------------------------------------------------------------------------------------ - return: - --------------------------------------------------------------------------------------------- */ - -static -void calcRefValFwd (CErRvlcInfo *pRvlc, - CAacDecoderChannelInfo *pAacDecoderChannelInfo, - int *refIsFwd, - int *refNrgFwd, - int *refScfFwd) -{ - int band,bnds,group,startBand; - int idIs,idNrg,idScf; - int conceal_min,conceal_group_min; - int MaximumScaleFactorBands; - - - if (GetWindowSequence(&pAacDecoderChannelInfo->icsInfo) == EightShortSequence) - MaximumScaleFactorBands = 16; - else - MaximumScaleFactorBands = 64; - - conceal_min = pRvlc->conceal_min % MaximumScaleFactorBands; - conceal_group_min = pRvlc->conceal_min / MaximumScaleFactorBands; - - /* calculate first reference value for approach in forward direction */ - idIs = idNrg = idScf = 1; - - /* set reference values */ - *refIsFwd = - SF_OFFSET; - *refNrgFwd = pAacDecoderChannelInfo->pDynData->RawDataInfo.GlobalGain - SF_OFFSET - 90 - 256; - *refScfFwd = pAacDecoderChannelInfo->pDynData->RawDataInfo.GlobalGain - SF_OFFSET; - - startBand = conceal_min-1; - for (group=conceal_group_min; group >= 0; group--) { - for (band=startBand; band >= 0; band--) { - bnds = 16*group+band; - switch (pAacDecoderChannelInfo->pDynData->aCodeBook[bnds]) { - case ZERO_HCB: - break; - case INTENSITY_HCB: - case INTENSITY_HCB2: - if (idIs) { - *refIsFwd = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds]; - idIs=0; /* reference value has been set */ - } - break; - case NOISE_HCB: - if (idNrg) { - *refNrgFwd = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds]; - idNrg=0; /* reference value has been set */ - } - break ; - default: - if (idScf) { - *refScfFwd = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds]; - idScf=0; /* reference value has been set */ - } - break; - } - } - startBand = pRvlc->maxSfbTransmitted-1; - } - -} - -/*--------------------------------------------------------------------------------------------- - function: calcRefValBwd - - description: The function determines the scalefactor which is closed to the scalefactorband - conceal_max. The same is done for intensity data and noise energies. ------------------------------------------------------------------------------------------------ - output: - reference value scf - - reference value internsity data - - reference value noise energy ------------------------------------------------------------------------------------------------ - return: - --------------------------------------------------------------------------------------------- */ - -static -void calcRefValBwd (CErRvlcInfo *pRvlc, - CAacDecoderChannelInfo *pAacDecoderChannelInfo, - int *refIsBwd, - int *refNrgBwd, - int *refScfBwd) -{ - int band,bnds,group,startBand; - int idIs,idNrg,idScf; - int conceal_max,conceal_group_max; - int MaximumScaleFactorBands; - - if (GetWindowSequence(&pAacDecoderChannelInfo->icsInfo) == EightShortSequence) - MaximumScaleFactorBands = 16; - else - MaximumScaleFactorBands = 64; - - conceal_max = pRvlc->conceal_max % MaximumScaleFactorBands; - conceal_group_max = pRvlc->conceal_max / MaximumScaleFactorBands; - - /* calculate first reference value for approach in backward direction */ - idIs = idNrg = idScf = 1; - - /* set reference values */ - *refIsBwd = pRvlc->dpcm_is_last_position - SF_OFFSET; - *refNrgBwd = pRvlc->rev_global_gain + pRvlc->dpcm_noise_last_position - SF_OFFSET - 90 - 256 + pRvlc->dpcm_noise_nrg; - *refScfBwd = pRvlc->rev_global_gain - SF_OFFSET; - - startBand=conceal_max+1; - - /* if needed, re-set reference values */ - for (group=conceal_group_max; group < pRvlc->numWindowGroups; group++) { - for (band=startBand; band < pRvlc->maxSfbTransmitted; band++) { - bnds = 16*group+band; - switch (pAacDecoderChannelInfo->pDynData->aCodeBook[bnds]) { - case ZERO_HCB: - break; - case INTENSITY_HCB: - case INTENSITY_HCB2: - if (idIs) { - *refIsBwd = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[bnds]; - idIs=0; /* reference value has been set */ - } - break; - case NOISE_HCB: - if (idNrg) { - *refNrgBwd = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[bnds]; - idNrg=0; /* reference value has been set */ - } - break ; - default: - if (idScf) { - *refScfBwd = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[bnds]; - idScf=0; /* reference value has been set */ - } - break; - } - } - startBand=0; - } - -} - - -/*--------------------------------------------------------------------------------------------- - function: BidirectionalEstimation_UseLowerScfOfCurrentFrame - - description: This approach by means of bidirectional estimation is generally performed when - a single bit error has been detected, the bit error can be isolated between - 'conceal_min' and 'conceal_max' and the 'sf_concealment' flag is not set. The - sets of scalefactors decoded in forward and backward direction are compared - with each other. The smaller scalefactor will be considered as the correct one - respectively. The reconstruction of the scalefactors with this approach archieve - good results in audio quality. The strategy must be applied to scalefactors, - intensity data and noise energy seperately. ------------------------------------------------------------------------------------------------ - output: Concealed scalefactor, noise energy and intensity data between conceal_min and - conceal_max ------------------------------------------------------------------------------------------------ - return: - --------------------------------------------------------------------------------------------- */ - -void BidirectionalEstimation_UseLowerScfOfCurrentFrame (CAacDecoderChannelInfo *pAacDecoderChannelInfo) -{ - CErRvlcInfo *pRvlc = &pAacDecoderChannelInfo->pComData->overlay.aac.erRvlcInfo; - int band,bnds,startBand,endBand,group; - int conceal_min,conceal_max; - int conceal_group_min,conceal_group_max; - int MaximumScaleFactorBands; - - if (GetWindowSequence(&pAacDecoderChannelInfo->icsInfo) == EightShortSequence) { - MaximumScaleFactorBands = 16; - } - else { - MaximumScaleFactorBands = 64; - } - - /* If an error was detected just in forward or backward direction, set the corresponding border for concealment to a - appropriate scalefactor band. The border is set to first or last sfb respectively, because the error will possibly - not follow directly after the corrupt bit but just after decoding some more (wrong) scalefactors. */ - if (pRvlc->conceal_min == CONCEAL_MIN_INIT) - pRvlc->conceal_min = 0; - - if (pRvlc->conceal_max == CONCEAL_MAX_INIT) - pRvlc->conceal_max = (pRvlc->numWindowGroups-1)*16+pRvlc->maxSfbTransmitted-1; - - conceal_min = pRvlc->conceal_min % MaximumScaleFactorBands; - conceal_group_min = pRvlc->conceal_min / MaximumScaleFactorBands; - conceal_max = pRvlc->conceal_max % MaximumScaleFactorBands; - conceal_group_max = pRvlc->conceal_max / MaximumScaleFactorBands; - - if (pRvlc->conceal_min == pRvlc->conceal_max) { - - int refIsFwd,refNrgFwd,refScfFwd; - int refIsBwd,refNrgBwd,refScfBwd; - - bnds = pRvlc->conceal_min; - calcRefValFwd(pRvlc,pAacDecoderChannelInfo,&refIsFwd,&refNrgFwd,&refScfFwd); - calcRefValBwd(pRvlc,pAacDecoderChannelInfo,&refIsBwd,&refNrgBwd,&refScfBwd); - - switch (pAacDecoderChannelInfo->pDynData->aCodeBook[bnds]) { - case ZERO_HCB: - break; - case INTENSITY_HCB: - case INTENSITY_HCB2: - if (refIsFwd < refIsBwd) - pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = refIsFwd; - else - pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = refIsBwd; - break; - case NOISE_HCB: - if (refNrgFwd < refNrgBwd) - pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = refNrgFwd; - else - pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = refNrgBwd; - break; - default: - if (refScfFwd < refScfBwd) - pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = refScfFwd; - else - pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = refScfBwd; - break; - } - } - else { - pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[pRvlc->conceal_max] = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[pRvlc->conceal_max]; - pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[pRvlc->conceal_min] = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[pRvlc->conceal_min]; - - /* consider the smaller of the forward and backward decoded value as the correct one */ - startBand = conceal_min; - if (conceal_group_min == conceal_group_max) - endBand = conceal_max; - else - endBand = pRvlc->maxSfbTransmitted-1; - - for (group=conceal_group_min; group <= conceal_group_max; group++) { - for (band=startBand; band <= endBand; band++) { - bnds = 16*group+band; - if (pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds] < pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[bnds]) - pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds]; - else - pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[bnds]; - } - startBand = 0; - if ((group+1) == conceal_group_max) - endBand = conceal_max; - } - } - - /* now copy all data to the output buffer which needs not to be concealed */ - if (conceal_group_min == 0) - endBand = conceal_min; - else - endBand = pRvlc->maxSfbTransmitted; - for (group=0; group <= conceal_group_min; group++) { - for (band=0; band < endBand; band++) { - bnds = 16*group+band; - pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds]; - } - if ((group+1) == conceal_group_min) - endBand = conceal_min; - } - - startBand = conceal_max+1; - for (group=conceal_group_max; group < pRvlc->numWindowGroups; group++) { - for (band=startBand; band < pRvlc->maxSfbTransmitted; band++) { - bnds = 16*group+band; - pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[bnds]; - } - startBand = 0; - } -} - -/*--------------------------------------------------------------------------------------------- - function: BidirectionalEstimation_UseScfOfPrevFrameAsReference - - description: This approach by means of bidirectional estimation is generally performed when - a single bit error has been detected, the bit error can be isolated between - 'conceal_min' and 'conceal_max', the 'sf_concealment' flag is set and the - previous frame has the same block type as the current frame. The scalefactor - decoded in forward and backward direction and the scalefactor of the previous - frame are compared with each other. The smaller scalefactor will be considered - as the correct one. At this the codebook of the previous and current frame must - be of the same set (scf, nrg, is) in each scalefactorband. Otherwise the - scalefactor of the previous frame is not considered in the minimum calculation. - The reconstruction of the scalefactors with this approach archieve good results - in audio quality. The strategy must be applied to scalefactors, intensity data - and noise energy seperately. ------------------------------------------------------------------------------------------------ - output: Concealed scalefactor, noise energy and intensity data between conceal_min and - conceal_max ------------------------------------------------------------------------------------------------ - return: - --------------------------------------------------------------------------------------------- */ - -void BidirectionalEstimation_UseScfOfPrevFrameAsReference ( - CAacDecoderChannelInfo *pAacDecoderChannelInfo, - CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo - ) -{ - CErRvlcInfo *pRvlc = &pAacDecoderChannelInfo->pComData->overlay.aac.erRvlcInfo; - int band,bnds,startBand,endBand,group; - int conceal_min,conceal_max; - int conceal_group_min,conceal_group_max; - int MaximumScaleFactorBands; - int commonMin; - - if (GetWindowSequence(&pAacDecoderChannelInfo->icsInfo) == EightShortSequence) { - MaximumScaleFactorBands = 16; - } - else { - MaximumScaleFactorBands = 64; - } - - /* If an error was detected just in forward or backward direction, set the corresponding border for concealment to a - appropriate scalefactor band. The border is set to first or last sfb respectively, because the error will possibly - not follow directly after the corrupt bit but just after decoding some more (wrong) scalefactors. */ - if (pRvlc->conceal_min == CONCEAL_MIN_INIT) - pRvlc->conceal_min = 0; - - if (pRvlc->conceal_max == CONCEAL_MAX_INIT) - pRvlc->conceal_max = (pRvlc->numWindowGroups-1)*16+pRvlc->maxSfbTransmitted-1; - - conceal_min = pRvlc->conceal_min % MaximumScaleFactorBands; - conceal_group_min = pRvlc->conceal_min / MaximumScaleFactorBands; - conceal_max = pRvlc->conceal_max % MaximumScaleFactorBands; - conceal_group_max = pRvlc->conceal_max / MaximumScaleFactorBands; - - pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[pRvlc->conceal_max] = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[pRvlc->conceal_max]; - pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[pRvlc->conceal_min] = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[pRvlc->conceal_min]; - - /* consider the smaller of the forward and backward decoded value as the correct one */ - startBand = conceal_min; - if (conceal_group_min == conceal_group_max) - endBand = conceal_max; - else - endBand = pRvlc->maxSfbTransmitted-1; - - for (group=conceal_group_min; group <= conceal_group_max; group++) { - for (band=startBand; band <= endBand; band++) { - bnds = 16*group+band; - switch (pAacDecoderChannelInfo->pDynData->aCodeBook[bnds]) { - case ZERO_HCB: - pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = 0; - break; - - case INTENSITY_HCB: - case INTENSITY_HCB2: - if ( (pAacDecoderStaticChannelInfo->concealmentInfo.aRvlcPreviousCodebook[bnds]==INTENSITY_HCB) || (pAacDecoderStaticChannelInfo->concealmentInfo.aRvlcPreviousCodebook[bnds]==INTENSITY_HCB2) ) { - commonMin = FDKmin(pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds],pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[bnds]); - pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = FDKmin(commonMin, pAacDecoderStaticChannelInfo->concealmentInfo.aRvlcPreviousScaleFactor[bnds]); - } - else { - pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = FDKmin(pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds],pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[bnds]); - } - break; - - case NOISE_HCB: - if ( pAacDecoderStaticChannelInfo->concealmentInfo.aRvlcPreviousCodebook[bnds]==NOISE_HCB ) { - commonMin = FDKmin(pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds],pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[bnds]); - pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = FDKmin(commonMin, pAacDecoderStaticChannelInfo->concealmentInfo.aRvlcPreviousScaleFactor[bnds]); - } else { - pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = FDKmin(pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds],pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[bnds]); - } - break; - - default: - if ( (pAacDecoderStaticChannelInfo->concealmentInfo.aRvlcPreviousCodebook[bnds]!=ZERO_HCB) - && (pAacDecoderStaticChannelInfo->concealmentInfo.aRvlcPreviousCodebook[bnds]!=NOISE_HCB) - && (pAacDecoderStaticChannelInfo->concealmentInfo.aRvlcPreviousCodebook[bnds]!=INTENSITY_HCB) - && (pAacDecoderStaticChannelInfo->concealmentInfo.aRvlcPreviousCodebook[bnds]!=INTENSITY_HCB2) ) - { - commonMin = FDKmin(pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds], pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[bnds]); - pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = FDKmin(commonMin, pAacDecoderStaticChannelInfo->concealmentInfo.aRvlcPreviousScaleFactor[bnds]); - } else { - pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = FDKmin(pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds],pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[bnds]); - } - break; - } - } - startBand = 0; - if ((group+1) == conceal_group_max) - endBand = conceal_max; - } - - /* now copy all data to the output buffer which needs not to be concealed */ - if (conceal_group_min == 0) - endBand = conceal_min; - else - endBand = pRvlc->maxSfbTransmitted; - for (group=0; group <= conceal_group_min; group++) { - for (band=0; band < endBand; band++) { - bnds = 16*group+band; - pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds]; - } - if ((group+1) == conceal_group_min) - endBand = conceal_min; - } - - startBand = conceal_max+1; - for (group=conceal_group_max; group < pRvlc->numWindowGroups; group++) { - for (band=startBand; band < pRvlc->maxSfbTransmitted; band++) { - bnds = 16*group+band; - pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[bnds]; - } - startBand = 0; - } -} - -/*--------------------------------------------------------------------------------------------- - function: StatisticalEstimation - - description: This approach by means of statistical estimation is generally performed when - both the start value and the end value are different and no further errors have - been detected. Considering the forward and backward decoded scalefactors, the - set with the lower scalefactors in sum will be considered as the correct one. - The scalefactors are differentially encoded. Normally it would reach to compare - one pair of the forward and backward decoded scalefactors to specify the lower - set. But having detected no further errors does not necessarily mean the absence - of errors. Therefore all scalefactors decoded in forward and backward direction - are summed up seperately. The set with the lower sum will be used. The strategy - must be applied to scalefactors, intensity data and noise energy seperately. ------------------------------------------------------------------------------------------------ - output: Concealed scalefactor, noise energy and intensity data ------------------------------------------------------------------------------------------------ - return: - --------------------------------------------------------------------------------------------- */ - -void StatisticalEstimation (CAacDecoderChannelInfo *pAacDecoderChannelInfo) -{ - CErRvlcInfo *pRvlc = &pAacDecoderChannelInfo->pComData->overlay.aac.erRvlcInfo; - int band,bnds,group; - int sumIsFwd,sumIsBwd; /* sum of intensity data forward/backward */ - int sumNrgFwd,sumNrgBwd; /* sum of noise energy data forward/backward */ - int sumScfFwd,sumScfBwd; /* sum of scalefactor data forward/backward */ - int useIsFwd,useNrgFwd,useScfFwd; /* the flags signals the elements which are used for the final result */ - int MaximumScaleFactorBands; - - if (GetWindowSequence(&pAacDecoderChannelInfo->icsInfo) == EightShortSequence) - MaximumScaleFactorBands = 16; - else - MaximumScaleFactorBands = 64; - - sumIsFwd = sumIsBwd = sumNrgFwd = sumNrgBwd = sumScfFwd = sumScfBwd = 0; - useIsFwd = useNrgFwd = useScfFwd = 0; - - /* calculate sum of each group (scf,nrg,is) of forward and backward direction */ - for (group=0; groupnumWindowGroups; group++) { - for (band=0; band < pRvlc->maxSfbTransmitted; band++) { - bnds = 16*group+band; - switch (pAacDecoderChannelInfo->pDynData->aCodeBook[bnds]) { - case ZERO_HCB: - break; - - case INTENSITY_HCB: - case INTENSITY_HCB2: - sumIsFwd += pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds]; - sumIsBwd += pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[bnds]; - break; - - case NOISE_HCB: - sumNrgFwd += pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds]; - sumNrgBwd += pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[bnds]; - break ; - - default: - sumScfFwd += pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds]; - sumScfBwd += pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[bnds]; - break; - } - } - } - - /* find for each group (scf,nrg,is) the correct direction */ - if ( sumIsFwd < sumIsBwd ) - useIsFwd = 1; - - if ( sumNrgFwd < sumNrgBwd ) - useNrgFwd = 1; - - if ( sumScfFwd < sumScfBwd ) - useScfFwd = 1; - - /* conceal each group (scf,nrg,is) */ - for (group=0; groupnumWindowGroups; group++) { - for (band=0; band < pRvlc->maxSfbTransmitted; band++) { - bnds = 16*group+band; - switch (pAacDecoderChannelInfo->pDynData->aCodeBook[bnds]) { - case ZERO_HCB: - break; - - case INTENSITY_HCB: - case INTENSITY_HCB2: - if (useIsFwd) - pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds]; - else - pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[bnds]; - break; - - case NOISE_HCB: - if (useNrgFwd) - pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds]; - else - pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[bnds]; - break ; - - default: - if (useScfFwd) - pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds]; - else - pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[bnds]; - break; - } - } - } -} - - -/*--------------------------------------------------------------------------------------------- - description: Approach by means of predictive interpolation - This approach by means of predictive estimation is generally performed when - the error cannot be isolated between 'conceal_min' and 'conceal_max', the - 'sf_concealment' flag is set and the previous frame has the same block type - as the current frame. Check for each scalefactorband if the same type of data - (scalefactor, internsity data, noise energies) is transmitted. If so use the - scalefactor (intensity data, noise energy) in the current frame. Otherwise set - the scalefactor (intensity data, noise energy) for this scalefactorband to zero. ------------------------------------------------------------------------------------------------ - output: Concealed scalefactor, noise energy and intensity data ------------------------------------------------------------------------------------------------ - return: - --------------------------------------------------------------------------------------------- */ - -void PredictiveInterpolation ( - CAacDecoderChannelInfo *pAacDecoderChannelInfo, - CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo - ) -{ - CErRvlcInfo *pRvlc = &pAacDecoderChannelInfo->pComData->overlay.aac.erRvlcInfo; - int band,bnds,group; - int MaximumScaleFactorBands; - int commonMin; - - if (GetWindowSequence(&pAacDecoderChannelInfo->icsInfo) == EightShortSequence) - MaximumScaleFactorBands = 16; - else - MaximumScaleFactorBands = 64; - - for (group=0; groupnumWindowGroups; group++) { - for (band=0; band < pRvlc->maxSfbTransmitted; band++) { - bnds = 16*group+band; - switch (pAacDecoderChannelInfo->pDynData->aCodeBook[bnds]) { - case ZERO_HCB: - pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = 0; - break; - - case INTENSITY_HCB: - case INTENSITY_HCB2: - if ( (pAacDecoderStaticChannelInfo->concealmentInfo.aRvlcPreviousCodebook[bnds]==INTENSITY_HCB) || (pAacDecoderStaticChannelInfo->concealmentInfo.aRvlcPreviousCodebook[bnds]==INTENSITY_HCB2) ) { - commonMin = FDKmin(pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds],pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[bnds]); - pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = FDKmin(commonMin, pAacDecoderStaticChannelInfo->concealmentInfo.aRvlcPreviousScaleFactor[bnds]); - } - else { - pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = -110; - } - break; - - case NOISE_HCB: - if ( pAacDecoderStaticChannelInfo->concealmentInfo.aRvlcPreviousCodebook[bnds]==NOISE_HCB ) { - commonMin = FDKmin(pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds],pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[bnds]); - pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = FDKmin(commonMin, pAacDecoderStaticChannelInfo->concealmentInfo.aRvlcPreviousScaleFactor[bnds]); - } - else { - pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = -110; - } - break; - - default: - if ( (pAacDecoderStaticChannelInfo->concealmentInfo.aRvlcPreviousCodebook[bnds]!=ZERO_HCB) - && (pAacDecoderStaticChannelInfo->concealmentInfo.aRvlcPreviousCodebook[bnds]!=NOISE_HCB) - && (pAacDecoderStaticChannelInfo->concealmentInfo.aRvlcPreviousCodebook[bnds]!=INTENSITY_HCB) - && (pAacDecoderStaticChannelInfo->concealmentInfo.aRvlcPreviousCodebook[bnds]!=INTENSITY_HCB2) ) { - commonMin = FDKmin(pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds],pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[bnds]); - pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = FDKmin(commonMin, pAacDecoderStaticChannelInfo->concealmentInfo.aRvlcPreviousScaleFactor[bnds]); - } - else { - pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = 0; - } - break; - } - } - } -} - diff --git a/libAACdec/src/rvlcconceal.h b/libAACdec/src/rvlcconceal.h deleted file mode 100644 index 750cbcd..0000000 --- a/libAACdec/src/rvlcconceal.h +++ /dev/null @@ -1,112 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief rvlc concealment - \author Josef Hoepfl -*/ - -#ifndef RVLCCONCEAL_H -#define RVLCCONCEAL_H - - - -#include "rvlc.h" - -void BidirectionalEstimation_UseLowerScfOfCurrentFrame(CAacDecoderChannelInfo *pAacDecoderChannelInfo); - -void BidirectionalEstimation_UseScfOfPrevFrameAsReference( - CAacDecoderChannelInfo *pAacDecoderChannelInfo, - CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo - ); - -void StatisticalEstimation (CAacDecoderChannelInfo *pAacDecoderChannelInfo); - -void PredictiveInterpolation ( - CAacDecoderChannelInfo *pAacDecoderChannelInfo, - CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo - ); - - -#endif /* RVLCCONCEAL_H */ diff --git a/libAACenc/include/aacenc_lib.h b/libAACenc/include/aacenc_lib.h deleted file mode 100644 index 63c3697..0000000 --- a/libAACenc/include/aacenc_lib.h +++ /dev/null @@ -1,1239 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/**************************** MPEG-4 HE-AAC Encoder ************************** - - Initial author: M. Lohwasser -******************************************************************************/ - -/** - * \file aacenc_lib.h - * \brief FDK AAC Encoder library interface header file. - * -\mainpage Introduction - -\section Scope - -This document describes the high-level interface and usage of the ISO/MPEG-2/4 AAC Encoder -library developed by the Fraunhofer Institute for Integrated Circuits (IIS). - -The library implements encoding on the basis of the MPEG-2 and MPEG-4 AAC Low-Complexity -standard, and depending on the library's configuration, MPEG-4 High-Efficiency AAC v2 and/or AAC-ELD standard. - -All references to SBR (Spectral Band Replication) are only applicable to HE-AAC or AAC-ELD versions -of the library. All references to PS (Parametric Stereo) are only applicable to HE-AAC v2 -versions of the library. - -\section encBasics Encoder Basics - -This document can only give a rough overview about the ISO/MPEG-2 and ISO/MPEG-4 AAC audio coding -standard. To understand all the terms in this document, you are encouraged to read the following documents. - -- ISO/IEC 13818-7 (MPEG-2 AAC), which defines the syntax of MPEG-2 AAC audio bitstreams. -- ISO/IEC 14496-3 (MPEG-4 AAC, subparts 1 and 4), which defines the syntax of MPEG-4 AAC audio bitstreams. -- Lutzky, Schuller, Gayer, Krämer, Wabnik, "A guideline to audio codec delay", 116th AES Convention, May 8, 2004 - -MPEG Advanced Audio Coding is based on a time-to-frequency mapping of the signal. The signal is -partitioned into overlapping portions and transformed into frequency domain. The spectral components -are then quantized and coded. \n -An MPEG-2 or MPEG-4 AAC audio bitstream is composed of frames. Contrary to MPEG-1/2 Layer-3 (mp3), the -length of individual frames is not restricted to a fixed number of bytes, but can take on any length -between 1 and 768 bytes. - - -\page LIBUSE Library Usage - -\section InterfaceDescription API Files - -All API header files are located in the folder /include of the release package. All header files -are provided for usage in C/C++ programs. The AAC encoder library API functions are located at -aacenc_lib.h. - -In binary releases the encoder core resides in statically linkable libraries called for example -libAACenc.a/libFDK.a (LINUX) or FDK_fastaaclib.lib (MS Visual C++) for the plain AAC-LC core encoder -and libSBRenc.a (LINUX) or FDK_sbrEncLib.lib (MS Visual C++) for the SBR (Spectral Band -Replication) and PS (Parametric Stereo) modules. - -\section CallingSequence Calling Sequence - -For encoding of ISO/MPEG-2/4 AAC bitstreams the following sequence is mandatory. Input read and output -write functions as well as the corresponding open and close functions are left out, since they may be -implemented differently according to the user's specific requirements. The example implementation in -main.cpp uses file-based input/output. - --# Call aacEncOpen() to allocate encoder instance with required \ref encOpen "configuration".\n -\dontinclude main.cpp -\skipline hAacEncoder = -\skipline aacEncOpen --# Call aacEncoder_SetParam() for each parameter to be set. AOT, samplingrate, channelMode, bitrate and transport type are \ref encParams "mandatory". -\code - ErrorStatus = aacEncoder_SetParam(hAacEncoder, parameter, value); -\endcode --# Call aacEncEncode() with NULL parameters to \ref encReconf "initialize" encoder instance with present parameter set. -\skipline aacEncEncode --# Call aacEncInfo() to retrieve a configuration data block to be transmitted out of band. This is required when using RFC3640 or RFC3016 like transport. -\dontinclude main.cpp -\skipline encInfo -\skipline aacEncInfo --# Encode input audio data in loop. -\skip Encode as long as -\skipline do -\until { -Feed \ref feedInBuf "input buffer" with new audio data and provide input/output \ref bufDes "arguments" to aacEncEncode(). -\skipline aacEncEncode -\until ; -Write \ref writeOutData "output data" to file or audio device. \skipline while --# Call aacEncClose() and destroy encoder instance. -\skipline aacEncClose - -\section encOpen Encoder Instance Allocation - -The assignment of the aacEncOpen() function is very flexible and can be used in the following way. -- If the amount of memory consumption is not an issue, the encoder instance can be allocated -for the maximum number of possible audio channels (for example 6 or 8) with the full functional range supported by the library. -This is the default open procedure for the AAC encoder if memory consumption does not need to be minimized. -\code aacEncOpen(&hAacEncoder,0,0) \endcode -- If the required MPEG-4 AOTs do not call for the full functional range of the library, encoder modules can be allocated selectively. -\verbatim ------------------------------------------------------- - AAC | SBR | PS | MD | FLAGS | value ------+-----+-----+----+-----------------------+------- - X | - | - | - | (0x01) | 0x01 - X | X | - | - | (0x01|0x02) | 0x03 - X | X | X | - | (0x01|0x02|0x04) | 0x07 - X | - | - | X | (0x01 |0x10) | 0x11 - X | X | - | X | (0x01|0x02 |0x10) | 0x13 - X | X | X | X | (0x01|0x02|0x04|0x10) | 0x17 ------------------------------------------------------- - - AAC: Allocate AAC Core Encoder module. - - SBR: Allocate Spectral Band Replication module. - - PS: Allocate Parametric Stereo module. - - MD: Allocate Meta Data module within AAC encoder. -\endverbatim -\code aacEncOpen(&hAacEncoder,value,0) \endcode -- Specifying the maximum number of channels to be supported in the encoder instance can be done as follows. - - For example allocate an encoder instance which supports 2 channels for all supported AOTs. - The library itself may be capable of encoding up to 6 or 8 channels but in this example only 2 channel encoding is required and thus only buffers for 2 channels are allocated to save data memory. -\code aacEncOpen(&hAacEncoder,0,2) \endcode - - Additionally the maximum number of supported channels in the SBR module can be denoted separately.\n - In this example the encoder instance provides a maximum of 6 channels out of which up to 2 channels support SBR. - This encoder instance can produce for example 5.1 channel AAC-LC streams or stereo HE-AAC (v2) streams. - HE-AAC 5.1 multi channel is not possible since only 2 out of 6 channels support SBR, which saves data memory. -\code aacEncOpen(&hAacEncoder,0,6|(2<<8)) \endcode -\n - -\section bufDes Input/Output Arguments - -\subsection allocIOBufs Provide Buffer Descriptors -In the present encoder API, the input and output buffers are described with \ref AACENC_BufDesc "buffer descriptors". This mechanism allows a flexible handling -of input and output buffers without impact to the actual encoding call. Optional buffers are necessary e.g. for ancillary data, meta data input or additional output -buffers describing superframing data in DAB+ or DRM+.\n -At least one input buffer for audio input data and one output buffer for bitstream data must be allocated. The input buffer size can be a user defined multiple -of the number of input channels. PCM input data will be copied from the user defined PCM buffer to an internal input buffer and so input data can be less than one AAC audio frame. -The output buffer size should be 6144 bits per channel excluding the LFE channel. -If the output data does not fit into the provided buffer, an AACENC_ERROR will be returned by aacEncEncode(). -\dontinclude main.cpp -\skipline inputBuffer -\until outputBuffer -All input and output buffer must be clustered in input and output buffer arrays. -\skipline inBuffer -\until outBufferElSize -Allocate buffer descriptors -\skipline AACENC_BufDesc -\skipline AACENC_BufDesc -Initialize input buffer descriptor -\skipline inBufDesc -\until bufElSizes -Initialize output buffer descriptor -\skipline outBufDesc -\until bufElSizes - -\subsection argLists Provide Input/Output Argument Lists -The input and output arguments of an aacEncEncode() call are described in argument structures. -\dontinclude main.cpp -\skipline AACENC_InArgs -\skipline AACENC_OutArgs - -\section feedInBuf Feed Input Buffer -The input buffer should be handled as a modulo buffer. New audio data in the form of pulse-code- -modulated samples (PCM) must be read from external and be fed to the input buffer depending on its -fill level. The required sample bitrate (represented by the data type INT_PCM which is 16, 24 or 32 -bits wide) is fixed and depends on library configuration (usually 16 bit). - -\dontinclude main.cpp -\skipline WAV_InputRead -\until ; -After the encoder's internal buffer is fed with incoming audio samples, and aacEncEncode() -processed the new input data, update/move remaining samples in input buffer, simulating a modulo buffer: -\skipline outargs.numInSamples>0 -\until } - -\section writeOutData Output Bitstream Data -If any AAC bitstream data is available, write it to output file or device. This can be done once the -following condition is true: -\dontinclude main.cpp -\skip Valid bitstream available -\skipline outargs - -\skipline outBytes>0 - -If you use file I/O then for example call mpegFileWrite_Write() from the library libMpegFileWrite - -\dontinclude main.cpp -\skipline mpegFileWrite_Write - -\section cfgMetaData Meta Data Configuration - -If the present library is configured with Metadata support, it is possible to insert meta data side info into the generated -audio bitstream while encoding. - -To work with meta data the encoder instance has to be \ref encOpen "allocated" with meta data support. The meta data mode must be be configured with -the ::AACENC_METADATA_MODE parameter and aacEncoder_SetParam() function. -\code aacEncoder_SetParam(hAacEncoder, AACENC_METADATA_MODE, 0-2); \endcode - -This configuration indicates how to embed meta data into bitstrem. Either no insertion, MPEG or ETSI style. -The meta data itself must be specified within the meta data setup structure AACENC_MetaData. - -Changing one of the AACENC_MetaData setup parameters can be achieved from outside the library within ::IN_METADATA_SETUP input -buffer. There is no need to supply meta data setup structure every frame. If there is no new meta setup data available, the -encoder uses the previous setup or the default configuration in initial state. - -In general the audio compressor and limiter within the encoder library can be configured with the ::AACENC_METADATA_DRC_PROFILE parameter -AACENC_MetaData::drc_profile and and AACENC_MetaData::comp_profile. -\n - -\section encReconf Encoder Reconfiguration - -The encoder library allows reconfiguration of the encoder instance with new settings -continuously between encoding frames. Each parameter to be changed must be set with -a single aacEncoder_SetParam() call. The internal status of each parameter can be -retrieved with an aacEncoder_GetParam() call.\n -There is no stand-alone reconfiguration function available. When parameters were -modified from outside the library, an internal control mechanism triggers the necessary -reconfiguration process which will be applied at the beginning of the following -aacEncEncode() call. This state can be observed from external via the AACENC_INIT_STATUS -and aacEncoder_GetParam() function. The reconfiguration process can also be applied -immediately when all parameters of an aacEncEncode() call are NULL with a valid encoder -handle.\n\n -The internal reconfiguration process can be controlled from extern with the following access. -\code aacEncoder_SetParam(hAacEncoder, AACENC_CONTROL_STATE, AACENC_CTRLFLAGS); \endcode - - -\section encParams Encoder Parametrization - -All parameteres listed in ::AACENC_PARAM can be modified within an encoder instance. - -\subsection encMandatory Mandatory Encoder Parameters -The following parameters must be specified when the encoder instance is initialized. -\code -aacEncoder_SetParam(hAacEncoder, AACENC_AOT, value); -aacEncoder_SetParam(hAacEncoder, AACENC_BITRATE, value); -aacEncoder_SetParam(hAacEncoder, AACENC_SAMPLERATE, value); -aacEncoder_SetParam(hAacEncoder, AACENC_CHANNELMODE, value); -\endcode -Beyond that is an internal auto mode which preinitizializes the ::AACENC_BITRATE parameter -if the parameter was not set from extern. The bitrate depends on the number of effective -channels and sampling rate and is determined as follows. -\code -AAC-LC (AOT_AAC_LC): 1.5 bits per sample -HE-AAC (AOT_SBR): 0.625 bits per sample (dualrate sbr) -HE-AAC (AOT_SBR): 1.125 bits per sample (downsampled sbr) -HE-AAC v2 (AOT_PS): 0.5 bits per sample -\endcode - -\subsection channelMode Channel Mode Configuration -The input audio data is described with the ::AACENC_CHANNELMODE parameter in the -aacEncoder_SetParam() call. It is not possible to use the encoder instance with a 'number of -input channels' argument. Instead, the channelMode must be set as follows. -\code aacEncoder_SetParam(hAacEncoder, AACENC_CHANNELMODE, value); \endcode -The parameter is specified in ::CHANNEL_MODE and can be mapped from the number of input channels -in the following way. -\dontinclude main.cpp -\skip CHANNEL_MODE chMode = MODE_INVALID; -\until return - -\subsection encQual Audio Quality Considerations -The default encoder configuration is suggested to be used. Encoder tools such as TNS and PNS -are activated by default and are internally controlled (see \ref BEHAVIOUR_TOOLS). - -There is an additional quality parameter called ::AACENC_AFTERBURNER. In the default -configuration this quality switch is deactivated because it would cause a workload -increase which might be significant. If workload is not an issue in the application -we recommended to activate this feature. -\code aacEncoder_SetParam(hAacEncoder, AACENC_AFTERBURNER, 1); \endcode - -\subsection encELD ELD Auto Configuration Mode -For ELD configuration a so called auto configurator is available which configures SBR and the SBR ratio by itself. -The configurator is used when the encoder parameter ::AACENC_SBR_MODE and ::AACENC_SBR_RATIO are not set explicitely. - -Based on sampling rate and chosen bitrate per channel a reasonable SBR configuration will be used. -\verbatim ------------------------------------------------------------- - Sampling Rate | Channel Bitrate | SBR | SBR Ratio ------------------+-----------------+------+----------------- - ]min, 16] kHz | min - 27999 | on | downsampled SBR - | 28000 - max | off | --- ------------------+-----------------+------+----------------- - ]16 - 24] kHz | min - 39999 | on | downsampled SBR - | 40000 - max | off | --- ------------------+-----------------+------+----------------- - ]24 - 32] kHz | min - 27999 | on | dualrate SBR - | 28000 - 55999 | on | downsampled SBR - | 56000 - max | off | --- ------------------+-----------------+------+----------------- - ]32 - 44.1] kHz | min - 63999 | on | dualrate SBR - | 64000 - max | off | --- ------------------+-----------------+------+----------------- - ]44.1 - 48] kHz | min - 63999 | on | dualrate SBR - | 64000 - max | off | --- ------------------------------------------------------------- -\endverbatim - - -\section audiochCfg Audio Channel Configuration -The MPEG standard refers often to the so-called Channel Configuration. This Channel Configuration is used for a fixed Channel -Mapping. The configurations 1-7 are predefined in MPEG standard and used for implicit signalling within the encoded bitstream. -For user defined Configurations the Channel Configuration is set to 0 and the Channel Mapping must be explecitly described with an appropriate -Program Config Element. The present Encoder implementation does not allow the user to configure this Channel Configuration from -extern. The Encoder implementation supports fixed Channel Modes which are mapped to Channel Configuration as follow. -\verbatim -------------------------------------------------------------------------------- - ChannelMode | ChCfg | front_El | side_El | back_El | lfe_El ------------------------+--------+---------------+----------+----------+-------- -MODE_1 | 1 | SCE | | | -MODE_2 | 2 | CPE | | | -MODE_1_2 | 3 | SCE, CPE | | | -MODE_1_2_1 | 4 | SCE, CPE | | SCE | -MODE_1_2_2 | 5 | SCE, CPE | | CPE | -MODE_1_2_2_1 | 6 | SCE, CPE | | CPE | LFE -MODE_1_2_2_2_1 | 7 | SCE, CPE, CPE | | CPE | LFE ------------------------+--------+---------------+----------+----------+-------- -MODE_7_1_REAR_SURROUND | 0 | SCE, CPE | | CPE, CPE | LFE -MODE_7_1_FRONT_CENTER | 0 | SCE, CPE, CPE | | CPE | LFE -------------------------------------------------------------------------------- - - SCE: Single Channel Element. - - CPE: Channel Pair. - - SCE: Low Frequency Element. -\endverbatim - -Moreover, the Table describes all fixed Channel Elements for each Channel Mode which are assigned to a speaker arrangement. The -arrangement includes front, side, back and lfe Audio Channel Elements.\n -This mapping of Audio Channel Elements is defined in MPEG standard for Channel Config 1-7. The Channel assignment for MODE_1_1, -MODE_2_2 and MODE_2_1 is used from the ARIB standard. All other configurations are defined as suggested in MPEG.\n -In case of Channel Config 0 or writing matrix mixdown coefficients, the encoder enables the writing of Program Config Element -itself as described in \ref encPCE. The configuration used in Program Config Element refers to the denoted Table.\n -Beside the Channel Element assignment the Channel Modes are resposible for audio input data channel mapping. The Channel Mapping -of the audio data depends on the selected ::AACENC_CHANNELORDER which can be MPEG or WAV like order.\n -Following Table describes the complete channel mapping for both Channel Order configurations. -\verbatim ---------------------------------------------------------------------------------------- -ChannelMode | MPEG-Channelorder | WAV-Channelorder ------------------------+---+---+---+---+---+---+---+---+---+---+---+---+---+---+---+--- -MODE_1 | 0 | | | | | | | | 0 | | | | | | | -MODE_2 | 0 | 1 | | | | | | | 0 | 1 | | | | | | -MODE_1_2 | 0 | 1 | 2 | | | | | | 2 | 0 | 1 | | | | | -MODE_1_2_1 | 0 | 1 | 2 | 3 | | | | | 2 | 0 | 1 | 3 | | | | -MODE_1_2_2 | 0 | 1 | 2 | 3 | 4 | | | | 2 | 0 | 1 | 3 | 4 | | | -MODE_1_2_2_1 | 0 | 1 | 2 | 3 | 4 | 5 | | | 2 | 0 | 1 | 4 | 5 | 3 | | -MODE_1_2_2_2_1 | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | 2 | 6 | 7 | 0 | 1 | 4 | 5 | 3 ------------------------+---+---+---+---+---+---+---+---+---+---+---+---+---+---+---+--- -MODE_7_1_REAR_SURROUND | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | 2 | 0 | 1 | 6 | 7 | 4 | 5 | 3 -MODE_7_1_FRONT_CENTER | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | 2 | 6 | 7 | 0 | 1 | 4 | 5 | 3 ---------------------------------------------------------------------------------------- -\endverbatim - -The denoted mapping is important for correct audio channel assignment when using MPEG or WAV ordering. The incoming audio -channels are distributed MPEG like starting at the front channels and ending at the back channels. The distribution is used as -described in Table concering Channel Config and fix channel elements. Please see the following example for clarification. - -\verbatim -Example: MODE_1_2_2_1 - WAV-Channelorder 5.1 ------------------------------------------- - Input Channel | Coder Channel ---------------------+--------------------- - 2 (front center) | 0 (SCE channel) - 0 (left center) | 1 (1st of 1st CPE) - 1 (right center) | 2 (2nd of 1st CPE) - 4 (left surround) | 3 (1st of 2nd CPE) - 5 (right surround) | 4 (2nd of 2nd CPE) - 3 (LFE) | 5 (LFE) ------------------------------------------- -\endverbatim - - -\section suppBitrates Supported Bitrates - -The FDK AAC Encoder provides a wide range of supported bitrates. -The minimum and maximum allowed bitrate depends on the Audio Object Type. For AAC-LC the minimum -bitrate is the bitrate that is required to write the most basic and minimal valid bitstream. -It consists of the bitstream format header information and other static/mandatory information -within the AAC payload. The maximum AAC framesize allowed by the MPEG-4 standard -determines the maximum allowed bitrate for AAC-LC. For HE-AAC and HE-AAC v2 a library internal -look-up table is used. - -A good working point in terms of audio quality, sampling rate and bitrate, is at 1 to 1.5 -bits/audio sample for AAC-LC, 0.625 bits/audio sample for dualrate HE-AAC, 1.125 bits/audio sample -for downsampled HE-AAC and 0.5 bits/audio sample for HE-AAC v2. -For example for one channel with a sampling frequency of 48 kHz, the range from -48 kbit/s to 72 kbit/s achieves reasonable audio quality for AAC-LC. - -For HE-AAC and HE-AAC v2 the lowest possible audio input sampling frequency is 16 kHz because then the -AAC-LC core encoder operates in dual rate mode at its lowest possible sampling frequency, which is 8 kHz. -HE-AAC v2 requires stereo input audio data. - -Please note that in HE-AAC or HE-AAC v2 mode the encoder supports much higher bitrates than are -appropriate for HE-AAC or HE-AAC v2. For example, at a bitrate of more than 64 kbit/s for a stereo -audio signal at 44.1 kHz it usually makes sense to use AAC-LC, which will produce better audio -quality at that bitrate than HE-AAC or HE-AAC v2. - -\section reommendedConfig Recommended Sampling Rate and Bitrate Combinations - -The following table provides an overview of recommended encoder configuration parameters -which we determined by virtue of numerous listening tests. - -\subsection reommendedConfigLC AAC-LC, HE-AAC, HE-AACv2 in Dualrate SBR mode. -\verbatim ------------------------------------------------------------------------------------ -Audio Object Type | Bit Rate Range | Supported | Preferred | No. of - | [bit/s] | Sampling Rates | Sampl. | Chan. - | | [kHz] | Rate | - | | | [kHz] | --------------------+------------------+-----------------------+------------+------- -AAC LC + SBR + PS | 8000 - 11999 | 22.05, 24.00 | 24.00 | 2 -AAC LC + SBR + PS | 12000 - 17999 | 32.00 | 32.00 | 2 -AAC LC + SBR + PS | 18000 - 39999 | 32.00, 44.10, 48.00 | 44.10 | 2 -AAC LC + SBR + PS | 40000 - 56000 | 32.00, 44.10, 48.00 | 48.00 | 2 --------------------+------------------+-----------------------+------------+------- -AAC LC + SBR | 8000 - 11999 | 22.05, 24.00 | 24.00 | 1 -AAC LC + SBR | 12000 - 17999 | 32.00 | 32.00 | 1 -AAC LC + SBR | 18000 - 39999 | 32.00, 44.10, 48.00 | 44.10 | 1 -AAC LC + SBR | 40000 - 56000 | 32.00, 44.10, 48.00 | 48.00 | 1 -AAC LC + SBR | 16000 - 27999 | 32.00, 44.10, 48.00 | 32.00 | 2 -AAC LC + SBR | 28000 - 63999 | 32.00, 44.10, 48.00 | 44.10 | 2 -AAC LC + SBR | 64000 - 128000 | 32.00, 44.10, 48.00 | 48.00 | 2 --------------------+------------------+-----------------------+------------+------- -AAC LC + SBR | 64000 - 69999 | 32.00, 44.10, 48.00 | 32.00 | 5, 5.1 -AAC LC + SBR | 70000 - 159999 | 32.00, 44.10, 48.00 | 44.10 | 5, 5.1 -AAC LC + SBR | 160000 - 245999 | 32.00, 44.10, 48.00 | 48.00 | 5 -AAC LC + SBR | 160000 - 265999 | 32.00, 44.10, 48.00 | 48.00 | 5.1 --------------------+------------------+-----------------------+------------+------- -AAC LC | 8000 - 15999 | 11.025, 12.00, 16.00 | 12.00 | 1 -AAC LC | 16000 - 23999 | 16.00 | 16.00 | 1 -AAC LC | 24000 - 31999 | 16.00, 22.05, 24.00 | 24.00 | 1 -AAC LC | 32000 - 55999 | 32.00 | 32.00 | 1 -AAC LC | 56000 - 160000 | 32.00, 44.10, 48.00 | 44.10 | 1 -AAC LC | 160001 - 288000 | 48.00 | 48.00 | 1 --------------------+------------------+-----------------------+------------+------- -AAC LC | 16000 - 23999 | 11.025, 12.00, 16.00 | 12.00 | 2 -AAC LC | 24000 - 31999 | 16.00 | 16.00 | 2 -AAC LC | 32000 - 39999 | 16.00, 22.05, 24.00 | 22.05 | 2 -AAC LC | 40000 - 95999 | 32.00 | 32.00 | 2 -AAC LC | 96000 - 111999 | 32.00, 44.10, 48.00 | 32.00 | 2 -AAC LC | 112000 - 320001 | 32.00, 44.10, 48.00 | 44.10 | 2 -AAC LC | 320002 - 576000 | 48.00 | 48.00 | 2 --------------------+------------------+-----------------------+------------+------- -AAC LC | 160000 - 239999 | 32.00 | 32.00 | 5, 5.1 -AAC LC | 240000 - 279999 | 32.00, 44.10, 48.00 | 32.00 | 5, 5.1 -AAC LC | 280000 - 800000 | 32.00, 44.10, 48.00 | 44.10 | 5, 5.1 ------------------------------------------------------------------------------------ -\endverbatim \n - -\subsection reommendedConfigLD AAC-LD, AAC-ELD, AAC-ELD with SBR in Dualrate SBR mode. -\verbatim ------------------------------------------------------------------------------------ -Audio Object Type | Bit Rate Range | Supported | Preferred | No. of - | [bit/s] | Sampling Rates | Sampl. | Chan. - | | [kHz] | Rate | - | | | [kHz] | --------------------+------------------+-----------------------+------------+------- -ELD + SBR | 18000 - 24999 | 32.00 - 44.10 | 32.00 | 1 -ELD + SBR | 25000 - 31999 | 32.00 - 48.00 | 32.00 | 1 -ELD + SBR | 32000 - 64000 | 32.00 - 48.00 | 48.00 | 1 --------------------+------------------+-----------------------+------------+------- -ELD + SBR | 32000 - 51999 | 32.00 - 48.00 | 44.10 | 2 -ELD + SBR | 52000 - 128000 | 32.00 - 48.00 | 48.00 | 2 --------------------+------------------+-----------------------+------------+------- -ELD + SBR | 72000 - 160000 | 44.10 - 48.00 | 48.00 | 3 --------------------+------------------+-----------------------+------------+------- -ELD + SBR | 96000 - 212000 | 44.10 - 48.00 | 48.00 | 4 --------------------+------------------+-----------------------+------------+------- -ELD + SBR | 120000 - 246000 | 44.10 - 48.00 | 48.00 | 5 --------------------+------------------+-----------------------+------------+------- -ELD + SBR | 120000 - 266000 | 44.10 - 48.00 | 48.00 | 5.1 --------------------+------------------+-----------------------+------------+------- -LD, ELD | 16000 - 19999 | 16.00 - 24.00 | 16.00 | 1 -LD, ELD | 20000 - 39999 | 16.00 - 32.00 | 24.00 | 1 -LD, ELD | 40000 - 49999 | 22.05 - 32.00 | 32.00 | 1 -LD, ELD | 50000 - 61999 | 24.00 - 44.10 | 32.00 | 1 -LD, ELD | 62000 - 84999 | 32.00 - 48.00 | 44.10 | 1 -LD, ELD | 85000 - 192000 | 44.10 - 48.00 | 48.00 | 1 --------------------+------------------+-----------------------+------------+------- -LD, ELD | 64000 - 75999 | 24.00 - 32.00 | 32.00 | 2 -LD, ELD | 76000 - 97999 | 24.00 - 44.10 | 32.00 | 2 -LD, ELD | 98000 - 135999 | 32.00 - 48.00 | 44.10 | 2 -LD, ELD | 136000 - 384000 | 44.10 - 48.00 | 48.00 | 2 --------------------+------------------+-----------------------+------------+------- -LD, ELD | 96000 - 113999 | 24.00 - 32.00 | 32.00 | 3 -LD, ELD | 114000 - 146999 | 24.00 - 44.10 | 32.00 | 3 -LD, ELD | 147000 - 203999 | 32.00 - 48.00 | 44.10 | 3 -LD, ELD | 204000 - 576000 | 44.10 - 48.00 | 48.00 | 3 --------------------+------------------+-----------------------+------------+------- -LD, ELD | 128000 - 151999 | 24.00 - 32.00 | 32.00 | 4 -LD, ELD | 152000 - 195999 | 24.00 - 44.10 | 32.00 | 4 -LD, ELD | 196000 - 271999 | 32.00 - 48.00 | 44.10 | 4 -LD, ELD | 272000 - 768000 | 44.10 - 48.00 | 48.00 | 4 --------------------+------------------+-----------------------+------------+------- -LD, ELD | 160000 - 189999 | 24.00 - 32.00 | 32.00 | 5 -LD, ELD | 190000 - 244999 | 24.00 - 44.10 | 32.00 | 5 -LD, ELD | 245000 - 339999 | 32.00 - 48.00 | 44.10 | 5 -LD, ELD | 340000 - 960000 | 44.10 - 48.00 | 48.00 | 5 ------------------------------------------------------------------------------------ -\endverbatim \n - -\subsection reommendedConfigELD AAC-ELD with SBR in Downsampled SBR mode. -\verbatim ------------------------------------------------------------------------------------ -Audio Object Type | Bit Rate Range | Supported | Preferred | No. of - | [bit/s] | Sampling Rates | Sampl. | Chan. - | | [kHz] | Rate | - | | | [kHz] | --------------------+------------------+-----------------------+------------+------- -ELD + SBR | 18000 - 24999 | 16.00 - 22.05 | 22.05 | 1 -(downsampled SBR) | 25000 - 35999 | 22.05 - 32.00 | 24.00 | 1 - | 36000 - 64000 | 32.00 - 48.00 | 32.00 | 1 ------------------------------------------------------------------------------------ -\endverbatim \n - - -\page ENCODERBEHAVIOUR Encoder Behaviour - -\section BEHAVIOUR_BANDWIDTH Bandwidth - -The FDK AAC encoder usually does not use the full frequency range of the input signal, but restricts the bandwidth -according to certain library-internal settings. They can be changed in the table "bandWidthTable" in the -file bandwidth.cpp (if available). - -The encoder API provides the ::AACENC_BANDWIDTH parameter to adjust the bandwidth explicitly. -\code -aacEncoder_SetParam(hAacEncoder, AACENC_BANDWIDTH, value); -\endcode - -However it is not recommended to change these settings, because they are based on numerious listening -tests and careful tweaks to ensure the best overall encoding quality. - -Theoretically a signal of for example 48 kHz can contain frequencies up to 24 kHz, but to use this full range -in an audio encoder usually does not make sense. Usually the encoder has a very limited amount of -bits to spend (typically 128 kbit/s for stereo 48 kHz content) and to allow full range bandwidth would -waste a lot of these bits for frequencies the human ear is hardly able to perceive anyway, if at all. Hence it -is wise to use the available bits for the really important frequency range and just skip the rest. -At lower bitrates (e. g. <= 80 kbit/s for stereo 48 kHz content) the encoder will choose an even smaller -bandwidth, because an encoded signal with smaller bandwidth and hence less artifacts sounds better than a signal -with higher bandwidth but then more coding artefacts across all frequencies. These artefacts would occur if -small bitrates and high bandwidths are chosen because the available bits are just not enough to encode all -frequencies well. - -Unfortunately some people evaluate encoding quality based on possible bandwidth as well, but it is a two-sided -sword considering the trade-off described above. - -Another aspect is workload consumption. The higher the allowed bandwidth, the more frequency lines have to be -processed, which in turn increases the workload. - -\section FRAMESIZES_AND_BIT_RESERVOIR Frame Sizes & Bit Reservoir - -For AAC there is a difference between constant bit rate and constant frame -length due to the so-called bit reservoir technique, which allows the encoder to use less -bits in an AAC frame for those audio signal sections which are easy to encode, -and then spend them at a later point in -time for more complex audio sections. The extent to which this "bit exchange" -is done is limited to allow for reliable and relatively low delay real time -streaming. -Over a longer period in time the bitrate will be constant in the AAC constant -bitrate mode, e.g. for ISDN transmission. This means that in AAC each bitstream -frame will in general have a different length in bytes but over time it -will reach the target bitrate. One could also make an MPEG compliant -AAC encoder which always produces constant length packages for each AAC frame, -but the audio quality would be considerably worse since the bit reservoir -technique would have to be switched off completely. A higher bit rate would have -to be used to get the same audio quality as with an enabled bit reservoir. - -The maximum AAC frame length, regardless of the available bit reservoir, is defined -as 6144 bits per channel. - -For mp3 by the way, the same bit reservoir technique exists, but there each bit -stream frame has a constant length for a given bit rate (ignoring the -padding byte). In mp3 there is a so-called "back pointer" which tells -the decoder which bits belong to the current mp3 frame - and in general some or -many bits have been transmitted in an earlier mp3 frame. Basically this leads to -the same "bit exchange between mp3 frames" as in AAC but with virtually constant -length frames. - -This variable frame length at "constant bit rate" is not something special -in this Fraunhofer IIS AAC encoder. AAC has been designed in that way. - -\subsection BEHAVIOUR_ESTIM_AVG_FRAMESIZES Estimating Average Frame Sizes - -A HE-AAC v1 or v2 audio frame contains 2048 PCM samples per channel (there is -also one mode with 1920 samples per channel but this is only for special purposes -such as DAB+ digital radio). - -The number of HE-AAC frames \f$N\_FRAMES\f$ per second at 44.1 kHz is: - -\f[ -N\_FRAMES = 44100 / 2048 = 21.5332 -\f] - -At a bit rate of 8 kbps the average number of bits per frame \f$N\_BITS\_PER\_FRAME\f$ is: - -\f[ -N\_BITS\_PER\_FRAME = 8000 / 21.5332 = 371.52 -\f] - -which is about 46.44 bytes per encoded frame. - -At a bit rate of 32 kbps, which is quite high for single channel HE-AAC v1, it is: - -\f[ -N\_BITS\_PER\_FRAME = 32000 / 21.5332 = 1486 -\f] - -which is about 185.76 bytes per encoded frame. - -These bits/frame figures are average figures where each AAC frame generally has a different -size in bytes. To calculate the same for AAC-LC just use 1024 instead of 2048 PCM samples per -frame and channel. -For AAC-LD/ELD it is either 480 or 512 PCM samples per frame and channel. - - -\section BEHAVIOUR_TOOLS Encoder Tools - -The AAC encoder supports TNS, PNS, MS, Intensity and activates these tools depending on the audio signal and -the encoder configuration (i.e. bitrate or AOT). It is not required to configure these tools manually. - -PNS improves encoding quality only for certain bitrates. Therefore it makes sense to activate PNS only for -these bitrates and save the processing power required for PNS (about 10 % of the encoder) when using other -bitrates. This is done automatically inside the encoder library. PNS is disabled inside the encoder library if -an MPEG-2 AOT is choosen since PNS is an MPEG-4 AAC feature. - -If SBR is activated, the encoder automatically deactivates PNS internally. If TNS is disabled but PNS is allowed, -the encoder deactivates PNS calculation internally. - -*/ - -#ifndef _AAC_ENC_LIB_H_ -#define _AAC_ENC_LIB_H_ - -#include "machine_type.h" -#include "FDK_audio.h" - -#define AACENCODER_LIB_VL0 3 -#define AACENCODER_LIB_VL1 4 -#define AACENCODER_LIB_VL2 22 - -/** - * AAC encoder error codes. - */ -typedef enum { - AACENC_OK = 0x0000, /*!< No error happened. All fine. */ - - AACENC_INVALID_HANDLE = 0x0020, /*!< Handle passed to function call was invalid. */ - AACENC_MEMORY_ERROR = 0x0021, /*!< Memory allocation failed. */ - AACENC_UNSUPPORTED_PARAMETER = 0x0022, /*!< Parameter not available. */ - AACENC_INVALID_CONFIG = 0x0023, /*!< Configuration not provided. */ - - AACENC_INIT_ERROR = 0x0040, /*!< General initialization error. */ - AACENC_INIT_AAC_ERROR = 0x0041, /*!< AAC library initialization error. */ - AACENC_INIT_SBR_ERROR = 0x0042, /*!< SBR library initialization error. */ - AACENC_INIT_TP_ERROR = 0x0043, /*!< Transport library initialization error. */ - AACENC_INIT_META_ERROR = 0x0044, /*!< Meta data library initialization error. */ - - AACENC_ENCODE_ERROR = 0x0060, /*!< The encoding process was interrupted by an unexpected error. */ - - AACENC_ENCODE_EOF = 0x0080 /*!< End of file reached. */ - -} AACENC_ERROR; - - -/** - * AAC encoder buffer descriptors identifier. - * This identifier are used within buffer descriptors AACENC_BufDesc::bufferIdentifiers. - */ -typedef enum { - /* Input buffer identifier. */ - IN_AUDIO_DATA = 0, /*!< Audio input buffer, interleaved INT_PCM samples. */ - IN_ANCILLRY_DATA = 1, /*!< Ancillary data to be embedded into bitstream. */ - IN_METADATA_SETUP = 2, /*!< Setup structure for embedding meta data. */ - - /* Output buffer identifier. */ - OUT_BITSTREAM_DATA = 3, /*!< Buffer holds bitstream output data. */ - OUT_AU_SIZES = 4 /*!< Buffer contains sizes of each access unit. This information - is necessary for superframing. */ - -} AACENC_BufferIdentifier; - - -/** - * AAC encoder handle. - */ -typedef struct AACENCODER *HANDLE_AACENCODER; - - -/** - * Provides some info about the encoder configuration. - */ -typedef struct { - - UINT maxOutBufBytes; /*!< Maximum number of encoder bitstream bytes within one frame. - Size depends on maximum number of supported channels in encoder instance. - For superframing (as used for example in DAB+), size has to be a multiple accordingly. */ - - UINT maxAncBytes; /*!< Maximum number of ancillary data bytes which can be inserted into - bitstream within one frame. */ - - UINT inBufFillLevel; /*!< Internal input buffer fill level in samples per channel. This parameter - will automatically be cleared if samplingrate or channel(Mode/Order) changes. */ - - UINT inputChannels; /*!< Number of input channels expected in encoding process. */ - - UINT frameLength; /*!< Amount of input audio samples consumed each frame per channel, depending - on audio object type configuration. */ - - UINT encoderDelay; /*!< Codec delay in PCM samples/channel. Depends on framelength and AOT. Does not - include framing delay for filling up encoder PCM input buffer. */ - - UCHAR confBuf[64]; /*!< Configuration buffer in binary format as an AudioSpecificConfig - or StreamMuxConfig according to the selected transport type. */ - - UINT confSize; /*!< Number of valid bytes in confBuf. */ - -} AACENC_InfoStruct; - - -/** - * Describes the input and output buffers for an aacEncEncode() call. - */ -typedef struct { - INT numBufs; /*!< Number of buffers. */ - void **bufs; /*!< Pointer to vector containing buffer addresses. */ - INT *bufferIdentifiers; /*!< Identifier of each buffer element. See ::AACENC_BufferIdentifier. */ - INT *bufSizes; /*!< Size of each buffer in 8-bit bytes. */ - INT *bufElSizes; /*!< Size of each buffer element in bytes. */ - -} AACENC_BufDesc; - - -/** - * Defines the input arguments for an aacEncEncode() call. - */ -typedef struct { - INT numInSamples; /*!< Number of valid input audio samples (multiple of input channels). */ - INT numAncBytes; /*!< Number of ancillary data bytes to be encoded. */ - -} AACENC_InArgs; - - -/** - * Defines the output arguments for an aacEncEncode() call. - */ -typedef struct { - INT numOutBytes; /*!< Number of valid bitstream bytes generated during aacEncEncode(). */ - INT numInSamples; /*!< Number of input audio samples consumed by the encoder. */ - INT numAncBytes; /*!< Number of ancillary data bytes consumed by the encoder. */ - -} AACENC_OutArgs; - - -/** - * Meta Data Compression Profiles. - */ -typedef enum { - AACENC_METADATA_DRC_NONE = 0, /*!< None. */ - AACENC_METADATA_DRC_FILMSTANDARD = 1, /*!< Film standard. */ - AACENC_METADATA_DRC_FILMLIGHT = 2, /*!< Film light. */ - AACENC_METADATA_DRC_MUSICSTANDARD = 3, /*!< Music standard. */ - AACENC_METADATA_DRC_MUSICLIGHT = 4, /*!< Music light. */ - AACENC_METADATA_DRC_SPEECH = 5 /*!< Speech. */ - -} AACENC_METADATA_DRC_PROFILE; - - -/** - * Meta Data setup structure. - */ -typedef struct { - - AACENC_METADATA_DRC_PROFILE drc_profile; /*!< MPEG DRC compression profile. See ::AACENC_METADATA_DRC_PROFILE. */ - AACENC_METADATA_DRC_PROFILE comp_profile; /*!< ETSI heavy compression profile. See ::AACENC_METADATA_DRC_PROFILE. */ - - INT drc_TargetRefLevel; /*!< Used to define expected level to: - Scaled with 16 bit. x*2^16. */ - INT comp_TargetRefLevel; /*!< Adjust limiter to avoid overload. - Scaled with 16 bit. x*2^16. */ - - INT prog_ref_level_present; /*!< Flag, if prog_ref_level is present */ - INT prog_ref_level; /*!< Programme Reference Level = Dialogue Level: - -31.75dB .. 0 dB ; stepsize: 0.25dB - Scaled with 16 bit. x*2^16.*/ - - UCHAR PCE_mixdown_idx_present; /*!< Flag, if dmx-idx should be written in programme config element */ - UCHAR ETSI_DmxLvl_present; /*!< Flag, if dmx-lvl should be written in ETSI-ancData */ - - SCHAR centerMixLevel; /*!< Center downmix level (0...7, according to table) */ - SCHAR surroundMixLevel; /*!< Surround downmix level (0...7, according to table) */ - - UCHAR dolbySurroundMode; /*!< Indication for Dolby Surround Encoding Mode. - - 0: Dolby Surround mode not indicated - - 1: 2-ch audio part is not Dolby surround encoded - - 2: 2-ch audio part is Dolby surround encoded */ -} AACENC_MetaData; - - -/** - * AAC encoder control flags. - * - * In interaction with the ::AACENC_CONTROL_STATE parameter it is possible to get information about the internal - * initialization process. It is also possible to overwrite the internal state from extern when necessary. - */ -typedef enum -{ - AACENC_INIT_NONE = 0x0000, /*!< Do not trigger initialization. */ - AACENC_INIT_CONFIG = 0x0001, /*!< Initialize all encoder modules configuration. */ - AACENC_INIT_STATES = 0x0002, /*!< Reset all encoder modules history buffer. */ - AACENC_INIT_TRANSPORT = 0x1000, /*!< Initialize transport lib with new parameters. */ - AACENC_RESET_INBUFFER = 0x2000, /*!< Reset fill level of internal input buffer. */ - AACENC_INIT_ALL = 0xFFFF /*!< Initialize all. */ -} -AACENC_CTRLFLAGS; - - -/** - * \brief AAC encoder setting parameters. - * - * Use aacEncoder_SetParam() function to configure, or use aacEncoder_GetParam() function to read - * the internal status of the following parameters. - */ -typedef enum -{ - AACENC_AOT = 0x0100, /*!< Audio object type. See ::AUDIO_OBJECT_TYPE in FDK_audio.h. - - 2: MPEG-4 AAC Low Complexity. - - 5: MPEG-4 AAC Low Complexity with Spectral Band Replication (HE-AAC). - - 29: MPEG-4 AAC Low Complexity with Spectral Band Replication and Parametric Stereo (HE-AAC v2). - This configuration can be used only with stereo input audio data. - - 23: MPEG-4 AAC Low-Delay. - - 39: MPEG-4 AAC Enhanced Low-Delay. Since there is no ::AUDIO_OBJECT_TYPE for ELD in - combination with SBR defined, enable SBR explicitely by ::AACENC_SBR_MODE parameter. */ - - AACENC_BITRATE = 0x0101, /*!< Total encoder bitrate. This parameter is mandatory and interacts with ::AACENC_BITRATEMODE. - - CBR: Bitrate in bits/second. - See \ref suppBitrates for details. */ - - AACENC_BITRATEMODE = 0x0102, /*!< Bitrate mode. Configuration can be different kind of bitrate configurations: - - 0: Constant bitrate, use bitrate according to ::AACENC_BITRATE. (default) - Within none LD/ELD ::AUDIO_OBJECT_TYPE, the CBR mode makes use of full allowed bitreservoir. - In contrast, at Low-Delay ::AUDIO_OBJECT_TYPE the bitreservoir is kept very small. - - 8: LD/ELD full bitreservoir for packet based transmission. */ - - AACENC_SAMPLERATE = 0x0103, /*!< Audio input data sampling rate. Encoder supports following sampling rates: - 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, 64000, 88200, 96000 */ - - AACENC_SBR_MODE = 0x0104, /*!< Configure SBR independently of the chosen Audio Object Type ::AUDIO_OBJECT_TYPE. - This parameter is for ELD audio object type only. - - -1: Use ELD SBR auto configurator (default). - - 0: Disable Spectral Band Replication. - - 1: Enable Spectral Band Replication. */ - - AACENC_GRANULE_LENGTH = 0x0105, /*!< Core encoder (AAC) audio frame length in samples: - - 1024: Default configuration. - - 512: Default LD/ELD configuration. - - 480: Optional length in LD/ELD configuration. */ - - AACENC_CHANNELMODE = 0x0106, /*!< Set explicit channel mode. Channel mode must match with number of input channels. - - 1-7 and 33,34: MPEG channel modes supported, see ::CHANNEL_MODE in FDK_audio.h. */ - - AACENC_CHANNELORDER = 0x0107, /*!< Input audio data channel ordering scheme: - - 0: MPEG channel ordering (e. g. 5.1: C, L, R, SL, SR, LFE). (default) - - 1: WAVE file format channel ordering (e. g. 5.1: L, R, C, LFE, SL, SR). */ - - AACENC_SBR_RATIO = 0x0108, /*!< Controls activation of downsampled SBR. With downsampled SBR, the delay will be - shorter. On the other hand, for achieving the same quality level, downsampled SBR - needs more bits than dual-rate SBR. - With downsampled SBR, the AAC encoder will work at the same sampling rate as the - SBR encoder (single rate). - Downsampled SBR is supported for AAC-ELD and HE-AACv1. - - 1: Downsampled SBR (default for ELD). - - 2: Dual-rate SBR (default for HE-AAC). */ - - AACENC_AFTERBURNER = 0x0200, /*!< This parameter controls the use of the afterburner feature. - The afterburner is a type of analysis by synthesis algorithm which increases the - audio quality but also the required processing power. It is recommended to always - activate this if additional memory consumption and processing power consumption - is not a problem. If increased MHz and memory consumption are an issue then the MHz - and memory cost of this optional module need to be evaluated against the improvement - in audio quality on a case by case basis. - - 0: Disable afterburner (default). - - 1: Enable afterburner. */ - - AACENC_BANDWIDTH = 0x0203, /*!< Core encoder audio bandwidth: - - 0: Determine bandwidth internally (default, see chapter \ref BEHAVIOUR_BANDWIDTH). - - 1 to fs/2: Frequency bandwidth in Hertz. (Experts only, better do not - touch this value to avoid degraded audio quality) */ - - AACENC_PEAK_BITRATE = 0x0207, /*!< Peak bitrate configuration parameter to adjust maximum bits per audio frame. Bitrate is in bits/second. - The peak bitrate will internally be limited to the chosen bitrate ::AACENC_BITRATE as lower limit - and the number_of_effective_channels*6144 bit as upper limit. - - Setting the peak bitrate equal to ::AACENC_BITRATE does not necessarily mean that the audio frames - will be of constant size. Since the peak bitate is in bits/second, the frame sizes can vary by - one byte in one or the other direction over various frames. However, it is not recommended to reduce - the peak pitrate to ::AACENC_BITRATE - it would disable the bitreservoir, which would affect the - audio quality by a large amount. */ - - AACENC_TRANSMUX = 0x0300, /*!< Transport type to be used. See ::TRANSPORT_TYPE in FDK_audio.h. Following - types can be configured in encoder library: - - 0: raw access units - - 1: ADIF bitstream format - - 2: ADTS bitstream format - - 6: Audio Mux Elements (LATM) with muxConfigPresent = 1 - - 7: Audio Mux Elements (LATM) with muxConfigPresent = 0, out of band StreamMuxConfig - - 10: Audio Sync Stream (LOAS) */ - - AACENC_HEADER_PERIOD = 0x0301, /*!< Frame count period for sending in-band configuration buffers within LATM/LOAS - transport layer. Additionally this parameter configures the PCE repetition period - in raw_data_block(). See \ref encPCE. - - 0xFF: auto-mode default 10 for TT_MP4_ADTS, TT_MP4_LOAS and TT_MP4_LATM_MCP1, otherwise 0. - - n: Frame count period. */ - - AACENC_SIGNALING_MODE = 0x0302, /*!< Signaling mode of the extension AOT: - - 0: Implicit backward compatible signaling (default for non-MPEG-4 based - AOT's and for the transport formats ADIF and ADTS) - - A stream that uses implicit signaling can be decoded by every AAC decoder, even AAC-LC-only decoders - - An AAC-LC-only decoder will only decode the low-frequency part of the stream, resulting in a band-limited output - - This method works with all transport formats - - This method does not work with downsampled SBR - - 1: Explicit backward compatible signaling - - A stream that uses explicit backward compatible signaling can be decoded by every AAC decoder, even AAC-LC-only decoders - - An AAC-LC-only decoder will only decode the low-frequency part of the stream, resulting in a band-limited output - - A decoder not capable of decoding PS will only decode the AAC-LC+SBR part. - If the stream contained PS, the result will be a a decoded mono downmix - - This method does not work with ADIF or ADTS. For LOAS/LATM, it only works with AudioMuxVersion==1 - - This method does work with downsampled SBR - - 2: Explicit hierarchical signaling (default for MPEG-4 based AOT's and for all transport formats excluding ADIF and ADTS) - - A stream that uses explicit hierarchical signaling can be decoded only by HE-AAC decoders - - An AAC-LC-only decoder will not decode a stream that uses explicit hierarchical signaling - - A decoder not capable of decoding PS will not decode the stream at all if it contained PS - - This method does not work with ADIF or ADTS. It works with LOAS/LATM and the MPEG-4 File format - - This method does work with downsampled SBR - - For making sure that the listener always experiences the best audio quality, - explicit hierarchical signaling should be used. - This makes sure that only a full HE-AAC-capable decoder will decode those streams. - The audio is played at full bandwidth. - For best backwards compatibility, it is recommended to encode with implicit SBR signaling. - A decoder capable of AAC-LC only will then only decode the AAC part, which means the decoded - audio will sound band-limited. - - For MPEG-2 transport types (ADTS,ADIF), only implicit signaling is possible. - - For LOAS and LATM, explicit backwards compatible signaling only works together with AudioMuxVersion==1. - The reason is that, for explicit backwards compatible signaling, additional information will be appended to the ASC. - A decoder that is only capable of decoding AAC-LC will skip this part. - Nevertheless, for jumping to the end of the ASC, it needs to know the ASC length. - Transmitting the length of the ASC is a feature of AudioMuxVersion==1, it is not possible to transmit the - length of the ASC with AudioMuxVersion==0, therefore an AAC-LC-only decoder will not be able to parse a - LOAS/LATM stream that was being encoded with AudioMuxVersion==0. - - For downsampled SBR, explicit signaling is mandatory. The reason for this is that the - extension sampling frequency (which is in case of SBR the sampling frequqncy of the SBR part) - can only be signaled in explicit mode. - - For AAC-ELD, the SBR information is transmitted in the ELDSpecific Config, which is part of the - AudioSpecificConfig. Therefore, the settings here will have no effect on AAC-ELD.*/ - - AACENC_TPSUBFRAMES = 0x0303, /*!< Number of sub frames in a transport frame for LOAS/LATM or ADTS (default 1). - - ADTS: Maximum number of sub frames restricted to 4. - - LOAS/LATM: Maximum number of sub frames restricted to 2.*/ - - AACENC_AUDIOMUXVER = 0x0304, /*!< AudioMuxVersion to be used for LATM. (AudioMuxVersionA, currently not implemented): - - 0: Default, no transmission of tara Buffer fullness, no ASC length and including actual latm Buffer fullnes. - - 1: Transmission of tara Buffer fullness, ASC length and actual latm Buffer fullness. - - 2: Transmission of tara Buffer fullness, ASC length and maximum level of latm Buffer fullness. */ - - AACENC_PROTECTION = 0x0306, /*!< Configure protection in tranpsort layer: - - 0: No protection. (default) - - 1: CRC active for ADTS bitstream format. */ - - AACENC_ANCILLARY_BITRATE = 0x0500, /*!< Constant ancillary data bitrate in bits/second. - - 0: Either no ancillary data or insert exact number of bytes, denoted via - input parameter, numAncBytes in AACENC_InArgs. - - else: Insert ancillary data with specified bitrate. */ - - AACENC_METADATA_MODE = 0x0600, /*!< Configure Meta Data. See ::AACENC_MetaData for further details: - - 0: Do not embed any metadata. - - 1: Embed MPEG defined metadata only. - - 2: Embed all metadata. */ - - AACENC_CONTROL_STATE = 0xFF00, /*!< There is an automatic process which internally reconfigures the encoder instance - when a configuration parameter changed or an error occured. This paramerter allows - overwriting or getting the control status of this process. See ::AACENC_CTRLFLAGS. */ - - AACENC_NONE = 0xFFFF /*!< ------ */ - -} AACENC_PARAM; - - -#ifdef __cplusplus -extern "C" { -#endif - -/** - * \brief Open an instance of the encoder. - * - * Allocate memory for an encoder instance with a functional range denoted by the function parameters. - * Preinitialize encoder instance with default configuration. - * - * \param phAacEncoder A pointer to an encoder handle. Initialized on return. - * \param encModules Specify encoder modules to be supported in this encoder instance: - * - 0x0: Allocate memory for all available encoder modules. - * - else: Select memory allocation regarding encoder modules. Following flags are possible and can be combined. - * - 0x01: AAC module. - * - 0x02: SBR module. - * - 0x04: PS module. - * - 0x10: Metadata module. - * - example: (0x01|0x02|0x04|0x10) allocates all modules and is equivalent to default configuration denotet by 0x0. - * \param maxChannels Number of channels to be allocated. This parameter can be used in different ways: - * - 0: Allocate maximum number of AAC and SBR channels as supported by the library. - * - nChannels: Use same maximum number of channels for allocating memory in AAC and SBR module. - * - nChannels | (nSbrCh<<8): Number of SBR channels can be different to AAC channels to save data memory. - * - * \return - * - AACENC_OK, on succes. - * - AACENC_INVALID_HANDLE, AACENC_MEMORY_ERROR, AACENC_INVALID_CONFIG, on failure. - */ -AACENC_ERROR aacEncOpen( - HANDLE_AACENCODER *phAacEncoder, - const UINT encModules, - const UINT maxChannels - ); - - -/** - * \brief Close the encoder instance. - * - * Deallocate encoder instance and free whole memory. - * - * \param phAacEncoder Pointer to the encoder handle to be deallocated. - * - * \return - * - AACENC_OK, on success. - * - AACENC_INVALID_HANDLE, on failure. - */ -AACENC_ERROR aacEncClose( - HANDLE_AACENCODER *phAacEncoder - ); - - -/** - * \brief Encode audio data. - * - * This function is mainly for encoding audio data. In addition the function can be used for an encoder (re)configuration - * process. - * - PCM input data will be retrieved from external input buffer until the fill level allows encoding a single frame. - * This functionality allows an external buffer with reduced size in comparison to the AAC or HE-AAC audio frame length. - * - If the value of the input samples argument is zero, just internal reinitialization will be applied if it is - * requested. - * - At the end of a file the flushing process can be triggerd via setting the value of the input samples argument to -1. - * The encoder delay lines are fully flushed when the encoder returns no valid bitstream data AACENC_OutArgs::numOutBytes. - * Furthermore the end of file is signaled by the return value AACENC_ENCODE_EOF. - * - If an error occured in the previous frame or any of the encoder parameters changed, an internal reinitialization - * process will be applied before encoding the incoming audio samples. - * - The function can also be used for an independent reconfiguration process without encoding. The first parameter has to be a - * valid encoder handle and all other parameters can be set to NULL. - * - If the size of the external bitbuffer in outBufDesc is not sufficient for writing the whole bitstream, an internal - * error will be the return value and a reconfiguration will be triggered. - * - * \param hAacEncoder A valid AAC encoder handle. - * \param inBufDesc Input buffer descriptor, see AACENC_BufDesc: - * - At least one input buffer with audio data is expected. - * - Optionally a second input buffer with ancillary data can be fed. - * \param outBufDesc Output buffer descriptor, see AACENC_BufDesc: - * - Provide one output buffer for the encoded bitstream. - * \param inargs Input arguments, see AACENC_InArgs. - * \param outargs Output arguments, AACENC_OutArgs. - * - * \return - * - AACENC_OK, on success. - * - AACENC_INVALID_HANDLE, AACENC_ENCODE_ERROR, on failure in encoding process. - * - AACENC_INVALID_CONFIG, AACENC_INIT_ERROR, AACENC_INIT_AAC_ERROR, AACENC_INIT_SBR_ERROR, AACENC_INIT_TP_ERROR, - * AACENC_INIT_META_ERROR, on failure in encoder initialization. - * - AACENC_ENCODE_EOF, when flushing fully concluded. - */ -AACENC_ERROR aacEncEncode( - const HANDLE_AACENCODER hAacEncoder, - const AACENC_BufDesc *inBufDesc, - const AACENC_BufDesc *outBufDesc, - const AACENC_InArgs *inargs, - AACENC_OutArgs *outargs - ); - - -/** - * \brief Acquire info about present encoder instance. - * - * This function retrieves information of the encoder configuration. In addition to informative internal states, - * a configuration data block of the current encoder settings will be returned. The format is either Audio Specific Config - * in case of Raw Packets transport format or StreamMuxConfig in case of LOAS/LATM transport format. The configuration - * data block is binary coded as specified in ISO/IEC 14496-3 (MPEG-4 audio), to be used directly for MPEG-4 File Format - * or RFC3016 or RFC3640 applications. - * - * \param hAacEncoder A valid AAC encoder handle. - * \param pInfo Pointer to AACENC_InfoStruct. Filled on return. - * - * \return - * - AACENC_OK, on succes. - * - AACENC_INIT_ERROR, on failure. - */ -AACENC_ERROR aacEncInfo( - const HANDLE_AACENCODER hAacEncoder, - AACENC_InfoStruct *pInfo - ); - - -/** - * \brief Set one single AAC encoder parameter. - * - * This function allows configuration of all encoder parameters specified in ::AACENC_PARAM. Each parameter must be - * set with a separate function call. An internal validation of the configuration value range will be done and an - * internal reconfiguration will be signaled. The actual configuration adoption is part of the subsequent aacEncEncode() call. - * - * \param hAacEncoder A valid AAC encoder handle. - * \param param Parameter to be set. See ::AACENC_PARAM. - * \param value Parameter value. See parameter description in ::AACENC_PARAM. - * - * \return - * - AACENC_OK, on success. - * - AACENC_INVALID_HANDLE, AACENC_UNSUPPORTED_PARAMETER, AACENC_INVALID_CONFIG, on failure. - */ -AACENC_ERROR aacEncoder_SetParam( - const HANDLE_AACENCODER hAacEncoder, - const AACENC_PARAM param, - const UINT value - ); - - -/** - * \brief Get one single AAC encoder parameter. - * - * This function is the complement to aacEncoder_SetParam(). After encoder reinitialization with user defined settings, - * the internal status can be obtained of each parameter, specified with ::AACENC_PARAM. - * - * \param hAacEncoder A valid AAC encoder handle. - * \param param Parameter to be returned. See ::AACENC_PARAM. - * - * \return Internal configuration value of specifed parameter ::AACENC_PARAM. - */ -UINT aacEncoder_GetParam( - const HANDLE_AACENCODER hAacEncoder, - const AACENC_PARAM param - ); - - -/** - * \brief Get information about encoder library build. - * - * Fill a given LIB_INFO structure with library version information. - * - * \param info Pointer to an allocated LIB_INFO struct. - * - * \return - * - AACENC_OK, on success. - * - AACENC_INVALID_HANDLE, AACENC_INIT_ERROR, on failure. - */ -AACENC_ERROR aacEncGetLibInfo( - LIB_INFO *info - ); - - -#ifdef __cplusplus -} -#endif - -#endif /* _AAC_ENC_LIB_H_ */ diff --git a/libAACenc/src/aacEnc_ram.cpp b/libAACenc/src/aacEnc_ram.cpp deleted file mode 100644 index be3eea2..0000000 --- a/libAACenc/src/aacEnc_ram.cpp +++ /dev/null @@ -1,194 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/****************************************************************************** - - Initial authors: M. Lohwasser, M. Gayer - Contents/description: - -******************************************************************************/ -/*! - \file - \brief Memory layout - \author Markus Lohwasser -*/ - -#include "aacEnc_ram.h" - - C_AALLOC_MEM (AACdynamic_RAM, FIXP_DBL, AAC_ENC_DYN_RAM_SIZE/sizeof(FIXP_DBL)) - -/* - Static memory areas, must not be overwritten in other sections of the decoder ! -*/ - -/* - The structure AacEncoder contains all Encoder structures. -*/ - -C_ALLOC_MEM (Ram_aacEnc_AacEncoder, AAC_ENC, 1) - - -/* - The structure PSY_INTERNAl contains all psych configuration and data pointer. - * PsyStatic holds last and current Psych data. - * PsyInputBuffer contains time input. Signal is needed at the beginning of Psych. - Memory can be reused after signal is in time domain. - * PsyData contains spectral, nrg and threshold information. Necessary data are - copied into PsyOut, so memory is available after leaving psych. - * TnsData, ChaosMeasure, PnsData are temporarily necessary, e.g. use memory from - PsyInputBuffer. -*/ - -C_ALLOC_MEM2 (Ram_aacEnc_PsyElement, PSY_ELEMENT, 1, (8)) - -C_ALLOC_MEM (Ram_aacEnc_PsyInternal, PSY_INTERNAL, 1) -C_ALLOC_MEM2 (Ram_aacEnc_PsyStatic, PSY_STATIC, 1, (8)) - -C_ALLOC_MEM2 (Ram_aacEnc_PsyInputBuffer, INT_PCM, MAX_INPUT_BUFFER_SIZE, (8)) - - PSY_DYNAMIC *GetRam_aacEnc_PsyDynamic (int n, UCHAR* dynamic_RAM) { - FDK_ASSERT(dynamic_RAM!=0); - return ((PSY_DYNAMIC*) (dynamic_RAM + P_BUF_1 + n*sizeof(PSY_DYNAMIC))); - } - - C_ALLOC_MEM (Ram_bsOutbuffer, UCHAR, OUTPUTBUFFER_SIZE) - -/* - The structure PSY_OUT holds all psychoaccoustic data needed - in quantization module -*/ -C_ALLOC_MEM2 (Ram_aacEnc_PsyOut, PSY_OUT, 1, (1)) - -C_ALLOC_MEM2 (Ram_aacEnc_PsyOutElements, PSY_OUT_ELEMENT, 1, (1)*(8)) -C_ALLOC_MEM2 (Ram_aacEnc_PsyOutChannel, PSY_OUT_CHANNEL, 1, (1)*(8)) - - -/* - The structure QC_STATE contains preinitialized settings and quantizer structures. - * AdjustThreshold structure contains element-wise settings. - * ElementBits contains elemnt-wise bit consumption settings. - * When CRC is active, lookup table is necessary for fast crc calculation. - * Bitcounter contains buffer to find optimal codebooks and minimal bit consumption. - Values are temporarily, so dynamic memory can be used. -*/ - -C_ALLOC_MEM (Ram_aacEnc_QCstate, QC_STATE, 1) -C_ALLOC_MEM (Ram_aacEnc_AdjustThreshold, ADJ_THR_STATE, 1) - -C_ALLOC_MEM2 (Ram_aacEnc_AdjThrStateElement, ATS_ELEMENT, 1, (8)) -C_ALLOC_MEM2 (Ram_aacEnc_ElementBits, ELEMENT_BITS, 1, (8)) -C_ALLOC_MEM (Ram_aacEnc_BitCntrState, BITCNTR_STATE, 1) - - INT *GetRam_aacEnc_BitLookUp(int n, UCHAR* dynamic_RAM) { - FDK_ASSERT(dynamic_RAM!=0); - return ((INT*) (dynamic_RAM + P_BUF_1)); - } - INT *GetRam_aacEnc_MergeGainLookUp(int n, UCHAR* dynamic_RAM) { - FDK_ASSERT(dynamic_RAM!=0); - return ((INT*) (dynamic_RAM + P_BUF_1 + sizeof(INT)*(MAX_SFB_LONG*(CODE_BOOK_ESC_NDX+1)))); - } - - -/* - The structure QC_OUT contains settings and structures holding all necessary information - needed in bitstreamwriter. -*/ - -C_ALLOC_MEM2 (Ram_aacEnc_QCout, QC_OUT, 1, (1)) -C_ALLOC_MEM2 (Ram_aacEnc_QCelement, QC_OUT_ELEMENT, (1), (8)) - QC_OUT_CHANNEL *GetRam_aacEnc_QCchannel (int n, UCHAR* dynamic_RAM) { - FDK_ASSERT(dynamic_RAM!=0); - return ((QC_OUT_CHANNEL*) (dynamic_RAM + P_BUF_0 + n*sizeof(QC_OUT_CHANNEL))); - } - - - - - - - - - - - - diff --git a/libAACenc/src/aacEnc_ram.h b/libAACenc/src/aacEnc_ram.h deleted file mode 100644 index cf7da7c..0000000 --- a/libAACenc/src/aacEnc_ram.h +++ /dev/null @@ -1,226 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/****************************************************************************** - - Initial authors: M. Lohwasser, M. Gayer - Contents/description: - -******************************************************************************/ - -/*! - \file - \brief Memory layout - \author Markus Lohwasser -*/ - -#ifndef AAC_ENC_RAM_H -#define AAC_ENC_RAM_H - -#include "common_fix.h" - -#include "aacenc.h" -#include "psy_data.h" -#include "interface.h" -#include "psy_main.h" -#include "bitenc.h" -#include "bit_cnt.h" -#include "psy_const.h" - - #define OUTPUTBUFFER_SIZE (8192) /*!< Output buffer size has to be at least 6144 bits per channel (768 bytes). FDK bitbuffer implementation expects buffer of size 2^n. */ - - -/* - Moved AAC_ENC struct definition from aac_enc.cpp into aacEnc_ram.h to get size and respective - static memory in aacEnc_ram.cpp. - aac_enc.h is the outward visible header file and putting the struct into would cause necessity - of additional visible header files outside library. -*/ - -/* define hBitstream size: max AAC framelength is 6144 bits/channel */ -/*#define BUFFER_BITSTR_SIZE ((6400*(8)/bbWordSize) +((bbWordSize - 1) / bbWordSize))*/ - -struct AAC_ENC { - - AACENC_CONFIG *config; - - INT ancillaryBitsPerFrame; /* ancillary bits per frame calculated from ancillary rate */ - - CHANNEL_MAPPING channelMapping; - - QC_STATE *qcKernel; - QC_OUT *qcOut[(1)]; - - PSY_OUT *psyOut[(1)]; - PSY_INTERNAL *psyKernel; - - /* lifetime vars */ - - CHANNEL_MODE encoderMode; - INT bandwidth90dB; - AACENC_BITRATE_MODE bitrateMode; - - INT dontWriteAdif; /* use: write ADIF header only before 1st frame */ - - FIXP_DBL *dynamic_RAM; - - - INT maxChannels; /* used while allocation */ - INT maxElements; - INT maxFrames; - - AUDIO_OBJECT_TYPE aot; /* AOT to be used while encoding. */ - -} ; - -#define maxSize(a,b) ( ((a)>(b)) ? (a) : (b) ) - -#define BIT_LOOK_UP_SIZE ( sizeof(INT)*(MAX_SFB_LONG*(CODE_BOOK_ESC_NDX+1)) ) -#define MERGE_GAIN_LOOK_UP_SIZE ( sizeof(INT)*MAX_SFB_LONG ) - - - -/* Dynamic RAM - Allocation */ -/* - ++++++++++++++++++++++++++++++++++++++++++++ - | P_BUF_0 | P_BUF_1 | - ++++++++++++++++++++++++++++++++++++++++++++ - | QC_OUT_CH | PSY_DYN | - ++++++++++++++++++++++++++++++++++++++++++++ - | | BitLookUp+MergeGainLookUp | - ++++++++++++++++++++++++++++++++++++++++++++ - | | Bitstream output buffer | - ++++++++++++++++++++++++++++++++++++++++++++ -*/ - -#define BUF_SIZE_0 ( ALIGN_SIZE(sizeof(QC_OUT_CHANNEL)*(8)) ) -#define BUF_SIZE_1 ( ALIGN_SIZE(maxSize(sizeof(PSY_DYNAMIC), \ - (BIT_LOOK_UP_SIZE+MERGE_GAIN_LOOK_UP_SIZE))) ) - -#define P_BUF_0 ( 0 ) -#define P_BUF_1 ( P_BUF_0 + BUF_SIZE_0 ) - -#define AAC_ENC_DYN_RAM_SIZE ( BUF_SIZE_0 + BUF_SIZE_1 ) - - - H_ALLOC_MEM (AACdynamic_RAM, FIXP_DBL) -/* - ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ -END - Dynamic RAM - Allocation */ - -/* - See further Memory Allocation details in aacEnc_ram.cpp -*/ - H_ALLOC_MEM (Ram_aacEnc_AacEncoder, AAC_ENC) - - H_ALLOC_MEM (Ram_aacEnc_PsyElement, PSY_ELEMENT) - - H_ALLOC_MEM (Ram_aacEnc_PsyInternal, PSY_INTERNAL) - H_ALLOC_MEM (Ram_aacEnc_PsyStatic, PSY_STATIC) - H_ALLOC_MEM (Ram_aacEnc_PsyInputBuffer, INT_PCM) - - PSY_DYNAMIC *GetRam_aacEnc_PsyDynamic (int n, UCHAR* dynamic_RAM); - H_ALLOC_MEM (Ram_bsOutbuffer, UCHAR) - - H_ALLOC_MEM (Ram_aacEnc_PsyOutChannel, PSY_OUT_CHANNEL) - - H_ALLOC_MEM (Ram_aacEnc_PsyOut, PSY_OUT) - H_ALLOC_MEM (Ram_aacEnc_PsyOutElements, PSY_OUT_ELEMENT) - - H_ALLOC_MEM (Ram_aacEnc_QCstate, QC_STATE) - H_ALLOC_MEM (Ram_aacEnc_AdjustThreshold, ADJ_THR_STATE) - - H_ALLOC_MEM (Ram_aacEnc_AdjThrStateElement, ATS_ELEMENT) - H_ALLOC_MEM (Ram_aacEnc_ElementBits, ELEMENT_BITS) - H_ALLOC_MEM (Ram_aacEnc_BitCntrState, BITCNTR_STATE) - - INT *GetRam_aacEnc_BitLookUp(int n, UCHAR* dynamic_RAM); - INT *GetRam_aacEnc_MergeGainLookUp(int n, UCHAR* dynamic_RAM); - QC_OUT_CHANNEL *GetRam_aacEnc_QCchannel (int n, UCHAR* dynamic_RAM); - - H_ALLOC_MEM (Ram_aacEnc_QCout, QC_OUT) - H_ALLOC_MEM (Ram_aacEnc_QCelement, QC_OUT_ELEMENT) - - -#endif /* #ifndef AAC_ENC_RAM_H */ - diff --git a/libAACenc/src/aacEnc_rom.cpp b/libAACenc/src/aacEnc_rom.cpp deleted file mode 100644 index c6477e3..0000000 --- a/libAACenc/src/aacEnc_rom.cpp +++ /dev/null @@ -1,1232 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/****************************************************************************** - - Initial authors: M. Lohwasser, M. Gayer - Contents/description: - -******************************************************************************/ - -#include "aacEnc_rom.h" - -/* - Huffman Tables -*/ -const ULONG FDKaacEnc_huff_ltab1_2[3][3][3][3]= -{ - { - { {0x000b0009,0x00090007,0x000b0009}, {0x000a0008,0x00070006,0x000a0008}, {0x000b0009,0x00090008,0x000b0009} }, - { {0x000a0008,0x00070006,0x000a0007}, {0x00070006,0x00050005,0x00070006}, {0x00090007,0x00070006,0x000a0008} }, - { {0x000b0009,0x00090007,0x000b0008}, {0x00090008,0x00070006,0x00090008}, {0x000b0009,0x00090007,0x000b0009} } - }, - { - { {0x00090008,0x00070006,0x00090007}, {0x00070006,0x00050005,0x00070006}, {0x00090007,0x00070006,0x00090008} }, - { {0x00070006,0x00050005,0x00070006}, {0x00050005,0x00010003,0x00050005}, {0x00070006,0x00050005,0x00070006} }, - { {0x00090008,0x00070006,0x00090007}, {0x00070006,0x00050005,0x00070006}, {0x00090008,0x00070006,0x00090008} } - }, - { - { {0x000b0009,0x00090007,0x000b0009}, {0x00090008,0x00070006,0x00090008}, {0x000b0008,0x00090007,0x000b0009} }, - { {0x000a0008,0x00070006,0x00090007}, {0x00070006,0x00050004,0x00070006}, {0x00090008,0x00070006,0x000a0007} }, - { {0x000b0009,0x00090007,0x000b0009}, {0x000a0007,0x00070006,0x00090008}, {0x000b0009,0x00090007,0x000b0009} } - } -}; - - -const ULONG FDKaacEnc_huff_ltab3_4[3][3][3][3]= -{ - { - { {0x00010004,0x00040005,0x00080008}, {0x00040005,0x00050004,0x00080008}, {0x00090009,0x00090008,0x000a000b} }, - { {0x00040005,0x00060005,0x00090008}, {0x00060005,0x00060004,0x00090008}, {0x00090008,0x00090007,0x000a000a} }, - { {0x00090009,0x000a0008,0x000d000b}, {0x00090008,0x00090008,0x000b000a}, {0x000b000b,0x000a000a,0x000c000b} } - }, - { - { {0x00040004,0x00060005,0x000a0008}, {0x00060004,0x00070004,0x000a0008}, {0x000a0008,0x000a0008,0x000c000a} }, - { {0x00050004,0x00070004,0x000b0008}, {0x00060004,0x00070004,0x000a0007}, {0x00090008,0x00090007,0x000b0009} }, - { {0x00090008,0x000a0008,0x000d000a}, {0x00080007,0x00090007,0x000c0009}, {0x000a000a,0x000b0009,0x000c000a} } - }, - { - { {0x00080008,0x000a0008,0x000f000b}, {0x00090008,0x000b0007,0x000f000a}, {0x000d000b,0x000e000a,0x0010000c} }, - { {0x00080008,0x000a0007,0x000e000a}, {0x00090007,0x000a0007,0x000e0009}, {0x000c000a,0x000c0009,0x000f000b} }, - { {0x000b000b,0x000c000a,0x0010000c}, {0x000a000a,0x000b0009,0x000f000b}, {0x000c000b,0x000c000a,0x000f000b} } - } -}; - -const ULONG FDKaacEnc_huff_ltab5_6[9][9]= -{ - {0x000d000b, 0x000c000a, 0x000b0009, 0x000b0009, 0x000a0009, 0x000b0009, 0x000b0009, 0x000c000a, 0x000d000b}, - {0x000c000a, 0x000b0009, 0x000a0008, 0x00090007, 0x00080007, 0x00090007, 0x000a0008, 0x000b0009, 0x000c000a}, - {0x000c0009, 0x000a0008, 0x00090006, 0x00080006, 0x00070006, 0x00080006, 0x00090006, 0x000a0008, 0x000b0009}, - {0x000b0009, 0x00090007, 0x00080006, 0x00050004, 0x00040004, 0x00050004, 0x00080006, 0x00090007, 0x000b0009}, - {0x000a0009, 0x00080007, 0x00070006, 0x00040004, 0x00010004, 0x00040004, 0x00070006, 0x00080007, 0x000b0009}, - {0x000b0009, 0x00090007, 0x00080006, 0x00050004, 0x00040004, 0x00050004, 0x00080006, 0x00090007, 0x000b0009}, - {0x000b0009, 0x000a0008, 0x00090006, 0x00080006, 0x00070006, 0x00080006, 0x00090006, 0x000a0008, 0x000b0009}, - {0x000c000a, 0x000b0009, 0x000a0008, 0x00090007, 0x00080007, 0x00090007, 0x000a0007, 0x000b0008, 0x000c000a}, - {0x000d000b, 0x000c000a, 0x000c0009, 0x000b0009, 0x000a0009, 0x000a0009, 0x000b0009, 0x000c000a, 0x000d000b} -}; - -const ULONG FDKaacEnc_huff_ltab7_8[8][8]= -{ - {0x00010005, 0x00030004, 0x00060005, 0x00070006, 0x00080007, 0x00090008, 0x000a0009, 0x000b000a}, - {0x00030004, 0x00040003, 0x00060004, 0x00070005, 0x00080006, 0x00080007, 0x00090007, 0x00090008}, - {0x00060005, 0x00060004, 0x00070004, 0x00080005, 0x00080006, 0x00090007, 0x00090007, 0x000a0008}, - {0x00070006, 0x00070005, 0x00080005, 0x00080006, 0x00090006, 0x00090007, 0x000a0008, 0x000a0008}, - {0x00080007, 0x00080006, 0x00090006, 0x00090006, 0x000a0007, 0x000a0007, 0x000a0008, 0x000b0009}, - {0x00090008, 0x00080007, 0x00090006, 0x00090007, 0x000a0007, 0x000a0008, 0x000b0008, 0x000b000a}, - {0x000a0009, 0x00090007, 0x00090007, 0x000a0008, 0x000a0008, 0x000b0008, 0x000c0009, 0x000c0009}, - {0x000b000a, 0x000a0008, 0x000a0008, 0x000a0008, 0x000b0009, 0x000b0009, 0x000c0009, 0x000c000a} -}; - -const ULONG FDKaacEnc_huff_ltab9_10[13][13]= -{ - {0x00010006, 0x00030005, 0x00060006, 0x00080006, 0x00090007, 0x000a0008, 0x000a0009, 0x000b000a, 0x000b000a, 0x000c000a, 0x000c000b, 0x000d000b, 0x000d000c}, - {0x00030005, 0x00040004, 0x00060004, 0x00070005, 0x00080006, 0x00080007, 0x00090007, 0x000a0008, 0x000a0008, 0x000a0009, 0x000b000a, 0x000c000a, 0x000c000b}, - {0x00060006, 0x00060004, 0x00070005, 0x00080005, 0x00080006, 0x00090006, 0x000a0007, 0x000a0008, 0x000a0008, 0x000b0009, 0x000c0009, 0x000c000a, 0x000c000a}, - {0x00080006, 0x00070005, 0x00080005, 0x00090005, 0x00090006, 0x000a0007, 0x000a0007, 0x000b0008, 0x000b0008, 0x000b0009, 0x000c0009, 0x000c000a, 0x000d000a}, - {0x00090007, 0x00080006, 0x00090006, 0x00090006, 0x000a0006, 0x000a0007, 0x000b0007, 0x000b0008, 0x000b0008, 0x000c0009, 0x000c0009, 0x000c000a, 0x000d000a}, - {0x000a0008, 0x00090007, 0x00090006, 0x000a0007, 0x000b0007, 0x000b0007, 0x000b0008, 0x000c0008, 0x000b0008, 0x000c0009, 0x000c000a, 0x000d000a, 0x000d000b}, - {0x000b0009, 0x00090007, 0x000a0007, 0x000b0007, 0x000b0007, 0x000b0008, 0x000c0008, 0x000c0009, 0x000c0009, 0x000c0009, 0x000d000a, 0x000d000a, 0x000d000b}, - {0x000b0009, 0x000a0008, 0x000a0008, 0x000b0008, 0x000b0008, 0x000c0008, 0x000c0009, 0x000d0009, 0x000d0009, 0x000d000a, 0x000d000a, 0x000d000b, 0x000d000b}, - {0x000b0009, 0x000a0008, 0x000a0008, 0x000b0008, 0x000b0008, 0x000b0008, 0x000c0009, 0x000c0009, 0x000d000a, 0x000d000a, 0x000e000a, 0x000d000b, 0x000e000b}, - {0x000b000a, 0x000a0009, 0x000b0009, 0x000b0009, 0x000c0009, 0x000c0009, 0x000c0009, 0x000c000a, 0x000d000a, 0x000d000a, 0x000e000b, 0x000e000b, 0x000e000c}, - {0x000c000a, 0x000b0009, 0x000b0009, 0x000c0009, 0x000c0009, 0x000c000a, 0x000d000a, 0x000d000a, 0x000d000a, 0x000e000b, 0x000e000b, 0x000e000b, 0x000f000c}, - {0x000c000b, 0x000b000a, 0x000c0009, 0x000c000a, 0x000c000a, 0x000d000a, 0x000d000a, 0x000d000a, 0x000d000b, 0x000e000b, 0x000e000b, 0x000f000b, 0x000f000c}, - {0x000d000b, 0x000c000a, 0x000c000a, 0x000c000a, 0x000d000a, 0x000d000a, 0x000d000a, 0x000d000b, 0x000e000b, 0x000e000c, 0x000e000c, 0x000e000c, 0x000f000c} -}; - -const UCHAR FDKaacEnc_huff_ltab11[17][17]= -{ - {0x04, 0x05, 0x06, 0x07, 0x08, 0x08, 0x09, 0x0a, 0x0a, 0x0a, 0x0b, 0x0b, 0x0c, 0x0b, 0x0c, 0x0c, 0x0a}, - {0x05, 0x04, 0x05, 0x06, 0x07, 0x07, 0x08, 0x08, 0x09, 0x09, 0x09, 0x0a, 0x0a, 0x0a, 0x0a, 0x0b, 0x08}, - {0x06, 0x05, 0x05, 0x06, 0x07, 0x07, 0x08, 0x08, 0x08, 0x09, 0x09, 0x09, 0x0a, 0x0a, 0x0a, 0x0a, 0x08}, - {0x07, 0x06, 0x06, 0x06, 0x07, 0x07, 0x08, 0x08, 0x08, 0x09, 0x09, 0x09, 0x0a, 0x0a, 0x0a, 0x0a, 0x08}, - {0x08, 0x07, 0x07, 0x07, 0x07, 0x08, 0x08, 0x08, 0x08, 0x09, 0x09, 0x09, 0x0a, 0x0a, 0x0a, 0x0a, 0x08}, - {0x08, 0x07, 0x07, 0x07, 0x07, 0x08, 0x08, 0x08, 0x09, 0x09, 0x09, 0x09, 0x0a, 0x0a, 0x0a, 0x0a, 0x08}, - {0x09, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x09, 0x09, 0x09, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x08}, - {0x09, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x09, 0x09, 0x09, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x08}, - {0x0a, 0x09, 0x08, 0x08, 0x09, 0x09, 0x09, 0x09, 0x09, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0b, 0x08}, - {0x0a, 0x09, 0x09, 0x09, 0x09, 0x09, 0x09, 0x09, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0b, 0x0b, 0x08}, - {0x0b, 0x09, 0x09, 0x09, 0x09, 0x09, 0x09, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0b, 0x0a, 0x0b, 0x0b, 0x08}, - {0x0b, 0x0a, 0x09, 0x09, 0x0a, 0x09, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x08}, - {0x0b, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x09}, - {0x0b, 0x0a, 0x09, 0x09, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x09}, - {0x0b, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x09}, - {0x0c, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x0c, 0x0c, 0x09}, - {0x09, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x09, 0x05} -}; - -const UCHAR FDKaacEnc_huff_ltabscf[121]= -{ - 0x12, 0x12, 0x12, 0x12, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, - 0x13, 0x13, 0x13, 0x12, 0x13, 0x12, 0x11, 0x11, 0x10, 0x11, 0x10, 0x10, 0x10, 0x10, 0x0f, 0x0f, - 0x0e, 0x0e, 0x0e, 0x0e, 0x0e, 0x0e, 0x0d, 0x0d, 0x0c, 0x0c, 0x0c, 0x0b, 0x0c, 0x0b, 0x0a, 0x0a, - 0x0a, 0x09, 0x09, 0x08, 0x08, 0x08, 0x07, 0x06, 0x06, 0x05, 0x04, 0x03, 0x01, 0x04, 0x04, 0x05, - 0x06, 0x06, 0x07, 0x07, 0x08, 0x08, 0x09, 0x09, 0x0a, 0x0a, 0x0a, 0x0b, 0x0b, 0x0b, 0x0b, 0x0c, - 0x0c, 0x0d, 0x0d, 0x0d, 0x0e, 0x0e, 0x10, 0x0f, 0x10, 0x0f, 0x12, 0x13, 0x13, 0x13, 0x13, 0x13, - 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, - 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13 -}; - - -const USHORT FDKaacEnc_huff_ctab1[3][3][3][3]= -{ - { - { {0x07f8,0x01f1,0x07fd}, {0x03f5,0x0068,0x03f0}, {0x07f7,0x01ec,0x07f5} }, - { {0x03f1,0x0072,0x03f4}, {0x0074,0x0011,0x0076}, {0x01eb,0x006c,0x03f6} }, - { {0x07fc,0x01e1,0x07f1}, {0x01f0,0x0061,0x01f6}, {0x07f2,0x01ea,0x07fb} } - }, - { - { {0x01f2,0x0069,0x01ed}, {0x0077,0x0017,0x006f}, {0x01e6,0x0064,0x01e5} }, - { {0x0067,0x0015,0x0062}, {0x0012,0x0000,0x0014}, {0x0065,0x0016,0x006d} }, - { {0x01e9,0x0063,0x01e4}, {0x006b,0x0013,0x0071}, {0x01e3,0x0070,0x01f3} } - }, - { - { {0x07fe,0x01e7,0x07f3}, {0x01ef,0x0060,0x01ee}, {0x07f0,0x01e2,0x07fa} }, - { {0x03f3,0x006a,0x01e8}, {0x0075,0x0010,0x0073}, {0x01f4,0x006e,0x03f7} }, - { {0x07f6,0x01e0,0x07f9}, {0x03f2,0x0066,0x01f5}, {0x07ff,0x01f7,0x07f4} } - } -}; - -const USHORT FDKaacEnc_huff_ctab2[3][3][3][3]= -{ - { - { {0x01f3,0x006f,0x01fd}, {0x00eb,0x0023,0x00ea}, {0x01f7,0x00e8,0x01fa} }, - { {0x00f2,0x002d,0x0070}, {0x0020,0x0006,0x002b}, {0x006e,0x0028,0x00e9} }, - { {0x01f9,0x0066,0x00f8}, {0x00e7,0x001b,0x00f1}, {0x01f4,0x006b,0x01f5} } - }, - { - { {0x00ec,0x002a,0x006c}, {0x002c,0x000a,0x0027}, {0x0067,0x001a,0x00f5} }, - { {0x0024,0x0008,0x001f}, {0x0009,0x0000,0x0007}, {0x001d,0x000b,0x0030} }, - { {0x00ef,0x001c,0x0064}, {0x001e,0x000c,0x0029}, {0x00f3,0x002f,0x00f0} } - }, - { - { {0x01fc,0x0071,0x01f2}, {0x00f4,0x0021,0x00e6}, {0x00f7,0x0068,0x01f8} }, - { {0x00ee,0x0022,0x0065}, {0x0031,0x0002,0x0026}, {0x00ed,0x0025,0x006a} }, - { {0x01fb,0x0072,0x01fe}, {0x0069,0x002e,0x00f6}, {0x01ff,0x006d,0x01f6} } - } -}; - -const USHORT FDKaacEnc_huff_ctab3[3][3][3][3]= -{ - { - { {0x0000,0x0009,0x00ef}, {0x000b,0x0019,0x00f0}, {0x01eb,0x01e6,0x03f2} }, - { {0x000a,0x0035,0x01ef}, {0x0034,0x0037,0x01e9}, {0x01ed,0x01e7,0x03f3} }, - { {0x01ee,0x03ed,0x1ffa}, {0x01ec,0x01f2,0x07f9}, {0x07f8,0x03f8,0x0ff8} } - }, - { - { {0x0008,0x0038,0x03f6}, {0x0036,0x0075,0x03f1}, {0x03eb,0x03ec,0x0ff4} }, - { {0x0018,0x0076,0x07f4}, {0x0039,0x0074,0x03ef}, {0x01f3,0x01f4,0x07f6} }, - { {0x01e8,0x03ea,0x1ffc}, {0x00f2,0x01f1,0x0ffb}, {0x03f5,0x07f3,0x0ffc} } - }, - { - { {0x00ee,0x03f7,0x7ffe}, {0x01f0,0x07f5,0x7ffd}, {0x1ffb,0x3ffa,0xffff} }, - { {0x00f1,0x03f0,0x3ffc}, {0x01ea,0x03ee,0x3ffb}, {0x0ff6,0x0ffa,0x7ffc} }, - { {0x07f2,0x0ff5,0xfffe}, {0x03f4,0x07f7,0x7ffb}, {0x0ff7,0x0ff9,0x7ffa} } - } -}; - -const USHORT FDKaacEnc_huff_ctab4[3][3][3][3]= -{ - { - { {0x0007,0x0016,0x00f6}, {0x0018,0x0008,0x00ef}, {0x01ef,0x00f3,0x07f8} }, - { {0x0019,0x0017,0x00ed}, {0x0015,0x0001,0x00e2}, {0x00f0,0x0070,0x03f0} }, - { {0x01ee,0x00f1,0x07fa}, {0x00ee,0x00e4,0x03f2}, {0x07f6,0x03ef,0x07fd} } - }, - { - { {0x0005,0x0014,0x00f2}, {0x0009,0x0004,0x00e5}, {0x00f4,0x00e8,0x03f4} }, - { {0x0006,0x0002,0x00e7}, {0x0003,0x0000,0x006b}, {0x00e3,0x0069,0x01f3} }, - { {0x00eb,0x00e6,0x03f6}, {0x006e,0x006a,0x01f4}, {0x03ec,0x01f0,0x03f9} } - }, - { - { {0x00f5,0x00ec,0x07fb}, {0x00ea,0x006f,0x03f7}, {0x07f9,0x03f3,0x0fff} }, - { {0x00e9,0x006d,0x03f8}, {0x006c,0x0068,0x01f5}, {0x03ee,0x01f2,0x07f4} }, - { {0x07f7,0x03f1,0x0ffe}, {0x03ed,0x01f1,0x07f5}, {0x07fe,0x03f5,0x07fc} } - } -}; -const USHORT FDKaacEnc_huff_ctab5[9][9]= -{ - {0x1fff, 0x0ff7, 0x07f4, 0x07e8, 0x03f1, 0x07ee, 0x07f9, 0x0ff8, 0x1ffd}, - {0x0ffd, 0x07f1, 0x03e8, 0x01e8, 0x00f0, 0x01ec, 0x03ee, 0x07f2, 0x0ffa}, - {0x0ff4, 0x03ef, 0x01f2, 0x00e8, 0x0070, 0x00ec, 0x01f0, 0x03ea, 0x07f3}, - {0x07eb, 0x01eb, 0x00ea, 0x001a, 0x0008, 0x0019, 0x00ee, 0x01ef, 0x07ed}, - {0x03f0, 0x00f2, 0x0073, 0x000b, 0x0000, 0x000a, 0x0071, 0x00f3, 0x07e9}, - {0x07ef, 0x01ee, 0x00ef, 0x0018, 0x0009, 0x001b, 0x00eb, 0x01e9, 0x07ec}, - {0x07f6, 0x03eb, 0x01f3, 0x00ed, 0x0072, 0x00e9, 0x01f1, 0x03ed, 0x07f7}, - {0x0ff6, 0x07f0, 0x03e9, 0x01ed, 0x00f1, 0x01ea, 0x03ec, 0x07f8, 0x0ff9}, - {0x1ffc, 0x0ffc, 0x0ff5, 0x07ea, 0x03f3, 0x03f2, 0x07f5, 0x0ffb, 0x1ffe} -}; - -const USHORT FDKaacEnc_huff_ctab6[9][9]= -{ - {0x07fe, 0x03fd, 0x01f1, 0x01eb, 0x01f4, 0x01ea, 0x01f0, 0x03fc, 0x07fd}, - {0x03f6, 0x01e5, 0x00ea, 0x006c, 0x0071, 0x0068, 0x00f0, 0x01e6, 0x03f7}, - {0x01f3, 0x00ef, 0x0032, 0x0027, 0x0028, 0x0026, 0x0031, 0x00eb, 0x01f7}, - {0x01e8, 0x006f, 0x002e, 0x0008, 0x0004, 0x0006, 0x0029, 0x006b, 0x01ee}, - {0x01ef, 0x0072, 0x002d, 0x0002, 0x0000, 0x0003, 0x002f, 0x0073, 0x01fa}, - {0x01e7, 0x006e, 0x002b, 0x0007, 0x0001, 0x0005, 0x002c, 0x006d, 0x01ec}, - {0x01f9, 0x00ee, 0x0030, 0x0024, 0x002a, 0x0025, 0x0033, 0x00ec, 0x01f2}, - {0x03f8, 0x01e4, 0x00ed, 0x006a, 0x0070, 0x0069, 0x0074, 0x00f1, 0x03fa}, - {0x07ff, 0x03f9, 0x01f6, 0x01ed, 0x01f8, 0x01e9, 0x01f5, 0x03fb, 0x07fc} -}; - -const USHORT FDKaacEnc_huff_ctab7[8][8]= -{ - {0x0000, 0x0005, 0x0037, 0x0074, 0x00f2, 0x01eb, 0x03ed, 0x07f7}, - {0x0004, 0x000c, 0x0035, 0x0071, 0x00ec, 0x00ee, 0x01ee, 0x01f5}, - {0x0036, 0x0034, 0x0072, 0x00ea, 0x00f1, 0x01e9, 0x01f3, 0x03f5}, - {0x0073, 0x0070, 0x00eb, 0x00f0, 0x01f1, 0x01f0, 0x03ec, 0x03fa}, - {0x00f3, 0x00ed, 0x01e8, 0x01ef, 0x03ef, 0x03f1, 0x03f9, 0x07fb}, - {0x01ed, 0x00ef, 0x01ea, 0x01f2, 0x03f3, 0x03f8, 0x07f9, 0x07fc}, - {0x03ee, 0x01ec, 0x01f4, 0x03f4, 0x03f7, 0x07f8, 0x0ffd, 0x0ffe}, - {0x07f6, 0x03f0, 0x03f2, 0x03f6, 0x07fa, 0x07fd, 0x0ffc, 0x0fff} -}; - -const USHORT FDKaacEnc_huff_ctab8[8][8]= -{ - {0x000e, 0x0005, 0x0010, 0x0030, 0x006f, 0x00f1, 0x01fa, 0x03fe}, - {0x0003, 0x0000, 0x0004, 0x0012, 0x002c, 0x006a, 0x0075, 0x00f8}, - {0x000f, 0x0002, 0x0006, 0x0014, 0x002e, 0x0069, 0x0072, 0x00f5}, - {0x002f, 0x0011, 0x0013, 0x002a, 0x0032, 0x006c, 0x00ec, 0x00fa}, - {0x0071, 0x002b, 0x002d, 0x0031, 0x006d, 0x0070, 0x00f2, 0x01f9}, - {0x00ef, 0x0068, 0x0033, 0x006b, 0x006e, 0x00ee, 0x00f9, 0x03fc}, - {0x01f8, 0x0074, 0x0073, 0x00ed, 0x00f0, 0x00f6, 0x01f6, 0x01fd}, - {0x03fd, 0x00f3, 0x00f4, 0x00f7, 0x01f7, 0x01fb, 0x01fc, 0x03ff} -}; - -const USHORT FDKaacEnc_huff_ctab9[13][13]= -{ - {0x0000, 0x0005, 0x0037, 0x00e7, 0x01de, 0x03ce, 0x03d9, 0x07c8, 0x07cd, 0x0fc8, 0x0fdd, 0x1fe4, 0x1fec}, - {0x0004, 0x000c, 0x0035, 0x0072, 0x00ea, 0x00ed, 0x01e2, 0x03d1, 0x03d3, 0x03e0, 0x07d8, 0x0fcf, 0x0fd5}, - {0x0036, 0x0034, 0x0071, 0x00e8, 0x00ec, 0x01e1, 0x03cf, 0x03dd, 0x03db, 0x07d0, 0x0fc7, 0x0fd4, 0x0fe4}, - {0x00e6, 0x0070, 0x00e9, 0x01dd, 0x01e3, 0x03d2, 0x03dc, 0x07cc, 0x07ca, 0x07de, 0x0fd8, 0x0fea, 0x1fdb}, - {0x01df, 0x00eb, 0x01dc, 0x01e6, 0x03d5, 0x03de, 0x07cb, 0x07dd, 0x07dc, 0x0fcd, 0x0fe2, 0x0fe7, 0x1fe1}, - {0x03d0, 0x01e0, 0x01e4, 0x03d6, 0x07c5, 0x07d1, 0x07db, 0x0fd2, 0x07e0, 0x0fd9, 0x0feb, 0x1fe3, 0x1fe9}, - {0x07c4, 0x01e5, 0x03d7, 0x07c6, 0x07cf, 0x07da, 0x0fcb, 0x0fda, 0x0fe3, 0x0fe9, 0x1fe6, 0x1ff3, 0x1ff7}, - {0x07d3, 0x03d8, 0x03e1, 0x07d4, 0x07d9, 0x0fd3, 0x0fde, 0x1fdd, 0x1fd9, 0x1fe2, 0x1fea, 0x1ff1, 0x1ff6}, - {0x07d2, 0x03d4, 0x03da, 0x07c7, 0x07d7, 0x07e2, 0x0fce, 0x0fdb, 0x1fd8, 0x1fee, 0x3ff0, 0x1ff4, 0x3ff2}, - {0x07e1, 0x03df, 0x07c9, 0x07d6, 0x0fca, 0x0fd0, 0x0fe5, 0x0fe6, 0x1feb, 0x1fef, 0x3ff3, 0x3ff4, 0x3ff5}, - {0x0fe0, 0x07ce, 0x07d5, 0x0fc6, 0x0fd1, 0x0fe1, 0x1fe0, 0x1fe8, 0x1ff0, 0x3ff1, 0x3ff8, 0x3ff6, 0x7ffc}, - {0x0fe8, 0x07df, 0x0fc9, 0x0fd7, 0x0fdc, 0x1fdc, 0x1fdf, 0x1fed, 0x1ff5, 0x3ff9, 0x3ffb, 0x7ffd, 0x7ffe}, - {0x1fe7, 0x0fcc, 0x0fd6, 0x0fdf, 0x1fde, 0x1fda, 0x1fe5, 0x1ff2, 0x3ffa, 0x3ff7, 0x3ffc, 0x3ffd, 0x7fff} -}; - -const USHORT FDKaacEnc_huff_ctab10[13][13]= -{ - {0x0022, 0x0008, 0x001d, 0x0026, 0x005f, 0x00d3, 0x01cf, 0x03d0, 0x03d7, 0x03ed, 0x07f0, 0x07f6, 0x0ffd}, - {0x0007, 0x0000, 0x0001, 0x0009, 0x0020, 0x0054, 0x0060, 0x00d5, 0x00dc, 0x01d4, 0x03cd, 0x03de, 0x07e7}, - {0x001c, 0x0002, 0x0006, 0x000c, 0x001e, 0x0028, 0x005b, 0x00cd, 0x00d9, 0x01ce, 0x01dc, 0x03d9, 0x03f1}, - {0x0025, 0x000b, 0x000a, 0x000d, 0x0024, 0x0057, 0x0061, 0x00cc, 0x00dd, 0x01cc, 0x01de, 0x03d3, 0x03e7}, - {0x005d, 0x0021, 0x001f, 0x0023, 0x0027, 0x0059, 0x0064, 0x00d8, 0x00df, 0x01d2, 0x01e2, 0x03dd, 0x03ee}, - {0x00d1, 0x0055, 0x0029, 0x0056, 0x0058, 0x0062, 0x00ce, 0x00e0, 0x00e2, 0x01da, 0x03d4, 0x03e3, 0x07eb}, - {0x01c9, 0x005e, 0x005a, 0x005c, 0x0063, 0x00ca, 0x00da, 0x01c7, 0x01ca, 0x01e0, 0x03db, 0x03e8, 0x07ec}, - {0x01e3, 0x00d2, 0x00cb, 0x00d0, 0x00d7, 0x00db, 0x01c6, 0x01d5, 0x01d8, 0x03ca, 0x03da, 0x07ea, 0x07f1}, - {0x01e1, 0x00d4, 0x00cf, 0x00d6, 0x00de, 0x00e1, 0x01d0, 0x01d6, 0x03d1, 0x03d5, 0x03f2, 0x07ee, 0x07fb}, - {0x03e9, 0x01cd, 0x01c8, 0x01cb, 0x01d1, 0x01d7, 0x01df, 0x03cf, 0x03e0, 0x03ef, 0x07e6, 0x07f8, 0x0ffa}, - {0x03eb, 0x01dd, 0x01d3, 0x01d9, 0x01db, 0x03d2, 0x03cc, 0x03dc, 0x03ea, 0x07ed, 0x07f3, 0x07f9, 0x0ff9}, - {0x07f2, 0x03ce, 0x01e4, 0x03cb, 0x03d8, 0x03d6, 0x03e2, 0x03e5, 0x07e8, 0x07f4, 0x07f5, 0x07f7, 0x0ffb}, - {0x07fa, 0x03ec, 0x03df, 0x03e1, 0x03e4, 0x03e6, 0x03f0, 0x07e9, 0x07ef, 0x0ff8, 0x0ffe, 0x0ffc, 0x0fff} -}; - -const USHORT FDKaacEnc_huff_ctab11[21][17]= -{ - {0x0000, 0x0006, 0x0019, 0x003d, 0x009c, 0x00c6, 0x01a7, 0x0390, 0x03c2, 0x03df, 0x07e6, 0x07f3, 0x0ffb, 0x07ec, 0x0ffa, 0x0ffe, 0x038e}, - {0x0005, 0x0001, 0x0008, 0x0014, 0x0037, 0x0042, 0x0092, 0x00af, 0x0191, 0x01a5, 0x01b5, 0x039e, 0x03c0, 0x03a2, 0x03cd, 0x07d6, 0x00ae}, - {0x0017, 0x0007, 0x0009, 0x0018, 0x0039, 0x0040, 0x008e, 0x00a3, 0x00b8, 0x0199, 0x01ac, 0x01c1, 0x03b1, 0x0396, 0x03be, 0x03ca, 0x009d}, - {0x003c, 0x0015, 0x0016, 0x001a, 0x003b, 0x0044, 0x0091, 0x00a5, 0x00be, 0x0196, 0x01ae, 0x01b9, 0x03a1, 0x0391, 0x03a5, 0x03d5, 0x0094}, - {0x009a, 0x0036, 0x0038, 0x003a, 0x0041, 0x008c, 0x009b, 0x00b0, 0x00c3, 0x019e, 0x01ab, 0x01bc, 0x039f, 0x038f, 0x03a9, 0x03cf, 0x0093}, - {0x00bf, 0x003e, 0x003f, 0x0043, 0x0045, 0x009e, 0x00a7, 0x00b9, 0x0194, 0x01a2, 0x01ba, 0x01c3, 0x03a6, 0x03a7, 0x03bb, 0x03d4, 0x009f}, - {0x01a0, 0x008f, 0x008d, 0x0090, 0x0098, 0x00a6, 0x00b6, 0x00c4, 0x019f, 0x01af, 0x01bf, 0x0399, 0x03bf, 0x03b4, 0x03c9, 0x03e7, 0x00a8}, - {0x01b6, 0x00ab, 0x00a4, 0x00aa, 0x00b2, 0x00c2, 0x00c5, 0x0198, 0x01a4, 0x01b8, 0x038c, 0x03a4, 0x03c4, 0x03c6, 0x03dd, 0x03e8, 0x00ad}, - {0x03af, 0x0192, 0x00bd, 0x00bc, 0x018e, 0x0197, 0x019a, 0x01a3, 0x01b1, 0x038d, 0x0398, 0x03b7, 0x03d3, 0x03d1, 0x03db, 0x07dd, 0x00b4}, - {0x03de, 0x01a9, 0x019b, 0x019c, 0x01a1, 0x01aa, 0x01ad, 0x01b3, 0x038b, 0x03b2, 0x03b8, 0x03ce, 0x03e1, 0x03e0, 0x07d2, 0x07e5, 0x00b7}, - {0x07e3, 0x01bb, 0x01a8, 0x01a6, 0x01b0, 0x01b2, 0x01b7, 0x039b, 0x039a, 0x03ba, 0x03b5, 0x03d6, 0x07d7, 0x03e4, 0x07d8, 0x07ea, 0x00ba}, - {0x07e8, 0x03a0, 0x01bd, 0x01b4, 0x038a, 0x01c4, 0x0392, 0x03aa, 0x03b0, 0x03bc, 0x03d7, 0x07d4, 0x07dc, 0x07db, 0x07d5, 0x07f0, 0x00c1}, - {0x07fb, 0x03c8, 0x03a3, 0x0395, 0x039d, 0x03ac, 0x03ae, 0x03c5, 0x03d8, 0x03e2, 0x03e6, 0x07e4, 0x07e7, 0x07e0, 0x07e9, 0x07f7, 0x0190}, - {0x07f2, 0x0393, 0x01be, 0x01c0, 0x0394, 0x0397, 0x03ad, 0x03c3, 0x03c1, 0x03d2, 0x07da, 0x07d9, 0x07df, 0x07eb, 0x07f4, 0x07fa, 0x0195}, - {0x07f8, 0x03bd, 0x039c, 0x03ab, 0x03a8, 0x03b3, 0x03b9, 0x03d0, 0x03e3, 0x03e5, 0x07e2, 0x07de, 0x07ed, 0x07f1, 0x07f9, 0x07fc, 0x0193}, - {0x0ffd, 0x03dc, 0x03b6, 0x03c7, 0x03cc, 0x03cb, 0x03d9, 0x03da, 0x07d3, 0x07e1, 0x07ee, 0x07ef, 0x07f5, 0x07f6, 0x0ffc, 0x0fff, 0x019d}, - {0x01c2, 0x00b5, 0x00a1, 0x0096, 0x0097, 0x0095, 0x0099, 0x00a0, 0x00a2, 0x00ac, 0x00a9, 0x00b1, 0x00b3, 0x00bb, 0x00c0, 0x018f, 0x0004}, - {0x0018, 0x002e, 0x0000, 0x005a, 0x00a5, 0x00f8, 0x00b7, 0x0094, 0x00f9, 0x004d, 0x0021, 0x002b, 0x004f, 0x007b, 0x00bc, 0x0046, 0x0015}, - {0x0042, 0x0037, 0x0078, 0x000d, 0x0068, 0x005f, 0x000d, 0x005e, 0x005a, 0x00be, 0x0063, 0x007e, 0x001f, 0x0092, 0x001a, 0x00ab, 0x0032}, - {0x00e6, 0x0037, 0x0000, 0x0058, 0x000b, 0x005a, 0x00e1, 0x005d, 0x0029, 0x0017, 0x007e, 0x0069, 0x00aa, 0x0054, 0x0029, 0x0032, 0x0041}, - {0x0046, 0x00ea, 0x0034, 0x00ea, 0x0011, 0x001b, 0x00a9, 0x0094, 0x00e2, 0x0031, 0x00d0, 0x00e5, 0x0007, 0x0070, 0x0069, 0x003e, 0x0021} -}; - -const ULONG FDKaacEnc_huff_ctabscf[121]= -{ - 0x0003ffe8, 0x0003ffe6, 0x0003ffe7, 0x0003ffe5, 0x0007fff5, 0x0007fff1, 0x0007ffed, 0x0007fff6, - 0x0007ffee, 0x0007ffef, 0x0007fff0, 0x0007fffc, 0x0007fffd, 0x0007ffff, 0x0007fffe, 0x0007fff7, - 0x0007fff8, 0x0007fffb, 0x0007fff9, 0x0003ffe4, 0x0007fffa, 0x0003ffe3, 0x0001ffef, 0x0001fff0, - 0x0000fff5, 0x0001ffee, 0x0000fff2, 0x0000fff3, 0x0000fff4, 0x0000fff1, 0x00007ff6, 0x00007ff7, - 0x00003ff9, 0x00003ff5, 0x00003ff7, 0x00003ff3, 0x00003ff6, 0x00003ff2, 0x00001ff7, 0x00001ff5, - 0x00000ff9, 0x00000ff7, 0x00000ff6, 0x000007f9, 0x00000ff4, 0x000007f8, 0x000003f9, 0x000003f7, - 0x000003f5, 0x000001f8, 0x000001f7, 0x000000fa, 0x000000f8, 0x000000f6, 0x00000079, 0x0000003a, - 0x00000038, 0x0000001a, 0x0000000b, 0x00000004, 0x00000000, 0x0000000a, 0x0000000c, 0x0000001b, - 0x00000039, 0x0000003b, 0x00000078, 0x0000007a, 0x000000f7, 0x000000f9, 0x000001f6, 0x000001f9, - 0x000003f4, 0x000003f6, 0x000003f8, 0x000007f5, 0x000007f4, 0x000007f6, 0x000007f7, 0x00000ff5, - 0x00000ff8, 0x00001ff4, 0x00001ff6, 0x00001ff8, 0x00003ff8, 0x00003ff4, 0x0000fff0, 0x00007ff4, - 0x0000fff6, 0x00007ff5, 0x0003ffe2, 0x0007ffd9, 0x0007ffda, 0x0007ffdb, 0x0007ffdc, 0x0007ffdd, - 0x0007ffde, 0x0007ffd8, 0x0007ffd2, 0x0007ffd3, 0x0007ffd4, 0x0007ffd5, 0x0007ffd6, 0x0007fff2, - 0x0007ffdf, 0x0007ffe7, 0x0007ffe8, 0x0007ffe9, 0x0007ffea, 0x0007ffeb, 0x0007ffe6, 0x0007ffe0, - 0x0007ffe1, 0x0007ffe2, 0x0007ffe3, 0x0007ffe4, 0x0007ffe5, 0x0007ffd7, 0x0007ffec, 0x0007fff4, - 0x0007fff3 -}; - -/* - table of (0.50000...1.00000) ^0.75 -*/ -const FIXP_QTD FDKaacEnc_mTab_3_4[MANT_SIZE] = -{ - QTC(0x4c1bf829), QTC(0x4c3880de), QTC(0x4c550603), QTC(0x4c71879c), QTC(0x4c8e05aa), QTC(0x4caa8030), QTC(0x4cc6f72f), QTC(0x4ce36aab), - QTC(0x4cffdaa4), QTC(0x4d1c471d), QTC(0x4d38b019), QTC(0x4d55159a), QTC(0x4d7177a1), QTC(0x4d8dd631), QTC(0x4daa314b), QTC(0x4dc688f3), - QTC(0x4de2dd2a), QTC(0x4dff2df2), QTC(0x4e1b7b4d), QTC(0x4e37c53d), QTC(0x4e540bc5), QTC(0x4e704ee6), QTC(0x4e8c8ea3), QTC(0x4ea8cafd), - QTC(0x4ec503f7), QTC(0x4ee13992), QTC(0x4efd6bd0), QTC(0x4f199ab4), QTC(0x4f35c640), QTC(0x4f51ee75), QTC(0x4f6e1356), QTC(0x4f8a34e4), - QTC(0x4fa65321), QTC(0x4fc26e10), QTC(0x4fde85b2), QTC(0x4ffa9a0a), QTC(0x5016ab18), QTC(0x5032b8e0), QTC(0x504ec362), QTC(0x506acaa1), - QTC(0x5086cea0), QTC(0x50a2cf5e), QTC(0x50becce0), QTC(0x50dac725), QTC(0x50f6be31), QTC(0x5112b205), QTC(0x512ea2a3), QTC(0x514a900d), - QTC(0x51667a45), QTC(0x5182614c), QTC(0x519e4524), QTC(0x51ba25cf), QTC(0x51d60350), QTC(0x51f1dda7), QTC(0x520db4d6), QTC(0x522988e0), - QTC(0x524559c6), QTC(0x52612789), QTC(0x527cf22d), QTC(0x5298b9b1), QTC(0x52b47e19), QTC(0x52d03f65), QTC(0x52ebfd98), QTC(0x5307b8b4), - QTC(0x532370b9), QTC(0x533f25aa), QTC(0x535ad789), QTC(0x53768656), QTC(0x53923215), QTC(0x53addac6), QTC(0x53c9806b), QTC(0x53e52306), - QTC(0x5400c298), QTC(0x541c5f24), QTC(0x5437f8ab), QTC(0x54538f2e), QTC(0x546f22af), QTC(0x548ab330), QTC(0x54a640b3), QTC(0x54c1cb38), - QTC(0x54dd52c2), QTC(0x54f8d753), QTC(0x551458eb), QTC(0x552fd78d), QTC(0x554b5339), QTC(0x5566cbf3), QTC(0x558241bb), QTC(0x559db492), - QTC(0x55b9247b), QTC(0x55d49177), QTC(0x55effb87), QTC(0x560b62ad), QTC(0x5626c6eb), QTC(0x56422842), QTC(0x565d86b4), QTC(0x5678e242), - QTC(0x56943aee), QTC(0x56af90b9), QTC(0x56cae3a4), QTC(0x56e633b2), QTC(0x570180e4), QTC(0x571ccb3b), QTC(0x573812b8), QTC(0x5753575e), - QTC(0x576e992e), QTC(0x5789d829), QTC(0x57a51450), QTC(0x57c04da6), QTC(0x57db842b), QTC(0x57f6b7e1), QTC(0x5811e8c9), QTC(0x582d16e6), - QTC(0x58484238), QTC(0x58636ac0), QTC(0x587e9081), QTC(0x5899b37c), QTC(0x58b4d3b1), QTC(0x58cff123), QTC(0x58eb0bd3), QTC(0x590623c2), - QTC(0x592138f2), QTC(0x593c4b63), QTC(0x59575b19), QTC(0x59726812), QTC(0x598d7253), QTC(0x59a879da), QTC(0x59c37eab), QTC(0x59de80c6), - QTC(0x59f9802d), QTC(0x5a147ce0), QTC(0x5a2f76e2), QTC(0x5a4a6e34), QTC(0x5a6562d6), QTC(0x5a8054cb), QTC(0x5a9b4414), QTC(0x5ab630b2), - QTC(0x5ad11aa6), QTC(0x5aec01f1), QTC(0x5b06e696), QTC(0x5b21c895), QTC(0x5b3ca7ef), QTC(0x5b5784a6), QTC(0x5b725ebc), QTC(0x5b8d3631), - QTC(0x5ba80b06), QTC(0x5bc2dd3e), QTC(0x5bddacd9), QTC(0x5bf879d8), QTC(0x5c13443d), QTC(0x5c2e0c09), QTC(0x5c48d13e), QTC(0x5c6393dc), - QTC(0x5c7e53e5), QTC(0x5c99115a), QTC(0x5cb3cc3c), QTC(0x5cce848d), QTC(0x5ce93a4e), QTC(0x5d03ed80), QTC(0x5d1e9e24), QTC(0x5d394c3b), - QTC(0x5d53f7c7), QTC(0x5d6ea0c9), QTC(0x5d894742), QTC(0x5da3eb33), QTC(0x5dbe8c9e), QTC(0x5dd92b84), QTC(0x5df3c7e5), QTC(0x5e0e61c3), - QTC(0x5e28f920), QTC(0x5e438dfc), QTC(0x5e5e2059), QTC(0x5e78b037), QTC(0x5e933d99), QTC(0x5eadc87e), QTC(0x5ec850e9), QTC(0x5ee2d6da), - QTC(0x5efd5a53), QTC(0x5f17db54), QTC(0x5f3259e0), QTC(0x5f4cd5f6), QTC(0x5f674f99), QTC(0x5f81c6c8), QTC(0x5f9c3b87), QTC(0x5fb6add4), - QTC(0x5fd11db3), QTC(0x5feb8b23), QTC(0x6005f626), QTC(0x60205ebd), QTC(0x603ac4e9), QTC(0x605528ac), QTC(0x606f8a05), QTC(0x6089e8f7), - QTC(0x60a44583), QTC(0x60be9fa9), QTC(0x60d8f76b), QTC(0x60f34cca), QTC(0x610d9fc7), QTC(0x6127f062), QTC(0x61423e9e), QTC(0x615c8a7a), - QTC(0x6176d3f9), QTC(0x61911b1b), QTC(0x61ab5fe1), QTC(0x61c5a24d), QTC(0x61dfe25f), QTC(0x61fa2018), QTC(0x62145b7a), QTC(0x622e9485), - QTC(0x6248cb3b), QTC(0x6262ff9d), QTC(0x627d31ab), QTC(0x62976167), QTC(0x62b18ed1), QTC(0x62cbb9eb), QTC(0x62e5e2b6), QTC(0x63000933), - QTC(0x631a2d62), QTC(0x63344f45), QTC(0x634e6edd), QTC(0x63688c2b), QTC(0x6382a730), QTC(0x639cbfec), QTC(0x63b6d661), QTC(0x63d0ea90), - QTC(0x63eafc7a), QTC(0x64050c1f), QTC(0x641f1982), QTC(0x643924a2), QTC(0x64532d80), QTC(0x646d341f), QTC(0x6487387e), QTC(0x64a13a9e), - QTC(0x64bb3a81), QTC(0x64d53828), QTC(0x64ef3393), QTC(0x65092cc4), QTC(0x652323bb), QTC(0x653d1879), QTC(0x65570b00), QTC(0x6570fb50), - QTC(0x658ae96b), QTC(0x65a4d550), QTC(0x65bebf01), QTC(0x65d8a680), QTC(0x65f28bcc), QTC(0x660c6ee8), QTC(0x66264fd3), QTC(0x66402e8f), - QTC(0x665a0b1c), QTC(0x6673e57d), QTC(0x668dbdb0), QTC(0x66a793b8), QTC(0x66c16795), QTC(0x66db3949), QTC(0x66f508d4), QTC(0x670ed636), - QTC(0x6728a172), QTC(0x67426a87), QTC(0x675c3177), QTC(0x6775f643), QTC(0x678fb8eb), QTC(0x67a97971), QTC(0x67c337d5), QTC(0x67dcf418), - QTC(0x67f6ae3b), QTC(0x6810663f), QTC(0x682a1c25), QTC(0x6843cfed), QTC(0x685d8199), QTC(0x68773129), QTC(0x6890de9f), QTC(0x68aa89fa), - QTC(0x68c4333d), QTC(0x68ddda67), QTC(0x68f77f7a), QTC(0x69112277), QTC(0x692ac35e), QTC(0x69446230), QTC(0x695dfeee), QTC(0x6977999a), - QTC(0x69913232), QTC(0x69aac8ba), QTC(0x69c45d31), QTC(0x69ddef98), QTC(0x69f77ff0), QTC(0x6a110e3a), QTC(0x6a2a9a77), QTC(0x6a4424a8), - QTC(0x6a5daccc), QTC(0x6a7732e6), QTC(0x6a90b6f6), QTC(0x6aaa38fd), QTC(0x6ac3b8fb), QTC(0x6add36f2), QTC(0x6af6b2e2), QTC(0x6b102ccd), - QTC(0x6b29a4b2), QTC(0x6b431a92), QTC(0x6b5c8e6f), QTC(0x6b76004a), QTC(0x6b8f7022), QTC(0x6ba8ddf9), QTC(0x6bc249d0), QTC(0x6bdbb3a7), - QTC(0x6bf51b80), QTC(0x6c0e815a), QTC(0x6c27e537), QTC(0x6c414718), QTC(0x6c5aa6fd), QTC(0x6c7404e7), QTC(0x6c8d60d7), QTC(0x6ca6bace), - QTC(0x6cc012cc), QTC(0x6cd968d2), QTC(0x6cf2bce1), QTC(0x6d0c0ef9), QTC(0x6d255f1d), QTC(0x6d3ead4b), QTC(0x6d57f985), QTC(0x6d7143cc), - QTC(0x6d8a8c21), QTC(0x6da3d283), QTC(0x6dbd16f5), QTC(0x6dd65976), QTC(0x6def9a08), QTC(0x6e08d8ab), QTC(0x6e221560), QTC(0x6e3b5027), - QTC(0x6e548902), QTC(0x6e6dbff1), QTC(0x6e86f4f5), QTC(0x6ea0280e), QTC(0x6eb9593e), QTC(0x6ed28885), QTC(0x6eebb5e3), QTC(0x6f04e15a), - QTC(0x6f1e0aea), QTC(0x6f373294), QTC(0x6f505859), QTC(0x6f697c39), QTC(0x6f829e35), QTC(0x6f9bbe4e), QTC(0x6fb4dc85), QTC(0x6fcdf8d9), - QTC(0x6fe7134d), QTC(0x70002be0), QTC(0x70194293), QTC(0x70325767), QTC(0x704b6a5d), QTC(0x70647b76), QTC(0x707d8ab1), QTC(0x70969811), - QTC(0x70afa394), QTC(0x70c8ad3d), QTC(0x70e1b50c), QTC(0x70fabb01), QTC(0x7113bf1d), QTC(0x712cc161), QTC(0x7145c1ce), QTC(0x715ec064), - QTC(0x7177bd24), QTC(0x7190b80f), QTC(0x71a9b124), QTC(0x71c2a866), QTC(0x71db9dd4), QTC(0x71f49170), QTC(0x720d8339), QTC(0x72267331), - QTC(0x723f6159), QTC(0x72584db0), QTC(0x72713838), QTC(0x728a20f1), QTC(0x72a307db), QTC(0x72bbecf9), QTC(0x72d4d049), QTC(0x72edb1ce), - QTC(0x73069187), QTC(0x731f6f75), QTC(0x73384b98), QTC(0x735125f3), QTC(0x7369fe84), QTC(0x7382d54d), QTC(0x739baa4e), QTC(0x73b47d89), - QTC(0x73cd4efd), QTC(0x73e61eab), QTC(0x73feec94), QTC(0x7417b8b8), QTC(0x74308319), QTC(0x74494bb6), QTC(0x74621291), QTC(0x747ad7aa), - QTC(0x74939b02), QTC(0x74ac5c98), QTC(0x74c51c6f), QTC(0x74ddda86), QTC(0x74f696de), QTC(0x750f5178), QTC(0x75280a54), QTC(0x7540c174), - QTC(0x755976d7), QTC(0x75722a7e), QTC(0x758adc69), QTC(0x75a38c9b), QTC(0x75bc3b12), QTC(0x75d4e7cf), QTC(0x75ed92d4), QTC(0x76063c21), - QTC(0x761ee3b6), QTC(0x76378994), QTC(0x76502dbc), QTC(0x7668d02e), QTC(0x768170eb), QTC(0x769a0ff3), QTC(0x76b2ad47), QTC(0x76cb48e7), - QTC(0x76e3e2d5), QTC(0x76fc7b10), QTC(0x7715119a), QTC(0x772da673), QTC(0x7746399b), QTC(0x775ecb13), QTC(0x77775adc), QTC(0x778fe8f6), - QTC(0x77a87561), QTC(0x77c1001f), QTC(0x77d98930), QTC(0x77f21095), QTC(0x780a964d), QTC(0x78231a5b), QTC(0x783b9cbd), QTC(0x78541d75), - QTC(0x786c9c84), QTC(0x788519e9), QTC(0x789d95a6), QTC(0x78b60fbb), QTC(0x78ce8828), QTC(0x78e6feef), QTC(0x78ff740f), QTC(0x7917e78a), - QTC(0x7930595f), QTC(0x7948c990), QTC(0x7961381d), QTC(0x7979a506), QTC(0x7992104c), QTC(0x79aa79f0), QTC(0x79c2e1f1), QTC(0x79db4852), - QTC(0x79f3ad11), QTC(0x7a0c1031), QTC(0x7a2471b0), QTC(0x7a3cd191), QTC(0x7a552fd3), QTC(0x7a6d8c76), QTC(0x7a85e77d), QTC(0x7a9e40e6), - QTC(0x7ab698b2), QTC(0x7aceeee3), QTC(0x7ae74378), QTC(0x7aff9673), QTC(0x7b17e7d2), QTC(0x7b303799), QTC(0x7b4885c5), QTC(0x7b60d259), - QTC(0x7b791d55), QTC(0x7b9166b9), QTC(0x7ba9ae86), QTC(0x7bc1f4bc), QTC(0x7bda395c), QTC(0x7bf27c66), QTC(0x7c0abddb), QTC(0x7c22fdbb), - QTC(0x7c3b3c07), QTC(0x7c5378c0), QTC(0x7c6bb3e5), QTC(0x7c83ed78), QTC(0x7c9c2579), QTC(0x7cb45be9), QTC(0x7ccc90c7), QTC(0x7ce4c414), - QTC(0x7cfcf5d2), QTC(0x7d152600), QTC(0x7d2d549f), QTC(0x7d4581b0), QTC(0x7d5dad32), QTC(0x7d75d727), QTC(0x7d8dff8f), QTC(0x7da6266a), - QTC(0x7dbe4bba), QTC(0x7dd66f7d), QTC(0x7dee91b6), QTC(0x7e06b264), QTC(0x7e1ed188), QTC(0x7e36ef22), QTC(0x7e4f0b34), QTC(0x7e6725bd), - QTC(0x7e7f3ebd), QTC(0x7e975636), QTC(0x7eaf6c28), QTC(0x7ec78093), QTC(0x7edf9378), QTC(0x7ef7a4d7), QTC(0x7f0fb4b1), QTC(0x7f27c307), - QTC(0x7f3fcfd8), QTC(0x7f57db25), QTC(0x7f6fe4ef), QTC(0x7f87ed36), QTC(0x7f9ff3fb), QTC(0x7fb7f93e), QTC(0x7fcffcff), QTC(0x7fe7ff40) -}; - -/* - table of pow(2.0,0.25*q)/2.0, q[0..4) -*/ -const FIXP_QTD FDKaacEnc_quantTableQ[4] = { QTC(0x40000000), QTC(0x4c1bf7ff), QTC(0x5a82797f), QTC(0x6ba27e7f) }; - -/* - table of pow(2.0,0.75*e)/8.0, e[0..4) -*/ -const FIXP_QTD FDKaacEnc_quantTableE[4] = { QTC(0x10000000), QTC(0x1ae89f99), QTC(0x2d413ccd), QTC(0x4c1bf828) }; - - -/* - table to count used number of bits -*/ -const SHORT FDKaacEnc_sideInfoTabLong[MAX_SFB_LONG + 1] = -{ - 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, - 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, - 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, - 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x000e, - 0x000e, 0x000e, 0x000e, 0x000e, 0x000e, 0x000e, 0x000e, 0x000e, - 0x000e, 0x000e, 0x000e, 0x000e, 0x000e, 0x000e, 0x000e, 0x000e, - 0x000e, 0x000e, 0x000e, 0x000e -}; - - -const SHORT FDKaacEnc_sideInfoTabShort[MAX_SFB_SHORT + 1] = -{ - 0x0007, 0x0007, 0x0007, 0x0007, 0x0007, 0x0007, 0x0007, 0x000a, - 0x000a, 0x000a, 0x000a, 0x000a, 0x000a, 0x000a, 0x000d, 0x000d -}; - - - - - - -/* - Psy Configuration constants -*/ - -const SFB_PARAM_LONG p_FDKaacEnc_8000_long_1024 = { - 40, - { 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 16, 16, 16, - 20, 20, 20, 20, 24, 24, 24, 28, 28, 32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80 } -}; -const SFB_PARAM_SHORT p_FDKaacEnc_8000_short_128 = { - 15, - { 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20 } -}; - -const SFB_PARAM_LONG p_FDKaacEnc_11025_long_1024 = { - 43, - { 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 12, 12, 12, 12, 12, 12, 12, 12, 12, - 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28, 32, 36, 40, 40, 44, 48, 52, 56, 60, - 64, 64, 64 } -}; -const SFB_PARAM_SHORT p_FDKaacEnc_11025_short_128 = { - 15, - { 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20 } -}; - -const SFB_PARAM_LONG p_FDKaacEnc_12000_long_1024 = { - 43, - { 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 12, 12, 12, 12, 12, 12, 12, 12, 12, - 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28, 32, 36, 40, 40, 44, 48, 52, 56, 60, - 64, 64, 64 } -}; -const SFB_PARAM_SHORT p_FDKaacEnc_12000_short_128 = { - 15, - { 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20 } -}; - -const SFB_PARAM_LONG p_FDKaacEnc_16000_long_1024 = { - 43, - { 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 12, 12, 12, 12, 12, 12, 12, 12, 12, - 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28, 32, 36, 40, 40, 44, 48, 52, 56, 60, - 64, 64, 64 } -}; -const SFB_PARAM_SHORT p_FDKaacEnc_16000_short_128 = { - 15, - { 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20 } -}; -const SFB_PARAM_LONG p_FDKaacEnc_22050_long_1024 = { - 47, - { 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 8, 8, - 8, 12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28, 32, 36, 36, 40, 44, 48, - 52, 52, 64, 64, 64, 64, 64 } -}; -const SFB_PARAM_SHORT p_FDKaacEnc_22050_short_128 = { - 15, - { 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20 } -}; -const SFB_PARAM_LONG p_FDKaacEnc_24000_long_1024 = { - 47, - { 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 8, 8, - 8, 12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28, 32, 36, 36, 40, 44, 48, - 52, 52, 64, 64, 64, 64, 64 } -}; -const SFB_PARAM_SHORT p_FDKaacEnc_24000_short_128 = { - 15, - { 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20 } -}; -const SFB_PARAM_LONG p_FDKaacEnc_32000_long_1024 = { - 51, - { 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 12, 12, 12, - 12, 16, 16, 20, 20, 24, 24, 28, 28, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, - 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32 } -}; -const SFB_PARAM_SHORT p_FDKaacEnc_32000_short_128 = { - 14, - { 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16 } -}; -const SFB_PARAM_LONG p_FDKaacEnc_44100_long_1024 = { - 49, - { 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 12, 12, 12, - 12, 16, 16, 20, 20, 24, 24, 28, 28, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, - 32, 32, 32, 32, 32, 32, 32, 32, 96 } -}; -const SFB_PARAM_SHORT p_FDKaacEnc_44100_short_128 = { - 14, - { 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16 } -}; -const SFB_PARAM_LONG p_FDKaacEnc_48000_long_1024 = { - 49, - { 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 12, 12, 12, - 12, 16, 16, 20, 20, 24, 24, 28, 28, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, - 32, 32, 32, 32, 32, 32, 32, 32, 96 } -}; -const SFB_PARAM_SHORT p_FDKaacEnc_48000_short_128 = { - 14, - { 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16 } -}; -const SFB_PARAM_LONG p_FDKaacEnc_64000_long_1024 = { - 47, - { 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 12, - 12, 16, 16, 16, 20, 24, 24, 28, 36, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, - 40, 40, 40, 40, 40, 40, 40 } -}; -const SFB_PARAM_SHORT p_FDKaacEnc_64000_short_128 = { - 12, - { 4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36 } -}; -const SFB_PARAM_LONG p_FDKaacEnc_88200_long_1024 = { - 41, - { 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 12, - 12, 12, 12, 12, 16, 16, 24, 28, 36, 44, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64, - 64 } -}; -const SFB_PARAM_SHORT p_FDKaacEnc_88200_short_128 = { - 12, - { 4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36 } -}; -const SFB_PARAM_LONG p_FDKaacEnc_96000_long_1024 = { - 41, - { 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 12, - 12, 12, 12, 12, 16, 16, 24, 28, 36, 44, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64, - 64 } -}; -const SFB_PARAM_SHORT p_FDKaacEnc_96000_short_128 = { - 12, - { 4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36 } -}; - - -/* - TNS filter coefficients -*/ - -/* - 3 bit resolution -*/ -const FIXP_DBL FDKaacEnc_tnsEncCoeff3[8]= -{ - (FIXP_DBL)0x81f1d201, (FIXP_DBL)0x91261481, (FIXP_DBL)0xadb92301, (FIXP_DBL)0xd438af00, (FIXP_DBL)0x00000000, (FIXP_DBL)0x37898080, (FIXP_DBL)0x64130dff, (FIXP_DBL)0x7cca6fff -}; -const FIXP_DBL FDKaacEnc_tnsCoeff3Borders[8]={ - (FIXP_DBL)0x80000001 /*-4*/, (FIXP_DBL)0x87b826df /*-3*/, (FIXP_DBL)0x9df24154 /*-2*/, (FIXP_DBL)0xbfffffe5 /*-1*/, - (FIXP_DBL)0xe9c5e578 /* 0*/, (FIXP_DBL)0x1c7b90f0 /* 1*/, (FIXP_DBL)0x4fce83a9 /* 2*/, (FIXP_DBL)0x7352f2c3 /* 3*/ -}; - -/* - 4 bit resolution -*/ -const FIXP_DBL FDKaacEnc_tnsEncCoeff4[16]= -{ - (FIXP_DBL)0x808bc881, (FIXP_DBL)0x84e2e581, (FIXP_DBL)0x8d6b4a01, (FIXP_DBL)0x99da9201, (FIXP_DBL)0xa9c45701, (FIXP_DBL)0xbc9dde81, (FIXP_DBL)0xd1c2d500, (FIXP_DBL)0xe87ae540, - (FIXP_DBL)0x00000000, (FIXP_DBL)0x1a9cd9c0, (FIXP_DBL)0x340ff240, (FIXP_DBL)0x4b3c8bff, (FIXP_DBL)0x5f1f5e7f, (FIXP_DBL)0x6ed9eb7f, (FIXP_DBL)0x79bc387f, (FIXP_DBL)0x7f4c7e7f -}; -const FIXP_DBL FDKaacEnc_tnsCoeff4Borders[16]= -{ - (FIXP_DBL)0x80000001 /*-8*/, (FIXP_DBL)0x822deff0 /*-7*/, (FIXP_DBL)0x88a4bfe6 /*-6*/, (FIXP_DBL)0x932c159d /*-5*/, - (FIXP_DBL)0xa16827c2 /*-4*/, (FIXP_DBL)0xb2dcde27 /*-3*/, (FIXP_DBL)0xc6f20b91 /*-2*/, (FIXP_DBL)0xdcf89c64 /*-1*/, - (FIXP_DBL)0xf4308ce1 /* 0*/, (FIXP_DBL)0x0d613054 /* 1*/, (FIXP_DBL)0x278dde80 /* 2*/, (FIXP_DBL)0x4000001b /* 3*/, - (FIXP_DBL)0x55a6127b /* 4*/, (FIXP_DBL)0x678dde8f /* 5*/, (FIXP_DBL)0x74ef0ed7 /* 6*/, (FIXP_DBL)0x7d33f0da /* 7*/ -}; -const FIXP_DBL FDKaacEnc_mTab_4_3Elc[512]={ - FL2FXCONST_DBL(0.3968502629920499),FL2FXCONST_DBL(0.3978840634868335),FL2FXCONST_DBL(0.3989185359354711),FL2FXCONST_DBL(0.3999536794661432), - FL2FXCONST_DBL(0.4009894932098531),FL2FXCONST_DBL(0.4020259763004115),FL2FXCONST_DBL(0.4030631278744227),FL2FXCONST_DBL(0.4041009470712695), - FL2FXCONST_DBL(0.4051394330330996),FL2FXCONST_DBL(0.4061785849048110),FL2FXCONST_DBL(0.4072184018340380),FL2FXCONST_DBL(0.4082588829711372), - FL2FXCONST_DBL(0.4093000274691739),FL2FXCONST_DBL(0.4103418344839078),FL2FXCONST_DBL(0.4113843031737798),FL2FXCONST_DBL(0.4124274326998980), - FL2FXCONST_DBL(0.4134712222260245),FL2FXCONST_DBL(0.4145156709185620),FL2FXCONST_DBL(0.4155607779465400),FL2FXCONST_DBL(0.4166065424816022), - FL2FXCONST_DBL(0.4176529636979932),FL2FXCONST_DBL(0.4187000407725452),FL2FXCONST_DBL(0.4197477728846652),FL2FXCONST_DBL(0.4207961592163222), - FL2FXCONST_DBL(0.4218451989520345),FL2FXCONST_DBL(0.4228948912788567),FL2FXCONST_DBL(0.4239452353863673),FL2FXCONST_DBL(0.4249962304666564), - FL2FXCONST_DBL(0.4260478757143130),FL2FXCONST_DBL(0.4271001703264124),FL2FXCONST_DBL(0.4281531135025046),FL2FXCONST_DBL(0.4292067044446017), - FL2FXCONST_DBL(0.4302609423571658),FL2FXCONST_DBL(0.4313158264470970),FL2FXCONST_DBL(0.4323713559237216),FL2FXCONST_DBL(0.4334275299987803), - FL2FXCONST_DBL(0.4344843478864161),FL2FXCONST_DBL(0.4355418088031630),FL2FXCONST_DBL(0.4365999119679339),FL2FXCONST_DBL(0.4376586566020096), - FL2FXCONST_DBL(0.4387180419290272),FL2FXCONST_DBL(0.4397780671749683),FL2FXCONST_DBL(0.4408387315681480),FL2FXCONST_DBL(0.4419000343392039), - FL2FXCONST_DBL(0.4429619747210847),FL2FXCONST_DBL(0.4440245519490388),FL2FXCONST_DBL(0.4450877652606038),FL2FXCONST_DBL(0.4461516138955953), - FL2FXCONST_DBL(0.4472160970960963),FL2FXCONST_DBL(0.4482812141064458),FL2FXCONST_DBL(0.4493469641732286),FL2FXCONST_DBL(0.4504133465452648), - FL2FXCONST_DBL(0.4514803604735984),FL2FXCONST_DBL(0.4525480052114875),FL2FXCONST_DBL(0.4536162800143939),FL2FXCONST_DBL(0.4546851841399719), - FL2FXCONST_DBL(0.4557547168480591),FL2FXCONST_DBL(0.4568248774006652),FL2FXCONST_DBL(0.4578956650619623),FL2FXCONST_DBL(0.4589670790982746), - FL2FXCONST_DBL(0.4600391187780688),FL2FXCONST_DBL(0.4611117833719430),FL2FXCONST_DBL(0.4621850721526184),FL2FXCONST_DBL(0.4632589843949278), - FL2FXCONST_DBL(0.4643335193758069),FL2FXCONST_DBL(0.4654086763742842),FL2FXCONST_DBL(0.4664844546714713),FL2FXCONST_DBL(0.4675608535505532), - FL2FXCONST_DBL(0.4686378722967790),FL2FXCONST_DBL(0.4697155101974522),FL2FXCONST_DBL(0.4707937665419216),FL2FXCONST_DBL(0.4718726406215713), - FL2FXCONST_DBL(0.4729521317298118),FL2FXCONST_DBL(0.4740322391620711),FL2FXCONST_DBL(0.4751129622157845),FL2FXCONST_DBL(0.4761943001903867), - FL2FXCONST_DBL(0.4772762523873015),FL2FXCONST_DBL(0.4783588181099338),FL2FXCONST_DBL(0.4794419966636599),FL2FXCONST_DBL(0.4805257873558190), - FL2FXCONST_DBL(0.4816101894957042),FL2FXCONST_DBL(0.4826952023945537),FL2FXCONST_DBL(0.4837808253655421),FL2FXCONST_DBL(0.4848670577237714), - FL2FXCONST_DBL(0.4859538987862632),FL2FXCONST_DBL(0.4870413478719488),FL2FXCONST_DBL(0.4881294043016621),FL2FXCONST_DBL(0.4892180673981298), - FL2FXCONST_DBL(0.4903073364859640),FL2FXCONST_DBL(0.4913972108916533),FL2FXCONST_DBL(0.4924876899435545),FL2FXCONST_DBL(0.4935787729718844), - FL2FXCONST_DBL(0.4946704593087116),FL2FXCONST_DBL(0.4957627482879484),FL2FXCONST_DBL(0.4968556392453423),FL2FXCONST_DBL(0.4979491315184684), - FL2FXCONST_DBL(0.4990432244467211),FL2FXCONST_DBL(0.5001379173713062),FL2FXCONST_DBL(0.5012332096352328),FL2FXCONST_DBL(0.5023291005833056), - FL2FXCONST_DBL(0.5034255895621171),FL2FXCONST_DBL(0.5045226759200399),FL2FXCONST_DBL(0.5056203590072181),FL2FXCONST_DBL(0.5067186381755611), - FL2FXCONST_DBL(0.5078175127787346),FL2FXCONST_DBL(0.5089169821721536),FL2FXCONST_DBL(0.5100170457129749),FL2FXCONST_DBL(0.5111177027600893), - FL2FXCONST_DBL(0.5122189526741143),FL2FXCONST_DBL(0.5133207948173868),FL2FXCONST_DBL(0.5144232285539552),FL2FXCONST_DBL(0.5155262532495726), - FL2FXCONST_DBL(0.5166298682716894),FL2FXCONST_DBL(0.5177340729894460),FL2FXCONST_DBL(0.5188388667736652),FL2FXCONST_DBL(0.5199442489968457), - FL2FXCONST_DBL(0.5210502190331544),FL2FXCONST_DBL(0.5221567762584198),FL2FXCONST_DBL(0.5232639200501247),FL2FXCONST_DBL(0.5243716497873989), - FL2FXCONST_DBL(0.5254799648510130),FL2FXCONST_DBL(0.5265888646233705),FL2FXCONST_DBL(0.5276983484885021),FL2FXCONST_DBL(0.5288084158320574), - FL2FXCONST_DBL(0.5299190660412995),FL2FXCONST_DBL(0.5310302985050975),FL2FXCONST_DBL(0.5321421126139198),FL2FXCONST_DBL(0.5332545077598274), - FL2FXCONST_DBL(0.5343674833364678),FL2FXCONST_DBL(0.5354810387390675),FL2FXCONST_DBL(0.5365951733644262),FL2FXCONST_DBL(0.5377098866109097), - FL2FXCONST_DBL(0.5388251778784438),FL2FXCONST_DBL(0.5399410465685075),FL2FXCONST_DBL(0.5410574920841272),FL2FXCONST_DBL(0.5421745138298695), - FL2FXCONST_DBL(0.5432921112118353),FL2FXCONST_DBL(0.5444102836376534),FL2FXCONST_DBL(0.5455290305164744),FL2FXCONST_DBL(0.5466483512589642), - FL2FXCONST_DBL(0.5477682452772976),FL2FXCONST_DBL(0.5488887119851529),FL2FXCONST_DBL(0.5500097507977050),FL2FXCONST_DBL(0.5511313611316194), - FL2FXCONST_DBL(0.5522535424050467),FL2FXCONST_DBL(0.5533762940376158),FL2FXCONST_DBL(0.5544996154504284),FL2FXCONST_DBL(0.5556235060660528), - FL2FXCONST_DBL(0.5567479653085183),FL2FXCONST_DBL(0.5578729926033087),FL2FXCONST_DBL(0.5589985873773569),FL2FXCONST_DBL(0.5601247490590389), - FL2FXCONST_DBL(0.5612514770781683),FL2FXCONST_DBL(0.5623787708659898),FL2FXCONST_DBL(0.5635066298551742),FL2FXCONST_DBL(0.5646350534798125), - FL2FXCONST_DBL(0.5657640411754097),FL2FXCONST_DBL(0.5668935923788799),FL2FXCONST_DBL(0.5680237065285404),FL2FXCONST_DBL(0.5691543830641059), - FL2FXCONST_DBL(0.5702856214266832),FL2FXCONST_DBL(0.5714174210587655),FL2FXCONST_DBL(0.5725497814042271),FL2FXCONST_DBL(0.5736827019083177), - FL2FXCONST_DBL(0.5748161820176573),FL2FXCONST_DBL(0.5759502211802304),FL2FXCONST_DBL(0.5770848188453810),FL2FXCONST_DBL(0.5782199744638067), - FL2FXCONST_DBL(0.5793556874875542),FL2FXCONST_DBL(0.5804919573700131),FL2FXCONST_DBL(0.5816287835659116),FL2FXCONST_DBL(0.5827661655313104), - FL2FXCONST_DBL(0.5839041027235979),FL2FXCONST_DBL(0.5850425946014850),FL2FXCONST_DBL(0.5861816406250000),FL2FXCONST_DBL(0.5873212402554834), - FL2FXCONST_DBL(0.5884613929555826),FL2FXCONST_DBL(0.5896020981892474),FL2FXCONST_DBL(0.5907433554217242),FL2FXCONST_DBL(0.5918851641195517), - FL2FXCONST_DBL(0.5930275237505556),FL2FXCONST_DBL(0.5941704337838434),FL2FXCONST_DBL(0.5953138936897999),FL2FXCONST_DBL(0.5964579029400819), - FL2FXCONST_DBL(0.5976024610076139),FL2FXCONST_DBL(0.5987475673665825),FL2FXCONST_DBL(0.5998932214924321),FL2FXCONST_DBL(0.6010394228618597), - FL2FXCONST_DBL(0.6021861709528106),FL2FXCONST_DBL(0.6033334652444733),FL2FXCONST_DBL(0.6044813052172748),FL2FXCONST_DBL(0.6056296903528761), - FL2FXCONST_DBL(0.6067786201341671),FL2FXCONST_DBL(0.6079280940452625),FL2FXCONST_DBL(0.6090781115714966),FL2FXCONST_DBL(0.6102286721994192), - FL2FXCONST_DBL(0.6113797754167908),FL2FXCONST_DBL(0.6125314207125777),FL2FXCONST_DBL(0.6136836075769482),FL2FXCONST_DBL(0.6148363355012674), - FL2FXCONST_DBL(0.6159896039780929),FL2FXCONST_DBL(0.6171434125011708),FL2FXCONST_DBL(0.6182977605654305),FL2FXCONST_DBL(0.6194526476669808), - FL2FXCONST_DBL(0.6206080733031054),FL2FXCONST_DBL(0.6217640369722584),FL2FXCONST_DBL(0.6229205381740598),FL2FXCONST_DBL(0.6240775764092919), - FL2FXCONST_DBL(0.6252351511798939),FL2FXCONST_DBL(0.6263932619889586),FL2FXCONST_DBL(0.6275519083407275),FL2FXCONST_DBL(0.6287110897405869), - FL2FXCONST_DBL(0.6298708056950635),FL2FXCONST_DBL(0.6310310557118203),FL2FXCONST_DBL(0.6321918392996523),FL2FXCONST_DBL(0.6333531559684823), - FL2FXCONST_DBL(0.6345150052293571),FL2FXCONST_DBL(0.6356773865944432),FL2FXCONST_DBL(0.6368402995770224),FL2FXCONST_DBL(0.6380037436914881), - FL2FXCONST_DBL(0.6391677184533411),FL2FXCONST_DBL(0.6403322233791856),FL2FXCONST_DBL(0.6414972579867254),FL2FXCONST_DBL(0.6426628217947594), - FL2FXCONST_DBL(0.6438289143231779),FL2FXCONST_DBL(0.6449955350929588),FL2FXCONST_DBL(0.6461626836261636),FL2FXCONST_DBL(0.6473303594459330), - FL2FXCONST_DBL(0.6484985620764839),FL2FXCONST_DBL(0.6496672910431047),FL2FXCONST_DBL(0.6508365458721518),FL2FXCONST_DBL(0.6520063260910459), - FL2FXCONST_DBL(0.6531766312282679),FL2FXCONST_DBL(0.6543474608133552),FL2FXCONST_DBL(0.6555188143768979),FL2FXCONST_DBL(0.6566906914505349), - FL2FXCONST_DBL(0.6578630915669509),FL2FXCONST_DBL(0.6590360142598715),FL2FXCONST_DBL(0.6602094590640603),FL2FXCONST_DBL(0.6613834255153149), - FL2FXCONST_DBL(0.6625579131504635),FL2FXCONST_DBL(0.6637329215073610),FL2FXCONST_DBL(0.6649084501248851),FL2FXCONST_DBL(0.6660844985429335), - FL2FXCONST_DBL(0.6672610663024197),FL2FXCONST_DBL(0.6684381529452691),FL2FXCONST_DBL(0.6696157580144163),FL2FXCONST_DBL(0.6707938810538011), - FL2FXCONST_DBL(0.6719725216083646),FL2FXCONST_DBL(0.6731516792240465),FL2FXCONST_DBL(0.6743313534477807),FL2FXCONST_DBL(0.6755115438274927), - FL2FXCONST_DBL(0.6766922499120955),FL2FXCONST_DBL(0.6778734712514865),FL2FXCONST_DBL(0.6790552073965435),FL2FXCONST_DBL(0.6802374578991223), - FL2FXCONST_DBL(0.6814202223120524),FL2FXCONST_DBL(0.6826035001891340),FL2FXCONST_DBL(0.6837872910851345),FL2FXCONST_DBL(0.6849715945557853), - FL2FXCONST_DBL(0.6861564101577784),FL2FXCONST_DBL(0.6873417374487629),FL2FXCONST_DBL(0.6885275759873420),FL2FXCONST_DBL(0.6897139253330697), - FL2FXCONST_DBL(0.6909007850464473),FL2FXCONST_DBL(0.6920881546889198),FL2FXCONST_DBL(0.6932760338228737),FL2FXCONST_DBL(0.6944644220116332), - FL2FXCONST_DBL(0.6956533188194565),FL2FXCONST_DBL(0.6968427238115332),FL2FXCONST_DBL(0.6980326365539813),FL2FXCONST_DBL(0.6992230566138435), - FL2FXCONST_DBL(0.7004139835590845),FL2FXCONST_DBL(0.7016054169585869),FL2FXCONST_DBL(0.7027973563821499),FL2FXCONST_DBL(0.7039898014004843), - FL2FXCONST_DBL(0.7051827515852106),FL2FXCONST_DBL(0.7063762065088554),FL2FXCONST_DBL(0.7075701657448483),FL2FXCONST_DBL(0.7087646288675196), - FL2FXCONST_DBL(0.7099595954520960),FL2FXCONST_DBL(0.7111550650746988),FL2FXCONST_DBL(0.7123510373123402),FL2FXCONST_DBL(0.7135475117429202), - FL2FXCONST_DBL(0.7147444879452244),FL2FXCONST_DBL(0.7159419654989200),FL2FXCONST_DBL(0.7171399439845538),FL2FXCONST_DBL(0.7183384229835486), - FL2FXCONST_DBL(0.7195374020782005),FL2FXCONST_DBL(0.7207368808516762),FL2FXCONST_DBL(0.7219368588880097),FL2FXCONST_DBL(0.7231373357720997), - FL2FXCONST_DBL(0.7243383110897066),FL2FXCONST_DBL(0.7255397844274496),FL2FXCONST_DBL(0.7267417553728043),FL2FXCONST_DBL(0.7279442235140992), - FL2FXCONST_DBL(0.7291471884405130),FL2FXCONST_DBL(0.7303506497420724),FL2FXCONST_DBL(0.7315546070096487),FL2FXCONST_DBL(0.7327590598349553), - FL2FXCONST_DBL(0.7339640078105445),FL2FXCONST_DBL(0.7351694505298055),FL2FXCONST_DBL(0.7363753875869610),FL2FXCONST_DBL(0.7375818185770647), - FL2FXCONST_DBL(0.7387887430959987),FL2FXCONST_DBL(0.7399961607404706),FL2FXCONST_DBL(0.7412040711080108),FL2FXCONST_DBL(0.7424124737969701), - FL2FXCONST_DBL(0.7436213684065166),FL2FXCONST_DBL(0.7448307545366334),FL2FXCONST_DBL(0.7460406317881158),FL2FXCONST_DBL(0.7472509997625686), - FL2FXCONST_DBL(0.7484618580624036),FL2FXCONST_DBL(0.7496732062908372),FL2FXCONST_DBL(0.7508850440518872),FL2FXCONST_DBL(0.7520973709503704), - FL2FXCONST_DBL(0.7533101865919009),FL2FXCONST_DBL(0.7545234905828862),FL2FXCONST_DBL(0.7557372825305252),FL2FXCONST_DBL(0.7569515620428062), - FL2FXCONST_DBL(0.7581663287285035),FL2FXCONST_DBL(0.7593815821971756),FL2FXCONST_DBL(0.7605973220591619),FL2FXCONST_DBL(0.7618135479255810), - FL2FXCONST_DBL(0.7630302594083277),FL2FXCONST_DBL(0.7642474561200708),FL2FXCONST_DBL(0.7654651376742505),FL2FXCONST_DBL(0.7666833036850760), - FL2FXCONST_DBL(0.7679019537675227),FL2FXCONST_DBL(0.7691210875373307),FL2FXCONST_DBL(0.7703407046110011),FL2FXCONST_DBL(0.7715608046057948), - FL2FXCONST_DBL(0.7727813871397293),FL2FXCONST_DBL(0.7740024518315765),FL2FXCONST_DBL(0.7752239983008605),FL2FXCONST_DBL(0.7764460261678551), - FL2FXCONST_DBL(0.7776685350535814),FL2FXCONST_DBL(0.7788915245798054),FL2FXCONST_DBL(0.7801149943690360),FL2FXCONST_DBL(0.7813389440445223), - FL2FXCONST_DBL(0.7825633732302513),FL2FXCONST_DBL(0.7837882815509458),FL2FXCONST_DBL(0.7850136686320621),FL2FXCONST_DBL(0.7862395340997874), - FL2FXCONST_DBL(0.7874658775810378),FL2FXCONST_DBL(0.7886926987034559),FL2FXCONST_DBL(0.7899199970954088),FL2FXCONST_DBL(0.7911477723859853), - FL2FXCONST_DBL(0.7923760242049944),FL2FXCONST_DBL(0.7936047521829623),FL2FXCONST_DBL(0.7948339559511308),FL2FXCONST_DBL(0.7960636351414546), - FL2FXCONST_DBL(0.7972937893865995),FL2FXCONST_DBL(0.7985244183199399),FL2FXCONST_DBL(0.7997555215755570),FL2FXCONST_DBL(0.8009870987882359), - FL2FXCONST_DBL(0.8022191495934644),FL2FXCONST_DBL(0.8034516736274301),FL2FXCONST_DBL(0.8046846705270185),FL2FXCONST_DBL(0.8059181399298110), - FL2FXCONST_DBL(0.8071520814740822),FL2FXCONST_DBL(0.8083864947987989),FL2FXCONST_DBL(0.8096213795436166),FL2FXCONST_DBL(0.8108567353488784), - FL2FXCONST_DBL(0.8120925618556127),FL2FXCONST_DBL(0.8133288587055308),FL2FXCONST_DBL(0.8145656255410253),FL2FXCONST_DBL(0.8158028620051674), - FL2FXCONST_DBL(0.8170405677417053),FL2FXCONST_DBL(0.8182787423950622),FL2FXCONST_DBL(0.8195173856103341),FL2FXCONST_DBL(0.8207564970332875), - FL2FXCONST_DBL(0.8219960763103580),FL2FXCONST_DBL(0.8232361230886477),FL2FXCONST_DBL(0.8244766370159234),FL2FXCONST_DBL(0.8257176177406150), - FL2FXCONST_DBL(0.8269590649118125),FL2FXCONST_DBL(0.8282009781792650),FL2FXCONST_DBL(0.8294433571933784),FL2FXCONST_DBL(0.8306862016052132), - FL2FXCONST_DBL(0.8319295110664831),FL2FXCONST_DBL(0.8331732852295520),FL2FXCONST_DBL(0.8344175237474336),FL2FXCONST_DBL(0.8356622262737878), - FL2FXCONST_DBL(0.8369073924629202),FL2FXCONST_DBL(0.8381530219697793),FL2FXCONST_DBL(0.8393991144499545),FL2FXCONST_DBL(0.8406456695596752), - FL2FXCONST_DBL(0.8418926869558079),FL2FXCONST_DBL(0.8431401662958544),FL2FXCONST_DBL(0.8443881072379507),FL2FXCONST_DBL(0.8456365094408642), - FL2FXCONST_DBL(0.8468853725639923),FL2FXCONST_DBL(0.8481346962673606),FL2FXCONST_DBL(0.8493844802116208),FL2FXCONST_DBL(0.8506347240580492), - FL2FXCONST_DBL(0.8518854274685442),FL2FXCONST_DBL(0.8531365901056253),FL2FXCONST_DBL(0.8543882116324307),FL2FXCONST_DBL(0.8556402917127157), - FL2FXCONST_DBL(0.8568928300108512),FL2FXCONST_DBL(0.8581458261918209),FL2FXCONST_DBL(0.8593992799212207),FL2FXCONST_DBL(0.8606531908652563), - FL2FXCONST_DBL(0.8619075586907414),FL2FXCONST_DBL(0.8631623830650962),FL2FXCONST_DBL(0.8644176636563452),FL2FXCONST_DBL(0.8656734001331161), - FL2FXCONST_DBL(0.8669295921646375),FL2FXCONST_DBL(0.8681862394207371),FL2FXCONST_DBL(0.8694433415718407),FL2FXCONST_DBL(0.8707008982889695), - FL2FXCONST_DBL(0.8719589092437391),FL2FXCONST_DBL(0.8732173741083574),FL2FXCONST_DBL(0.8744762925556232),FL2FXCONST_DBL(0.8757356642589241), - FL2FXCONST_DBL(0.8769954888922352),FL2FXCONST_DBL(0.8782557661301171),FL2FXCONST_DBL(0.8795164956477146),FL2FXCONST_DBL(0.8807776771207545), - FL2FXCONST_DBL(0.8820393102255443),FL2FXCONST_DBL(0.8833013946389704),FL2FXCONST_DBL(0.8845639300384969),FL2FXCONST_DBL(0.8858269161021629), - FL2FXCONST_DBL(0.8870903525085819),FL2FXCONST_DBL(0.8883542389369399),FL2FXCONST_DBL(0.8896185750669933),FL2FXCONST_DBL(0.8908833605790678), - FL2FXCONST_DBL(0.8921485951540565),FL2FXCONST_DBL(0.8934142784734187),FL2FXCONST_DBL(0.8946804102191776),FL2FXCONST_DBL(0.8959469900739191), - FL2FXCONST_DBL(0.8972140177207906),FL2FXCONST_DBL(0.8984814928434985),FL2FXCONST_DBL(0.8997494151263077),FL2FXCONST_DBL(0.9010177842540390), - FL2FXCONST_DBL(0.9022865999120682),FL2FXCONST_DBL(0.9035558617863242),FL2FXCONST_DBL(0.9048255695632878),FL2FXCONST_DBL(0.9060957229299895), - FL2FXCONST_DBL(0.9073663215740092),FL2FXCONST_DBL(0.9086373651834729),FL2FXCONST_DBL(0.9099088534470528),FL2FXCONST_DBL(0.9111807860539647), - FL2FXCONST_DBL(0.9124531626939672),FL2FXCONST_DBL(0.9137259830573594),FL2FXCONST_DBL(0.9149992468349805),FL2FXCONST_DBL(0.9162729537182071), - FL2FXCONST_DBL(0.9175471033989524),FL2FXCONST_DBL(0.9188216955696648),FL2FXCONST_DBL(0.9200967299233258),FL2FXCONST_DBL(0.9213722061534494), - FL2FXCONST_DBL(0.9226481239540795),FL2FXCONST_DBL(0.9239244830197896),FL2FXCONST_DBL(0.9252012830456805),FL2FXCONST_DBL(0.9264785237273793), - FL2FXCONST_DBL(0.9277562047610376),FL2FXCONST_DBL(0.9290343258433305),FL2FXCONST_DBL(0.9303128866714547),FL2FXCONST_DBL(0.9315918869431275), - FL2FXCONST_DBL(0.9328713263565848),FL2FXCONST_DBL(0.9341512046105802),FL2FXCONST_DBL(0.9354315214043836),FL2FXCONST_DBL(0.9367122764377792), - FL2FXCONST_DBL(0.9379934694110648),FL2FXCONST_DBL(0.9392751000250497),FL2FXCONST_DBL(0.9405571679810542),FL2FXCONST_DBL(0.9418396729809072), - FL2FXCONST_DBL(0.9431226147269456),FL2FXCONST_DBL(0.9444059929220124),FL2FXCONST_DBL(0.9456898072694558),FL2FXCONST_DBL(0.9469740574731275), - FL2FXCONST_DBL(0.9482587432373810),FL2FXCONST_DBL(0.9495438642670713),FL2FXCONST_DBL(0.9508294202675522),FL2FXCONST_DBL(0.9521154109446763), - FL2FXCONST_DBL(0.9534018360047926),FL2FXCONST_DBL(0.9546886951547455),FL2FXCONST_DBL(0.9559759881018738),FL2FXCONST_DBL(0.9572637145540087), - FL2FXCONST_DBL(0.9585518742194732),FL2FXCONST_DBL(0.9598404668070802),FL2FXCONST_DBL(0.9611294920261317),FL2FXCONST_DBL(0.9624189495864168), - FL2FXCONST_DBL(0.9637088391982110),FL2FXCONST_DBL(0.9649991605722750),FL2FXCONST_DBL(0.9662899134198524),FL2FXCONST_DBL(0.9675810974526697), - FL2FXCONST_DBL(0.9688727123829343),FL2FXCONST_DBL(0.9701647579233330),FL2FXCONST_DBL(0.9714572337870316),FL2FXCONST_DBL(0.9727501396876727), - FL2FXCONST_DBL(0.9740434753393749),FL2FXCONST_DBL(0.9753372404567313),FL2FXCONST_DBL(0.9766314347548087),FL2FXCONST_DBL(0.9779260579491460), - FL2FXCONST_DBL(0.9792211097557527),FL2FXCONST_DBL(0.9805165898911081),FL2FXCONST_DBL(0.9818124980721600),FL2FXCONST_DBL(0.9831088340163232), - FL2FXCONST_DBL(0.9844055974414786),FL2FXCONST_DBL(0.9857027880659716),FL2FXCONST_DBL(0.9870004056086111),FL2FXCONST_DBL(0.9882984497886684), - FL2FXCONST_DBL(0.9895969203258759),FL2FXCONST_DBL(0.9908958169404255),FL2FXCONST_DBL(0.9921951393529680),FL2FXCONST_DBL(0.9934948872846116), - FL2FXCONST_DBL(0.9947950604569206),FL2FXCONST_DBL(0.9960956585919144),FL2FXCONST_DBL(0.9973966814120665),FL2FXCONST_DBL(0.9986981286403025) -}; - -const FIXP_DBL FDKaacEnc_specExpMantTableCombElc[4][14] = -{ - {FL2FXCONST_DBL(0.5000000000000000), FL2FXCONST_DBL(0.6299605249474366), FL2FXCONST_DBL(0.7937005259840998), FL2FXCONST_DBL(0.5000000000000000), - FL2FXCONST_DBL(0.6299605249474366), FL2FXCONST_DBL(0.7937005259840998), FL2FXCONST_DBL(0.5000000000000000), FL2FXCONST_DBL(0.6299605249474366), - FL2FXCONST_DBL(0.7937005259840998), FL2FXCONST_DBL(0.5000000000000000), FL2FXCONST_DBL(0.6299605249474366), FL2FXCONST_DBL(0.7937005259840998), - FL2FXCONST_DBL(0.5000000000000000), FL2FXCONST_DBL(0.6299605249474366)}, - - {FL2FXCONST_DBL(0.5946035575013605), FL2FXCONST_DBL(0.7491535384383408), FL2FXCONST_DBL(0.9438743126816935), FL2FXCONST_DBL(0.5946035575013605), - FL2FXCONST_DBL(0.7491535384383408), FL2FXCONST_DBL(0.9438743126816935), FL2FXCONST_DBL(0.5946035575013605), FL2FXCONST_DBL(0.7491535384383408), - FL2FXCONST_DBL(0.9438743126816935), FL2FXCONST_DBL(0.5946035575013605), FL2FXCONST_DBL(0.7491535384383408), FL2FXCONST_DBL(0.9438743126816935), - FL2FXCONST_DBL(0.5946035575013605), FL2FXCONST_DBL(0.7491535384383408)}, - - {FL2FXCONST_DBL(0.7071067811865476), FL2FXCONST_DBL(0.8908987181403393), FL2FXCONST_DBL(0.5612310241546865), FL2FXCONST_DBL(0.7071067811865476), - FL2FXCONST_DBL(0.8908987181403393), FL2FXCONST_DBL(0.5612310241546865), FL2FXCONST_DBL(0.7071067811865476), FL2FXCONST_DBL(0.8908987181403393), - FL2FXCONST_DBL(0.5612310241546865), FL2FXCONST_DBL(0.7071067811865476), FL2FXCONST_DBL(0.8908987181403393), FL2FXCONST_DBL(0.5612310241546865), - FL2FXCONST_DBL(0.7071067811865476), FL2FXCONST_DBL(0.8908987181403393)}, - - {FL2FXCONST_DBL(0.8408964152537145), FL2FXCONST_DBL(0.5297315471796477), FL2FXCONST_DBL(0.6674199270850172), FL2FXCONST_DBL(0.8408964152537145), - FL2FXCONST_DBL(0.5297315471796477), FL2FXCONST_DBL(0.6674199270850172), FL2FXCONST_DBL(0.8408964152537145), FL2FXCONST_DBL(0.5297315471796477), - FL2FXCONST_DBL(0.6674199270850172), FL2FXCONST_DBL(0.8408964152537145), FL2FXCONST_DBL(0.5297315471796477), FL2FXCONST_DBL(0.6674199270850172), - FL2FXCONST_DBL(0.8408964152537145), FL2FXCONST_DBL(0.5297315471796477)} -}; - -const UCHAR FDKaacEnc_specExpTableComb[4][14] = -{ - {1, 2, 3, 5, 6, 7, 9, 10, 11, 13, 14, 15, 17, 18}, - {1, 2, 3, 5, 6, 7, 9, 10, 11, 13, 14, 15, 17, 18}, - {1, 2, 4, 5, 6, 8, 9, 10, 12, 13, 14, 16, 17, 18}, - {1, 3, 4, 5, 7, 8, 9, 11, 12, 13, 15, 16, 17, 19} -}; - - -#define WTS0 1 -#define WTS1 0 -#define WTS2 -2 - -const FIXP_WTB ELDAnalysis512[1536] = { - /* part 0 */ - WTC0(0xfac5a770), WTC0(0xfaafbab8), WTC0(0xfa996a40), WTC0(0xfa82bbd0), WTC0(0xfa6bb538), WTC0(0xfa545c38), WTC0(0xfa3cb698), WTC0(0xfa24ca28), - WTC0(0xfa0c9ca8), WTC0(0xf9f433e8), WTC0(0xf9db9580), WTC0(0xf9c2c298), WTC0(0xf9a9b800), WTC0(0xf9907250), WTC0(0xf976ee38), WTC0(0xf95d2b88), - WTC0(0xf9432d10), WTC0(0xf928f5c0), WTC0(0xf90e8868), WTC0(0xf8f3e400), WTC0(0xf8d903a0), WTC0(0xf8bde238), WTC0(0xf8a27af0), WTC0(0xf886cde8), - WTC0(0xf86ae020), WTC0(0xf84eb6c0), WTC0(0xf83256f8), WTC0(0xf815c4b8), WTC0(0xf7f902c0), WTC0(0xf7dc13b0), WTC0(0xf7befa60), WTC0(0xf7a1ba40), - WTC0(0xf78457c0), WTC0(0xf766d780), WTC0(0xf7493d90), WTC0(0xf72b8990), WTC0(0xf70db5f0), WTC0(0xf6efbd30), WTC0(0xf6d19a20), WTC0(0xf6b352e0), - WTC0(0xf694f8c0), WTC0(0xf6769da0), WTC0(0xf6585310), WTC0(0xf63a28d0), WTC0(0xf61c2c60), WTC0(0xf5fe6b10), WTC0(0xf5e0f250), WTC0(0xf5c3ceb0), - WTC0(0xf5a70be0), WTC0(0xf58ab5a0), WTC0(0xf56ed7b0), WTC0(0xf5537e40), WTC0(0xf538b610), WTC0(0xf51e8bf0), WTC0(0xf5050c90), WTC0(0xf4ec4330), - WTC0(0xf4d439b0), WTC0(0xf4bcf9b0), WTC0(0xf4a68ce0), WTC0(0xf490fa80), WTC0(0xf47c4760), WTC0(0xf4687830), WTC0(0xf4558f00), WTC0(0xf4434fc0), - WTC0(0xf4314070), WTC0(0xf41ee450), WTC0(0xf40bc130), WTC0(0xf3f799c0), WTC0(0xf3e26d30), WTC0(0xf3cc3d70), WTC0(0xf3b50c80), WTC0(0xf39cdd60), - WTC0(0xf383b440), WTC0(0xf3699550), WTC0(0xf34e84c0), WTC0(0xf33286b0), WTC0(0xf3159f10), WTC0(0xf2f7d1b0), WTC0(0xf2d92290), WTC0(0xf2b994d0), - WTC0(0xf2992ad0), WTC0(0xf277e6d0), WTC0(0xf255cb60), WTC0(0xf232dd00), WTC0(0xf20f2240), WTC0(0xf1eaa1d0), WTC0(0xf1c56240), WTC0(0xf19f63d0), - WTC0(0xf178a0f0), WTC0(0xf15113a0), WTC0(0xf128b5c0), WTC0(0xf0ff7fd0), WTC0(0xf0d56860), WTC0(0xf0aa6610), WTC0(0xf07e6fd0), WTC0(0xf0518190), - WTC0(0xf0239cd0), WTC0(0xeff4c320), WTC0(0xefc4f720), WTC0(0xef945080), WTC0(0xef62fce0), WTC0(0xef312a40), WTC0(0xeeff05c0), WTC0(0xeecca2c0), - WTC0(0xee99faa0), WTC0(0xee6705a0), WTC0(0xee33bb60), WTC0(0xee000060), WTC0(0xedcba660), WTC0(0xed967e80), WTC0(0xed605b80), WTC0(0xed293b40), - WTC0(0xecf146a0), WTC0(0xecb8a8a0), WTC0(0xec7f8bc0), WTC0(0xec461260), WTC0(0xec0c5720), WTC0(0xebd27440), WTC0(0xeb988220), WTC0(0xeb5e7040), - WTC0(0xeb2404c0), WTC0(0xeae90440), WTC0(0xeaad33c0), WTC0(0xea7066c0), WTC0(0xea327f60), WTC0(0xe9f36000), WTC0(0xe9b2ed60), WTC0(0xe9713920), - WTC0(0xe92e81e0), WTC0(0xe8eb08c0), WTC0(0xe8a70e60), WTC0(0xe862d8e0), WTC0(0xe81eb340), WTC0(0xe7dae8a0), WTC0(0xe797c1a0), WTC0(0xe7554ca0), - WTC0(0xe7135dc0), WTC0(0xe6d1c6a0), WTC0(0xe6905720), WTC0(0xe64eb9c0), WTC0(0xe60c7300), WTC0(0xe5c90600), WTC0(0xe583f920), WTC0(0xe53d1ce0), - WTC0(0xe4f48c80), WTC0(0xe4aa6640), WTC0(0xe45ecaa0), WTC0(0xe4120be0), WTC0(0xe3c4ae60), WTC0(0xe3773860), WTC0(0xe32a2ea0), WTC0(0xe2ddeea0), - WTC0(0xe292af00), WTC0(0xe248a4a0), WTC0(0xe2000140), WTC0(0xe1b8b640), WTC0(0xe1727440), WTC0(0xe12ce900), WTC0(0xe0e7c280), WTC0(0xe0a2b420), - WTC0(0xe05d76c0), WTC0(0xe017c360), WTC0(0xdfd15440), WTC0(0xdf8a0540), WTC0(0xdf41d300), WTC0(0xdef8bb40), WTC0(0xdeaebd40), WTC0(0xde63e7c0), - WTC0(0xde185940), WTC0(0xddcc3180), WTC0(0xdd7f9000), WTC0(0xdd329e80), WTC0(0xdce58e80), WTC0(0xdc989300), WTC0(0xdc4bde40), WTC0(0xdbff96c0), - WTC0(0xdbb3d780), WTC0(0xdb68bb80), WTC0(0xdb1e5c80), WTC0(0xdad4c380), WTC0(0xda8be840), WTC0(0xda43c1c0), WTC0(0xd9fc4740), WTC0(0xd9b56640), - WTC0(0xd96f0440), WTC0(0xd9290600), WTC0(0xd8e35080), WTC0(0xd89dcd40), WTC0(0xd8586b40), WTC0(0xd8131940), WTC0(0xd7cdc640), WTC0(0xd7886180), - WTC0(0xd742dc80), WTC0(0xd6fd2780), WTC0(0xd6b73400), WTC0(0xd670fd80), WTC0(0xd62a8a40), WTC0(0xd5e3e080), WTC0(0xd59d0840), WTC0(0xd5562b80), - WTC0(0xd50f9540), WTC0(0xd4c992c0), WTC0(0xd4846f80), WTC0(0xd4405a80), WTC0(0xd3fd6580), WTC0(0xd3bba140), WTC0(0xd37b1c80), WTC0(0xd33bb780), - WTC0(0xd2fd2400), WTC0(0xd2bf1240), WTC0(0xd2813300), WTC0(0xd2435ac0), WTC0(0xd2057fc0), WTC0(0xd1c79a00), WTC0(0xd189a240), WTC0(0xd14b9dc0), - WTC0(0xd10d9e00), WTC0(0xd0cfb580), WTC0(0xd091f6c0), WTC0(0xd0548100), WTC0(0xd0177f40), WTC0(0xcfdb1cc0), WTC0(0xcf9f84c0), WTC0(0xcf64d780), - WTC0(0xcf2b2b00), WTC0(0xcef29440), WTC0(0xcebb2640), WTC0(0xce84c000), WTC0(0xce4f0bc0), WTC0(0xce19b200), WTC0(0xcde45d40), WTC0(0xcdaeedc0), - WTC0(0xcd7979c0), WTC0(0xcd4419c0), WTC0(0xcd0ee6c0), WTC0(0xccda0540), WTC0(0xcca5a500), WTC0(0xcc71f640), WTC0(0xcc3f2800), WTC0(0xcc0d4300), - WTC0(0xcbdc2a00), WTC0(0xcbabbe80), WTC0(0xcb7be200), WTC0(0xcb4c8200), WTC0(0xcb1d9800), WTC0(0xcaef1d40), WTC0(0xcac10bc0), WTC0(0xca936440), - WTC0(0xca662d00), WTC0(0xca396d40), WTC0(0xca0d2b80), WTC0(0xc9e16f80), WTC0(0xc9b63f80), WTC0(0xc98ba2c0), WTC0(0xc961a000), WTC0(0xc9383ec0), - WTC0(0xc90a0440), WTC0(0xc8e0d280), WTC0(0xc8b73b80), WTC0(0xc88d4900), WTC0(0xc86304c0), WTC0(0xc83878c0), WTC0(0xc80dae80), WTC0(0xc7e2afc0), - WTC0(0xc7b78640), WTC0(0xc78c3c40), WTC0(0xc760da80), WTC0(0xc7356640), WTC0(0xc709de40), WTC0(0xc6de41c0), WTC0(0xc6b28fc0), WTC0(0xc686bd40), - WTC0(0xc65ab600), WTC0(0xc62e6580), WTC0(0xc601b880), WTC0(0xc5d4bac0), WTC0(0xc5a79640), WTC0(0xc57a76c0), WTC0(0xc54d8780), WTC0(0xc520e840), - WTC0(0xc4f4acc0), WTC0(0xc4c8e880), WTC0(0xc49dad80), WTC0(0xc472e640), WTC0(0xc44856c0), WTC0(0xc41dc140), WTC0(0xc3f2e940), WTC0(0xc3c7bc00), - WTC0(0xc39c4f00), WTC0(0xc370b9c0), WTC0(0xc34513c0), WTC0(0xc3197940), WTC0(0xc2ee0a00), WTC0(0xc2c2e640), WTC0(0xc2982d80), WTC0(0xc26df5c0), - WTC0(0xc2444b00), WTC0(0xc21b3940), WTC0(0xc1f2cbc0), WTC0(0xc1cb05c0), WTC0(0xc1a3e340), WTC0(0xc17d5f00), WTC0(0xc15773c0), WTC0(0xc1320940), - WTC0(0xc10cf480), WTC0(0xc0e80a00), WTC0(0xc0c31f00), WTC0(0xc09e2640), WTC0(0xc0792ec0), WTC0(0xc0544940), WTC0(0xc02f86c0), WTC0(0xc00b04c0), - WTC0(0xbfe6ed01), WTC0(0xbfc36a01), WTC0(0xbfa0a581), WTC0(0xbf7eb581), WTC0(0xbf5d9a81), WTC0(0xbf3d5501), WTC0(0xbf1de601), WTC0(0xbeff4801), - WTC0(0xbee17201), WTC0(0xbec45881), WTC0(0xbea7f301), WTC0(0xbe8c3781), WTC0(0xbe712001), WTC0(0xbe56a381), WTC0(0xbe3cbc01), WTC0(0xbe236001), - WTC0(0xbe0a8581), WTC0(0xbdf22181), WTC0(0xbdda2a01), WTC0(0xbdc29a81), WTC0(0xbdab7181), WTC0(0xbd94b001), WTC0(0xbd7e5581), WTC0(0xbd686681), - WTC0(0xbd52eb01), WTC0(0xbd3deb81), WTC0(0xbd297181), WTC0(0xbd158801), WTC0(0xbd023f01), WTC0(0xbcefa601), WTC0(0xbcddcc81), WTC0(0xbcccbd01), - WTC0(0xbcbc7e01), WTC0(0xbcad1501), WTC0(0xbc9e8801), WTC0(0xbc90d481), WTC0(0xbc83f201), WTC0(0xbc77d601), WTC0(0xbc6c7781), WTC0(0xbc61c401), - WTC0(0xbc57a301), WTC0(0xbc4dfb81), WTC0(0xbc44b481), WTC0(0xbc3bbc01), WTC0(0xbc330781), WTC0(0xbc2a8c81), WTC0(0xbc224181), WTC0(0xbc1a2401), - WTC0(0xbc123b81), WTC0(0xbc0a8f01), WTC0(0xbc032601), WTC0(0xbbfc0f81), WTC0(0xbbf56181), WTC0(0xbbef3301), WTC0(0xbbe99981), WTC0(0xbbe49d01), - WTC0(0xbbe03801), WTC0(0xbbdc6481), WTC0(0xbbd91b81), WTC0(0xbbd64d01), WTC0(0xbbd3e101), WTC0(0xbbd1bd81), WTC0(0xbbcfca81), WTC0(0xbbce0601), - WTC0(0xbbcc8201), WTC0(0xbbcb5301), WTC0(0xbbca8d01), WTC0(0xbbca5081), WTC0(0xbbcaca01), WTC0(0xbbcc2681), WTC0(0xbbce9181), WTC0(0xbbd21281), - WTC0(0xbbd68c81), WTC0(0xbbdbe201), WTC0(0xbbe1f401), WTC0(0xbbe89901), WTC0(0xbbef9b81), WTC0(0xbbf6c601), WTC0(0xbbfde481), WTC0(0xbc04e381), - WTC0(0xbc0bcf81), WTC0(0xbc12b801), WTC0(0xbc19ab01), WTC0(0xbc20ae01), WTC0(0xbc27bd81), WTC0(0xbc2ed681), WTC0(0xbc35f501), WTC0(0xbc3d1801), - WTC0(0xbc444081), WTC0(0xbc4b6e81), WTC0(0xbc52a381), WTC0(0xbc59df81), WTC0(0xbc612301), WTC0(0xbc686e01), WTC0(0xbc6fc101), WTC0(0xbc771c01), - WTC0(0xbc7e7e01), WTC0(0xbc85e801), WTC0(0xbc8d5901), WTC0(0xbc94d201), WTC0(0xbc9c5281), WTC0(0xbca3db01), WTC0(0xbcab6c01), WTC0(0xbcb30601), - WTC0(0xbcbaa801), WTC0(0xbcc25181), WTC0(0xbcca0301), WTC0(0xbcd1bb81), WTC0(0xbcd97c81), WTC0(0xbce14601), WTC0(0xbce91801), WTC0(0xbcf0f381), - WTC0(0xbcf8d781), WTC0(0xbd00c381), WTC0(0xbd08b781), WTC0(0xbd10b381), WTC0(0xbd18b781), WTC0(0xbd20c401), WTC0(0xbd28d981), WTC0(0xbd30f881), - WTC0(0xbd391f81), WTC0(0xbd414f01), WTC0(0xbd498601), WTC0(0xbd51c481), WTC0(0xbd5a0b01), WTC0(0xbd625981), WTC0(0xbd6ab101), WTC0(0xbd731081), - WTC0(0xbd7b7781), WTC0(0xbd83e681), WTC0(0xbd8c5c01), WTC0(0xbd94d801), WTC0(0xbd9d5b81), WTC0(0xbda5e601), WTC0(0xbdae7881), WTC0(0xbdb71201), - WTC0(0xbdbfb281), WTC0(0xbdc85981), WTC0(0xbdd10681), WTC0(0xbdd9b981), WTC0(0xbde27201), WTC0(0xbdeb3101), WTC0(0xbdf3f701), WTC0(0xbdfcc301), - WTC0(0xbe059481), WTC0(0xbe0e6c01), WTC0(0xbe174781), WTC0(0xbe202801), WTC0(0xbe290d01), WTC0(0xbe31f701), WTC0(0xbe3ae601), WTC0(0xbe43da81), - WTC0(0xbe4cd381), WTC0(0xbe55d001), WTC0(0xbe5ed081), WTC0(0xbe67d381), WTC0(0xbe70da01), WTC0(0xbe79e481), WTC0(0xbe82f301), WTC0(0xbe8c0501), - WTC0(0xbe951a81), WTC0(0xbe9e3281), WTC0(0xbea74c81), WTC0(0xbeb06881), WTC0(0xbeb98681), WTC0(0xbec2a781), WTC0(0xbecbca81), WTC0(0xbed4f081), - WTC0(0xbede1901), WTC0(0xbee74281), WTC0(0xbef06d01), WTC0(0xbef99901), WTC0(0xbf02c581), WTC0(0xbf0bf381), WTC0(0xbf152381), WTC0(0xbf1e5501), - WTC0(0xbf278801), WTC0(0xbf30bb01), WTC0(0xbf39ee81), WTC0(0xbf432281), WTC0(0xbf4c5681), WTC0(0xbf558b01), WTC0(0xbf5ec101), WTC0(0xbf67f801), - WTC0(0xbf712f01), WTC0(0xbf7a6681), WTC0(0xbf839d81), WTC0(0xbf8cd481), WTC0(0xbf960b01), WTC0(0xbf9f4181), WTC0(0xbfa87901), WTC0(0xbfb1b101), - WTC0(0xbfbae981), WTC0(0xbfc42201), WTC0(0xbfcd5a01), WTC0(0xbfd69101), WTC0(0xbfdfc781), WTC0(0xbfe8fc01), WTC0(0xbff22f81), WTC0(0xbffb6081), - /* part 1 */ - WTC1(0x80093e01), WTC1(0x801b9b01), WTC1(0x802df701), WTC1(0x80405101), WTC1(0x8052a881), WTC1(0x8064fc81), WTC1(0x80774c81), WTC1(0x80899881), - WTC1(0x809bdf01), WTC1(0x80ae1f81), WTC1(0x80c05a01), WTC1(0x80d28d81), WTC1(0x80e4bb81), WTC1(0x80f6e481), WTC1(0x81090981), WTC1(0x811b2981), - WTC1(0x812d4481), WTC1(0x813f5981), WTC1(0x81516701), WTC1(0x81636d81), WTC1(0x81756d81), WTC1(0x81876781), WTC1(0x81995c01), WTC1(0x81ab4b01), - WTC1(0x81bd3401), WTC1(0x81cf1581), WTC1(0x81e0ee81), WTC1(0x81f2bf81), WTC1(0x82048881), WTC1(0x82164a81), WTC1(0x82280581), WTC1(0x8239b981), - WTC1(0x824b6601), WTC1(0x825d0901), WTC1(0x826ea201), WTC1(0x82803101), WTC1(0x8291b601), WTC1(0x82a33281), WTC1(0x82b4a601), WTC1(0x82c61101), - WTC1(0x82d77201), WTC1(0x82e8c801), WTC1(0x82fa1181), WTC1(0x830b4f81), WTC1(0x831c8101), WTC1(0x832da781), WTC1(0x833ec381), WTC1(0x834fd481), - WTC1(0x8360d901), WTC1(0x8371d081), WTC1(0x8382ba01), WTC1(0x83939501), WTC1(0x83a46181), WTC1(0x83b52101), WTC1(0x83c5d381), WTC1(0x83d67881), - WTC1(0x83e70f01), WTC1(0x83f79681), WTC1(0x84080d81), WTC1(0x84187401), WTC1(0x8428ca01), WTC1(0x84391081), WTC1(0x84494881), WTC1(0x84597081), - WTC1(0x84698881), WTC1(0x84798f81), WTC1(0x84898481), WTC1(0x84996701), WTC1(0x84a93801), WTC1(0x84b8f801), WTC1(0x84c8a701), WTC1(0x84d84601), - WTC1(0x84e7d381), WTC1(0x84f74e01), WTC1(0x8506b581), WTC1(0x85160981), WTC1(0x85254a81), WTC1(0x85347901), WTC1(0x85439601), WTC1(0x8552a181), - WTC1(0x85619a01), WTC1(0x85707f81), WTC1(0x857f5101), WTC1(0x858e0e01), WTC1(0x859cb781), WTC1(0x85ab4f01), WTC1(0x85b9d481), WTC1(0x85c84801), - WTC1(0x85d6a981), WTC1(0x85e4f801), WTC1(0x85f33281), WTC1(0x86015981), WTC1(0x860f6e01), WTC1(0x861d7081), WTC1(0x862b6201), WTC1(0x86394301), - WTC1(0x86471281), WTC1(0x8654d001), WTC1(0x86627b01), WTC1(0x86701381), WTC1(0x867d9a81), WTC1(0x868b1001), WTC1(0x86987581), WTC1(0x86a5ca81), - WTC1(0x86b30f01), WTC1(0x86c04381), WTC1(0x86cd6681), WTC1(0x86da7901), WTC1(0x86e77b81), WTC1(0x86f46d81), WTC1(0x87014f81), WTC1(0x870e2301), - WTC1(0x871ae981), WTC1(0x8727a381), WTC1(0x87345381), WTC1(0x8740f681), WTC1(0x874d8681), WTC1(0x8759fd01), WTC1(0x87665481), WTC1(0x87729701), - WTC1(0x877ede01), WTC1(0x878b4301), WTC1(0x8797dd81), WTC1(0x87a48b01), WTC1(0x87b0ef01), WTC1(0x87bcab81), WTC1(0x87c76201), WTC1(0x87d0ca81), - WTC1(0x87fdd781), WTC1(0x881dd301), WTC1(0x88423301), WTC1(0x886a8a81), WTC1(0x88962981), WTC1(0x88c45e81), WTC1(0x88f47901), WTC1(0x8925f101), - WTC1(0x89586901), WTC1(0x898b8301), WTC1(0x89bee581), WTC1(0x89f26101), WTC1(0x8a25f301), WTC1(0x8a599a81), WTC1(0x8a8d5801), WTC1(0x8ac13381), - WTC1(0x8af53e81), WTC1(0x8b298b81), WTC1(0x8b5e2c81), WTC1(0x8b933001), WTC1(0x8bc8a401), WTC1(0x8bfe9401), WTC1(0x8c350d01), WTC1(0x8c6c1b01), - WTC1(0x8ca3cb01), WTC1(0x8cdc2901), WTC1(0x8d154081), WTC1(0x8d4f1b01), WTC1(0x8d89be81), WTC1(0x8dc53001), WTC1(0x8e017581), WTC1(0x8e3e9481), - WTC1(0x8e7c9301), WTC1(0x8ebb7581), WTC1(0x8efb4181), WTC1(0x8f3bfb01), WTC1(0x8f7da401), WTC1(0x8fc03f01), WTC1(0x9003ce81), WTC1(0x90485401), - WTC1(0x908dd101), WTC1(0x90d44781), WTC1(0x911bb981), WTC1(0x91642781), WTC1(0x91ad9281), WTC1(0x91f7f981), WTC1(0x92435d01), WTC1(0x928fbe01), - WTC1(0x92dd1b01), WTC1(0x932b7501), WTC1(0x937acb01), WTC1(0x93cb1c81), WTC1(0x941c6901), WTC1(0x946eaf81), WTC1(0x94c1ee01), WTC1(0x95162381), - WTC1(0x956b4f81), WTC1(0x95c17081), WTC1(0x96188501), WTC1(0x96708b81), WTC1(0x96c98381), WTC1(0x97236b01), WTC1(0x977e4181), WTC1(0x97da0481), - WTC1(0x9836b201), WTC1(0x98944901), WTC1(0x98f2c601), WTC1(0x99522801), WTC1(0x99b26c81), WTC1(0x9a139101), WTC1(0x9a759301), WTC1(0x9ad87081), - WTC1(0x9b3c2801), WTC1(0x9ba0b701), WTC1(0x9c061b81), WTC1(0x9c6c5481), WTC1(0x9cd35f81), WTC1(0x9d3b3b81), WTC1(0x9da3e601), WTC1(0x9e0d5e01), - WTC1(0x9e779f81), WTC1(0x9ee2a901), WTC1(0x9f4e7801), WTC1(0x9fbb0981), WTC1(0xa0285d81), WTC1(0xa0967201), WTC1(0xa1054701), WTC1(0xa174da81), - WTC1(0xa1e52a81), WTC1(0xa2563501), WTC1(0xa2c7f801), WTC1(0xa33a7201), WTC1(0xa3ada281), WTC1(0xa4218801), WTC1(0xa4962181), WTC1(0xa50b6e81), - WTC1(0xa5816e81), WTC1(0xa5f81f81), WTC1(0xa66f8201), WTC1(0xa6e79401), WTC1(0xa7605601), WTC1(0xa7d9c681), WTC1(0xa853e501), WTC1(0xa8ceb201), - WTC1(0xa94a2c01), WTC1(0xa9c65401), WTC1(0xaa432981), WTC1(0xaac0ad01), WTC1(0xab3edf01), WTC1(0xabbdc001), WTC1(0xac3d5001), WTC1(0xacbd9081), - WTC1(0xad3e8101), WTC1(0xadc02281), WTC1(0xae427481), WTC1(0xaec57801), WTC1(0xaf492f01), WTC1(0xafcd9a81), WTC1(0xb052bc01), WTC1(0xb0d89401), - WTC1(0xb15f2381), WTC1(0xb1e66a01), WTC1(0xb26e6881), WTC1(0xb2f71f01), WTC1(0xb3808d81), WTC1(0xb40ab501), WTC1(0xb4959501), WTC1(0xb5212e81), - WTC1(0x4a6cf67f), WTC1(0x49dffeff), WTC1(0x495265ff), WTC1(0x48c4277f), WTC1(0x4835407f), WTC1(0x47a5aeff), WTC1(0x471570ff), WTC1(0x468484ff), - WTC1(0x45f2eaff), WTC1(0x4560a2ff), WTC1(0x44cdad7f), WTC1(0x443a0c7f), WTC1(0x43a5c07f), WTC1(0x4310caff), WTC1(0x427b2bff), WTC1(0x41e4e3ff), - WTC1(0x414df2ff), WTC1(0x40b6557f), WTC1(0x401e06ff), WTC1(0x3f8503c0), WTC1(0x3eeb4e00), WTC1(0x3e50ebc0), WTC1(0x3db5e680), WTC1(0x3d1a4680), - WTC1(0x3c7e10c0), WTC1(0x3be14cc0), WTC1(0x3b4402c0), WTC1(0x3aa63800), WTC1(0x3a07e840), WTC1(0x39690880), WTC1(0x38c98700), WTC1(0x38295b40), - WTC1(0x37888a80), WTC1(0x36e71d40), WTC1(0x36451d80), WTC1(0x35a29400), WTC1(0x34ff8800), WTC1(0x345c04c0), WTC1(0x33b81940), WTC1(0x3313d200), - WTC1(0x326f3800), WTC1(0x31ca5600), WTC1(0x31253840), WTC1(0x307fe8c0), WTC1(0x2fda6e40), WTC1(0x2f34ce40), WTC1(0x2e8f0e40), WTC1(0x2de92ec0), - WTC1(0x2d432780), WTC1(0x2c9cea40), WTC1(0x2bf66300), WTC1(0x2b4f88c0), WTC1(0x2aa864c0), WTC1(0x2a010240), WTC1(0x29596e40), WTC1(0x28b1ba80), - WTC1(0x2809ff40), WTC1(0x27625b80), WTC1(0x26baf580), WTC1(0x2613e7c0), WTC1(0x256d3dc0), WTC1(0x24c70300), WTC1(0x24214380), WTC1(0x237c0800), - WTC1(0x22d75400), WTC1(0x22332a80), WTC1(0x218f8cc0), WTC1(0x20ec7e40), WTC1(0x204a04c0), WTC1(0x1fa82540), WTC1(0x1f06e300), WTC1(0x1e664000), - WTC1(0x1dc63bc0), WTC1(0x1d26d3c0), WTC1(0x1c8803a0), WTC1(0x1be9cc40), WTC1(0x1b4c34c0), WTC1(0x1aaf4480), WTC1(0x1a130260), WTC1(0x197774a0), - WTC1(0x18dca260), WTC1(0x184294e0), WTC1(0x17a95840), WTC1(0x1710fd80), WTC1(0x16799ce0), WTC1(0x15e35340), WTC1(0x154e41a0), WTC1(0x14ba8360), - WTC1(0x14282be0), WTC1(0x13975100), WTC1(0x13080aa0), WTC1(0x127a6240), WTC1(0x11ee50a0), WTC1(0x1163cc80), WTC1(0x10dacb20), WTC1(0x105333a0), - WTC1(0x0fccdb30), WTC1(0x0f478f40), WTC1(0x0ec31700), WTC1(0x0e3f4e80), WTC1(0x0dbc27f0), WTC1(0x0d399000), WTC1(0x0cb76d00), WTC1(0x0c359d50), - WTC1(0x0bb3fd50), WTC1(0x0b326bd0), WTC1(0x0ab0ca80), WTC1(0x0a2f0dc0), WTC1(0x09ad40c0), WTC1(0x092b7a90), WTC1(0x08a9db80), WTC1(0x08285c80), - WTC1(0x07a6c7b8), WTC1(0x0724e4e0), WTC1(0x06a27b80), WTC1(0x061f52f8), WTC1(0x059b2ad0), WTC1(0x0515b568), WTC1(0x048ea058), WTC1(0x04066408), - WTC1(0x037e52d8), WTC1(0x02f7d3c8), WTC1(0x0274614c), WTC1(0x01f63008), WTC1(0x0180403a), WTC1(0x0115c442), WTC1(0x00ba09e2), WTC1(0x006f077c), - WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), - WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), - WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), - WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), - WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), - WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), - WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), - WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), - WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), - WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), - WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), - WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), - WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), - WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), - WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), - WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), - /* part 2 */ - WTC2(0xfff36be1), WTC2(0xffdafbc1), WTC2(0xffc28035), WTC2(0xffa9fe8a), WTC2(0xff917c08), WTC2(0xff78fdfc), WTC2(0xff6089af), WTC2(0xff48246c), - WTC2(0xff2fd37f), WTC2(0xff179c31), WTC2(0xfeff83b6), WTC2(0xfee78d18), WTC2(0xfecfb93e), WTC2(0xfeb808f2), WTC2(0xfea07d06), WTC2(0xfe8916b4), - WTC2(0xfe71d7a0), WTC2(0xfe5ac174), WTC2(0xfe43d5d6), WTC2(0xfe2d167e), WTC2(0xfe16852e), WTC2(0xfe0023a6), WTC2(0xfde9f3f8), WTC2(0xfdd3ff7c), - WTC2(0xfdbe56c0), WTC2(0xfda90aa8), WTC2(0xfd942b78), WTC2(0xfd7fbb20), WTC2(0xfd6bad50), WTC2(0xfd57f510), WTC2(0xfd44857c), WTC2(0xfd3153fc), - WTC2(0xfd1e5840), WTC2(0xfd0b8a0c), WTC2(0xfcf8e180), WTC2(0xfce65eec), WTC2(0xfcd40ad0), WTC2(0xfcc1ee0c), WTC2(0xfcb011e8), WTC2(0xfc9e896c), - WTC2(0xfc8d716c), WTC2(0xfc7ce720), WTC2(0xfc6d072c), WTC2(0xfc5de09c), WTC2(0xfc4f74e8), WTC2(0xfc41c4e8), WTC2(0xfc34d0dc), WTC2(0xfc288a68), - WTC2(0xfc1cd49c), WTC2(0xfc1191e0), WTC2(0xfc06a4d0), WTC2(0xfbfbf3e8), WTC2(0xfbf16990), WTC2(0xfbe6f068), WTC2(0xfbdc7428), WTC2(0xfbd1fc68), - WTC2(0xfbc7ac50), WTC2(0xfbbda868), WTC2(0xfbb41500), WTC2(0xfbab1438), WTC2(0xfba2c5f8), WTC2(0xfb9b4a00), WTC2(0xfb94bfa8), WTC2(0xfb8f3b48), - WTC2(0xfb8ac638), WTC2(0xfb876970), WTC2(0xfb852d20), WTC2(0xfb840ae0), WTC2(0xfb83ed60), WTC2(0xfb84bec0), WTC2(0xfb866918), WTC2(0xfb88d4a8), - WTC2(0xfb8be810), WTC2(0xfb8f89d0), WTC2(0xfb93a080), WTC2(0xfb981418), WTC2(0xfb9ccdf0), WTC2(0xfba1b770), WTC2(0xfba6bae0), WTC2(0xfbabd5c0), - WTC2(0xfbb118d8), WTC2(0xfbb695c0), WTC2(0xfbbc5e90), WTC2(0xfbc29030), WTC2(0xfbc95268), WTC2(0xfbd0cd78), WTC2(0xfbd929c8), WTC2(0xfbe294d0), - WTC2(0xfbed4108), WTC2(0xfbf96118), WTC2(0xfc0726c8), WTC2(0xfc16b064), WTC2(0xfc280890), WTC2(0xfc3b3920), WTC2(0xfc504a98), WTC2(0xfc67271c), - WTC2(0xfc7f9a74), WTC2(0xfc996f18), WTC2(0xfcb46eb8), WTC2(0xfcd050b0), WTC2(0xfcecba24), WTC2(0xfd094f64), WTC2(0xfd25b720), WTC2(0xfd41ce40), - WTC2(0xfd5da7f8), WTC2(0xfd7959d8), WTC2(0xfd94fb74), WTC2(0xfdb0d3fc), WTC2(0xfdcd5a34), WTC2(0xfdeb06e4), WTC2(0xfe0a5184), WTC2(0xfe2b92c4), - WTC2(0xfe4f0486), WTC2(0xfe74df54), WTC2(0xfe9d5886), WTC2(0xfec85b92), WTC2(0xfef58a16), WTC2(0xff248275), WTC2(0xff54e401), WTC2(0xff866330), - WTC2(0xffb8c99b), WTC2(0xffebe1c9), WTC2(0x001f786a), WTC2(0x00538bf9), WTC2(0x00884cbc), WTC2(0x00bded23), WTC2(0x00f49f54), WTC2(0x012c8ee4), - WTC2(0x0165e0d2), WTC2(0x01a0b9d6), WTC2(0x01dd3d80), WTC2(0x021b74d4), WTC2(0x025b4e48), WTC2(0x029cb730), WTC2(0x02df9d0c), WTC2(0x0323f1a4), - WTC2(0x0369ab00), WTC2(0x03b0bf5c), WTC2(0x03f925a0), WTC2(0x0442e3d8), WTC2(0x048e0f40), WTC2(0x04dabdb0), WTC2(0x05290430), WTC2(0x0578e428), - WTC2(0x05ca4b60), WTC2(0x061d26c0), WTC2(0x067163d8), WTC2(0x06c6ff10), WTC2(0x071e03b0), WTC2(0x07767da0), WTC2(0x07d07918), WTC2(0x082c08e0), - WTC2(0x08894660), WTC2(0x08e84b70), WTC2(0x094930b0), WTC2(0x09abf8d0), WTC2(0x0a109020), WTC2(0x0a76e210), WTC2(0x0adeda50), WTC2(0x0b486b80), - WTC2(0x0bb38f00), WTC2(0x0c203e80), WTC2(0x0c8e73e0), WTC2(0x0cfe2c30), WTC2(0x0d6f6820), WTC2(0x0de22850), WTC2(0x0e566d90), WTC2(0x0ecc3dd0), - WTC2(0x0f43a3a0), WTC2(0x0fbca9f0), WTC2(0x10375b80), WTC2(0x10b3be20), WTC2(0x1131d280), WTC2(0x11b19960), WTC2(0x123313a0), WTC2(0x12b64380), - WTC2(0x133b2d00), WTC2(0x13c1d440), WTC2(0x144a3d60), WTC2(0x14d46900), WTC2(0x15605480), WTC2(0x15edfd20), WTC2(0x167d6040), WTC2(0x170e7e80), - WTC2(0x17a15b80), WTC2(0x1835fb00), WTC2(0x18cc60a0), WTC2(0x19648dc0), WTC2(0x19fe80e0), WTC2(0x1a9a38a0), WTC2(0x1b37b3e0), WTC2(0x1bd6f400), - WTC2(0x1c77fd20), WTC2(0x1d1ad400), WTC2(0x1dbf7c80), WTC2(0x1e65f820), WTC2(0x1f0e4540), WTC2(0x1fb861e0), WTC2(0x20644cc0), WTC2(0x21120640), - WTC2(0x21c19240), WTC2(0x2272f480), WTC2(0x23263000), WTC2(0x23db4580), WTC2(0x24923340), WTC2(0x254af700), WTC2(0x26058e80), WTC2(0x26c1fa00), - WTC2(0x27803d00), WTC2(0x28405a40), WTC2(0x29025500), WTC2(0x29c62d40), WTC2(0x2a8be0c0), WTC2(0x2b536cc0), WTC2(0x2c1ccf80), WTC2(0x2ce80840), - WTC2(0x2db519c0), WTC2(0x2e840600), WTC2(0x2f54cf80), WTC2(0x302775c0), WTC2(0x30fbf640), WTC2(0x31d24e00), WTC2(0x32aa7a00), WTC2(0x338479c0), - WTC2(0x34604e40), WTC2(0x353df900), WTC2(0x361d7ac0), WTC2(0x36fed200), WTC2(0x37e1fb40), WTC2(0x38c6f240), WTC2(0x39adb2c0), WTC2(0x3a963a00), - WTC2(0x3b808740), WTC2(0x3c6c9880), WTC2(0x3d5a6cc0), WTC2(0x3e4a0040), WTC2(0x3f3b4bc0), WTC2(0x402e48ff), WTC2(0x4122f17f), WTC2(0x42193f7f), - WTC2(0x43112eff), WTC2(0x440abbff), WTC2(0x4505e2ff), WTC2(0x46029e7f), WTC2(0x4700e9ff), WTC2(0x4800bfff), WTC2(0x49021bff), WTC2(0x4a050eff), - WTC2(0x4b09bc7f), WTC2(0x4c104aff), WTC2(0x4d18df7f), WTC2(0x4e23a07f), WTC2(0x4f30b2ff), WTC2(0x50403c7f), WTC2(0x515262ff), WTC2(0x52674b7f), - WTC2(0x001678b2), WTC2(0x00061a3b), WTC2(0xfffb4622), WTC2(0xfff5ea94), WTC2(0xfff5f5b9), WTC2(0xfffb55bd), WTC2(0x0005f8cb), WTC2(0x0015cd0c), - WTC2(0x002ac0ac), WTC2(0x0044c1d5), WTC2(0x0063beb2), WTC2(0x0087a56d), WTC2(0x00b06431), WTC2(0x00dde929), WTC2(0x01102280), WTC2(0x0146fe5e), - WTC2(0x01826af2), WTC2(0x01c25662), WTC2(0x0206aedc), WTC2(0x024f6288), WTC2(0x029c5f94), WTC2(0x02ed9424), WTC2(0x0342ee6c), WTC2(0x039c5c90), - WTC2(0x03f9ccbc), WTC2(0x045b2d18), WTC2(0x04c06bd8), WTC2(0x05297718), WTC2(0x05963d10), WTC2(0x0606abe8), WTC2(0x067ab1c0), WTC2(0x06f23cd0), - WTC2(0x076d3b40), WTC2(0x07eb9b38), WTC2(0x086d4ae0), WTC2(0x08f23860), WTC2(0x097a51f0), WTC2(0x0a0585b0), WTC2(0x0a93c1d0), WTC2(0x0b24f470), - WTC2(0x0bb90bc0), WTC2(0x0c4ff5f0), WTC2(0x0ce9a130), WTC2(0x0d85fb90), WTC2(0x0e24f360), WTC2(0x0ec676b0), WTC2(0x0f6a73b0), WTC2(0x1010d880), - WTC2(0x10b99360), WTC2(0x11649280), WTC2(0x1211c400), WTC2(0x12c115e0), WTC2(0x137276a0), WTC2(0x1425d420), WTC2(0x14db1ca0), WTC2(0x15923e60), - WTC2(0x164b2780), WTC2(0x1705c620), WTC2(0x17c20860), WTC2(0x187fdca0), WTC2(0x193f30e0), WTC2(0x19fff340), WTC2(0x1ac21200), WTC2(0x1b857b40), - WTC2(0x1c4a1d40), WTC2(0x1d0fe600), WTC2(0x1dd6c3e0), WTC2(0x1e9ea4e0), WTC2(0x1f677740), WTC2(0x20312940), WTC2(0x20fba8c0), WTC2(0x21c6e440), - WTC2(0x2292c9c0), WTC2(0x235f4780), WTC2(0x242c4b80), WTC2(0x24f9c400), WTC2(0x25c79f40), WTC2(0x2695cb40), WTC2(0x27643680), WTC2(0x2832cec0), - WTC2(0x29018240), WTC2(0x29d03f80), WTC2(0x2a9ef480), WTC2(0x2b6d8f00), WTC2(0x2c3bfdc0), WTC2(0x2d0a2ec0), WTC2(0x2dd81000), WTC2(0x2ea58fc0), - WTC2(0x2f729c40), WTC2(0x303f2380), WTC2(0x310b1400), WTC2(0x31d65b80), WTC2(0x32a0e840), WTC2(0x336aa8c0), WTC2(0x34338ac0), WTC2(0x34fb7cc0), - WTC2(0x35c26cc0), WTC2(0x36884900), WTC2(0x374cff80), WTC2(0x38107e80), WTC2(0x38d2b440), WTC2(0x39938ec0), WTC2(0x3a52fc40), WTC2(0x3b10eb00), - WTC2(0x3bcd4900), WTC2(0x3c880480), WTC2(0x3d410bc0), WTC2(0x3df84d00), WTC2(0x3eadb600), WTC2(0x3f613540), WTC2(0x4012b8ff), WTC2(0x40c22eff), - WTC2(0x416f85ff), WTC2(0x421aab7f), WTC2(0x42c38e7f), WTC2(0x436a1c7f), WTC2(0x440e437f), WTC2(0x44aff27f), WTC2(0x454f167f), WTC2(0x45eb9eff), - WTC2(0x468578ff), WTC2(0x471c937f), WTC2(0x47b0dc7f), WTC2(0x484241ff), WTC2(0x48d0b1ff), WTC2(0x495c1a7f), WTC2(0x49e46a7f), WTC2(0x4a698f7f), - WTC2(0x4aeb77ff), WTC2(0x4b6a11ff), WTC2(0x4be54b7f), WTC2(0x4c5d12ff), WTC2(0x4cd155ff), WTC2(0x4d4203ff), WTC2(0x4daf09ff), WTC2(0x4e18567f), - WTC2(0x4e7dd77f), WTC2(0x4edf7b7f), WTC2(0x4f3d307f), WTC2(0x4f96e47f), WTC2(0x4fec85ff), WTC2(0x503e02ff), WTC2(0x508b497f), WTC2(0x50d447ff), - WTC2(0x5118ec7f), WTC2(0x515924ff), WTC2(0x5194dfff), WTC2(0x51cc0b7f), WTC2(0x51fe95ff), WTC2(0x522c6cff), WTC2(0x52557eff), WTC2(0x5279b9ff), - WTC2(0x52990c7f), WTC2(0x52b364ff), WTC2(0x52c8b07f), WTC2(0x52d8ddff), WTC2(0x52e3db7f), WTC2(0x52e996ff), WTC2(0x52e9ff7f), WTC2(0x52e501ff), - WTC2(0x52da8cff), WTC2(0x52ca8f7f), WTC2(0x52b4f67f), WTC2(0x5299b07f), WTC2(0x5278ac7f), WTC2(0x5251d77f), WTC2(0x52251fff), WTC2(0x51f274ff), - WTC2(0x51b9c37f), WTC2(0x517af9ff), WTC2(0x5136077f), WTC2(0x50ead8ff), WTC2(0x50995cff), WTC2(0x504181ff), WTC2(0x4fe335ff), WTC2(0x4f7e677f), - WTC2(0x4f1303ff), WTC2(0x4ea0f9ff), WTC2(0x4e2837ff), WTC2(0x4da8ab7f), WTC2(0x4d2242ff), WTC2(0x4c94ecff), WTC2(0x4c0096ff), WTC2(0x4b652f7f), - WTC2(0x4ac2a4ff), WTC2(0x4a18e4ff), WTC2(0x4967ddff), WTC2(0x48af7e7f), WTC2(0x47efb3ff), WTC2(0x47286cff), WTC2(0x4659ad7f), WTC2(0x45856f7f), - WTC2(0x44afa3ff), WTC2(0x43dc507f), WTC2(0x430f657f), WTC2(0x424ad47f), WTC2(0x418e927f), WTC2(0x40da7bff), WTC2(0x402e6f7f), WTC2(0x3f8a3100), - WTC2(0x3eed6f40), WTC2(0x3e57d700), WTC2(0x3dc914c0), WTC2(0x3d40cc40), WTC2(0x3cbe98c0), WTC2(0x3c421540), WTC2(0x3bcadbc0), WTC2(0x3b588880), - WTC2(0x3aeab780), WTC2(0x3a810540), WTC2(0x3a1b0e00), WTC2(0x39b86d00), WTC2(0x3958bcc0), WTC2(0x38fb9700), WTC2(0x38a095c0), WTC2(0x38473d80), - WTC2(0x37eeff40), WTC2(0x37974b40), WTC2(0x373f9500), WTC2(0x36e7ae00), WTC2(0x368fc4c0), WTC2(0x36380b80), WTC2(0x35e0b300), WTC2(0x3589c140), - WTC2(0x35331180), WTC2(0x34dc7c80), WTC2(0x3485dc80), WTC2(0x342f1600), WTC2(0x33d81780), WTC2(0x3380d0c0), WTC2(0x33293100), WTC2(0x32d11800), - WTC2(0x32785780), WTC2(0x321ec0c0), WTC2(0x31c42680), WTC2(0x316885c0), WTC2(0x310c0580), WTC2(0x30aecec0), WTC2(0x30510940), WTC2(0x2ff2b8c0), - WTC2(0x2f93bf40), WTC2(0x2f33fc00), WTC2(0x2ed350c0), WTC2(0x2e71ba80), WTC2(0x2e0f5340), WTC2(0x2dac35c0), WTC2(0x2d487c80), WTC2(0x2ce431c0), - WTC2(0x2c7f4fc0), WTC2(0x2c19d080), WTC2(0x2bb3ad80), WTC2(0x2b4ce080), WTC2(0x2ae56340), WTC2(0x2a7d2f80), WTC2(0x2a143f00), WTC2(0x29aa8b40) -}; - -const FIXP_WTB ELDAnalysis480[1440] = { - WTC0(0xfacfbef0), WTC0(0xfab88c18), WTC0(0xfaa0e520), WTC0(0xfa88d110), WTC0(0xfa7056e8), WTC0(0xfa577db0), WTC0(0xfa3e4c70), WTC0(0xfa24ca28), - WTC0(0xfa0afde0), WTC0(0xf9f0eea0), WTC0(0xf9d6a2c8), WTC0(0xf9bc1ab8), WTC0(0xf9a15230), WTC0(0xf9864510), WTC0(0xf96af058), WTC0(0xf94f55c0), - WTC0(0xf93378e0), WTC0(0xf9175d80), WTC0(0xf8fb0468), WTC0(0xf8de68b8), WTC0(0xf8c18438), WTC0(0xf8a450d8), WTC0(0xf886cde8), WTC0(0xf8690148), - WTC0(0xf84af148), WTC0(0xf82ca410), WTC0(0xf80e1e18), WTC0(0xf7ef62a0), WTC0(0xf7d074e0), WTC0(0xf7b15870), WTC0(0xf7921240), WTC0(0xf772a7a0), - WTC0(0xf7531e50), WTC0(0xf7337820), WTC0(0xf713afd0), WTC0(0xf6f3bea0), WTC0(0xf6d39dc0), WTC0(0xf6b352e0), WTC0(0xf692f280), WTC0(0xf6729250), - WTC0(0xf65247a0), WTC0(0xf63224c0), WTC0(0xf6123a00), WTC0(0xf5f297c0), WTC0(0xf5d34dd0), WTC0(0xf5b46b10), WTC0(0xf595fd90), WTC0(0xf5781390), - WTC0(0xf55abba0), WTC0(0xf53e0510), WTC0(0xf521ff70), WTC0(0xf506ba30), WTC0(0xf4ec4330), WTC0(0xf4d2a680), WTC0(0xf4b9efe0), WTC0(0xf4a22ac0), - WTC0(0xf48b5f70), WTC0(0xf4759310), WTC0(0xf460cde0), WTC0(0xf44cfcc0), WTC0(0xf439aff0), WTC0(0xf4264e00), WTC0(0xf4123d90), WTC0(0xf3fd1370), - WTC0(0xf3e6be00), WTC0(0xf3cf41a0), WTC0(0xf3b6a030), WTC0(0xf39cdd60), WTC0(0xf381fe00), WTC0(0xf3660760), WTC0(0xf348fe70), WTC0(0xf32ae820), - WTC0(0xf30bc940), WTC0(0xf2eba690), WTC0(0xf2ca8480), WTC0(0xf2a86670), WTC0(0xf2854f40), WTC0(0xf2614190), WTC0(0xf23c41e0), WTC0(0xf21657a0), - WTC0(0xf1ef8ae0), WTC0(0xf1c7e3e0), WTC0(0xf19f63d0), WTC0(0xf1760450), WTC0(0xf14bbdf0), WTC0(0xf1208960), WTC0(0xf0f45cd0), WTC0(0xf0c72ce0), - WTC0(0xf098ee00), WTC0(0xf06996f0), WTC0(0xf0392620), WTC0(0xf0079e10), WTC0(0xefd4ffc0), WTC0(0xefa15ca0), WTC0(0xef6ce600), WTC0(0xef37d460), - WTC0(0xef025f80), WTC0(0xeecca2c0), WTC0(0xee969760), WTC0(0xee603440), WTC0(0xee296d20), WTC0(0xedf21c00), WTC0(0xedba07e0), WTC0(0xed80f640), - WTC0(0xed46bf40), WTC0(0xed0b7b00), WTC0(0xeccf5fc0), WTC0(0xec92a120), WTC0(0xec556d60), WTC0(0xec17e700), WTC0(0xebda2d40), WTC0(0xeb9c5fa0), - WTC0(0xeb5e7040), WTC0(0xeb201b20), WTC0(0xeae117c0), WTC0(0xeaa12000), WTC0(0xea600180), WTC0(0xea1d9940), WTC0(0xe9d9c160), WTC0(0xe99468a0), - WTC0(0xe94dc040), WTC0(0xe9061940), WTC0(0xe8bdc140), WTC0(0xe8750ae0), WTC0(0xe82c4fa0), WTC0(0xe7e3ea40), WTC0(0xe79c35e0), WTC0(0xe7554ca0), - WTC0(0xe70efc00), WTC0(0xe6c90c20), WTC0(0xe6833f00), WTC0(0xe63d2300), WTC0(0xe5f620a0), WTC0(0xe5ad9dc0), WTC0(0xe5632080), WTC0(0xe5169da0), - WTC0(0xe4c83e60), WTC0(0xe4782400), WTC0(0xe4269840), WTC0(0xe3d42dc0), WTC0(0xe38188c0), WTC0(0xe32f4be0), WTC0(0xe2ddeea0), WTC0(0xe28db520), - WTC0(0xe23ee000), WTC0(0xe1f1a580), WTC0(0xe1a5e3a0), WTC0(0xe15b35a0), WTC0(0xe1113860), WTC0(0xe0c78a00), WTC0(0xe07dd0e0), WTC0(0xe033b7c0), - WTC0(0xdfe8e680), WTC0(0xdf9d1fc0), WTC0(0xdf5055c0), WTC0(0xdf0287c0), WTC0(0xdeb3b340), WTC0(0xde63e7c0), WTC0(0xde134a00), WTC0(0xddc20000), - WTC0(0xdd703180), WTC0(0xdd1e1280), WTC0(0xdccbe080), WTC0(0xdc79d980), WTC0(0xdc283600), WTC0(0xdbd71e00), WTC0(0xdb86b140), WTC0(0xdb3710c0), - WTC0(0xdae850c0), WTC0(0xda9a6bc0), WTC0(0xda4d5640), WTC0(0xda010640), WTC0(0xd9b56640), WTC0(0xd96a5700), WTC0(0xd91fb700), WTC0(0xd8d56600), - WTC0(0xd88b4a40), WTC0(0xd8414f00), WTC0(0xd7f75f80), WTC0(0xd7ad6740), WTC0(0xd76352c0), WTC0(0xd7191040), WTC0(0xd6ce8c80), WTC0(0xd683bd00), - WTC0(0xd638a5c0), WTC0(0xd5ed4f80), WTC0(0xd5a1c240), WTC0(0xd5562b80), WTC0(0xd50ae500), WTC0(0xd4c04c80), WTC0(0xd476bb40), WTC0(0xd42e62c0), - WTC0(0xd3e75680), WTC0(0xd3a1ad00), WTC0(0xd35d6780), WTC0(0xd31a4300), WTC0(0xd2d7dc00), WTC0(0xd295d080), WTC0(0xd253d8c0), WTC0(0xd211df40), - WTC0(0xd1cfdbc0), WTC0(0xd18dc480), WTC0(0xd14b9dc0), WTC0(0xd1097c80), WTC0(0xd0c77700), WTC0(0xd085a500), WTC0(0xd0442f40), WTC0(0xd0034a80), - WTC0(0xcfc32c00), WTC0(0xcf840400), WTC0(0xcf45f400), WTC0(0xcf0913c0), WTC0(0xcecd8000), WTC0(0xce932c80), WTC0(0xce59bf40), WTC0(0xce20cd40), - WTC0(0xcde7ec40), WTC0(0xcdaeedc0), WTC0(0xcd75ea00), WTC0(0xcd3cfec0), WTC0(0xcd044b40), WTC0(0xcccbff00), WTC0(0xcc945480), WTC0(0xcc5d8780), - WTC0(0xcc27c3c0), WTC0(0xcbf2fc40), WTC0(0xcbbf0a00), WTC0(0xcb8bc7c0), WTC0(0xcb591880), WTC0(0xcb26f0c0), WTC0(0xcaf54980), WTC0(0xcac41ac0), - WTC0(0xca936440), WTC0(0xca632d80), WTC0(0xca337f00), WTC0(0xca046180), WTC0(0xc9d5dd40), WTC0(0xc9a7fa80), WTC0(0xc97ac200), WTC0(0xc94e3c00), - WTC0(0xc91d1840), WTC0(0xc8f15980), WTC0(0xc8c52340), WTC0(0xc8988100), WTC0(0xc86b7f00), WTC0(0xc83e28c0), WTC0(0xc8108a80), WTC0(0xc7e2afc0), - WTC0(0xc7b4a480), WTC0(0xc7867480), WTC0(0xc7582b40), WTC0(0xc729cc80), WTC0(0xc6fb5700), WTC0(0xc6ccca40), WTC0(0xc69e2180), WTC0(0xc66f49c0), - WTC0(0xc64029c0), WTC0(0xc610a740), WTC0(0xc5e0bfc0), WTC0(0xc5b09e80), WTC0(0xc5807900), WTC0(0xc5508440), WTC0(0xc520e840), WTC0(0xc4f1bdc0), - WTC0(0xc4c31d00), WTC0(0xc4951780), WTC0(0xc4678a00), WTC0(0xc43a28c0), WTC0(0xc40ca800), WTC0(0xc3deccc0), WTC0(0xc3b09940), WTC0(0xc3822c00), - WTC0(0xc353a0c0), WTC0(0xc3251740), WTC0(0xc2f6b500), WTC0(0xc2c8a140), WTC0(0xc29b02c0), WTC0(0xc26df5c0), WTC0(0xc2418940), WTC0(0xc215cbc0), - WTC0(0xc1eaca00), WTC0(0xc1c08680), WTC0(0xc196fb00), WTC0(0xc16e22c0), WTC0(0xc145f040), WTC0(0xc11e3a80), WTC0(0xc0f6cc00), WTC0(0xc0cf6ec0), - WTC0(0xc0a802c0), WTC0(0xc0809280), WTC0(0xc0593340), WTC0(0xc031f880), WTC0(0xc00b04c0), WTC0(0xbfe48981), WTC0(0xbfbebb81), WTC0(0xbf99cb01), - WTC0(0xbf75cc81), WTC0(0xbf52c101), WTC0(0xbf30a901), WTC0(0xbf0f8301), WTC0(0xbeef4601), WTC0(0xbecfe601), WTC0(0xbeb15701), WTC0(0xbe938c81), - WTC0(0xbe767e81), WTC0(0xbe5a2301), WTC0(0xbe3e7201), WTC0(0xbe236001), WTC0(0xbe08e181), WTC0(0xbdeee981), WTC0(0xbdd56b81), WTC0(0xbdbc6381), - WTC0(0xbda3d081), WTC0(0xbd8bb281), WTC0(0xbd740b81), WTC0(0xbd5ce281), WTC0(0xbd464281), WTC0(0xbd303581), WTC0(0xbd1ac801), WTC0(0xbd060c81), - WTC0(0xbcf21601), WTC0(0xbcdef701), WTC0(0xbcccbd01), WTC0(0xbcbb7001), WTC0(0xbcab1781), WTC0(0xbc9bb901), WTC0(0xbc8d5101), WTC0(0xbc7fd301), - WTC0(0xbc733401), WTC0(0xbc676501), WTC0(0xbc5c4c81), WTC0(0xbc51cb01), WTC0(0xbc47c281), WTC0(0xbc3e1981), WTC0(0xbc34c081), WTC0(0xbc2bab01), - WTC0(0xbc22cd81), WTC0(0xbc1a2401), WTC0(0xbc11b681), WTC0(0xbc098d81), WTC0(0xbc01b381), WTC0(0xbbfa3c01), WTC0(0xbbf34281), WTC0(0xbbece281), - WTC0(0xbbe73201), WTC0(0xbbe23281), WTC0(0xbbdddb01), WTC0(0xbbda2501), WTC0(0xbbd70201), WTC0(0xbbd45601), WTC0(0xbbd20301), WTC0(0xbbcfea81), - WTC0(0xbbce0601), WTC0(0xbbcc6b01), WTC0(0xbbcb3201), WTC0(0xbbca7481), WTC0(0xbbca5d01), WTC0(0xbbcb2281), WTC0(0xbbccfc81), WTC0(0xbbd01301), - WTC0(0xbbd45881), WTC0(0xbbd9a781), WTC0(0xbbdfdb81), WTC0(0xbbe6c801), WTC0(0xbbee2f81), WTC0(0xbbf5d181), WTC0(0xbbfd6c01), WTC0(0xbc04e381), - WTC0(0xbc0c4581), WTC0(0xbc13a481), WTC0(0xbc1b1081), WTC0(0xbc228f01), WTC0(0xbc2a1a81), WTC0(0xbc31af01), WTC0(0xbc394901), WTC0(0xbc40e881), - WTC0(0xbc488e81), WTC0(0xbc503b81), WTC0(0xbc57f101), WTC0(0xbc5fae81), WTC0(0xbc677501), WTC0(0xbc6f4401), WTC0(0xbc771c01), WTC0(0xbc7efc81), - WTC0(0xbc86e581), WTC0(0xbc8ed701), WTC0(0xbc96d101), WTC0(0xbc9ed481), WTC0(0xbca6e101), WTC0(0xbcaef701), WTC0(0xbcb71701), WTC0(0xbcbf4001), - WTC0(0xbcc77181), WTC0(0xbccfac01), WTC0(0xbcd7ef01), WTC0(0xbce03b81), WTC0(0xbce89281), WTC0(0xbcf0f381), WTC0(0xbcf95e81), WTC0(0xbd01d281), - WTC0(0xbd0a4f81), WTC0(0xbd12d581), WTC0(0xbd1b6501), WTC0(0xbd23ff01), WTC0(0xbd2ca281), WTC0(0xbd355081), WTC0(0xbd3e0801), WTC0(0xbd46c801), - WTC0(0xbd4f9101), WTC0(0xbd586281), WTC0(0xbd613d81), WTC0(0xbd6a2201), WTC0(0xbd731081), WTC0(0xbd7c0781), WTC0(0xbd850701), WTC0(0xbd8e0e01), - WTC0(0xbd971c81), WTC0(0xbda03381), WTC0(0xbda95301), WTC0(0xbdb27b01), WTC0(0xbdbbab01), WTC0(0xbdc4e301), WTC0(0xbdce2181), WTC0(0xbdd76701), - WTC0(0xbde0b301), WTC0(0xbdea0681), WTC0(0xbdf36101), WTC0(0xbdfcc301), WTC0(0xbe062b81), WTC0(0xbe0f9a01), WTC0(0xbe190d81), WTC0(0xbe228681), - WTC0(0xbe2c0501), WTC0(0xbe358901), WTC0(0xbe3f1381), WTC0(0xbe48a301), WTC0(0xbe523781), WTC0(0xbe5bd001), WTC0(0xbe656c01), WTC0(0xbe6f0c01), - WTC0(0xbe78b001), WTC0(0xbe825801), WTC0(0xbe8c0501), WTC0(0xbe95b581), WTC0(0xbe9f6901), WTC0(0xbea91f01), WTC0(0xbeb2d681), WTC0(0xbebc9181), - WTC0(0xbec64e81), WTC0(0xbed00f81), WTC0(0xbed9d281), WTC0(0xbee39801), WTC0(0xbeed5f01), WTC0(0xbef72681), WTC0(0xbf00ef81), WTC0(0xbf0aba01), - WTC0(0xbf148681), WTC0(0xbf1e5501), WTC0(0xbf282501), WTC0(0xbf31f501), WTC0(0xbf3bc601), WTC0(0xbf459681), WTC0(0xbf4f6801), WTC0(0xbf593a01), - WTC0(0xbf630d81), WTC0(0xbf6ce201), WTC0(0xbf76b701), WTC0(0xbf808b81), WTC0(0xbf8a5f81), WTC0(0xbf943301), WTC0(0xbf9e0701), WTC0(0xbfa7dc01), - WTC0(0xbfb1b101), WTC0(0xbfbb8701), WTC0(0xbfc55c81), WTC0(0xbfcf3181), WTC0(0xbfd90601), WTC0(0xbfe2d901), WTC0(0xbfecaa81), WTC0(0xbff67a01), - /* part 1 */ - WTC1(0x80130981), WTC1(0x80269f81), WTC1(0x803a3381), WTC1(0x804dc481), WTC1(0x80615281), WTC1(0x8074dc01), WTC1(0x80886081), WTC1(0x809bdf01), - WTC1(0x80af5701), WTC1(0x80c2c781), WTC1(0x80d63101), WTC1(0x80e99401), WTC1(0x80fcf181), WTC1(0x81104a01), WTC1(0x81239d81), WTC1(0x8136ea01), - WTC1(0x814a2f81), WTC1(0x815d6c01), WTC1(0x8170a181), WTC1(0x8183cf81), WTC1(0x8196f781), WTC1(0x81aa1981), WTC1(0x81bd3401), WTC1(0x81d04681), - WTC1(0x81e34f81), WTC1(0x81f64f01), WTC1(0x82094581), WTC1(0x821c3401), WTC1(0x822f1b01), WTC1(0x8241fa01), WTC1(0x8254cf01), WTC1(0x82679901), - WTC1(0x827a5801), WTC1(0x828d0b01), WTC1(0x829fb401), WTC1(0x82b25301), WTC1(0x82c4e801), WTC1(0x82d77201), WTC1(0x82e9ef01), WTC1(0x82fc5f01), - WTC1(0x830ec081), WTC1(0x83211501), WTC1(0x83335c81), WTC1(0x83459881), WTC1(0x8357c701), WTC1(0x8369e781), WTC1(0x837bf801), WTC1(0x838df801), - WTC1(0x839fe801), WTC1(0x83b1c881), WTC1(0x83c39a81), WTC1(0x83d55d01), WTC1(0x83e70f01), WTC1(0x83f8b001), WTC1(0x840a3e81), WTC1(0x841bb981), - WTC1(0x842d2281), WTC1(0x843e7a81), WTC1(0x844fc081), WTC1(0x8460f581), WTC1(0x84721701), WTC1(0x84832481), WTC1(0x84941d81), WTC1(0x84a50201), - WTC1(0x84b5d301), WTC1(0x84c69101), WTC1(0x84d73c01), WTC1(0x84e7d381), WTC1(0x84f85581), WTC1(0x8508c181), WTC1(0x85191801), WTC1(0x85295881), - WTC1(0x85398481), WTC1(0x85499d01), WTC1(0x8559a081), WTC1(0x85698e81), WTC1(0x85796601), WTC1(0x85892681), WTC1(0x8598d081), WTC1(0x85a86581), - WTC1(0x85b7e601), WTC1(0x85c75201), WTC1(0x85d6a981), WTC1(0x85e5eb81), WTC1(0x85f51681), WTC1(0x86042c01), WTC1(0x86132c01), WTC1(0x86221801), - WTC1(0x8630f181), WTC1(0x863fb701), WTC1(0x864e6901), WTC1(0x865d0581), WTC1(0x866b8d81), WTC1(0x867a0081), WTC1(0x86886001), WTC1(0x8696ad01), - WTC1(0x86a4e781), WTC1(0x86b30f01), WTC1(0x86c12401), WTC1(0x86cf2601), WTC1(0x86dd1481), WTC1(0x86eaf081), WTC1(0x86f8ba81), WTC1(0x87067281), - WTC1(0x87141b01), WTC1(0x8721b481), WTC1(0x872f4201), WTC1(0x873cc201), WTC1(0x874a2f01), WTC1(0x87578181), WTC1(0x8764b101), WTC1(0x8771c601), - WTC1(0x877ede01), WTC1(0x878c1881), WTC1(0x87998f01), WTC1(0x87a70e81), WTC1(0x87b42481), WTC1(0x87c05e81), WTC1(0x87cb5101), WTC1(0x87d4ac81), - WTC1(0x87e73d81), WTC1(0x88124281), WTC1(0x88353501), WTC1(0x885f8481), WTC1(0x888d3181), WTC1(0x88be1681), WTC1(0x88f13801), WTC1(0x8925f101), - WTC1(0x895bcd01), WTC1(0x89925a81), WTC1(0x89c92f81), WTC1(0x8a001f01), WTC1(0x8a372881), WTC1(0x8a6e4a01), WTC1(0x8aa58681), WTC1(0x8adcee01), - WTC1(0x8b149701), WTC1(0x8b4c9701), WTC1(0x8b850281), WTC1(0x8bbde981), WTC1(0x8bf75b01), WTC1(0x8c316681), WTC1(0x8c6c1b01), WTC1(0x8ca78781), - WTC1(0x8ce3ba81), WTC1(0x8d20c301), WTC1(0x8d5eaa01), WTC1(0x8d9d7781), WTC1(0x8ddd3201), WTC1(0x8e1de001), WTC1(0x8e5f8881), WTC1(0x8ea23201), - WTC1(0x8ee5e301), WTC1(0x8f2aa101), WTC1(0x8f706f01), WTC1(0x8fb74f81), WTC1(0x8fff4601), WTC1(0x90485401), WTC1(0x90927b81), WTC1(0x90ddc001), - WTC1(0x912a2201), WTC1(0x9177a301), WTC1(0x91c64301), WTC1(0x92160301), WTC1(0x9266e281), WTC1(0x92b8e101), WTC1(0x930bff81), WTC1(0x93603d01), - WTC1(0x93b59901), WTC1(0x940c1281), WTC1(0x9463a881), WTC1(0x94bc5981), WTC1(0x95162381), WTC1(0x95710601), WTC1(0x95ccff01), WTC1(0x962a0c81), - WTC1(0x96882e01), WTC1(0x96e76101), WTC1(0x9747a481), WTC1(0x97a8f681), WTC1(0x980b5501), WTC1(0x986ebd81), WTC1(0x98d32d81), WTC1(0x9938a281), - WTC1(0x999f1981), WTC1(0x9a069001), WTC1(0x9a6f0381), WTC1(0x9ad87081), WTC1(0x9b42d581), WTC1(0x9bae2f81), WTC1(0x9c1a7c81), WTC1(0x9c87ba81), - WTC1(0x9cf5e701), WTC1(0x9d650081), WTC1(0x9dd50481), WTC1(0x9e45f081), WTC1(0x9eb7c101), WTC1(0x9f2a7281), WTC1(0x9f9e0301), WTC1(0xa0127081), - WTC1(0xa087b981), WTC1(0xa0fddd81), WTC1(0xa174da81), WTC1(0xa1ecae01), WTC1(0xa2655581), WTC1(0xa2dece81), WTC1(0xa3591801), WTC1(0xa3d43001), - WTC1(0xa4501601), WTC1(0xa4ccc901), WTC1(0xa54a4701), WTC1(0xa5c89001), WTC1(0xa647a301), WTC1(0xa6c77e01), WTC1(0xa7482101), WTC1(0xa7c98b01), - WTC1(0xa84bbb81), WTC1(0xa8ceb201), WTC1(0xa9526d81), WTC1(0xa9d6ef01), WTC1(0xaa5c3601), WTC1(0xaae24301), WTC1(0xab691681), WTC1(0xabf0b181), - WTC1(0xac791401), WTC1(0xad023f01), WTC1(0xad8c3301), WTC1(0xae16f001), WTC1(0xaea27681), WTC1(0xaf2ec901), WTC1(0xafbbe801), WTC1(0xb049d601), - WTC1(0xb0d89401), WTC1(0xb1682281), WTC1(0xb1f88181), WTC1(0xb289b181), WTC1(0xb31bb301), WTC1(0xb3ae8601), WTC1(0xb4422b81), WTC1(0xb4d6a381), - WTC1(0x4a5a327f), WTC1(0x49c4adff), WTC1(0x492e637f), WTC1(0x48974f7f), WTC1(0x47ff6d7f), WTC1(0x4766baff), WTC1(0x46cd35ff), WTC1(0x4632dd7f), - WTC1(0x4597b0ff), WTC1(0x44fbb1ff), WTC1(0x445eeaff), WTC1(0x43c165ff), WTC1(0x4323227f), WTC1(0x4284277f), WTC1(0x41e48aff), WTC1(0x4144557f), - WTC1(0x40a3867f), WTC1(0x4001f5ff), WTC1(0x3f5f5d80), WTC1(0x3ebbad00), WTC1(0x3e16ee40), WTC1(0x3d713d00), WTC1(0x3ccab700), WTC1(0x3c236500), - WTC1(0x3b7b5800), WTC1(0x3ad2ecc0), WTC1(0x3a2a6540), WTC1(0x3981b7c0), WTC1(0x38d8ba00), WTC1(0x382f01c0), WTC1(0x37846240), WTC1(0x36d8eb00), - WTC1(0x362c9ec0), WTC1(0x357f7a00), WTC1(0x34d18340), WTC1(0x3422c900), WTC1(0x33736c40), WTC1(0x32c39040), WTC1(0x32134280), WTC1(0x31629280), - WTC1(0x30b1a000), WTC1(0x30008380), WTC1(0x2f4f4240), WTC1(0x2e9df180), WTC1(0x2decc780), WTC1(0x2d3bd640), WTC1(0x2c8b0cc0), WTC1(0x2bda3080), - WTC1(0x2b28ec80), WTC1(0x2a773500), WTC1(0x29c51b40), WTC1(0x291293c0), WTC1(0x285f9280), WTC1(0x27ac35c0), WTC1(0x26f8ab40), WTC1(0x26454c00), - WTC1(0x25925600), WTC1(0x24dfd580), WTC1(0x242ddd40), WTC1(0x237c87c0), WTC1(0x22cbe240), WTC1(0x221bef40), WTC1(0x216cb040), WTC1(0x20be2800), - WTC1(0x20105c80), WTC1(0x1f6352a0), WTC1(0x1eb71240), WTC1(0x1e0ba140), WTC1(0x1d60fe40), WTC1(0x1cb723e0), WTC1(0x1c0e0300), WTC1(0x1b6596c0), - WTC1(0x1abde8a0), WTC1(0x1a16fbe0), WTC1(0x1970c680), WTC1(0x18cb4840), WTC1(0x18268e20), WTC1(0x1782a0c0), WTC1(0x16df8960), WTC1(0x163d6300), - WTC1(0x159c52c0), WTC1(0x14fc87e0), WTC1(0x145e2c80), WTC1(0x13c15b60), WTC1(0x13263240), WTC1(0x128cd9a0), WTC1(0x11f562a0), WTC1(0x115fc1c0), - WTC1(0x10cbf160), WTC1(0x1039f200), WTC1(0x0fa9a080), WTC1(0x0f1abd90), WTC1(0x0e8d01d0), WTC1(0x0e003330), WTC1(0x0d743590), WTC1(0x0ce8ef40), - WTC1(0x0c5e1900), WTC1(0x0bd35d70), WTC1(0x0b488eb0), WTC1(0x0abd8410), WTC1(0x0a320a00), WTC1(0x09a60e70), WTC1(0x0919ab00), WTC1(0x088d0de0), - WTC1(0x080065e0), WTC1(0x07739710), WTC1(0x06e65808), WTC1(0x06588348), WTC1(0x05ca0ae0), WTC1(0x053aaaf8), WTC1(0x04a9faf0), WTC1(0x0417f698), - WTC1(0x03859ff4), WTC1(0x02f49be4), WTC1(0x0266b668), WTC1(0x01de554e), WTC1(0x015f50ca), WTC1(0x00eb7e5d), WTC1(0x00904f24), WTC1(0x00212889), - WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), - WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), - WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), - WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), - WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), - WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), - WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), - WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), - WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), - WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), - WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), - WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), - WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), - WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), - WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), - /* part 2 */ - WTC2(0xfffece02), WTC2(0xffe4c3df), WTC2(0xffcaaa55), WTC2(0xffb087d1), WTC2(0xff9662bf), WTC2(0xff7c418b), WTC2(0xff622aa0), WTC2(0xff48246c), - WTC2(0xff2e355a), WTC2(0xff1463db), WTC2(0xfefab608), WTC2(0xfee12f0a), WTC2(0xfec7cfd2), WTC2(0xfeae995a), WTC2(0xfe958cc4), WTC2(0xfe7cabce), - WTC2(0xfe63f882), WTC2(0xfe4b74e0), WTC2(0xfe3322f6), WTC2(0xfe1b04dc), WTC2(0xfe031ccc), WTC2(0xfdeb6cf0), WTC2(0xfdd3ff7c), WTC2(0xfdbce834), - WTC2(0xfda63bb8), WTC2(0xfd900c68), WTC2(0xfd7a590c), WTC2(0xfd6511b4), WTC2(0xfd5026c0), WTC2(0xfd3b8954), WTC2(0xfd272df0), WTC2(0xfd130adc), - WTC2(0xfcff15ac), WTC2(0xfceb4a68), WTC2(0xfcd7b110), WTC2(0xfcc454d0), WTC2(0xfcb14064), WTC2(0xfc9e896c), WTC2(0xfc8c5264), WTC2(0xfc7abef0), - WTC2(0xfc69f078), WTC2(0xfc59f5e8), WTC2(0xfc4acfec), WTC2(0xfc3c8060), WTC2(0xfc2f0264), WTC2(0xfc223b7c), WTC2(0xfc160714), WTC2(0xfc0a4150), - WTC2(0xfbfec920), WTC2(0xfbf38320), WTC2(0xfbe855d0), WTC2(0xfbdd2740), WTC2(0xfbd1fc68), WTC2(0xfbc6fea0), WTC2(0xfbbc5a48), WTC2(0xfbb23b48), - WTC2(0xfba8ca78), WTC2(0xfba02e50), WTC2(0xfb988de0), WTC2(0xfb920b40), WTC2(0xfb8cb870), WTC2(0xfb889f68), WTC2(0xfb85cbe8), WTC2(0xfb843dd0), - WTC2(0xfb83df78), WTC2(0xfb8495d0), WTC2(0xfb864660), WTC2(0xfb88d4a8), WTC2(0xfb8c21e8), WTC2(0xfb900f28), WTC2(0xfb947dc0), WTC2(0xfb9950c0), - WTC2(0xfb9e6d08), WTC2(0xfba3b658), WTC2(0xfba91908), WTC2(0xfbae9e08), WTC2(0xfbb45bd0), WTC2(0xfbba66f8), WTC2(0xfbc0dcf0), WTC2(0xfbc7ead8), - WTC2(0xfbcfc200), WTC2(0xfbd89330), WTC2(0xfbe294d0), WTC2(0xfbee03d0), WTC2(0xfbfb1de8), WTC2(0xfc0a1da4), WTC2(0xfc1b22e0), WTC2(0xfc2e38f0), - WTC2(0xfc436d48), WTC2(0xfc5abf7c), WTC2(0xfc74024c), WTC2(0xfc8ef2e8), WTC2(0xfcab51ac), WTC2(0xfcc8d024), WTC2(0xfce704f0), WTC2(0xfd0580cc), - WTC2(0xfd23d4d0), WTC2(0xfd41ce40), WTC2(0xfd5f81b0), WTC2(0xfd7d08f0), WTC2(0xfd9a8560), WTC2(0xfdb85938), WTC2(0xfdd71798), WTC2(0xfdf753b8), - WTC2(0xfe1993ee), WTC2(0xfe3e30f8), WTC2(0xfe656cba), WTC2(0xfe8f8fdc), WTC2(0xfebca8a4), WTC2(0xfeec590e), WTC2(0xff1e285c), WTC2(0xff51a0b7), - WTC2(0xff866330), WTC2(0xffbc2cbb), WTC2(0xfff2bbff), WTC2(0x0029d79d), WTC2(0x00618a22), WTC2(0x009a1185), WTC2(0x00d3aa8c), WTC2(0x010e8ff6), - WTC2(0x014af29e), WTC2(0x0188fe56), WTC2(0x01c8e108), WTC2(0x020ab3c4), WTC2(0x024e68a8), WTC2(0x0293e824), WTC2(0x02db1bc8), WTC2(0x0323f1a4), - WTC2(0x036e5d6c), WTC2(0x03ba5320), WTC2(0x0407c938), WTC2(0x0456cad0), WTC2(0x04a77288), WTC2(0x04f9db88), WTC2(0x054e1888), WTC2(0x05a41ef0), - WTC2(0x05fbd6e0), WTC2(0x065528c0), WTC2(0x06b00838), WTC2(0x070c7ee0), WTC2(0x076a9bb0), WTC2(0x07ca6d10), WTC2(0x082c08e0), WTC2(0x088f8da0), - WTC2(0x08f51ac0), WTC2(0x095ccc20), WTC2(0x09c69f70), WTC2(0x0a327b40), WTC2(0x0aa046d0), WTC2(0x0b0febb0), WTC2(0x0b815dd0), WTC2(0x0bf49600), - WTC2(0x0c698c50), WTC2(0x0ce03ba0), WTC2(0x0d58a380), WTC2(0x0dd2c510), WTC2(0x0e4ea110), WTC2(0x0ecc3dd0), WTC2(0x0f4ba800), WTC2(0x0fcced10), - WTC2(0x10501960), WTC2(0x10d532a0), WTC2(0x115c39c0), WTC2(0x11e52fa0), WTC2(0x12701560), WTC2(0x12fcef20), WTC2(0x138bc200), WTC2(0x141c9300), - WTC2(0x14af64a0), WTC2(0x154434e0), WTC2(0x15db0020), WTC2(0x1673c360), WTC2(0x170e7e80), WTC2(0x17ab35e0), WTC2(0x1849ee40), WTC2(0x18eaaba0), - WTC2(0x198d6f00), WTC2(0x1a3236a0), WTC2(0x1ad90080), WTC2(0x1b81cc60), WTC2(0x1c2c9da0), WTC2(0x1cd97980), WTC2(0x1d8865c0), WTC2(0x1e396540), - WTC2(0x1eec7700), WTC2(0x1fa198c0), WTC2(0x2058c840), WTC2(0x21120640), WTC2(0x21cd5700), WTC2(0x228abec0), WTC2(0x234a4180), WTC2(0x240bdf80), - WTC2(0x24cf95c0), WTC2(0x259561c0), WTC2(0x265d4200), WTC2(0x27273840), WTC2(0x27f348c0), WTC2(0x28c17700), WTC2(0x2991c500), WTC2(0x2a643080), - WTC2(0x2b38b680), WTC2(0x2c0f53c0), WTC2(0x2ce80840), WTC2(0x2dc2d680), WTC2(0x2e9fc100), WTC2(0x2f7ecac0), WTC2(0x305ff280), WTC2(0x314334c0), - WTC2(0x32288e00), WTC2(0x330ffb80), WTC2(0x33f97d80), WTC2(0x34e515c0), WTC2(0x35d2c5c0), WTC2(0x36c28d00), WTC2(0x37b467c0), WTC2(0x38a85080), - WTC2(0x399e4240), WTC2(0x3a963a00), WTC2(0x3b903600), WTC2(0x3c8c3480), WTC2(0x3d8a3380), WTC2(0x3e8a2dc0), WTC2(0x3f8c1b40), WTC2(0x408ff2ff), - WTC2(0x4195ae7f), WTC2(0x429d477f), WTC2(0x43a6b87f), WTC2(0x44b1fdff), WTC2(0x45bf11ff), WTC2(0x46cdee7f), WTC2(0x47de8cff), WTC2(0x48f0e77f), - WTC2(0x4a050eff), WTC2(0x4b1b2dff), WTC2(0x4c3372ff), WTC2(0x4d4e0bff), WTC2(0x4e6b257f), WTC2(0x4f8aedff), WTC2(0x50ad92ff), WTC2(0x51d341ff), - WTC2(0x002006a9), WTC2(0x000bfb36), WTC2(0xfffe45ac), WTC2(0xfff6d064), WTC2(0xfff585bc), WTC2(0xfffa500d), WTC2(0x000519b4), WTC2(0x0015cd0c), - WTC2(0x002c5470), WTC2(0x00489a3b), WTC2(0x006a88c8), WTC2(0x00920a74), WTC2(0x00bf0999), WTC2(0x00f17092), WTC2(0x012929bc), WTC2(0x01661f70), - WTC2(0x01a83c0c), WTC2(0x01ef69e8), WTC2(0x023b9364), WTC2(0x028ca2d4), WTC2(0x02e2829c), WTC2(0x033d1d10), WTC2(0x039c5c90), WTC2(0x04002b78), - WTC2(0x04687418), WTC2(0x04d520e0), WTC2(0x05461c18), WTC2(0x05bb5020), WTC2(0x0634a758), WTC2(0x06b20c20), WTC2(0x073368c8), WTC2(0x07b8a7b0), - WTC2(0x0841b340), WTC2(0x08ce75b0), WTC2(0x095ed980), WTC2(0x09f2c900), WTC2(0x0a8a2e80), WTC2(0x0b24f470), WTC2(0x0bc30510), WTC2(0x0c644ad0), - WTC2(0x0d08b010), WTC2(0x0db01f10), WTC2(0x0e5a8250), WTC2(0x0f07c400), WTC2(0x0fb7cea0), WTC2(0x106a8c80), WTC2(0x111fe800), WTC2(0x11d7cb60), - WTC2(0x12922120), WTC2(0x134ed3a0), WTC2(0x140dcd00), WTC2(0x14cef7e0), WTC2(0x15923e60), WTC2(0x16578b00), WTC2(0x171ec820), WTC2(0x17e7e020), - WTC2(0x18b2bd20), WTC2(0x197f49c0), WTC2(0x1a4d7040), WTC2(0x1b1d1b00), WTC2(0x1bee3460), WTC2(0x1cc0a6a0), WTC2(0x1d945c40), WTC2(0x1e693f80), - WTC2(0x1f3f3ac0), WTC2(0x20163880), WTC2(0x20ee22c0), WTC2(0x21c6e440), WTC2(0x22a06740), WTC2(0x237a9600), WTC2(0x24555ac0), WTC2(0x2530a040), - WTC2(0x260c5080), WTC2(0x26e85600), WTC2(0x27c49b00), WTC2(0x28a10a00), WTC2(0x297d8d80), WTC2(0x2a5a0f80), WTC2(0x2b367a80), WTC2(0x2c12b8c0), - WTC2(0x2ceeb500), WTC2(0x2dca5940), WTC2(0x2ea58fc0), WTC2(0x2f804340), WTC2(0x305a5dc0), WTC2(0x3133ca00), WTC2(0x320c7200), WTC2(0x32e44000), - WTC2(0x33bb1ec0), WTC2(0x3490f880), WTC2(0x3565b7c0), WTC2(0x36394640), WTC2(0x370b8f00), WTC2(0x37dc7c00), WTC2(0x38abf7c0), WTC2(0x3979ecc0), - WTC2(0x3a464500), WTC2(0x3b10eb00), WTC2(0x3bd9c940), WTC2(0x3ca0c9c0), WTC2(0x3d65d740), WTC2(0x3e28dc00), WTC2(0x3ee9c240), WTC2(0x3fa87480), - WTC2(0x4064dcff), WTC2(0x411ee67f), WTC2(0x41d67a7f), WTC2(0x428b847f), WTC2(0x433ded7f), WTC2(0x43eda0ff), WTC2(0x449a887f), WTC2(0x45448f7f), - WTC2(0x45eb9eff), WTC2(0x468fa1ff), WTC2(0x473082ff), WTC2(0x47ce2c7f), WTC2(0x4868887f), WTC2(0x48ff80ff), WTC2(0x499300ff), WTC2(0x4a22f2ff), - WTC2(0x4aaf407f), WTC2(0x4b37d47f), WTC2(0x4bbc997f), WTC2(0x4c3d78ff), WTC2(0x4cba5e7f), WTC2(0x4d33337f), WTC2(0x4da7e27f), WTC2(0x4e18567f), - WTC2(0x4e8478ff), WTC2(0x4eec347f), WTC2(0x4f4f737f), WTC2(0x4fae20ff), WTC2(0x500825ff), WTC2(0x505d6dff), WTC2(0x50ade37f), WTC2(0x50f96f7f), - WTC2(0x513ffdff), WTC2(0x518177ff), WTC2(0x51bdc87f), WTC2(0x51f4d9ff), WTC2(0x5226967f), WTC2(0x5252e87f), WTC2(0x5279b9ff), WTC2(0x529af5ff), - WTC2(0x52b6867f), WTC2(0x52cc55ff), WTC2(0x52dc4eff), WTC2(0x52e65aff), WTC2(0x52ea657f), WTC2(0x52e857ff), WTC2(0x52e01d7f), WTC2(0x52d19fff), - WTC2(0x52bcc9ff), WTC2(0x52a1857f), WTC2(0x527fbd7f), WTC2(0x52575b7f), WTC2(0x52284a7f), WTC2(0x51f274ff), WTC2(0x51b5c47f), WTC2(0x5172247f), - WTC2(0x51277dff), WTC2(0x50d5bc7f), WTC2(0x507cc9ff), WTC2(0x501c90ff), WTC2(0x4fb4fb7f), WTC2(0x4f45f3ff), WTC2(0x4ecf64ff), WTC2(0x4e5138ff), - WTC2(0x4dcb597f), WTC2(0x4d3db1ff), WTC2(0x4ca82bff), WTC2(0x4c0ab27f), WTC2(0x4b652f7f), WTC2(0x4ab78d7f), WTC2(0x4a01b67f), WTC2(0x4943957f), - WTC2(0x487d12ff), WTC2(0x47ae1f7f), WTC2(0x46d68f7f), WTC2(0x45f7187f), WTC2(0x4513597f), WTC2(0x4430467f), WTC2(0x4352d2ff), WTC2(0x427e6bff), - WTC2(0x41b390ff), WTC2(0x40f2077f), WTC2(0x4039a87f), WTC2(0x3f8a3100), WTC2(0x3ee33e00), WTC2(0x3e446ac0), WTC2(0x3dad5180), WTC2(0x3d1d7fc0), - WTC2(0x3c947b00), WTC2(0x3c11c7c0), WTC2(0x3b94ebc0), WTC2(0x3b1d6dc0), WTC2(0x3aaad480), WTC2(0x3a3ca740), WTC2(0x39d26c40), WTC2(0x396ba8c0), - WTC2(0x3907e080), WTC2(0x38a69800), WTC2(0x38473d80), WTC2(0x37e923c0), WTC2(0x378b9b80), WTC2(0x372e0380), WTC2(0x36d03a80), WTC2(0x36727f00), - WTC2(0x36150e40), WTC2(0x35b81540), WTC2(0x355b8000), WTC2(0x34ff1dc0), WTC2(0x34a2bfc0), WTC2(0x34463e80), WTC2(0x33e982c0), WTC2(0x338c7880), - WTC2(0x332f0bc0), WTC2(0x32d11800), WTC2(0x327265c0), WTC2(0x3212bbc0), WTC2(0x31b1e740), WTC2(0x314fef00), WTC2(0x30ed0540), WTC2(0x30895c80), - WTC2(0x30251880), WTC2(0x2fc02880), WTC2(0x2f5a6480), WTC2(0x2ef3a480), WTC2(0x2e8bd640), WTC2(0x2e231100), WTC2(0x2db97680), WTC2(0x2d4f2700), - WTC2(0x2ce431c0), WTC2(0x2c789080), WTC2(0x2c0c3bc0), WTC2(0x2b9f2bc0), WTC2(0x2b315940), WTC2(0x2ac2bc00), WTC2(0x2a534cc0), WTC2(0x29e303c0) -}; - - diff --git a/libAACenc/src/aacEnc_rom.h b/libAACenc/src/aacEnc_rom.h deleted file mode 100644 index 862417f..0000000 --- a/libAACenc/src/aacEnc_rom.h +++ /dev/null @@ -1,203 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/****************************************************************************** - - Initial authors: M. Lohwasser, M. Gayer - Contents/description: - -******************************************************************************/ -/*! - \file - \brief Memory layout - \author Markus Lohwasser -*/ - -#ifndef AAC_ENC_ROM_H -#define AAC_ENC_ROM_H - -#include "common_fix.h" - -#include "psy_const.h" -#include "psy_configuration.h" -#include "FDK_tools_rom.h" - -/* - Huffman Tables -*/ -extern const ULONG FDKaacEnc_huff_ltab1_2[3][3][3][3]; -extern const ULONG FDKaacEnc_huff_ltab3_4[3][3][3][3]; -extern const ULONG FDKaacEnc_huff_ltab5_6[9][9]; -extern const ULONG FDKaacEnc_huff_ltab7_8[8][8]; -extern const ULONG FDKaacEnc_huff_ltab9_10[13][13]; -extern const UCHAR FDKaacEnc_huff_ltab11[17][17]; -extern const UCHAR FDKaacEnc_huff_ltabscf[121]; -extern const USHORT FDKaacEnc_huff_ctab1[3][3][3][3]; -extern const USHORT FDKaacEnc_huff_ctab2[3][3][3][3]; -extern const USHORT FDKaacEnc_huff_ctab3[3][3][3][3]; -extern const USHORT FDKaacEnc_huff_ctab4[3][3][3][3]; -extern const USHORT FDKaacEnc_huff_ctab5[9][9]; -extern const USHORT FDKaacEnc_huff_ctab6[9][9]; -extern const USHORT FDKaacEnc_huff_ctab7[8][8]; -extern const USHORT FDKaacEnc_huff_ctab8[8][8]; -extern const USHORT FDKaacEnc_huff_ctab9[13][13]; -extern const USHORT FDKaacEnc_huff_ctab10[13][13]; -extern const USHORT FDKaacEnc_huff_ctab11[21][17]; -extern const ULONG FDKaacEnc_huff_ctabscf[121]; - -/* - quantizer -*/ -#define MANT_DIGITS 9 -#define MANT_SIZE (1<>shift)) / (samplingRate>>shift); -} - -INT FDKaacEnc_CalcBitrate( - const INT bitsPerFrame, - const INT frameLength, - const INT samplingRate - ) -{ - int shift = 0; - while ((frameLength & ~((1 << (shift + 1)) - 1)) == frameLength - && (samplingRate & ~((1 << (shift + 1)) - 1)) == samplingRate) - { - shift++; - } - - return (bitsPerFrame * (samplingRate>>shift)) / ( frameLength>>shift) ; - -} - -static AAC_ENCODER_ERROR FDKaacEnc_InitCheckAncillary(INT bitRate, - INT framelength, - INT ancillaryRate, - INT *ancillaryBitsPerFrame, - INT sampleRate); - -INT FDKaacEnc_LimitBitrate( - HANDLE_TRANSPORTENC hTpEnc, - INT coreSamplingRate, - INT frameLength, - INT nChannels, - INT nChannelsEff, - INT bitRate, - INT averageBits, - INT *pAverageBitsPerFrame, - INT bitrateMode, - INT nSubFrames - ) -{ - INT transportBits, prevBitRate, averageBitsPerFrame, shift = 0, iter=0; - - while ( (frameLength & ~((1<<(shift+1))-1)) == frameLength - && (coreSamplingRate & ~((1<<(shift+1))-1)) == coreSamplingRate ) - { - shift ++; - } - - do { - prevBitRate = bitRate; - averageBitsPerFrame = (bitRate*(frameLength>>shift)) / (coreSamplingRate>>shift) / nSubFrames; - - if (pAverageBitsPerFrame != NULL) { - *pAverageBitsPerFrame = averageBitsPerFrame; - } - - if (hTpEnc != NULL) { - transportBits = transportEnc_GetStaticBits(hTpEnc, averageBitsPerFrame); - } else { - /* Assume some worst case */ - transportBits = 208; - } - - bitRate = FDKmax(bitRate, ((((40 * nChannels) + transportBits) * (coreSamplingRate)) / frameLength) ); - FDK_ASSERT(bitRate >= 0); - - bitRate = FDKmin(bitRate, ((nChannelsEff * MIN_BUFSIZE_PER_EFF_CHAN)*(coreSamplingRate>>shift)) / (frameLength>>shift)) ; - FDK_ASSERT(bitRate >= 0); - - } while (prevBitRate != bitRate && iter++ < 3) ; - - return bitRate; -} - - -typedef struct -{ - AACENC_BITRATE_MODE bitrateMode; - int chanBitrate[2]; /* mono/stereo settings */ -} CONFIG_TAB_ENTRY_VBR; - -static const CONFIG_TAB_ENTRY_VBR configTabVBR[] = { - {AACENC_BR_MODE_CBR, { 0, 0}} , - {AACENC_BR_MODE_VBR_1, { 32000, 20000}} , - {AACENC_BR_MODE_VBR_2, { 40000, 32000}} , - {AACENC_BR_MODE_VBR_3, { 56000, 48000}} , - {AACENC_BR_MODE_VBR_4, { 72000, 64000}} , - {AACENC_BR_MODE_VBR_5, {112000, 96000}} -}; - -/*----------------------------------------------------------------------------- - - functionname: FDKaacEnc_GetVBRBitrate - description: Get VBR bitrate from vbr quality - input params: int vbrQuality (VBR0, VBR1, VBR2) - channelMode - returns: vbr bitrate - - ------------------------------------------------------------------------------*/ -INT FDKaacEnc_GetVBRBitrate(INT bitrateMode, CHANNEL_MODE channelMode) -{ - INT bitrate = 0; - INT monoStereoMode = 0; /* default mono */ - - if (FDKaacEnc_GetMonoStereoMode(channelMode)==EL_MODE_STEREO) { - monoStereoMode = 1; - } - - switch((AACENC_BITRATE_MODE)bitrateMode){ - case AACENC_BR_MODE_VBR_1: - case AACENC_BR_MODE_VBR_2: - case AACENC_BR_MODE_VBR_3: - case AACENC_BR_MODE_VBR_4: - case AACENC_BR_MODE_VBR_5: - bitrate = configTabVBR[bitrateMode].chanBitrate[monoStereoMode]; - break; - case AACENC_BR_MODE_INVALID: - case AACENC_BR_MODE_CBR: - case AACENC_BR_MODE_SFR: - case AACENC_BR_MODE_FF: - default: - bitrate = 0; - break; - } - - /* convert channel bitrate to overall bitrate*/ - bitrate *= FDKaacEnc_GetChannelModeConfiguration(channelMode)->nChannelsEff; - - return bitrate; -} - -/** - * \brief Convert encoder bitreservoir value for transport library. - * - * \param hAacEnc Encoder handle - * - * \return Corrected bitreservoir level used in transport library. - */ -static INT FDKaacEnc_EncBitresToTpBitres( - const HANDLE_AAC_ENC hAacEnc - ) -{ - INT transporBitreservoir = 0; - - switch (hAacEnc->bitrateMode) { - case AACENC_BR_MODE_CBR: - transporBitreservoir = hAacEnc->qcKernel->bitResTot; /* encoder bitreservoir level */ - break; - case AACENC_BR_MODE_VBR_1: - case AACENC_BR_MODE_VBR_2: - case AACENC_BR_MODE_VBR_3: - case AACENC_BR_MODE_VBR_4: - case AACENC_BR_MODE_VBR_5: - transporBitreservoir = FDK_INT_MAX; /* signal variable bitrate */ - break; - case AACENC_BR_MODE_FF: - case AACENC_BR_MODE_SFR: - transporBitreservoir = 0; /* super framing and fixed framing */ - break; /* without bitreservoir signaling */ - default: - case AACENC_BR_MODE_INVALID: - transporBitreservoir = 0; /* invalid configuration*/ - FDK_ASSERT(0); - } - - if (hAacEnc->config->audioMuxVersion==2) { - transporBitreservoir = MIN_BUFSIZE_PER_EFF_CHAN * hAacEnc->channelMapping.nChannelsEff; - } - - return transporBitreservoir; -} - -/*----------------------------------------------------------------------------- - - functionname: FDKaacEnc_AacInitDefaultConfig - description: gives reasonable default configuration - returns: --- - - ------------------------------------------------------------------------------*/ -void FDKaacEnc_AacInitDefaultConfig(AACENC_CONFIG *config) -{ - /* make thepre initialization of the structs flexible */ - FDKmemclear(config, sizeof(AACENC_CONFIG)); - - /* default ancillary */ - config->anc_Rate = 0; /* no ancillary data */ - config->ancDataBitRate = 0; /* no additional consumed bitrate */ - - /* default configurations */ - config->bitRate = -1; /* bitrate must be set*/ - config->averageBits = -1; /* instead of bitrate/s we can configure bits/superframe */ - config->bitrateMode = 0; - config->bandWidth = 0; /* get bandwidth from table */ - config->useTns = TNS_ENABLE_MASK; /* tns enabled completly */ - config->usePns = 1; /* depending on channelBitrate this might be set to 0 later */ - config->useIS = 1; /* Intensity Stereo Configuration */ - config->framelength = -1; /* Framesize not configured */ - config->syntaxFlags = 0; /* default syntax with no specialities */ - config->epConfig = -1; /* no ER syntax -> no additional error protection */ - config->nSubFrames = 1; /* default, no sub frames */ - config->channelOrder = CH_ORDER_MPEG; /* Use MPEG channel ordering. */ - config->channelMode = MODE_UNKNOWN; - config->minBitsPerFrame = -1; /* minum number of bits in each AU */ - config->maxBitsPerFrame = -1; /* minum number of bits in each AU */ - config->bitreservoir = -1; /* default, uninitialized value */ - config->audioMuxVersion = -1; /* audio mux version not configured */ - - /* init tabs in fixpoint_math */ - InitLdInt(); - InitInvSqrtTab(); -} - - -/*--------------------------------------------------------------------------- - - functionname: FDKaacEnc_Open - description: allocate and initialize a new encoder instance - returns: error code - - ---------------------------------------------------------------------------*/ -AAC_ENCODER_ERROR FDKaacEnc_Open(HANDLE_AAC_ENC *phAacEnc, - const INT nElements, - const INT nChannels, - const INT nSubFrames) -{ - AAC_ENCODER_ERROR ErrorStatus; - AAC_ENC *hAacEnc = NULL; - UCHAR *dynamicRAM = NULL; - - if (phAacEnc==NULL) { - return AAC_ENC_INVALID_HANDLE; - } - - /* allocate encoder structure */ - hAacEnc = GetRam_aacEnc_AacEncoder(); - if (hAacEnc == NULL) { - ErrorStatus = AAC_ENC_NO_MEMORY; - goto bail; - } - FDKmemclear(hAacEnc, sizeof(AAC_ENC)); - - hAacEnc->dynamic_RAM = GetAACdynamic_RAM(); - dynamicRAM = (UCHAR*)hAacEnc->dynamic_RAM; - - /* allocate the Psy aud Psy Out structure */ - ErrorStatus = FDKaacEnc_PsyNew(&hAacEnc->psyKernel, - nElements, - nChannels - ,dynamicRAM - ); - if (ErrorStatus != AAC_ENC_OK) - goto bail; - - ErrorStatus = FDKaacEnc_PsyOutNew(hAacEnc->psyOut, - nElements, - nChannels, - nSubFrames - ,dynamicRAM - ); - if (ErrorStatus != AAC_ENC_OK) - goto bail; - - /* allocate the Q&C Out structure */ - ErrorStatus = FDKaacEnc_QCOutNew(hAacEnc->qcOut, - nElements, - nChannels, - nSubFrames - ,dynamicRAM - ); - if (ErrorStatus != AAC_ENC_OK) - goto bail; - - /* allocate the Q&C kernel */ - ErrorStatus = FDKaacEnc_QCNew(&hAacEnc->qcKernel, - nElements - ,dynamicRAM - ); - if (ErrorStatus != AAC_ENC_OK) - goto bail; - - hAacEnc->maxChannels = nChannels; - hAacEnc->maxElements = nElements; - hAacEnc->maxFrames = nSubFrames; - -bail: - *phAacEnc = hAacEnc; - return ErrorStatus; -} - - -AAC_ENCODER_ERROR FDKaacEnc_Initialize(HANDLE_AAC_ENC hAacEnc, - AACENC_CONFIG *config, /* pre-initialized config struct */ - HANDLE_TRANSPORTENC hTpEnc, - ULONG initFlags) -{ - AAC_ENCODER_ERROR ErrorStatus; - INT psyBitrate, tnsMask; //INT profile = 1; - CHANNEL_MAPPING *cm = NULL; - - INT qmbfac, qbw; - FIXP_DBL mbfac, bw_ratio; - QC_INIT qcInit; - INT averageBitsPerFrame = 0; - - if (config==NULL) - return AAC_ENC_INVALID_HANDLE; - - /******************* sanity checks *******************/ - - /* check config structure */ - if (config->nChannels < 1 || config->nChannels > (8)) { - return AAC_ENC_UNSUPPORTED_CHANNELCONFIG; - } - - /* check sample rate */ - switch (config->sampleRate) - { - case 8000: - case 11025: - case 12000: - case 16000: - case 22050: - case 24000: - case 32000: - case 44100: - case 48000: - case 64000: - case 88200: - case 96000: - break; - default: - return AAC_ENC_UNSUPPORTED_SAMPLINGRATE; - } - - /* bitrate has to be set */ - if (config->bitRate==-1) { - return AAC_ENC_UNSUPPORTED_BITRATE; - } - - /* check bit rate */ - - if (FDKaacEnc_LimitBitrate( - hTpEnc, - config->sampleRate, - config->framelength, - config->nChannels, - FDKaacEnc_GetChannelModeConfiguration(config->channelMode)->nChannelsEff, - config->bitRate, - config->averageBits, - &averageBitsPerFrame, - config->bitrateMode, - config->nSubFrames - ) != config->bitRate - && !((config->bitrateMode>=1) && (config->bitrateMode<=5)) - ) - { - return AAC_ENC_UNSUPPORTED_BITRATE; - } - - if (config->syntaxFlags & AC_ER_VCB11) { - return AAC_ENC_UNSUPPORTED_ER_FORMAT; - } - if (config->syntaxFlags & AC_ER_HCR) { - return AAC_ENC_UNSUPPORTED_ER_FORMAT; - } - - /* check frame length */ - switch (config->framelength) - { - case 1024: - if ( config->audioObjectType == AOT_ER_AAC_LD - || config->audioObjectType == AOT_ER_AAC_ELD ) - { - return AAC_ENC_INVALID_FRAME_LENGTH; - } - break; - case 512: - case 480: - if ( config->audioObjectType != AOT_ER_AAC_LD - && config->audioObjectType != AOT_ER_AAC_ELD ) - { - return AAC_ENC_INVALID_FRAME_LENGTH; - } - break; - default: - return AAC_ENC_INVALID_FRAME_LENGTH; - } - - if (config->anc_Rate != 0) { - - ErrorStatus = FDKaacEnc_InitCheckAncillary(config->bitRate, - config->framelength, - config->anc_Rate, - &hAacEnc->ancillaryBitsPerFrame, - config->sampleRate); - if (ErrorStatus != AAC_ENC_OK) - goto bail; - - - /* update estimated consumed bitrate */ - config->ancDataBitRate += ( (hAacEnc->ancillaryBitsPerFrame * config->sampleRate) / config->framelength ); - - } - - /* maximal allowed DSE bytes in frame */ - { - /* fixpoint calculation*/ - INT q_res, encBitrate, sc; - FIXP_DBL tmp = fDivNorm(config->framelength, config->sampleRate, &q_res); - encBitrate = (config->bitRate/*-config->ancDataBitRate*/)- (INT)(config->nChannels*8000); - sc = CountLeadingBits(encBitrate); - config->maxAncBytesPerAU = FDKmin( (256), FDKmax(0,(INT)(fMultDiv2(tmp, (FIXP_DBL)(encBitrate<>(-q_res+sc-1+3))) ); - } - - /* bind config to hAacEnc->config */ - hAacEnc->config = config; - - /* set hAacEnc->bitrateMode */ - hAacEnc->bitrateMode = (AACENC_BITRATE_MODE)config->bitrateMode; - - hAacEnc->encoderMode = config->channelMode; - - ErrorStatus = FDKaacEnc_InitChannelMapping(hAacEnc->encoderMode, config->channelOrder, &hAacEnc->channelMapping); - if (ErrorStatus != AAC_ENC_OK) - goto bail; - - cm = &hAacEnc->channelMapping; - - ErrorStatus = FDKaacEnc_DetermineBandWidth(&hAacEnc->config->bandWidth, - config->bandWidth, - config->bitRate - config->ancDataBitRate, - hAacEnc->bitrateMode, - config->sampleRate, - config->framelength, - cm, - hAacEnc->encoderMode); - if (ErrorStatus != AAC_ENC_OK) - goto bail; - - hAacEnc->bandwidth90dB = (INT)hAacEnc->config->bandWidth; - - tnsMask = config->useTns ? TNS_ENABLE_MASK : 0x0; - psyBitrate = config->bitRate - config->ancDataBitRate; - - ErrorStatus = FDKaacEnc_psyInit(hAacEnc->psyKernel, - hAacEnc->psyOut, - hAacEnc->maxFrames, - hAacEnc->maxChannels, - config->audioObjectType, - cm); - if (ErrorStatus != AAC_ENC_OK) - goto bail; - - ErrorStatus = FDKaacEnc_psyMainInit(hAacEnc->psyKernel, - config->audioObjectType, - cm, - config->sampleRate, - config->framelength, - psyBitrate, - tnsMask, - hAacEnc->bandwidth90dB, - config->usePns, - config->useIS, - config->syntaxFlags, - initFlags); - if (ErrorStatus != AAC_ENC_OK) - goto bail; - - ErrorStatus = FDKaacEnc_QCOutInit(hAacEnc->qcOut, hAacEnc->maxFrames, cm); - if (ErrorStatus != AAC_ENC_OK) - goto bail; - - - - qcInit.channelMapping = &hAacEnc->channelMapping; - qcInit.sceCpe = 0; - - if ((config->bitrateMode>=1) && (config->bitrateMode<=5)) { - qcInit.averageBits = (averageBitsPerFrame+7)&~7; - qcInit.bitRes = MIN_BUFSIZE_PER_EFF_CHAN*cm->nChannelsEff; - qcInit.maxBits = MIN_BUFSIZE_PER_EFF_CHAN*cm->nChannelsEff; - qcInit.maxBits = (config->maxBitsPerFrame!=-1) ? fixMin(qcInit.maxBits, config->maxBitsPerFrame) : qcInit.maxBits; - qcInit.maxBits = fixMax(qcInit.maxBits, (averageBitsPerFrame+7)&~7); - qcInit.minBits = (config->minBitsPerFrame!=-1) ? config->minBitsPerFrame : 0; - qcInit.minBits = fixMin(qcInit.minBits, averageBitsPerFrame&~7); - } - else - { - int maxBitres; - qcInit.averageBits = (averageBitsPerFrame+7)&~7; - maxBitres = (MIN_BUFSIZE_PER_EFF_CHAN*cm->nChannelsEff) - qcInit.averageBits; - qcInit.bitRes = (config->bitreservoir!=-1) ? FDKmin(config->bitreservoir, maxBitres) : maxBitres; - - qcInit.maxBits = fixMin(MIN_BUFSIZE_PER_EFF_CHAN*cm->nChannelsEff, ((averageBitsPerFrame+7)&~7)+qcInit.bitRes); - qcInit.maxBits = (config->maxBitsPerFrame!=-1) ? fixMin(qcInit.maxBits, config->maxBitsPerFrame) : qcInit.maxBits; - qcInit.maxBits = fixMin(MIN_BUFSIZE_PER_EFF_CHAN*cm->nChannelsEff, fixMax(qcInit.maxBits, (averageBitsPerFrame+7+8)&~7)); - - qcInit.minBits = fixMax(0, ((averageBitsPerFrame-1)&~7)-qcInit.bitRes-transportEnc_GetStaticBits(hTpEnc, ((averageBitsPerFrame+7)&~7)+qcInit.bitRes)); - qcInit.minBits = (config->minBitsPerFrame!=-1) ? fixMax(qcInit.minBits, config->minBitsPerFrame) : qcInit.minBits; - qcInit.minBits = fixMin(qcInit.minBits, (averageBitsPerFrame - transportEnc_GetStaticBits(hTpEnc, qcInit.maxBits))&~7); - } - - qcInit.sampleRate = config->sampleRate; - qcInit.advancedBitsToPe = isLowDelay(config->audioObjectType) ? 1 : 0 ; - qcInit.nSubFrames = config->nSubFrames; - qcInit.padding.paddingRest = config->sampleRate; - - /* Calc meanPe: qcInit.meanPe = 10.0f * FRAME_LEN_LONG * hAacEnc->bandwidth90dB/(config->sampleRate/2.0f); */ - bw_ratio = fDivNorm((FIXP_DBL)(10*config->framelength*hAacEnc->bandwidth90dB), (FIXP_DBL)(config->sampleRate), &qbw); - qcInit.meanPe = FDKmax((INT)scaleValue(bw_ratio, qbw+1-(DFRACT_BITS-1)), 1); - - /* Calc maxBitFac */ - mbfac = fDivNorm((MIN_BUFSIZE_PER_EFF_CHAN-744)*cm->nChannelsEff, qcInit.averageBits/qcInit.nSubFrames, &qmbfac); - qmbfac = DFRACT_BITS-1-qmbfac; - qcInit.maxBitFac = (qmbfac > 24) ? (mbfac >> (qmbfac - 24)):(mbfac << (24 - qmbfac)); - - switch(config->bitrateMode){ - case AACENC_BR_MODE_CBR: - qcInit.bitrateMode = QCDATA_BR_MODE_CBR; - break; - case AACENC_BR_MODE_VBR_1: - qcInit.bitrateMode = QCDATA_BR_MODE_VBR_1; - break; - case AACENC_BR_MODE_VBR_2: - qcInit.bitrateMode = QCDATA_BR_MODE_VBR_2; - break; - case AACENC_BR_MODE_VBR_3: - qcInit.bitrateMode = QCDATA_BR_MODE_VBR_3; - break; - case AACENC_BR_MODE_VBR_4: - qcInit.bitrateMode = QCDATA_BR_MODE_VBR_4; - break; - case AACENC_BR_MODE_VBR_5: - qcInit.bitrateMode = QCDATA_BR_MODE_VBR_5; - break; - case AACENC_BR_MODE_SFR: - qcInit.bitrateMode = QCDATA_BR_MODE_SFR; - break; - case AACENC_BR_MODE_FF: - qcInit.bitrateMode = QCDATA_BR_MODE_FF; - break; - default: - ErrorStatus = AAC_ENC_UNSUPPORTED_BITRATE_MODE; - goto bail; - } - - qcInit.invQuant = (config->useRequant)?2:0; - - /* maxIterations should be set to the maximum number of requantization iterations that are - * allowed before the crash recovery functionality is activated. This setting should be adjusted - * to the processing power available, i.e. to the processing power headroom in one frame that is - * still left after normal encoding without requantization. Please note that if activated this - * functionality is used most likely only in cases where the encoder is operating beyond - * recommended settings, i.e. the audio quality is suboptimal anyway. Activating the crash - * recovery does not further reduce audio quality significantly in these cases. */ - if ( (config->audioObjectType == AOT_ER_AAC_LD) || (config->audioObjectType == AOT_ER_AAC_ELD) ) { - qcInit.maxIterations = 2; - } - else - { - qcInit.maxIterations = 5; - } - - qcInit.bitrate = config->bitRate - config->ancDataBitRate; - - qcInit.staticBits = transportEnc_GetStaticBits(hTpEnc, qcInit.averageBits/qcInit.nSubFrames); - - ErrorStatus = FDKaacEnc_QCInit(hAacEnc->qcKernel, &qcInit); - if (ErrorStatus != AAC_ENC_OK) - goto bail; - - hAacEnc->aot = hAacEnc->config->audioObjectType; - - /* common things */ - - return AAC_ENC_OK; - -bail: - - return ErrorStatus; -} - - -/*--------------------------------------------------------------------------- - - functionname: FDKaacEnc_EncodeFrame - description: encodes one frame - returns: error code - - ---------------------------------------------------------------------------*/ -AAC_ENCODER_ERROR FDKaacEnc_EncodeFrame( HANDLE_AAC_ENC hAacEnc, /* encoder handle */ - HANDLE_TRANSPORTENC hTpEnc, - INT_PCM* RESTRICT inputBuffer, - INT* nOutBytes, - AACENC_EXT_PAYLOAD extPayload[MAX_TOTAL_EXT_PAYLOADS] - ) -{ - AAC_ENCODER_ERROR ErrorStatus; - int el, n, c=0; - UCHAR extPayloadUsed[MAX_TOTAL_EXT_PAYLOADS]; - - CHANNEL_MAPPING *cm = &hAacEnc->channelMapping; - - - - PSY_OUT *psyOut = hAacEnc->psyOut[c]; - QC_OUT *qcOut = hAacEnc->qcOut[c]; - - FDKmemclear(extPayloadUsed, MAX_TOTAL_EXT_PAYLOADS * sizeof(UCHAR)); - - qcOut->elementExtBits = 0; /* sum up all extended bit of each element */ - qcOut->staticBits = 0; /* sum up side info bits of each element */ - qcOut->totalNoRedPe = 0; /* sum up PE */ - - /* advance psychoacoustics */ - for (el=0; elnElements; el++) { - ELEMENT_INFO elInfo = cm->elInfo[el]; - - if ( (elInfo.elType == ID_SCE) - || (elInfo.elType == ID_CPE) - || (elInfo.elType == ID_LFE) ) - { - int ch; - - /* update pointer!*/ - for(ch=0;chpsyOutElement[el]->psyOutChannel[ch]; - QC_OUT_CHANNEL *qcOutChan = qcOut->qcElement[el]->qcOutChannel[ch]; - - psyOutChan->mdctSpectrum = qcOutChan->mdctSpectrum; - psyOutChan->sfbSpreadEnergy = qcOutChan->sfbSpreadEnergy; - psyOutChan->sfbEnergy = qcOutChan->sfbEnergy; - psyOutChan->sfbEnergyLdData = qcOutChan->sfbEnergyLdData; - psyOutChan->sfbMinSnrLdData = qcOutChan->sfbMinSnrLdData; - psyOutChan->sfbThresholdLdData = qcOutChan->sfbThresholdLdData; - - } - - FDKaacEnc_psyMain(elInfo.nChannelsInEl, - hAacEnc->psyKernel->psyElement[el], - hAacEnc->psyKernel->psyDynamic, - hAacEnc->psyKernel->psyConf, - psyOut->psyOutElement[el], - inputBuffer, - cm->elInfo[el].ChannelIndex, - cm->nChannels - - ); - - /* FormFactor, Pe and staticBitDemand calculation */ - ErrorStatus = FDKaacEnc_QCMainPrepare(&elInfo, - hAacEnc->qcKernel->hAdjThr->adjThrStateElem[el], - psyOut->psyOutElement[el], - qcOut->qcElement[el], - hAacEnc->aot, - hAacEnc->config->syntaxFlags, - hAacEnc->config->epConfig); - - if (ErrorStatus != AAC_ENC_OK) - return ErrorStatus; - - /*-------------------------------------------- */ - - qcOut->qcElement[el]->extBitsUsed = 0; - qcOut->qcElement[el]->nExtensions = 0; - /* reset extension payload */ - FDKmemclear(&qcOut->qcElement[el]->extension, (1)*sizeof(QC_OUT_EXTENSION)); - - for ( n = 0; n < MAX_TOTAL_EXT_PAYLOADS; n++ ) { - if ( !extPayloadUsed[n] - && (extPayload[n].associatedChElement == el) - && (extPayload[n].dataSize > 0) - && (extPayload[n].pData != NULL) ) - { - int idx = qcOut->qcElement[el]->nExtensions++; - - qcOut->qcElement[el]->extension[idx].type = extPayload[n].dataType; /* Perform a sanity check on the type? */ - qcOut->qcElement[el]->extension[idx].nPayloadBits = extPayload[n].dataSize; - qcOut->qcElement[el]->extension[idx].pPayload = extPayload[n].pData; - /* Now ask the bitstream encoder how many bits we need to encode the data with the current bitstream syntax: */ - qcOut->qcElement[el]->extBitsUsed += - FDKaacEnc_writeExtensionData( NULL, - &qcOut->qcElement[el]->extension[idx], - 0, 0, - hAacEnc->config->syntaxFlags, - hAacEnc->aot, - hAacEnc->config->epConfig ); - extPayloadUsed[n] = 1; - } - } - - /* sum up extension and static bits for all channel elements */ - qcOut->elementExtBits += qcOut->qcElement[el]->extBitsUsed; - qcOut->staticBits += qcOut->qcElement[el]->staticBitsUsed; - - /* sum up pe */ - qcOut->totalNoRedPe += qcOut->qcElement[el]->peData.pe; - } - } - - qcOut->nExtensions = 0; - qcOut->globalExtBits = 0; - - /* reset extension payload */ - FDKmemclear(&qcOut->extension, (2+2)*sizeof(QC_OUT_EXTENSION)); - - /* Add extension payload not assigned to an channel element - (Ancillary data is the only supported type up to now) */ - for ( n = 0; n < MAX_TOTAL_EXT_PAYLOADS; n++ ) { - if ( !extPayloadUsed[n] - && (extPayload[n].associatedChElement == -1) - && (extPayload[n].pData != NULL) ) - { - UINT payloadBits = 0; - - if (extPayload[n].dataType == EXT_DATA_ELEMENT) { - if (hAacEnc->ancillaryBitsPerFrame) { - /* granted frame dse bitrate */ - payloadBits = hAacEnc->ancillaryBitsPerFrame; - } - else { - /* write anc data if bitrate constraint fulfilled */ - if ((extPayload[n].dataSize>>3) <= hAacEnc->config->maxAncBytesPerAU) { - payloadBits = extPayload[n].dataSize; - } - } - payloadBits = fixMin( extPayload[n].dataSize, payloadBits ); - } else { - payloadBits = extPayload[n].dataSize; - } - - if (payloadBits > 0) - { - int idx = qcOut->nExtensions++; - - qcOut->extension[idx].type = extPayload[n].dataType; /* Perform a sanity check on the type? */ - qcOut->extension[idx].nPayloadBits = payloadBits; - qcOut->extension[idx].pPayload = extPayload[n].pData; - /* Now ask the bitstream encoder how many bits we need to encode the data with the current bitstream syntax: */ - qcOut->globalExtBits += FDKaacEnc_writeExtensionData( NULL, - &qcOut->extension[idx], - 0, 0, - hAacEnc->config->syntaxFlags, - hAacEnc->aot, - hAacEnc->config->epConfig ); - if (extPayload[n].dataType == EXT_DATA_ELEMENT) { - /* substract the processed bits */ - extPayload[n].dataSize -= payloadBits; - } - extPayloadUsed[n] = 1; - } - } - } - - if (!(hAacEnc->config->syntaxFlags & (AC_SCALABLE|AC_ER))) { - qcOut->globalExtBits += EL_ID_BITS; /* add bits for ID_END */ - } - - /* build bitstream all nSubFrames */ - { - INT totalBits = 0; /* Total AU bits */; - INT avgTotalBits = 0; - - /*-------------------------------------------- */ - /* Get average total bits */ - /*-------------------------------------------- */ - { - /* frame wise bitrate adaption */ - FDKaacEnc_AdjustBitrate(hAacEnc->qcKernel, - cm, - &avgTotalBits, - hAacEnc->config->bitRate, - hAacEnc->config->sampleRate, - hAacEnc->config->framelength); - - /* adjust super frame bitrate */ - avgTotalBits *= hAacEnc->config->nSubFrames; - } - - /* Make first estimate of transport header overhead. - Take maximum possible frame size into account to prevent bitreservoir underrun. */ - hAacEnc->qcKernel->globHdrBits = transportEnc_GetStaticBits(hTpEnc, avgTotalBits + hAacEnc->qcKernel->bitResTot); - - - /*-------------------------------------------- */ - /*-------------------------------------------- */ - /*-------------------------------------------- */ - - ErrorStatus = FDKaacEnc_QCMain(hAacEnc->qcKernel, - hAacEnc->psyOut, - hAacEnc->qcOut, - avgTotalBits, - cm - ,hAacEnc->aot, - hAacEnc->config->syntaxFlags, - hAacEnc->config->epConfig); - - if (ErrorStatus != AAC_ENC_OK) - return ErrorStatus; - /*-------------------------------------------- */ - - /*-------------------------------------------- */ - ErrorStatus = FDKaacEnc_updateFillBits(cm, - hAacEnc->qcKernel, - hAacEnc->qcKernel->elementBits, - hAacEnc->qcOut); - if (ErrorStatus != AAC_ENC_OK) - return ErrorStatus; - - /*-------------------------------------------- */ - ErrorStatus = FDKaacEnc_FinalizeBitConsumption(cm, - hAacEnc->qcKernel, - qcOut, - qcOut->qcElement, - hTpEnc, - hAacEnc->aot, - hAacEnc->config->syntaxFlags, - hAacEnc->config->epConfig); - if (ErrorStatus != AAC_ENC_OK) - return ErrorStatus; - /*-------------------------------------------- */ - totalBits += qcOut->totalBits; - - - /*-------------------------------------------- */ - FDKaacEnc_updateBitres(cm, - hAacEnc->qcKernel, - hAacEnc->qcOut); - - /*-------------------------------------------- */ - - /* for ( all sub frames ) ... */ - /* write bitstream header */ - transportEnc_WriteAccessUnit( - hTpEnc, - totalBits, - FDKaacEnc_EncBitresToTpBitres(hAacEnc), - cm->nChannelsEff); - - /* write bitstream */ - ErrorStatus = FDKaacEnc_WriteBitstream( - hTpEnc, - cm, - qcOut, - psyOut, - hAacEnc->qcKernel, - hAacEnc->aot, - hAacEnc->config->syntaxFlags, - hAacEnc->config->epConfig); - - if (ErrorStatus != AAC_ENC_OK) - return ErrorStatus; - - /* transportEnc_EndAccessUnit() is being called inside FDKaacEnc_WriteBitstream() */ - transportEnc_GetFrame(hTpEnc, nOutBytes); - - } /* -end- if (curFrame==hAacEnc->qcKernel->nSubFrames) */ - - - /*-------------------------------------------- */ - return AAC_ENC_OK; -} - -/*--------------------------------------------------------------------------- - - functionname:FDKaacEnc_Close - description: delete encoder instance - returns: - - ---------------------------------------------------------------------------*/ - -void FDKaacEnc_Close( HANDLE_AAC_ENC* phAacEnc) /* encoder handle */ -{ - if (*phAacEnc == NULL) { - return; - } - AAC_ENC *hAacEnc = (AAC_ENC*)*phAacEnc; - - if (hAacEnc->dynamic_RAM != NULL) - FreeAACdynamic_RAM(&hAacEnc->dynamic_RAM); - - FDKaacEnc_PsyClose(&hAacEnc->psyKernel,hAacEnc->psyOut); - - FDKaacEnc_QCClose(&hAacEnc->qcKernel, hAacEnc->qcOut); - - FreeRam_aacEnc_AacEncoder(phAacEnc); -} - - -/* The following functions are in this source file only for convenience and */ -/* need not be visible outside of a possible encoder library. */ - -/* basic defines for ancillary data */ -#define MAX_ANCRATE 19200 /* ancillary rate >= 19200 isn't valid */ - -/*--------------------------------------------------------------------------- - - functionname: FDKaacEnc_InitCheckAncillary - description: initialize and check ancillary data struct - return: if success or NULL if error - - ---------------------------------------------------------------------------*/ -static AAC_ENCODER_ERROR FDKaacEnc_InitCheckAncillary(INT bitRate, - INT framelength, - INT ancillaryRate, - INT *ancillaryBitsPerFrame, - INT sampleRate) -{ - INT diffToByteAlign; - - /* don't use negative ancillary rates */ - if ( ancillaryRate < -1 ) - return AAC_ENC_UNSUPPORTED_ANC_BITRATE; - - /* check if ancillary rate is ok */ - if ( (ancillaryRate != (-1)) && (ancillaryRate != 0) ) { - /* ancRate <= 15% of bitrate && ancRate < 19200 */ - if ( ( ancillaryRate >= MAX_ANCRATE ) || - ( (ancillaryRate * 20) > (bitRate * 3) ) ) { - return AAC_ENC_UNSUPPORTED_ANC_BITRATE; - } - } - else if (ancillaryRate == -1) { - /* if no special ancRate is requested but a ancillary file is - stated, then generate a ancillary rate matching to the bitrate */ - if (bitRate >= (MAX_ANCRATE * 10)) { - /* ancillary rate is 19199 */ - ancillaryRate = (MAX_ANCRATE - 1); - } - else { /* 10% of bitrate */ - ancillaryRate = bitRate / 10; - } - } - - /* make ancillaryBitsPerFrame byte align */ - *ancillaryBitsPerFrame = (ancillaryRate * framelength ) / sampleRate; - diffToByteAlign = *ancillaryBitsPerFrame % 8; - *ancillaryBitsPerFrame = *ancillaryBitsPerFrame - diffToByteAlign; - - return AAC_ENC_OK; -} diff --git a/libAACenc/src/aacenc.h b/libAACenc/src/aacenc.h deleted file mode 100644 index dd09ed9..0000000 --- a/libAACenc/src/aacenc.h +++ /dev/null @@ -1,360 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/************************* Fast MPEG AAC Audio Encoder ********************** - - Initial author: M. Schug / A. Groeschel - contents/description: fast aac coder interface library functions - -******************************************************************************/ - -#ifndef _aacenc_h_ -#define _aacenc_h_ - -#include "common_fix.h" -#include "FDK_audio.h" - -#include "tpenc_lib.h" - -#include "sbr_encoder.h" - -#define BITRES_MAX_LD 4000 -#define BITRES_MIN_LD 500 -#define BITRATE_MAX_LD 70000 /* Max assumed bitrate for bitres calculation */ -#define BITRATE_MIN_LD 12000 /* Min assumed bitrate for bitres calculation */ - -#ifdef __cplusplus -extern "C" { -#endif - -/* - * AAC-LC error codes. - */ -typedef enum { - AAC_ENC_OK = 0x0000, /*!< All fine. */ - - AAC_ENC_UNKNOWN = 0x0002, /*!< Error condition is of unknown reason, or from another module. */ - - /* initialization errors */ - aac_enc_init_error_start = 0x2000, - AAC_ENC_INVALID_HANDLE = 0x2020, /*!< The handle passed to the function call was invalid (probably NULL). */ - AAC_ENC_INVALID_FRAME_LENGTH = 0x2080, /*!< Invalid frame length. */ - AAC_ENC_INVALID_N_CHANNELS = 0x20e0, /*!< Invalid amount of audio input channels. */ - AAC_ENC_INVALID_SFB_TABLE = 0x2140, /*!< Internal encoder error. */ - - AAC_ENC_UNSUPPORTED_AOT = 0x3000, /*!< The Audio Object Type (AOT) is not supported. */ - AAC_ENC_UNSUPPORTED_BITRATE = 0x3020, /*!< The chosen bitrate is not supported. */ - AAC_ENC_UNSUPPORTED_BITRATE_MODE = 0x3028, /*!< Unsupported bit rate mode (CBR or VBR). */ - AAC_ENC_UNSUPPORTED_ANC_BITRATE = 0x3040, /*!< Unsupported ancillay bitrate. */ - AAC_ENC_UNSUPPORTED_ANC_MODE = 0x3060, - AAC_ENC_UNSUPPORTED_TRANSPORT_TYPE = 0x3080, /*!< The bitstream format is not supported. */ - AAC_ENC_UNSUPPORTED_ER_FORMAT = 0x30a0, /*!< The error resilience tool format is not supported. */ - AAC_ENC_UNSUPPORTED_EPCONFIG = 0x30c0, /*!< The error protection format is not supported. */ - AAC_ENC_UNSUPPORTED_CHANNELCONFIG = 0x30e0, /*!< The channel configuration (either number or arrangement) is not supported. */ - AAC_ENC_UNSUPPORTED_SAMPLINGRATE = 0x3100, /*!< Sample rate of audio input is not supported. */ - AAC_ENC_NO_MEMORY = 0x3120, /*!< Could not allocate memory. */ - AAC_ENC_PE_INIT_TABLE_NOT_FOUND = 0x3140, /*!< Internal encoder error. */ - - aac_enc_init_error_end, - - /* encode errors */ - aac_enc_error_start = 0x4000, - AAC_ENC_QUANT_ERROR = 0x4020, /*!< Too many bits used in quantization. */ - AAC_ENC_WRITTEN_BITS_ERROR = 0x4040, /*!< Unexpected number of written bits, differs to - calculated number of bits. */ - AAC_ENC_PNS_TABLE_ERROR = 0x4060, /*!< PNS level out of range. */ - AAC_ENC_GLOBAL_GAIN_TOO_HIGH = 0x4080, /*!< Internal quantizer error. */ - AAC_ENC_BITRES_TOO_LOW = 0x40a0, /*!< Too few bits in bit reservoir. */ - AAC_ENC_BITRES_TOO_HIGH = 0x40a1, /*!< Too many bits in bit reservoir. */ - AAC_ENC_INVALID_CHANNEL_BITRATE = 0x4100, - AAC_ENC_INVALID_ELEMENTINFO_TYPE = 0x4120, /*!< Internal encoder error. */ - - AAC_ENC_WRITE_SCAL_ERROR = 0x41e0, /*!< Error writing scalefacData. */ - AAC_ENC_WRITE_SEC_ERROR = 0x4200, /*!< Error writing sectionData. */ - AAC_ENC_WRITE_SPEC_ERROR = 0x4220, /*!< Error writing spectralData. */ - aac_enc_error_end - -} AAC_ENCODER_ERROR; -/*-------------------------- defines --------------------------------------*/ - -#define ANC_DATA_BUFFERSIZE 1024 /* ancBuffer size */ - -#define MAX_TOTAL_EXT_PAYLOADS (((8) * (1)) + (2+2)) - - -typedef enum { - AACENC_BR_MODE_INVALID = -1, /*!< Invalid bitrate mode. */ - AACENC_BR_MODE_CBR = 0, /*!< Constant bitrate mode. */ - AACENC_BR_MODE_VBR_1 = 1, /*!< Variable bitrate mode, about 32 kbps/channel. */ - AACENC_BR_MODE_VBR_2 = 2, /*!< Variable bitrate mode, about 40 kbps/channel. */ - AACENC_BR_MODE_VBR_3 = 3, /*!< Variable bitrate mode, about 48-56 kbps/channel. */ - AACENC_BR_MODE_VBR_4 = 4, /*!< Variable bitrate mode, about 64 kbps/channel. */ - AACENC_BR_MODE_VBR_5 = 5, /*!< Variable bitrate mode, about 80-96 kbps/channel. */ - AACENC_BR_MODE_FF = 6, /*!< Fixed frame mode. */ - AACENC_BR_MODE_SFR = 7 /*!< Superframe mode. */ - -} AACENC_BITRATE_MODE; - -typedef enum { - - CH_ORDER_MPEG = 0, /*!< MPEG channel ordering (e. g. 5.1: C, L, R, SL, SR, LFE) */ - CH_ORDER_WAV, /*!< WAV fileformat channel ordering (e. g. 5.1: L, R, C, LFE, SL, SR) */ - CH_ORDER_WG4 /*!< WG4 fileformat channel ordering (e. g. 5.1: L, R, SL, SR, C, LFE) */ - -} CHANNEL_ORDER; - -/*-------------------- structure definitions ------------------------------*/ - -struct AACENC_CONFIG { - INT sampleRate; /* encoder sample rate */ - INT bitRate; /* encoder bit rate in bits/sec */ - INT ancDataBitRate; /* additional bits consumed by anc data or sbr have to be consiedered while configuration */ - - INT nSubFrames; /* number of frames in super frame (not ADTS/LATM subframes !) */ - AUDIO_OBJECT_TYPE audioObjectType; /* Audio Object Type */ - - INT averageBits; /* encoder bit rate in bits/superframe */ - INT bitrateMode; /* encoder bitrate mode (CBR/VBR) */ - INT nChannels; /* number of channels to process */ - CHANNEL_ORDER channelOrder; /* Input Channel ordering scheme. */ - INT bandWidth; /* targeted audio bandwidth in Hz */ - CHANNEL_MODE channelMode; /* encoder channel mode configuration */ - INT framelength; /* used frame size */ - - UINT syntaxFlags; /* bitstreams syntax configuration */ - SCHAR epConfig; /* error protection configuration */ - - INT anc_Rate; /* ancillary rate, 0 (disabled), -1 (default) else desired rate */ - UINT maxAncBytesPerAU; - INT minBitsPerFrame; /* minimum number of bits in AU */ - INT maxBitsPerFrame; /* maximum number of bits in AU */ - INT bitreservoir; /* size of bitreservoir */ - - INT audioMuxVersion; /* audio mux version in loas/latm transport format */ - - UINT sbrRatio; /* sbr sampling rate ratio: dual- or single-rate */ - - UCHAR useTns; /* flag: use temporal noise shaping */ - UCHAR usePns; /* flag: use perceptual noise substitution */ - UCHAR useIS; /* flag: use intensity coding */ - - UCHAR useRequant; /* flag: use afterburner */ -}; - -typedef struct { - UCHAR *pData; /* pointer to extension payload data */ - UINT dataSize; /* extension payload data size in bits */ - EXT_PAYLOAD_TYPE dataType; /* extension payload data type */ - INT associatedChElement; /* number of the channel element the data is assigned to */ -} AACENC_EXT_PAYLOAD; - -typedef struct AAC_ENC *HANDLE_AAC_ENC; - -/** - * \brief Calculate framesize in bits for given bit rate, frame length and sampling rate. - * - * \param bitRate Ttarget bitrate in bits per second. - * \param frameLength Number of audio samples in one frame. - * \param samplingRate Sampling rate in Hz. - * - * \return Framesize in bits per frame. -*/ -INT FDKaacEnc_CalcBitsPerFrame( - const INT bitRate, - const INT frameLength, - const INT samplingRate - ); - -/** - * \brief Calculate bitrate in bits per second for given framesize, frame length and sampling rate. - * - * \param bitsPerFrame Framesize in bits per frame. - * \param frameLength Number of audio samples in one frame. - * \param samplingRate Sampling rate in Hz. - * - * \return Bitrate in bits per second. -*/ -INT FDKaacEnc_CalcBitrate( - const INT bitsPerFrame, - const INT frameLength, - const INT samplingRate - ); - -/** - * \brief Limit given bit rate to a valid value - * \param hTpEnc transport encoder handle - * \param coreSamplingRate the sample rate to be used for the AAC encoder - * \param frameLength the frameLength to be used for the AAC encoder - * \param nChannels number of total channels - * \param nChannelsEff number of effective channels - * \param bitRate the initial bit rate value for which the closest valid bit rate value is searched for - * \param averageBits average bits per frame for fixed framing. Set to -1 if not available. - * \param optional pointer where the current bits per frame are stored into. - * \param bitrateMode the current bit rate mode - * \param nSubFrames number of sub frames for super framing (not transport frames). - * \return a valid bit rate value as close as possible or identical to bitRate - */ -INT FDKaacEnc_LimitBitrate( - HANDLE_TRANSPORTENC hTpEnc, - INT coreSamplingRate, - INT frameLength, - INT nChannels, - INT nChannelsEff, - INT bitRate, - INT averageBits, - INT *pAverageBitsPerFrame, - INT bitrateMode, - INT nSubFrames - ); - - /*----------------------------------------------------------------------------- - - functionname: FDKaacEnc_GetVBRBitrate - description: Get VBR bitrate from vbr quality - input params: int vbrQuality (VBR0, VBR1, VBR2) - channelMode - returns: vbr bitrate - - ------------------------------------------------------------------------------*/ - INT FDKaacEnc_GetVBRBitrate(INT bitrateMode, CHANNEL_MODE channelMode); - - -/*----------------------------------------------------------------------------- - - functionname: FDKaacEnc_AacInitDefaultConfig - description: gives reasonable default configuration - returns: --- - - ------------------------------------------------------------------------------*/ -void FDKaacEnc_AacInitDefaultConfig(AACENC_CONFIG *config); - -/*--------------------------------------------------------------------------- - - functionname:FDKaacEnc_Open - description: allocate and initialize a new encoder instance - returns: 0 if success - - ---------------------------------------------------------------------------*/ -AAC_ENCODER_ERROR FDKaacEnc_Open(HANDLE_AAC_ENC *phAacEnc, /* pointer to an encoder handle, initialized on return */ - const INT nElements, /* number of maximal elements in instance to support */ - const INT nChannels, /* number of maximal channels in instance to support */ - const INT nSubFrames); /* support superframing in instance */ - - -AAC_ENCODER_ERROR FDKaacEnc_Initialize(HANDLE_AAC_ENC hAacEncoder, /* pointer to an encoder handle, initialized on return */ - AACENC_CONFIG *config, /* pre-initialized config struct */ - HANDLE_TRANSPORTENC hTpEnc, - ULONG initFlags); - - -/*--------------------------------------------------------------------------- - - functionname: FDKaacEnc_EncodeFrame - description: encode one frame - returns: 0 if success - - ---------------------------------------------------------------------------*/ - -AAC_ENCODER_ERROR FDKaacEnc_EncodeFrame( HANDLE_AAC_ENC hAacEnc, /* encoder handle */ - HANDLE_TRANSPORTENC hTpEnc, - INT_PCM* inputBuffer, - INT* numOutBytes, - AACENC_EXT_PAYLOAD extPayload[MAX_TOTAL_EXT_PAYLOADS] - ); - -/*--------------------------------------------------------------------------- - - functionname:FDKaacEnc_Close - description: delete encoder instance - returns: - - ---------------------------------------------------------------------------*/ - -void FDKaacEnc_Close( HANDLE_AAC_ENC* phAacEnc); /* encoder handle */ - -#ifdef __cplusplus -} -#endif - -#endif /* _aacenc_h_ */ - diff --git a/libAACenc/src/aacenc_lib.cpp b/libAACenc/src/aacenc_lib.cpp deleted file mode 100644 index fc58d6d..0000000 --- a/libAACenc/src/aacenc_lib.cpp +++ /dev/null @@ -1,2139 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/**************************** MPEG-4 HE-AAC Encoder ************************* - - Initial author: M. Lohwasser - contents/description: FDK HE-AAC Encoder interface library functions - -****************************************************************************/ - -#include "aacenc_lib.h" -#include "FDK_audio.h" -#include "aacenc.h" - -#include "aacEnc_ram.h" -#include "FDK_core.h" /* FDK_tools versioning info */ - -/* Encoder library info */ -#define AACENCODER_LIB_VL0 3 -#define AACENCODER_LIB_VL1 4 -#define AACENCODER_LIB_VL2 22 -#define AACENCODER_LIB_TITLE "AAC Encoder" -#ifdef __ANDROID__ -#define AACENCODER_LIB_BUILD_DATE "" -#define AACENCODER_LIB_BUILD_TIME "" -#else -#define AACENCODER_LIB_BUILD_DATE __DATE__ -#define AACENCODER_LIB_BUILD_TIME __TIME__ -#endif - - -#include "sbr_encoder.h" -#include "../src/sbr_ram.h" -#include "channel_map.h" - -#include "psy_const.h" -#include "bitenc.h" - -#include "tpenc_lib.h" - -#include "metadata_main.h" - -#define SBL(fl) (fl/8) /*!< Short block length (hardcoded to 8 short blocks per long block) */ -#define BSLA(fl) (4*SBL(fl)+SBL(fl)/2) /*!< AAC block switching look-ahead */ -#define DELAY_AAC(fl) (fl+BSLA(fl)) /*!< MDCT + blockswitching */ -#define DELAY_AACELD(fl) ((fl)/2) /*!< ELD FB delay (no framing delay included) */ - -#define INPUTBUFFER_SIZE (1537+100+2048) - -#define DEFAULT_HEADER_PERIOD_REPETITION_RATE 10 /*!< Default header repetition rate used in transport library and for SBR header. */ - -//////////////////////////////////////////////////////////////////////////////////// -/** - * Flags to characterize encoder modules to be supported in present instance. - */ -enum { - ENC_MODE_FLAG_AAC = 0x0001, - ENC_MODE_FLAG_SBR = 0x0002, - ENC_MODE_FLAG_PS = 0x0004, - ENC_MODE_FLAG_SAC = 0x0008, - ENC_MODE_FLAG_META = 0x0010 -}; - -//////////////////////////////////////////////////////////////////////////////////// -typedef struct { - AUDIO_OBJECT_TYPE userAOT; /*!< Audio Object Type. */ - UINT userSamplerate; /*!< Sampling frequency. */ - UINT nChannels; /*!< will be set via channelMode. */ - CHANNEL_MODE userChannelMode; - UINT userBitrate; - UINT userBitrateMode; - UINT userBandwidth; - UINT userAfterburner; - UINT userFramelength; - UINT userAncDataRate; - UINT userPeakBitrate; - - UCHAR userTns; /*!< Use TNS coding. */ - UCHAR userPns; /*!< Use PNS coding. */ - UCHAR userIntensity; /*!< Use Intensity coding. */ - - TRANSPORT_TYPE userTpType; /*!< Transport type */ - UCHAR userTpSignaling; /*!< Extension AOT signaling mode. */ - UCHAR userTpNsubFrames; /*!< Number of sub frames in a transport frame for LOAS/LATM or ADTS (default 1). */ - UCHAR userTpAmxv; /*!< AudioMuxVersion to be used for LATM (default 0). */ - UCHAR userTpProtection; - UCHAR userTpHeaderPeriod; /*!< Parameter used to configure LATM/LOAS SMC rate. Moreover this parameters is - used to configure repetition rate of PCE in raw_data_block. */ - - UCHAR userErTools; /*!< Use VCB11, HCR and/or RVLC ER tool. */ - UINT userPceAdditions; /*!< Configure additional bits in PCE. */ - - UCHAR userMetaDataMode; /*!< Meta data library configuration. */ - - UCHAR userSbrEnabled; /*!< Enable SBR for ELD. */ - UINT userSbrRatio; /*!< SBR sampling rate ratio. Dual- or single-rate. */ - -} USER_PARAM; - -//////////////////////////////////////////////////////////////////////////////////// - -/**************************************************************************** - Structure Definitions -****************************************************************************/ - -typedef struct AACENC_CONFIG *HANDLE_AACENC_CONFIG; - - -struct AACENCODER -{ - USER_PARAM extParam; - CODER_CONFIG coderConfig; - - /* AAC */ - AACENC_CONFIG aacConfig; - HANDLE_AAC_ENC hAacEnc; - - /* SBR */ - HANDLE_SBR_ENCODER hEnvEnc; - - /* Meta Data */ - HANDLE_FDK_METADATA_ENCODER hMetadataEnc; - INT metaDataAllowed; /* Signal whether chosen configuration allows metadata. Necessary for delay - compensation. Metadata mode is a separate parameter. */ - - /* Transport */ - HANDLE_TRANSPORTENC hTpEnc; - - /* Output */ - UCHAR *outBuffer; /* Internal bitstream buffer */ - INT outBufferInBytes; /* Size of internal bitstream buffer*/ - - /* Input */ - INT_PCM *inputBuffer; /* Internal input buffer. Input source for AAC encoder */ - INT inputBufferOffset; /* Where to write new input samples. */ - - INT nSamplesToRead; /* number of input samples neeeded for encoding one frame */ - INT nSamplesRead; /* number of input samples already in input buffer */ - INT nZerosAppended; /* appended zeros at end of file*/ - INT nDelay; /* encoder delay */ - - AACENC_EXT_PAYLOAD extPayload [MAX_TOTAL_EXT_PAYLOADS]; - /* Extension payload */ - UCHAR extPayloadData [(1)][(8)][MAX_PAYLOAD_SIZE]; - UINT extPayloadSize [(1)][(8)]; /* payload sizes in bits */ - - ULONG InitFlags; /* internal status to treggier re-initialization */ - - - /* Memory allocation info. */ - INT nMaxAacElements; - INT nMaxAacChannels; - INT nMaxSbrElements; - INT nMaxSbrChannels; - UINT nMaxSubFrames; - - UINT encoder_modis; - - /* Capability flags */ - UINT CAPF_tpEnc; - -} ; - -typedef struct -{ - ULONG samplingRate; /*!< Encoder output sampling rate. */ - ULONG bitrateRange; /*!< Lower bitrate range for config entry. */ - - UCHAR lowDelaySbr; /*!< 0: ELD sbr off, - 1: ELD sbr on */ - - UCHAR downsampledSbr; /*!< 0: ELD with dualrate sbr, - 1: ELD with downsampled sbr */ - -} ELD_SBR_CONFIGURATOR; - -/** - * \brief This table defines ELD/SBR default configurations. - */ -static const ELD_SBR_CONFIGURATOR eldSbrAutoConfigTab[] = -{ - { 48000, 0, 1, 0 }, - { 48000, 64001, 0, 0 }, - - { 44100, 0, 1, 0 }, - { 44100, 64001, 0, 0 }, - - { 32000, 0, 1, 0 }, - { 32000, 28000, 1, 1 }, - { 32000, 56000, 0, 0 }, - - { 24000, 0, 1, 1 }, - { 24000, 40000, 0, 0 }, - - { 16000, 0, 1, 1 }, - { 16000, 28000, 0, 0 } - -}; - -/* - * \brief Configure SBR for ELD configuration. - * - * This function finds default SBR configuration for ELD based on sampling rate and channel bitrate. - * Outputparameters are SBR on/off, and SBR ratio. - * - * \param samplingRate Audio signal sampling rate. - * \param channelMode Channel configuration to be used. - * \param totalBitrate Overall bitrate. - * \param eldSbr Pointer to eldSbr parameter, filled on return. - * \param eldSbrRatio Pointer to eldSbrRatio parameter, filled on return. - * - * \return - AACENC_OK, all fine. - * - AACENC_INVALID_CONFIG, on failure. - */ -static AACENC_ERROR eldSbrConfigurator( - const ULONG samplingRate, - const CHANNEL_MODE channelMode, - const ULONG totalBitrate, - UINT * const eldSbr, - UINT * const eldSbrRatio - ) -{ - AACENC_ERROR err = AACENC_OK; - int i, cfgIdx = -1; - const ULONG channelBitrate = totalBitrate / FDKaacEnc_GetChannelModeConfiguration(channelMode)->nChannelsEff; - - for (i=0; i<(int)(sizeof(eldSbrAutoConfigTab)/sizeof(ELD_SBR_CONFIGURATOR)); i++) { - if ( (samplingRate <= eldSbrAutoConfigTab[i].samplingRate) - && (channelBitrate >= eldSbrAutoConfigTab[i].bitrateRange) ) - { - cfgIdx = i; - } - } - - if (cfgIdx != -1) { - *eldSbr = (eldSbrAutoConfigTab[cfgIdx].lowDelaySbr==0) ? 0 : 1; - *eldSbrRatio = (eldSbrAutoConfigTab[cfgIdx].downsampledSbr==0) ? 2 : 1; - } - else { - err = AACENC_INVALID_CONFIG; /* no default configuration for eld-sbr available. */ - } - - return err; -} - -static inline INT isSbrActive(const HANDLE_AACENC_CONFIG hAacConfig) -{ - INT sbrUsed = 0; - - if ( (hAacConfig->audioObjectType==AOT_SBR) || (hAacConfig->audioObjectType==AOT_PS) ) - { - sbrUsed = 1; - } - if (hAacConfig->audioObjectType == AOT_ER_AAC_ELD && (hAacConfig->syntaxFlags & AC_SBR_PRESENT)) - { - sbrUsed = 1; - } - - return ( sbrUsed ); -} - -static inline INT isPsActive(const AUDIO_OBJECT_TYPE audioObjectType) -{ - INT psUsed = 0; - - if ( (audioObjectType==AOT_PS) ) - { - psUsed = 1; - } - - return ( psUsed ); -} - -static SBR_PS_SIGNALING getSbrSignalingMode( - const AUDIO_OBJECT_TYPE audioObjectType, - const TRANSPORT_TYPE transportType, - const UCHAR transportSignaling, - const UINT sbrRatio - ) - -{ - SBR_PS_SIGNALING sbrSignaling; - - if (transportType==TT_UNKNOWN || sbrRatio==0) { - sbrSignaling = SIG_UNKNOWN; /* Needed parameters have not been set */ - return sbrSignaling; - } else { - sbrSignaling = SIG_IMPLICIT; /* default: implicit signaling */ - } - - if ( (audioObjectType==AOT_AAC_LC) || (audioObjectType==AOT_SBR) || (audioObjectType==AOT_PS) ) { - switch (transportType) { - case TT_MP4_ADIF: - case TT_MP4_ADTS: - sbrSignaling = SIG_IMPLICIT; /* For MPEG-2 transport types, only implicit signaling is possible */ - break; - - case TT_MP4_RAW: - case TT_MP4_LATM_MCP1: - case TT_MP4_LATM_MCP0: - case TT_MP4_LOAS: - default: - if ( transportSignaling==0xFF ) { - /* Defaults */ - if ( sbrRatio==1 ) { - sbrSignaling = SIG_EXPLICIT_HIERARCHICAL; /* For downsampled SBR, explicit signaling is mandatory */ - } else { - sbrSignaling = SIG_IMPLICIT; /* For dual-rate SBR, implicit signaling is default */ - } - } else { - /* User set parameters */ - /* Attention: Backward compatible explicit signaling does only work with AMV1 for LATM/LOAS */ - sbrSignaling = (SBR_PS_SIGNALING)transportSignaling; - } - break; - } - } - - return sbrSignaling; -} - -/**************************************************************************** - Allocate Encoder -****************************************************************************/ - -H_ALLOC_MEM (_AacEncoder, AACENCODER) -C_ALLOC_MEM (_AacEncoder, AACENCODER, 1) - - - - -/* - * Map Encoder specific config structures to CODER_CONFIG. - */ -static void FDKaacEnc_MapConfig( - CODER_CONFIG *const cc, - const USER_PARAM *const extCfg, - const SBR_PS_SIGNALING sbrSignaling, - const HANDLE_AACENC_CONFIG hAacConfig - ) -{ - AUDIO_OBJECT_TYPE transport_AOT = AOT_NULL_OBJECT; - FDKmemclear(cc, sizeof(CODER_CONFIG)); - - cc->flags = 0; - - transport_AOT = hAacConfig->audioObjectType; - - if (hAacConfig->audioObjectType == AOT_ER_AAC_ELD) { - cc->flags |= (hAacConfig->syntaxFlags & AC_SBR_PRESENT) ? CC_SBR : 0; - } - - /* transport type is usually AAC-LC. */ - if ( (transport_AOT == AOT_SBR) || (transport_AOT == AOT_PS) ) { - cc->aot = AOT_AAC_LC; - } - else { - cc->aot = transport_AOT; - } - - /* Configure extension aot. */ - if (sbrSignaling==SIG_IMPLICIT) { - cc->extAOT = AOT_NULL_OBJECT; /* implicit */ - } - else { - if ( (sbrSignaling==SIG_EXPLICIT_BW_COMPATIBLE) && ( (transport_AOT==AOT_SBR) || (transport_AOT==AOT_PS) ) ) { - cc->extAOT = AOT_SBR; /* explicit backward compatible */ - } - else { - cc->extAOT = transport_AOT; /* explicit hierarchical */ - } - } - - if ( (transport_AOT==AOT_SBR) || (transport_AOT==AOT_PS) ) { - cc->sbrPresent=1; - if (transport_AOT==AOT_PS) { - cc->psPresent=1; - } - } - cc->sbrSignaling = sbrSignaling; - - cc->extSamplingRate = extCfg->userSamplerate; - cc->bitRate = hAacConfig->bitRate; - cc->noChannels = hAacConfig->nChannels; - cc->flags |= CC_IS_BASELAYER; - cc->channelMode = hAacConfig->channelMode; - - cc->nSubFrames = (hAacConfig->nSubFrames > 1 && extCfg->userTpNsubFrames == 1) - ? hAacConfig->nSubFrames - : extCfg->userTpNsubFrames; - - cc->flags |= (extCfg->userTpProtection) ? CC_PROTECTION : 0; - - if (extCfg->userTpHeaderPeriod!=0xFF) { - cc->headerPeriod = extCfg->userTpHeaderPeriod; - } - else { /* auto-mode */ - switch (extCfg->userTpType) { - case TT_MP4_ADTS: - case TT_MP4_LOAS: - case TT_MP4_LATM_MCP1: - cc->headerPeriod = DEFAULT_HEADER_PERIOD_REPETITION_RATE; - break; - default: - cc->headerPeriod = 0; - } - } - - cc->samplesPerFrame = hAacConfig->framelength; - cc->samplingRate = hAacConfig->sampleRate; - - /* Mpeg-4 signaling for transport library. */ - cc->flags |= CC_MPEG_ID; - - /* ER-tools signaling. */ - cc->flags |= (hAacConfig->syntaxFlags & AC_ER_VCB11) ? CC_VCB11 : 0; - cc->flags |= (hAacConfig->syntaxFlags & AC_ER_HCR) ? CC_HCR : 0; - cc->flags |= (hAacConfig->syntaxFlags & AC_ER_RVLC) ? CC_RVLC : 0; - - /* Matrix mixdown coefficient configuration. */ - if ( (extCfg->userPceAdditions&0x1) && (hAacConfig->epConfig==-1) - && ((cc->channelMode==MODE_1_2_2)||(cc->channelMode==MODE_1_2_2_1)) ) - { - cc->matrixMixdownA = ((extCfg->userPceAdditions>>1)&0x3)+1; - cc->flags |= (extCfg->userPceAdditions>>3)&0x1 ? CC_PSEUDO_SURROUND : 0; - } - else { - cc->matrixMixdownA = 0; - } -} - -/* - * Examine buffer descriptor regarding choosen identifier. - * - * \param pBufDesc Pointer to buffer descriptor - * \param identifier Buffer identifier to look for. - - * \return - Buffer descriptor index. - * -1, if there is no entry available. - */ -static INT getBufDescIdx( - const AACENC_BufDesc *pBufDesc, - const AACENC_BufferIdentifier identifier -) -{ - INT i, idx = -1; - - for (i=0; inumBufs; i++) { - if ( (AACENC_BufferIdentifier)pBufDesc->bufferIdentifiers[i] == identifier ) { - idx = i; - break; - } - } - return idx; -} - - -/**************************************************************************** - Function Declarations -****************************************************************************/ - -AAC_ENCODER_ERROR aacEncDefaultConfig(HANDLE_AACENC_CONFIG hAacConfig, - USER_PARAM *config) -{ - /* make reasonable default settings */ - FDKaacEnc_AacInitDefaultConfig (hAacConfig); - - /* clear configuration structure and copy default settings */ - FDKmemclear(config, sizeof(USER_PARAM)); - - /* copy encoder configuration settings */ - config->nChannels = hAacConfig->nChannels; - config->userAOT = hAacConfig->audioObjectType = AOT_AAC_LC; - config->userSamplerate = hAacConfig->sampleRate; - config->userChannelMode = hAacConfig->channelMode; - config->userBitrate = hAacConfig->bitRate; - config->userBitrateMode = hAacConfig->bitrateMode; - config->userPeakBitrate = (UINT)-1; - config->userBandwidth = hAacConfig->bandWidth; - config->userTns = hAacConfig->useTns; - config->userPns = hAacConfig->usePns; - config->userIntensity = hAacConfig->useIS; - config->userAfterburner = hAacConfig->useRequant; - config->userFramelength = (UINT)-1; - - if (hAacConfig->syntaxFlags & AC_ER_VCB11) { - config->userErTools |= 0x01; - } - if (hAacConfig->syntaxFlags & AC_ER_HCR) { - config->userErTools |= 0x02; - } - - /* initialize transport parameters */ - config->userTpType = TT_UNKNOWN; - config->userTpAmxv = 0; - config->userTpSignaling = 0xFF; /* choose signaling automatically */ - config->userTpNsubFrames = 1; - config->userTpProtection = 0; /* not crc protected*/ - config->userTpHeaderPeriod = 0xFF; /* header period in auto mode */ - config->userPceAdditions = 0; /* no matrix mixdown coefficient */ - config->userMetaDataMode = 0; /* do not embed any meta data info */ - - config->userAncDataRate = 0; - - /* SBR rate is set to 0 here, which means it should be set automatically - in FDKaacEnc_AdjustEncSettings() if the user did not set a rate - expilicitely. */ - config->userSbrRatio = 0; - - /* SBR enable set to -1 means to inquire ELD audio configurator for reasonable configuration. */ - config->userSbrEnabled = -1; - - return AAC_ENC_OK; -} - -static -void aacEncDistributeSbrBits(CHANNEL_MAPPING *channelMapping, SBR_ELEMENT_INFO *sbrElInfo, INT bitRate) -{ - INT codebits = bitRate; - int el; - - /* Copy Element info */ - for (el=0; elnElements; el++) { - sbrElInfo[el].ChannelIndex[0] = channelMapping->elInfo[el].ChannelIndex[0]; - sbrElInfo[el].ChannelIndex[1] = channelMapping->elInfo[el].ChannelIndex[1]; - sbrElInfo[el].elType = channelMapping->elInfo[el].elType; - sbrElInfo[el].bitRate = (INT)(fMultNorm(channelMapping->elInfo[el].relativeBits, (FIXP_DBL)bitRate)); - sbrElInfo[el].instanceTag = channelMapping->elInfo[el].instanceTag; - sbrElInfo[el].nChannelsInEl = channelMapping->elInfo[el].nChannelsInEl; - - codebits -= sbrElInfo[el].bitRate; - } - sbrElInfo[0].bitRate += codebits; -} - - -static -INT aacEncoder_LimitBitrate( - const HANDLE_TRANSPORTENC hTpEnc, - const INT samplingRate, - const INT frameLength, - const INT nChannels, - const CHANNEL_MODE channelMode, - INT bitRate, - const INT nSubFrames, - const INT sbrActive, - const INT sbrDownSampleRate, - const AUDIO_OBJECT_TYPE aot - ) -{ - INT coreSamplingRate; - CHANNEL_MAPPING cm; - - FDKaacEnc_InitChannelMapping(channelMode, CH_ORDER_MPEG, &cm); - - if (sbrActive) { - coreSamplingRate = samplingRate >> (sbrEncoder_IsSingleRatePossible(aot) ? (sbrDownSampleRate-1):1); - } else { - coreSamplingRate = samplingRate; - } - - /* Consider bandwidth channel bit rate limit (see bandwidth.cpp: GetBandwidthEntry()) */ - if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD) { - bitRate = FDKmin(360000*nChannels, bitRate); - bitRate = FDKmax(8000*nChannels, bitRate); - } - - if (aot == AOT_AAC_LC || aot == AOT_SBR || aot == AOT_PS) { - bitRate = FDKmin(576000*nChannels, bitRate); - /*bitRate = FDKmax(0*nChannels, bitRate);*/ - } - - - /* Limit bit rate in respect to the core coder */ - bitRate = FDKaacEnc_LimitBitrate( - hTpEnc, - coreSamplingRate, - frameLength, - nChannels, - cm.nChannelsEff, - bitRate, - -1, - NULL, - -1, - nSubFrames - ); - - /* Limit bit rate in respect to available SBR modes if active */ - if (sbrActive) - { - int numIterations = 0; - INT initialBitrate, adjustedBitrate; - initialBitrate = adjustedBitrate = bitRate; - - /* Find total bitrate which provides valid configuration for each SBR element. */ - do { - int e; - SBR_ELEMENT_INFO sbrElInfo[(8)]; - FDK_ASSERT(cm.nElements <= (8)); - - initialBitrate = adjustedBitrate; - - /* Get bit rate for each SBR element */ - aacEncDistributeSbrBits(&cm, sbrElInfo, initialBitrate); - - for (e=0; e sbrBitRateOut) { - adjustedBitrate = fMin(initialBitrate, (INT)fDivNorm((FIXP_DBL)(sbrBitRateOut-8), cm.elInfo[e].relativeBits)); - break; - } - - } /* sbrElementBitRateIn != sbrBitRateOut */ - - } /* elements */ - - numIterations++; /* restrict iteration to worst case of num elements */ - - } while ( (initialBitrate!=adjustedBitrate) && (numIterations<=cm.nElements) ); - - /* Unequal bitrates mean that no reasonable bitrate configuration found. */ - bitRate = (initialBitrate==adjustedBitrate) ? adjustedBitrate : 0; - } - - FDK_ASSERT(bitRate > 0); - - return bitRate; -} - -/* - * \brief Consistency check of given USER_PARAM struct and - * copy back configuration from public struct into internal - * encoder configuration struct. - * - * \hAacEncoder Internal encoder config which is to be updated - * \param config User provided config (public struct) - * \return ´returns always AAC_ENC_OK - */ -static -AACENC_ERROR FDKaacEnc_AdjustEncSettings(HANDLE_AACENCODER hAacEncoder, - USER_PARAM *config) -{ - AACENC_ERROR err = AACENC_OK; - - /* Get struct pointers. */ - HANDLE_AACENC_CONFIG hAacConfig = &hAacEncoder->aacConfig; - - hAacConfig->nChannels = config->nChannels; - - /* Encoder settings update. */ - hAacConfig->sampleRate = config->userSamplerate; - hAacConfig->useTns = config->userTns; - hAacConfig->usePns = config->userPns; - hAacConfig->useIS = config->userIntensity; - hAacConfig->bitRate = config->userBitrate; - hAacConfig->channelMode = config->userChannelMode; - hAacConfig->bitrateMode = config->userBitrateMode; - hAacConfig->bandWidth = config->userBandwidth; - hAacConfig->useRequant = config->userAfterburner; - - hAacConfig->audioObjectType = config->userAOT; - hAacConfig->anc_Rate = config->userAncDataRate; - hAacConfig->syntaxFlags = 0; - hAacConfig->epConfig = -1; - - if (config->userTpType==TT_MP4_LATM_MCP1 || config->userTpType==TT_MP4_LATM_MCP0 || config->userTpType==TT_MP4_LOAS) { - hAacConfig->audioMuxVersion = config->userTpAmxv; - } - else { - hAacConfig->audioMuxVersion = -1; - } - - /* Adapt internal AOT when necessary. */ - switch ( hAacConfig->audioObjectType ) { - case AOT_AAC_LC: - case AOT_SBR: - case AOT_PS: - config->userTpType = (config->userTpType!=TT_UNKNOWN) ? config->userTpType : TT_MP4_ADTS; - hAacConfig->framelength = (config->userFramelength!=(UINT)-1) ? config->userFramelength : 1024; - if (hAacConfig->framelength != 1024) { - return AACENC_INVALID_CONFIG; - } - break; - case AOT_ER_AAC_LD: - hAacConfig->epConfig = 0; - hAacConfig->syntaxFlags |= AC_ER|AC_LD; - hAacConfig->syntaxFlags |= ((config->userErTools & 0x1) ? AC_ER_VCB11 : 0); - hAacConfig->syntaxFlags |= ((config->userErTools & 0x2) ? AC_ER_HCR : 0); - hAacConfig->syntaxFlags |= ((config->userErTools & 0x4) ? AC_ER_RVLC : 0); - config->userTpType = (config->userTpType!=TT_UNKNOWN) ? config->userTpType : TT_MP4_LOAS; - hAacConfig->framelength = (config->userFramelength!=(UINT)-1) ? config->userFramelength : 512; - if (hAacConfig->framelength != 512 && hAacConfig->framelength != 480) { - return AACENC_INVALID_CONFIG; - } - break; - case AOT_ER_AAC_ELD: - hAacConfig->epConfig = 0; - hAacConfig->syntaxFlags |= AC_ER|AC_ELD; - hAacConfig->syntaxFlags |= ((config->userErTools & 0x1) ? AC_ER_VCB11 : 0); - hAacConfig->syntaxFlags |= ((config->userErTools & 0x2) ? AC_ER_HCR : 0); - hAacConfig->syntaxFlags |= ((config->userErTools & 0x4) ? AC_ER_RVLC : 0); - hAacConfig->syntaxFlags |= ((config->userSbrEnabled==1) ? AC_SBR_PRESENT : 0); - config->userTpType = (config->userTpType!=TT_UNKNOWN) ? config->userTpType : TT_MP4_LOAS; - hAacConfig->framelength = (config->userFramelength!=(UINT)-1) ? config->userFramelength : 512; - if (hAacConfig->framelength != 512 && hAacConfig->framelength != 480) { - return AACENC_INVALID_CONFIG; - } - break; - default: - break; - } - - switch ( hAacConfig->audioObjectType ) { - case AOT_ER_AAC_LD: - case AOT_ER_AAC_ELD: - if (config->userBitrateMode==0) { - /* bitreservoir = (maxBitRes-minBitRes)/(maxBitRate-minBitrate)*(bitRate-minBitrate)+minBitRes; */ - if ( isLowDelay(hAacConfig->audioObjectType) ) { - INT bitreservoir; - INT brPerChannel = hAacConfig->bitRate/hAacConfig->nChannels; - brPerChannel = fMin(BITRATE_MAX_LD, fMax(BITRATE_MIN_LD, brPerChannel)); - FIXP_DBL slope = fDivNorm((brPerChannel-BITRATE_MIN_LD), BITRATE_MAX_LD-BITRATE_MIN_LD); /* calc slope for interpolation */ - bitreservoir = fMultI(slope, (INT)(BITRES_MAX_LD-BITRES_MIN_LD)) + BITRES_MIN_LD; /* interpolate */ - hAacConfig->bitreservoir = bitreservoir & ~7; /* align to bytes */ - } - } - if (hAacConfig->bitrateMode!=0) { - return AACENC_INVALID_CONFIG; - } - break; - default: - break; - } - - hAacConfig->bitRate = config->userBitrate; - - /* get bitrate in VBR configuration */ - if ( (hAacConfig->bitrateMode>=1) && (hAacConfig->bitrateMode<=5) ) { - /* In VBR mode; SBR-modul depends on bitrate, core encoder on bitrateMode. */ - hAacConfig->bitRate = FDKaacEnc_GetVBRBitrate(hAacConfig->bitrateMode, hAacConfig->channelMode); - } - - - - /* Set default bitrate if no external bitrate declared. */ - if ( (hAacConfig->bitrateMode==0) && (config->userBitrate==(UINT)-1) ) { - INT bitrate = FDKaacEnc_GetChannelModeConfiguration(hAacConfig->channelMode)->nChannelsEff * hAacConfig->sampleRate; - - if ( isPsActive(hAacConfig->audioObjectType) ) { - hAacConfig->bitRate = (bitrate>>1); /* 0.5 bit per sample */ - } - else if ( isSbrActive(hAacConfig) ) - { - if ( (config->userSbrRatio==2) || ((config->userSbrRatio==0)&&(hAacConfig->audioObjectType!=AOT_ER_AAC_ELD)) ) { - hAacConfig->bitRate = (bitrate + (bitrate>>2))>>1; /* 0.625 bits per sample */ - } - if ( (config->userSbrRatio==1) || ((config->userSbrRatio==0)&&(hAacConfig->audioObjectType==AOT_ER_AAC_ELD)) ) { - hAacConfig->bitRate = (bitrate + (bitrate>>3)); /* 1.125 bits per sample */ - } - } else - { - hAacConfig->bitRate = bitrate + (bitrate>>1); /* 1.5 bits per sample */ - } - } - - if ((hAacConfig->bitrateMode >= 0) && (hAacConfig->bitrateMode <= 5)) { - if ((INT)config->userPeakBitrate != -1) { - hAacConfig->maxBitsPerFrame = (FDKaacEnc_CalcBitsPerFrame(fMax(hAacConfig->bitRate, (INT)config->userPeakBitrate), hAacConfig->framelength, hAacConfig->sampleRate) + 7)&~7; - } - else { - hAacConfig->maxBitsPerFrame = -1; - } - if (hAacConfig->audioMuxVersion==2) { - hAacConfig->minBitsPerFrame = fMin(32*8, FDKaacEnc_CalcBitsPerFrame(hAacConfig->bitRate, hAacConfig->framelength, hAacConfig->sampleRate))&~7; - } - } - - /* Initialize SBR parameters */ - if ( (hAacConfig->audioObjectType==AOT_ER_AAC_ELD) - && (config->userSbrEnabled == (UCHAR)-1) && (config->userSbrRatio==0) ) - { - UINT eldSbr = 0; - UINT eldSbrRatio = 0; - - if ( AACENC_OK!=(err=eldSbrConfigurator( - hAacConfig->sampleRate, - hAacConfig->channelMode, - hAacConfig->bitRate, - &eldSbr, - &eldSbrRatio)) ) - { - return err; - } - - hAacConfig->syntaxFlags |= ((eldSbr) ? AC_SBR_PRESENT : 0); - hAacConfig->sbrRatio = eldSbrRatio; - } - else - if ( (config->userSbrRatio==0) && (isSbrActive(hAacConfig)) ) { - /* Automatic SBR ratio configuration - * - downsampled SBR for ELD - * - otherwise always dualrate SBR - */ - hAacConfig->sbrRatio = (hAacConfig->audioObjectType==AOT_ER_AAC_ELD) ? 1 : 2; - } - else { - /* SBR ratio has been set by the user, so use it. */ - hAacConfig->sbrRatio = isSbrActive(hAacConfig) ? config->userSbrRatio : 0; - } - - { - UCHAR tpSignaling=getSbrSignalingMode(hAacConfig->audioObjectType, config->userTpType, config->userTpSignaling, hAacConfig->sbrRatio); - - if ( (hAacConfig->audioObjectType==AOT_AAC_LC || hAacConfig->audioObjectType==AOT_SBR || hAacConfig->audioObjectType==AOT_PS) && - (config->userTpType==TT_MP4_LATM_MCP1 || config->userTpType==TT_MP4_LATM_MCP0 || config->userTpType==TT_MP4_LOAS) && - (tpSignaling==1) && (config->userTpAmxv==0) ) { - /* For backward compatible explicit signaling, AMV1 has to be active */ - return AACENC_INVALID_CONFIG; - } - - if ( (hAacConfig->audioObjectType==AOT_AAC_LC || hAacConfig->audioObjectType==AOT_SBR || hAacConfig->audioObjectType==AOT_PS) && - (tpSignaling==0) && (hAacConfig->sbrRatio==1)) { - /* Downsampled SBR has to be signaled explicitely (for transmission of SBR sampling fequency) */ - return AACENC_INVALID_CONFIG; - } - } - - - - /* We need the frame length to call aacEncoder_LimitBitrate() */ - hAacConfig->bitRate = aacEncoder_LimitBitrate( - NULL, - hAacConfig->sampleRate, - hAacConfig->framelength, - hAacConfig->nChannels, - hAacConfig->channelMode, - hAacConfig->bitRate, - hAacConfig->nSubFrames, - isSbrActive(hAacConfig), - hAacConfig->sbrRatio, - hAacConfig->audioObjectType - ); - - /* Configure PNS */ - if ( ((hAacConfig->bitrateMode>=1) && (hAacConfig->bitrateMode<=5)) /* VBR without PNS. */ - || (hAacConfig->useTns == 0) ) /* TNS required. */ - { - hAacConfig->usePns = 0; - } - - if (hAacConfig->epConfig >= 0) { - hAacConfig->syntaxFlags |= AC_ER; - if (((INT)hAacConfig->channelMode < 1) || ((INT)hAacConfig->channelMode > 7)) { - return AACENC_INVALID_CONFIG; /* Cannel config 0 not supported. */ - } - } - - if ( FDKaacEnc_DetermineEncoderMode(&hAacConfig->channelMode, hAacConfig->nChannels) != AAC_ENC_OK) { - return AACENC_INVALID_CONFIG; /* nChannels doesn't match chMode, this is just a check-up */ - } - - if ( (hAacConfig->nChannels > hAacEncoder->nMaxAacChannels) - || ( (FDKaacEnc_GetChannelModeConfiguration(hAacConfig->channelMode)->nChannelsEff > hAacEncoder->nMaxSbrChannels) && - isSbrActive(hAacConfig) ) - ) - { - return AACENC_INVALID_CONFIG; /* not enough channels allocated */ - } - - /* Meta data restriction. */ - switch (hAacConfig->audioObjectType) - { - /* Allow metadata support */ - case AOT_AAC_LC: - case AOT_SBR: - case AOT_PS: - hAacEncoder->metaDataAllowed = 1; - if (((INT)hAacConfig->channelMode < 1) || ((INT)hAacConfig->channelMode > 7)) { - config->userMetaDataMode = 0; - } - break; - /* Prohibit metadata support */ - default: - hAacEncoder->metaDataAllowed = 0; - } - - return err; -} - -static -INT aacenc_SbrCallback( - void * self, - HANDLE_FDK_BITSTREAM hBs, - const INT sampleRateIn, - const INT sampleRateOut, - const INT samplesPerFrame, - const AUDIO_OBJECT_TYPE coreCodec, - const MP4_ELEMENT_ID elementID, - const INT elementIndex - ) -{ - HANDLE_AACENCODER hAacEncoder = (HANDLE_AACENCODER)self; - - sbrEncoder_GetHeader(hAacEncoder->hEnvEnc, hBs, elementIndex, 0); - - return 0; -} - -static AACENC_ERROR aacEncInit(HANDLE_AACENCODER hAacEncoder, - ULONG InitFlags, - USER_PARAM *config) -{ - AACENC_ERROR err = AACENC_OK; - - INT aacBufferOffset = 0; - HANDLE_SBR_ENCODER *hSbrEncoder = &hAacEncoder->hEnvEnc; - HANDLE_AACENC_CONFIG hAacConfig = &hAacEncoder->aacConfig; - - hAacEncoder->nZerosAppended = 0; /* count appended zeros */ - - INT frameLength = hAacConfig->framelength; - - if ( (InitFlags & AACENC_INIT_CONFIG) ) - { - CHANNEL_MODE prevChMode = hAacConfig->channelMode; - - /* Verify settings and update: config -> heAacEncoder */ - if ( (err=FDKaacEnc_AdjustEncSettings(hAacEncoder, config)) != AACENC_OK ) { - return err; - } - frameLength = hAacConfig->framelength; /* adapt temporal framelength */ - - /* Seamless channel reconfiguration in sbr not fully implemented */ - if ( (prevChMode!=hAacConfig->channelMode) && isSbrActive(hAacConfig) ) { - InitFlags |= AACENC_INIT_STATES; - } - } - - /* Clear input buffer */ - if ( InitFlags == AACENC_INIT_ALL ) { - FDKmemclear(hAacEncoder->inputBuffer, sizeof(INT_PCM)*hAacEncoder->nMaxAacChannels*INPUTBUFFER_SIZE); - } - - if ( (InitFlags & AACENC_INIT_CONFIG) ) - { - aacBufferOffset = 0; - if (hAacConfig->audioObjectType == AOT_ER_AAC_ELD) { - hAacEncoder->nDelay = DELAY_AACELD(hAacConfig->framelength); - } else - { - hAacEncoder->nDelay = DELAY_AAC(hAacConfig->framelength); /* AAC encoder delay */ - } - hAacConfig->ancDataBitRate = 0; - } - - if ( isSbrActive(hAacConfig) && - ((InitFlags & AACENC_INIT_CONFIG) || (InitFlags & AACENC_INIT_STATES)) ) - { - INT sbrError; - SBR_ELEMENT_INFO sbrElInfo[(8)]; - CHANNEL_MAPPING channelMapping; - - if ( FDKaacEnc_InitChannelMapping(hAacConfig->channelMode, - hAacConfig->channelOrder, - &channelMapping) != AAC_ENC_OK ) - { - return AACENC_INIT_ERROR; - } - - /* Check return value and if the SBR encoder can handle enough elements */ - if (channelMapping.nElements > (8)) { - return AACENC_INIT_ERROR; - } - - aacEncDistributeSbrBits(&channelMapping, sbrElInfo, hAacConfig->bitRate); - - UINT initFlag = 0; - initFlag += (InitFlags & AACENC_INIT_STATES) ? 1 : 0; - - /* Let the SBR encoder take a look at the configuration and change if required. */ - sbrError = sbrEncoder_Init( - *hSbrEncoder, - sbrElInfo, - channelMapping.nElements, - hAacEncoder->inputBuffer, - &hAacConfig->bandWidth, - &aacBufferOffset, - &hAacConfig->nChannels, - &hAacConfig->sampleRate, - &hAacConfig->sbrRatio, - &frameLength, - hAacConfig->audioObjectType, - &hAacEncoder->nDelay, - (hAacConfig->audioObjectType == AOT_ER_AAC_ELD) ? 1 : TRANS_FAC, - (config->userTpHeaderPeriod!=0xFF) ? config->userTpHeaderPeriod : DEFAULT_HEADER_PERIOD_REPETITION_RATE, - initFlag - ); - - /* Suppress AOT reconfiguration and check error status. */ - if (sbrError) { - return AACENC_INIT_SBR_ERROR; - } - - if (hAacConfig->nChannels == 1) { - hAacConfig->channelMode = MODE_1; - } - - /* Never use PNS if SBR is active */ - if ( hAacConfig->usePns ) { - hAacConfig->usePns = 0; - } - - /* estimated bitrate consumed by SBR or PS */ - hAacConfig->ancDataBitRate = sbrEncoder_GetEstimateBitrate(*hSbrEncoder) ; - - } /* sbr initialization */ - - - /* - * Initialize Transport - Module. - */ - if ( (InitFlags & AACENC_INIT_TRANSPORT) ) - { - UINT flags = 0; - - FDKaacEnc_MapConfig( - &hAacEncoder->coderConfig, - config, - getSbrSignalingMode(hAacConfig->audioObjectType, config->userTpType, config->userTpSignaling, hAacConfig->sbrRatio), - hAacConfig); - - /* create flags for transport encoder */ - if (config->userTpAmxv != 0) { - flags |= TP_FLAG_LATM_AMV; - } - /* Clear output buffer */ - FDKmemclear(hAacEncoder->outBuffer, hAacEncoder->outBufferInBytes*sizeof(UCHAR)); - - /* Initialize Bitstream encoder */ - if ( transportEnc_Init(hAacEncoder->hTpEnc, hAacEncoder->outBuffer, hAacEncoder->outBufferInBytes, config->userTpType, &hAacEncoder->coderConfig, flags) != 0) { - return AACENC_INIT_TP_ERROR; - } - - } /* transport initialization */ - - /* - * Initialize AAC - Core. - */ - if ( (InitFlags & AACENC_INIT_CONFIG) || - (InitFlags & AACENC_INIT_STATES) ) - { - AAC_ENCODER_ERROR err; - err = FDKaacEnc_Initialize(hAacEncoder->hAacEnc, - hAacConfig, - hAacEncoder->hTpEnc, - (InitFlags & AACENC_INIT_STATES) ? 1 : 0); - - if (err != AAC_ENC_OK) { - return AACENC_INIT_AAC_ERROR; - } - - } /* aac initialization */ - - /* - * Initialize Meta Data - Encoder. - */ - if ( hAacEncoder->hMetadataEnc && (hAacEncoder->metaDataAllowed!=0) && - ((InitFlags & AACENC_INIT_CONFIG) ||(InitFlags & AACENC_INIT_STATES)) ) - { - INT inputDataDelay = DELAY_AAC(hAacConfig->framelength); - - if ( isSbrActive(hAacConfig) && hSbrEncoder!=NULL) { - inputDataDelay = hAacConfig->sbrRatio*inputDataDelay + sbrEncoder_GetInputDataDelay(*hSbrEncoder); - } - - if ( FDK_MetadataEnc_Init(hAacEncoder->hMetadataEnc, - ((InitFlags&AACENC_INIT_STATES) ? 1 : 0), - config->userMetaDataMode, - inputDataDelay, - frameLength, - config->userSamplerate, - config->nChannels, - config->userChannelMode, - hAacConfig->channelOrder) != 0) - { - return AACENC_INIT_META_ERROR; - } - - hAacEncoder->nDelay += FDK_MetadataEnc_GetDelay(hAacEncoder->hMetadataEnc); - } - - /* - * Update pointer to working buffer. - */ - if ( (InitFlags & AACENC_INIT_CONFIG) ) - { - hAacEncoder->inputBufferOffset = aacBufferOffset; - - hAacEncoder->nSamplesToRead = frameLength * config->nChannels; - - /* Make nDelay comparison compatible with config->nSamplesRead */ - hAacEncoder->nDelay *= config->nChannels; - - } /* parameter changed */ - - return AACENC_OK; -} - - -AACENC_ERROR aacEncOpen( - HANDLE_AACENCODER *phAacEncoder, - const UINT encModules, - const UINT maxChannels - ) -{ - AACENC_ERROR err = AACENC_OK; - HANDLE_AACENCODER hAacEncoder = NULL; - - if (phAacEncoder == NULL) { - err = AACENC_INVALID_HANDLE; - goto bail; - } - - /* allocate memory */ - hAacEncoder = Get_AacEncoder(); - - if (hAacEncoder == NULL) { - err = AACENC_MEMORY_ERROR; - goto bail; - } - - FDKmemclear(hAacEncoder, sizeof(AACENCODER)); - - /* Specify encoder modules to be allocated. */ - if (encModules==0) { - hAacEncoder->encoder_modis = ENC_MODE_FLAG_AAC; - hAacEncoder->encoder_modis |= ENC_MODE_FLAG_SBR; - hAacEncoder->encoder_modis |= ENC_MODE_FLAG_PS; - hAacEncoder->encoder_modis |= ENC_MODE_FLAG_META; - } - else { - /* consider SAC and PS module */ - hAacEncoder->encoder_modis = encModules; - } - - /* Determine max channel configuration. */ - if (maxChannels==0) { - hAacEncoder->nMaxAacChannels = (8); - hAacEncoder->nMaxSbrChannels = (8); - } - else { - hAacEncoder->nMaxAacChannels = (maxChannels&0x00FF); - if ( (hAacEncoder->encoder_modis&ENC_MODE_FLAG_SBR) ) { - hAacEncoder->nMaxSbrChannels = (maxChannels&0xFF00) ? (maxChannels>>8) : hAacEncoder->nMaxAacChannels; - } - - if ( (hAacEncoder->nMaxAacChannels>(8)) || (hAacEncoder->nMaxSbrChannels>(8)) ) { - err = AACENC_INVALID_CONFIG; - goto bail; - } - } /* maxChannels==0 */ - - /* Max number of elements could be tuned any more. */ - hAacEncoder->nMaxAacElements = fixMin((8), hAacEncoder->nMaxAacChannels); - hAacEncoder->nMaxSbrElements = fixMin((8), hAacEncoder->nMaxSbrChannels); - hAacEncoder->nMaxSubFrames = (1); - - - /* In case of memory overlay, allocate memory out of libraries */ - - hAacEncoder->inputBuffer = (INT_PCM*)FDKcalloc(hAacEncoder->nMaxAacChannels*INPUTBUFFER_SIZE, sizeof(INT_PCM)); - - /* Open SBR Encoder */ - if (hAacEncoder->encoder_modis&ENC_MODE_FLAG_SBR) { - if ( sbrEncoder_Open(&hAacEncoder->hEnvEnc, - hAacEncoder->nMaxSbrElements, - hAacEncoder->nMaxSbrChannels, - (hAacEncoder->encoder_modis&ENC_MODE_FLAG_PS) ? 1 : 0 ) ) - { - err = AACENC_MEMORY_ERROR; - goto bail; - } - } /* (encoder_modis&ENC_MODE_FLAG_SBR) */ - - - /* Open Aac Encoder */ - if ( FDKaacEnc_Open(&hAacEncoder->hAacEnc, - hAacEncoder->nMaxAacElements, - hAacEncoder->nMaxAacChannels, - (1)) != AAC_ENC_OK ) - { - err = AACENC_MEMORY_ERROR; - goto bail; - } - - { /* Get bitstream outputbuffer size */ - UINT ld_M; - for (ld_M=1; (UINT)(1<nMaxSubFrames*hAacEncoder->nMaxAacChannels*6144)>>3; ld_M++) ; - hAacEncoder->outBufferInBytes = (1<outBuffer = GetRam_bsOutbuffer(); - if (OUTPUTBUFFER_SIZE < hAacEncoder->outBufferInBytes ) { - err = AACENC_MEMORY_ERROR; - goto bail; - } - - /* Open Meta Data Encoder */ - if (hAacEncoder->encoder_modis&ENC_MODE_FLAG_META) { - if ( FDK_MetadataEnc_Open(&hAacEncoder->hMetadataEnc) ) - { - err = AACENC_MEMORY_ERROR; - goto bail; - } - } /* (encoder_modis&ENC_MODE_FLAG_META) */ - - /* Open Transport Encoder */ - if ( transportEnc_Open(&hAacEncoder->hTpEnc) != 0 ) - { - err = AACENC_MEMORY_ERROR; - goto bail; - } - else { - C_ALLOC_SCRATCH_START(pLibInfo, LIB_INFO, FDK_MODULE_LAST); - - FDKinitLibInfo( pLibInfo); - transportEnc_GetLibInfo( pLibInfo ); - - /* Get capabilty flag for transport encoder. */ - hAacEncoder->CAPF_tpEnc = FDKlibInfo_getCapabilities( pLibInfo, FDK_TPENC); - - C_ALLOC_SCRATCH_END(pLibInfo, LIB_INFO, FDK_MODULE_LAST); - } - if ( transportEnc_RegisterSbrCallback(hAacEncoder->hTpEnc, aacenc_SbrCallback, hAacEncoder) != 0 ) { - err = AACENC_INIT_TP_ERROR; - goto bail; - } - - /* Initialize encoder instance with default parameters. */ - aacEncDefaultConfig(&hAacEncoder->aacConfig, &hAacEncoder->extParam); - - /* Initialize headerPeriod in coderConfig for aacEncoder_GetParam(). */ - hAacEncoder->coderConfig.headerPeriod = hAacEncoder->extParam.userTpHeaderPeriod; - - /* All encoder modules have to be initialized */ - hAacEncoder->InitFlags = AACENC_INIT_ALL; - - /* Return encoder instance */ - *phAacEncoder = hAacEncoder; - - return err; - -bail: - aacEncClose(&hAacEncoder); - - return err; -} - - - -AACENC_ERROR aacEncClose(HANDLE_AACENCODER *phAacEncoder) -{ - AACENC_ERROR err = AACENC_OK; - - if (phAacEncoder == NULL) { - err = AACENC_INVALID_HANDLE; - goto bail; - } - - if (*phAacEncoder != NULL) { - HANDLE_AACENCODER hAacEncoder = *phAacEncoder; - - - if (hAacEncoder->inputBuffer!=NULL) { - FDKfree(hAacEncoder->inputBuffer); - hAacEncoder->inputBuffer = NULL; - } - - if (hAacEncoder->outBuffer) { - FreeRam_bsOutbuffer(&hAacEncoder->outBuffer); - } - - if (hAacEncoder->hEnvEnc) { - sbrEncoder_Close (&hAacEncoder->hEnvEnc); - } - if (hAacEncoder->hAacEnc) { - FDKaacEnc_Close (&hAacEncoder->hAacEnc); - } - - transportEnc_Close(&hAacEncoder->hTpEnc); - - if (hAacEncoder->hMetadataEnc) { - FDK_MetadataEnc_Close (&hAacEncoder->hMetadataEnc); - } - - Free_AacEncoder(phAacEncoder); - } - -bail: - return err; -} - -AACENC_ERROR aacEncEncode( - const HANDLE_AACENCODER hAacEncoder, - const AACENC_BufDesc *inBufDesc, - const AACENC_BufDesc *outBufDesc, - const AACENC_InArgs *inargs, - AACENC_OutArgs *outargs - ) -{ - AACENC_ERROR err = AACENC_OK; - INT i, nBsBytes = 0; - INT outBytes[(1)]; - int nExtensions = 0; - int ancDataExtIdx = -1; - - /* deal with valid encoder handle */ - if (hAacEncoder==NULL) { - err = AACENC_INVALID_HANDLE; - goto bail; - } - - - /* - * Adjust user settings and trigger reinitialization. - */ - if (hAacEncoder->InitFlags!=0) { - - err = aacEncInit(hAacEncoder, - hAacEncoder->InitFlags, - &hAacEncoder->extParam); - - if (err!=AACENC_OK) { - /* keep init flags alive! */ - goto bail; - } - hAacEncoder->InitFlags = AACENC_INIT_NONE; - } - - if (outargs!=NULL) { - FDKmemclear(outargs, sizeof(AACENC_OutArgs)); - } - - if (outBufDesc!=NULL) { - for (i=0; inumBufs; i++) { - if (outBufDesc->bufs[i]!=NULL) { - FDKmemclear(outBufDesc->bufs[i], outBufDesc->bufSizes[i]); - } - } - } - - /* - * If only encoder handle given, independent (re)initialization can be triggered. - */ - if ( (hAacEncoder!=NULL) & (inBufDesc==NULL) && (outBufDesc==NULL) && (inargs==NULL) && (outargs==NULL) ) { - goto bail; - } - - /* reset buffer wich signals number of valid bytes in output bitstream buffer */ - FDKmemclear(outBytes, hAacEncoder->aacConfig.nSubFrames*sizeof(INT)); - - /* - * Manage incoming audio samples. - */ - if ( (inargs->numInSamples > 0) && (getBufDescIdx(inBufDesc,IN_AUDIO_DATA) != -1) ) - { - /* Fetch data until nSamplesToRead reached */ - INT idx = getBufDescIdx(inBufDesc,IN_AUDIO_DATA); - INT newSamples = fixMax(0,fixMin(inargs->numInSamples, hAacEncoder->nSamplesToRead-hAacEncoder->nSamplesRead)); - INT_PCM *pIn = hAacEncoder->inputBuffer+hAacEncoder->inputBufferOffset+hAacEncoder->nSamplesRead; - - /* Copy new input samples to internal buffer */ - if (inBufDesc->bufElSizes[idx]==(INT)sizeof(INT_PCM)) { - FDKmemcpy(pIn, (INT_PCM*)inBufDesc->bufs[idx], newSamples*sizeof(INT_PCM)); /* Fast copy. */ - } - else if (inBufDesc->bufElSizes[idx]>(INT)sizeof(INT_PCM)) { - for (i=0; ibufs[idx])[i]>>16); /* Convert 32 to 16 bit. */ - } - } - else { - for (i=0; ibufs[idx])[i]))<<16; /* Convert 16 to 32 bit. */ - } - } - hAacEncoder->nSamplesRead += newSamples; - - /* Number of fetched input buffer samples. */ - outargs->numInSamples = newSamples; - } - - /* input buffer completely filled ? */ - if (hAacEncoder->nSamplesRead < hAacEncoder->nSamplesToRead) - { - /* - eof reached and flushing enabled, or - - return to main and wait for further incoming audio samples */ - if (inargs->numInSamples==-1) - { - if ( (hAacEncoder->nZerosAppended < hAacEncoder->nDelay) - ) - { - int nZeros = hAacEncoder->nSamplesToRead - hAacEncoder->nSamplesRead; - - FDK_ASSERT(nZeros >= 0); - - /* clear out until end-of-buffer */ - if (nZeros) { - FDKmemclear(hAacEncoder->inputBuffer+hAacEncoder->inputBufferOffset+hAacEncoder->nSamplesRead, sizeof(INT_PCM)*nZeros ); - hAacEncoder->nZerosAppended += nZeros; - hAacEncoder->nSamplesRead = hAacEncoder->nSamplesToRead; - } - } - else { /* flushing completed */ - err = AACENC_ENCODE_EOF; /* eof reached */ - goto bail; - } - } - else { /* inargs->numInSamples!= -1 */ - goto bail; /* not enough samples in input buffer and no flushing enabled */ - } - } - - /* init payload */ - FDKmemclear(hAacEncoder->extPayload, sizeof(AACENC_EXT_PAYLOAD) * MAX_TOTAL_EXT_PAYLOADS); - for (i = 0; i < MAX_TOTAL_EXT_PAYLOADS; i++) { - hAacEncoder->extPayload[i].associatedChElement = -1; - } - FDKmemclear(hAacEncoder->extPayloadData, sizeof(hAacEncoder->extPayloadData)); - FDKmemclear(hAacEncoder->extPayloadSize, sizeof(hAacEncoder->extPayloadSize)); - - - /* - * Calculate Meta Data info. - */ - if ( (hAacEncoder->hMetadataEnc!=NULL) && (hAacEncoder->metaDataAllowed!=0) ) { - - const AACENC_MetaData *pMetaData = NULL; - AACENC_EXT_PAYLOAD *pMetaDataExtPayload = NULL; - UINT nMetaDataExtensions = 0; - INT matrix_mixdown_idx = 0; - - /* New meta data info available ? */ - if ( getBufDescIdx(inBufDesc,IN_METADATA_SETUP) != -1 ) { - pMetaData = (AACENC_MetaData*)inBufDesc->bufs[getBufDescIdx(inBufDesc,IN_METADATA_SETUP)]; - } - - FDK_MetadataEnc_Process(hAacEncoder->hMetadataEnc, - hAacEncoder->inputBuffer+hAacEncoder->inputBufferOffset, - hAacEncoder->nSamplesRead, - pMetaData, - &pMetaDataExtPayload, - &nMetaDataExtensions, - &matrix_mixdown_idx - ); - - for (i=0; i<(INT)nMetaDataExtensions; i++) { /* Get meta data extension payload. */ - hAacEncoder->extPayload[nExtensions++] = pMetaDataExtPayload[i]; - } - - if ( (matrix_mixdown_idx!=-1) - && ((hAacEncoder->extParam.userChannelMode==MODE_1_2_2)||(hAacEncoder->extParam.userChannelMode==MODE_1_2_2_1)) ) - { - /* Set matrix mixdown coefficient. */ - UINT pceValue = (UINT)( (0<<3) | ((matrix_mixdown_idx&0x3)<<1) | 1 ); - if (hAacEncoder->extParam.userPceAdditions != pceValue) { - hAacEncoder->extParam.userPceAdditions = pceValue; - hAacEncoder->InitFlags |= AACENC_INIT_TRANSPORT; - } - } - } - - - if ( isSbrActive(&hAacEncoder->aacConfig) ) { - - INT nPayload = 0; - - /* - * Encode SBR data. - */ - if (sbrEncoder_EncodeFrame(hAacEncoder->hEnvEnc, - hAacEncoder->inputBuffer, - hAacEncoder->extParam.nChannels, - hAacEncoder->extPayloadSize[nPayload], - hAacEncoder->extPayloadData[nPayload] -#if defined(EVAL_PACKAGE_SILENCE) || defined(EVAL_PACKAGE_SBR_SILENCE) - ,hAacEncoder->hAacEnc->clearOutput -#endif - )) - { - err = AACENC_ENCODE_ERROR; - goto bail; - } - else { - /* Add SBR extension payload */ - for (i = 0; i < (8); i++) { - if (hAacEncoder->extPayloadSize[nPayload][i] > 0) { - hAacEncoder->extPayload[nExtensions].pData = hAacEncoder->extPayloadData[nPayload][i]; - { - hAacEncoder->extPayload[nExtensions].dataSize = hAacEncoder->extPayloadSize[nPayload][i]; - hAacEncoder->extPayload[nExtensions].associatedChElement = i; - } - hAacEncoder->extPayload[nExtensions].dataType = EXT_SBR_DATA; /* Once SBR Encoder supports SBR CRC set EXT_SBR_DATA_CRC */ - nExtensions++; /* or EXT_SBR_DATA according to configuration. */ - FDK_ASSERT(nExtensions<=MAX_TOTAL_EXT_PAYLOADS); - } - } - nPayload++; - } - } /* sbrEnabled */ - - if ( (inargs->numAncBytes > 0) && ( getBufDescIdx(inBufDesc,IN_ANCILLRY_DATA)!=-1 ) ) { - INT idx = getBufDescIdx(inBufDesc,IN_ANCILLRY_DATA); - hAacEncoder->extPayload[nExtensions].dataSize = inargs->numAncBytes * 8; - hAacEncoder->extPayload[nExtensions].pData = (UCHAR*)inBufDesc->bufs[idx]; - hAacEncoder->extPayload[nExtensions].dataType = EXT_DATA_ELEMENT; - hAacEncoder->extPayload[nExtensions].associatedChElement = -1; - ancDataExtIdx = nExtensions; /* store index */ - nExtensions++; - } - - /* - * Encode AAC - Core. - */ - if ( FDKaacEnc_EncodeFrame( hAacEncoder->hAacEnc, - hAacEncoder->hTpEnc, - hAacEncoder->inputBuffer, - outBytes, - hAacEncoder->extPayload - ) != AAC_ENC_OK ) - { - err = AACENC_ENCODE_ERROR; - goto bail; - } - - if (ancDataExtIdx >= 0) { - outargs->numAncBytes = inargs->numAncBytes - (hAacEncoder->extPayload[ancDataExtIdx].dataSize>>3); - } - - /* samples exhausted */ - hAacEncoder->nSamplesRead -= hAacEncoder->nSamplesToRead; - - /* - * Delay balancing buffer handling - */ - if (isSbrActive(&hAacEncoder->aacConfig)) { - sbrEncoder_UpdateBuffers(hAacEncoder->hEnvEnc, hAacEncoder->inputBuffer); - } - - /* - * Make bitstream public - */ - if (outBufDesc->numBufs>=1) { - - INT bsIdx = getBufDescIdx(outBufDesc,OUT_BITSTREAM_DATA); - INT auIdx = getBufDescIdx(outBufDesc,OUT_AU_SIZES); - - for (i=0,nBsBytes=0; iaacConfig.nSubFrames; i++) { - nBsBytes += outBytes[i]; - - if (auIdx!=-1) { - ((INT*)outBufDesc->bufs[auIdx])[i] = outBytes[i]; - } - } - - if ( (bsIdx!=-1) && (outBufDesc->bufSizes[bsIdx]>=nBsBytes) ) { - FDKmemcpy(outBufDesc->bufs[bsIdx], hAacEncoder->outBuffer, sizeof(UCHAR)*nBsBytes); - outargs->numOutBytes = nBsBytes; - } - else { - /* output buffer too small, can't write valid bitstream */ - err = AACENC_ENCODE_ERROR; - goto bail; - } - } - -bail: - if (err == AACENC_ENCODE_ERROR) { - /* All encoder modules have to be initialized */ - hAacEncoder->InitFlags = AACENC_INIT_ALL; - } - - return err; -} - -static -AAC_ENCODER_ERROR aacEncGetConf(HANDLE_AACENCODER hAacEncoder, - UINT *size, - UCHAR *confBuffer) -{ - FDK_BITSTREAM tmpConf; - UINT confType; - UCHAR buf[64]; - int err; - - /* Init bit buffer */ - FDKinitBitStream(&tmpConf, buf, 64, 0, BS_WRITER); - - /* write conf in tmp buffer */ - err = transportEnc_GetConf(hAacEncoder->hTpEnc, &hAacEncoder->coderConfig, &tmpConf, &confType); - - /* copy data to outbuffer: length in bytes */ - FDKbyteAlign(&tmpConf, 0); - - /* Check buffer size */ - if (FDKgetValidBits(&tmpConf) > ((*size)<<3)) - return AAC_ENC_UNKNOWN; - - FDKfetchBuffer(&tmpConf, confBuffer, size); - - if (err != 0) - return AAC_ENC_UNKNOWN; - else - return AAC_ENC_OK; -} - - -AACENC_ERROR aacEncGetLibInfo(LIB_INFO *info) -{ - int i = 0; - - if (info == NULL) { - return AACENC_INVALID_HANDLE; - } - - FDK_toolsGetLibInfo( info ); - transportEnc_GetLibInfo( info ); - - sbrEncoder_GetLibInfo( info ); - - /* search for next free tab */ - for (i = 0; i < FDK_MODULE_LAST; i++) { - if (info[i].module_id == FDK_NONE) break; - } - if (i == FDK_MODULE_LAST) { - return AACENC_INIT_ERROR; - } - - info[i].module_id = FDK_AACENC; - info[i].build_date = (char*)AACENCODER_LIB_BUILD_DATE; - info[i].build_time = (char*)AACENCODER_LIB_BUILD_TIME; - info[i].title = (char*)AACENCODER_LIB_TITLE; - info[i].version = LIB_VERSION(AACENCODER_LIB_VL0, AACENCODER_LIB_VL1, AACENCODER_LIB_VL2);; - LIB_VERSION_STRING(&info[i]); - - /* Capability flags */ - info[i].flags = 0 - | CAPF_AAC_1024 | CAPF_AAC_LC - | CAPF_AAC_512 - | CAPF_AAC_480 - | CAPF_AAC_DRC - ; - /* End of flags */ - - return AACENC_OK; -} - -AACENC_ERROR aacEncoder_SetParam( - const HANDLE_AACENCODER hAacEncoder, - const AACENC_PARAM param, - const UINT value - ) -{ - AACENC_ERROR err = AACENC_OK; - USER_PARAM *settings = &hAacEncoder->extParam; - - /* check encoder handle */ - if (hAacEncoder == NULL) { - err = AACENC_INVALID_HANDLE; - goto bail; - } - - /* apply param value */ - switch (param) - { - case AACENC_AOT: - if (settings->userAOT != (AUDIO_OBJECT_TYPE)value) { - /* check if AOT matches the allocated modules */ - switch ( value ) { - case AOT_PS: - if (!(hAacEncoder->encoder_modis & (ENC_MODE_FLAG_PS))) { - err = AACENC_INVALID_CONFIG; - goto bail; - } - case AOT_SBR: - if (!(hAacEncoder->encoder_modis & (ENC_MODE_FLAG_SBR))) { - err = AACENC_INVALID_CONFIG; - goto bail; - } - case AOT_AAC_LC: - case AOT_ER_AAC_LD: - case AOT_ER_AAC_ELD: - if (!(hAacEncoder->encoder_modis & (ENC_MODE_FLAG_AAC))) { - err = AACENC_INVALID_CONFIG; - goto bail; - } - break; - default: - err = AACENC_INVALID_CONFIG; - goto bail; - }/* switch value */ - settings->userAOT = (AUDIO_OBJECT_TYPE)value; - hAacEncoder->InitFlags |= AACENC_INIT_CONFIG | AACENC_INIT_STATES | AACENC_INIT_TRANSPORT; - } - break; - case AACENC_BITRATE: - if (settings->userBitrate != value) { - settings->userBitrate = value; - hAacEncoder->InitFlags |= AACENC_INIT_CONFIG | AACENC_INIT_TRANSPORT; - } - break; - case AACENC_BITRATEMODE: - if (settings->userBitrateMode != value) { - switch ( value ) { - case 0: - case 1: case 2: case 3: case 4: case 5: - case 8: - settings->userBitrateMode = value; - hAacEncoder->InitFlags |= AACENC_INIT_CONFIG | AACENC_INIT_TRANSPORT; - break; - default: - err = AACENC_INVALID_CONFIG; - break; - } /* switch value */ - } - break; - case AACENC_SAMPLERATE: - if (settings->userSamplerate != value) { - if ( !( (value==8000) || (value==11025) || (value==12000) || (value==16000) || (value==22050) || (value==24000) || - (value==32000) || (value==44100) || (value==48000) || (value==64000) || (value==88200) || (value==96000) ) ) - { - err = AACENC_INVALID_CONFIG; - break; - } - settings->userSamplerate = value; - hAacEncoder->nSamplesRead = 0; /* reset internal inputbuffer */ - hAacEncoder->InitFlags |= AACENC_INIT_CONFIG | AACENC_INIT_STATES | AACENC_INIT_TRANSPORT; - } - break; - case AACENC_CHANNELMODE: - if (settings->userChannelMode != (CHANNEL_MODE)value) { - const CHANNEL_MODE_CONFIG_TAB* pConfig = FDKaacEnc_GetChannelModeConfiguration((CHANNEL_MODE)value); - if (pConfig==NULL) { - err = AACENC_INVALID_CONFIG; - break; - } - if ( (pConfig->nElements > hAacEncoder->nMaxAacElements) - || (pConfig->nChannelsEff > hAacEncoder->nMaxAacChannels) - || !(((value>=1) && (value<=7))||((value>=33) && (value<=34))) - ) - { - err = AACENC_INVALID_CONFIG; - break; - } - - settings->userChannelMode = (CHANNEL_MODE)value; - settings->nChannels = pConfig->nChannels; - hAacEncoder->nSamplesRead = 0; /* reset internal inputbuffer */ - hAacEncoder->InitFlags |= AACENC_INIT_CONFIG | AACENC_INIT_TRANSPORT; - } - break; - case AACENC_BANDWIDTH: - if (settings->userBandwidth != value) { - settings->userBandwidth = value; - hAacEncoder->InitFlags |= AACENC_INIT_CONFIG; - } - break; - case AACENC_CHANNELORDER: - if (hAacEncoder->aacConfig.channelOrder != (CHANNEL_ORDER)value) { - if (! ((value==0) || (value==1) || (value==2)) ) { - err = AACENC_INVALID_CONFIG; - break; - } - hAacEncoder->aacConfig.channelOrder = (CHANNEL_ORDER)value; - hAacEncoder->nSamplesRead = 0; /* reset internal inputbuffer */ - hAacEncoder->InitFlags |= AACENC_INIT_CONFIG | AACENC_INIT_STATES | AACENC_INIT_TRANSPORT; - } - break; - case AACENC_AFTERBURNER: - if (settings->userAfterburner != value) { - if (! ((value==0) || (value==1)) ) { - err = AACENC_INVALID_CONFIG; - break; - } - settings->userAfterburner = value; - hAacEncoder->InitFlags |= AACENC_INIT_CONFIG; - } - break; - case AACENC_GRANULE_LENGTH: - if (settings->userFramelength != value) { - switch (value) { - case 1024: - case 512: - case 480: - settings->userFramelength = value; - hAacEncoder->InitFlags |= AACENC_INIT_CONFIG | AACENC_INIT_TRANSPORT; - break; - default: - err = AACENC_INVALID_CONFIG; - break; - } - } - break; - case AACENC_SBR_RATIO: - if (settings->userSbrRatio != value) { - if (! ((value==0) || (value==1) || (value==2)) ) { - err = AACENC_INVALID_CONFIG; - break; - } - settings->userSbrRatio = value; - hAacEncoder->InitFlags |= AACENC_INIT_CONFIG | AACENC_INIT_STATES | AACENC_INIT_TRANSPORT; - } - break; - case AACENC_SBR_MODE: - if (settings->userSbrEnabled != value) { - settings->userSbrEnabled = value; - hAacEncoder->InitFlags |= AACENC_INIT_CONFIG | AACENC_INIT_STATES | AACENC_INIT_TRANSPORT; - } - break; - case AACENC_TRANSMUX: - if (settings->userTpType != (TRANSPORT_TYPE)value) { - - TRANSPORT_TYPE type = (TRANSPORT_TYPE)value; - UINT flags = hAacEncoder->CAPF_tpEnc; - - if ( !( ((type==TT_MP4_ADIF) && (flags&CAPF_ADIF)) - || ((type==TT_MP4_ADTS) && (flags&CAPF_ADTS)) - || ((type==TT_MP4_LATM_MCP0) && ((flags&CAPF_LATM) && (flags&CAPF_RAWPACKETS))) - || ((type==TT_MP4_LATM_MCP1) && ((flags&CAPF_LATM) && (flags&CAPF_RAWPACKETS))) - || ((type==TT_MP4_LOAS) && (flags&CAPF_LOAS)) - || ((type==TT_MP4_RAW) && (flags&CAPF_RAWPACKETS)) - ) ) - { - err = AACENC_INVALID_CONFIG; - break; - } - settings->userTpType = (TRANSPORT_TYPE)value; - hAacEncoder->InitFlags |= AACENC_INIT_TRANSPORT; - } - break; - case AACENC_SIGNALING_MODE: - if (settings->userTpSignaling != value) { - if ( !((value==0) || (value==1) || (value==2)) ) { - err = AACENC_INVALID_CONFIG; - break; - } - settings->userTpSignaling = value; - hAacEncoder->InitFlags |= AACENC_INIT_TRANSPORT; - } - break; - case AACENC_PROTECTION: - if (settings->userTpProtection != value) { - if ( !((value==0) || (value==1)) ) { - err = AACENC_INVALID_CONFIG; - break; - } - settings->userTpProtection = value; - hAacEncoder->InitFlags |= AACENC_INIT_TRANSPORT; - } - break; - case AACENC_HEADER_PERIOD: - if (settings->userTpHeaderPeriod != value) { - settings->userTpHeaderPeriod = value; - hAacEncoder->InitFlags |= AACENC_INIT_TRANSPORT; - } - break; - case AACENC_AUDIOMUXVER: - if (settings->userTpAmxv != value) { - if ( !((value==0) || (value==1) || (value==2)) ) { - err = AACENC_INVALID_CONFIG; - break; - } - settings->userTpAmxv = value; - hAacEncoder->InitFlags |= AACENC_INIT_TRANSPORT; - } - break; - case AACENC_TPSUBFRAMES: - if (settings->userTpNsubFrames != value) { - if (! ( (value>=1) && (value<=4) ) ) { - err = AACENC_INVALID_CONFIG; - break; - } - settings->userTpNsubFrames = value; - hAacEncoder->InitFlags |= AACENC_INIT_TRANSPORT; - } - break; - case AACENC_ANCILLARY_BITRATE: - if (settings->userAncDataRate != value) { - settings->userAncDataRate = value; - } - break; - case AACENC_CONTROL_STATE: - if (hAacEncoder->InitFlags != value) { - if (value&AACENC_RESET_INBUFFER) { - hAacEncoder->nSamplesRead = 0; - } - hAacEncoder->InitFlags = value; - } - break; - case AACENC_METADATA_MODE: - if ((UINT)settings->userMetaDataMode != value) { - if ( !(((INT)value>=0) && ((INT)value<=2)) ) { - err = AACENC_INVALID_CONFIG; - break; - } - settings->userMetaDataMode = value; - hAacEncoder->InitFlags |= AACENC_INIT_CONFIG; - } - break; - case AACENC_PEAK_BITRATE: - if (settings->userPeakBitrate != value) { - settings->userPeakBitrate = value; - hAacEncoder->InitFlags |= AACENC_INIT_CONFIG | AACENC_INIT_TRANSPORT; - } - break; - default: - err = AACENC_UNSUPPORTED_PARAMETER; - break; - } /* switch(param) */ - -bail: - return err; -} - -UINT aacEncoder_GetParam( - const HANDLE_AACENCODER hAacEncoder, - const AACENC_PARAM param - ) -{ - UINT value = 0; - USER_PARAM *settings = &hAacEncoder->extParam; - - /* check encoder handle */ - if (hAacEncoder == NULL) { - goto bail; - } - - /* apply param value */ - switch (param) - { - case AACENC_AOT: - value = (UINT)hAacEncoder->aacConfig.audioObjectType; - break; - case AACENC_BITRATE: - value = (UINT)((hAacEncoder->aacConfig.bitrateMode==AACENC_BR_MODE_CBR) ? hAacEncoder->aacConfig.bitRate : -1); - break; - case AACENC_BITRATEMODE: - value = (UINT)hAacEncoder->aacConfig.bitrateMode; - break; - case AACENC_SAMPLERATE: - value = (UINT)hAacEncoder->coderConfig.extSamplingRate; - break; - case AACENC_CHANNELMODE: - value = (UINT)hAacEncoder->aacConfig.channelMode; - break; - case AACENC_BANDWIDTH: - value = (UINT)hAacEncoder->aacConfig.bandWidth; - break; - case AACENC_CHANNELORDER: - value = (UINT)hAacEncoder->aacConfig.channelOrder; - break; - case AACENC_AFTERBURNER: - value = (UINT)hAacEncoder->aacConfig.useRequant; - break; - case AACENC_GRANULE_LENGTH: - value = (UINT)hAacEncoder->aacConfig.framelength; - break; - case AACENC_SBR_RATIO: - value = isSbrActive(&hAacEncoder->aacConfig) ? hAacEncoder->aacConfig.sbrRatio : 0; - break; - case AACENC_SBR_MODE: - value = (UINT) (hAacEncoder->aacConfig.syntaxFlags & AC_SBR_PRESENT) ? 1 : 0; - break; - case AACENC_TRANSMUX: - value = (UINT)settings->userTpType; - break; - case AACENC_SIGNALING_MODE: - value = (UINT)getSbrSignalingMode(hAacEncoder->aacConfig.audioObjectType, settings->userTpType, settings->userTpSignaling, hAacEncoder->aacConfig.sbrRatio); - break; - case AACENC_PROTECTION: - value = (UINT)settings->userTpProtection; - break; - case AACENC_HEADER_PERIOD: - value = (UINT)hAacEncoder->coderConfig.headerPeriod; - break; - case AACENC_AUDIOMUXVER: - value = (UINT)hAacEncoder->aacConfig.audioMuxVersion; - break; - case AACENC_TPSUBFRAMES: - value = (UINT)settings->userTpNsubFrames; - break; - case AACENC_ANCILLARY_BITRATE: - value = (UINT)hAacEncoder->aacConfig.anc_Rate; - break; - case AACENC_CONTROL_STATE: - value = (UINT)hAacEncoder->InitFlags; - break; - case AACENC_METADATA_MODE: - value = (hAacEncoder->metaDataAllowed==0) ? 0 : (UINT)settings->userMetaDataMode; - break; - case AACENC_PEAK_BITRATE: - value = (UINT)-1; /* peak bitrate parameter is meaningless */ - if ( ((INT)hAacEncoder->extParam.userPeakBitrate!=-1) ) { - value = (UINT)(fMax((INT)hAacEncoder->extParam.userPeakBitrate, hAacEncoder->aacConfig.bitRate)); /* peak bitrate parameter is in use */ - } - break; - default: - //err = MPS_INVALID_PARAMETER; - break; - } /* switch(param) */ - -bail: - return value; -} - -AACENC_ERROR aacEncInfo( - const HANDLE_AACENCODER hAacEncoder, - AACENC_InfoStruct *pInfo - ) -{ - AACENC_ERROR err = AACENC_OK; - - FDKmemclear(pInfo, sizeof(AACENC_InfoStruct)); - pInfo->confSize = 64; /* pre-initialize */ - - pInfo->maxOutBufBytes = ((hAacEncoder->nMaxAacChannels*6144)+7)>>3; - pInfo->maxAncBytes = hAacEncoder->aacConfig.maxAncBytesPerAU; - pInfo->inBufFillLevel = hAacEncoder->nSamplesRead/hAacEncoder->extParam.nChannels; - pInfo->inputChannels = hAacEncoder->extParam.nChannels; - pInfo->frameLength = hAacEncoder->nSamplesToRead/hAacEncoder->extParam.nChannels; - pInfo->encoderDelay = hAacEncoder->nDelay/hAacEncoder->extParam.nChannels; - - /* Get encoder configuration */ - if ( aacEncGetConf(hAacEncoder, &pInfo->confSize, &pInfo->confBuf[0]) != AAC_ENC_OK) { - err = AACENC_INIT_ERROR; - goto bail; - } -bail: - return err; -} - diff --git a/libAACenc/src/aacenc_pns.cpp b/libAACenc/src/aacenc_pns.cpp deleted file mode 100644 index b9640d9..0000000 --- a/libAACenc/src/aacenc_pns.cpp +++ /dev/null @@ -1,591 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M. Lohwasser - contents/description: pns.c - -******************************************************************************/ - -#include "aacenc_pns.h" -#include "psy_data.h" -#include "pnsparam.h" -#include "noisedet.h" -#include "bit_cnt.h" -#include "interface.h" - - -/* minCorrelationEnergy = (1.0e-10f)^2 ~ 2^-67 = 2^-47 * 2^-20 */ -static const FIXP_DBL minCorrelationEnergy = FL2FXCONST_DBL(0.0); /* FL2FXCONST_DBL((float)FDKpow(2.0,-47)); */ -/* noiseCorrelationThresh = 0.6^2 */ -static const FIXP_DBL noiseCorrelationThresh = FL2FXCONST_DBL(0.36); - -static void FDKaacEnc_FDKaacEnc_noiseDetection( PNS_CONFIG *pnsConf, - PNS_DATA *pnsData, - const INT sfbActive, - const INT *sfbOffset, - INT tnsOrder, - INT tnsPredictionGain, - INT tnsActive, - FIXP_DBL *mdctSpectrum, - INT *sfbMaxScaleSpec, - FIXP_SGL *sfbtonality ); - -static void FDKaacEnc_CalcNoiseNrgs( const INT sfbActive, - INT *pnsFlag, - FIXP_DBL *sfbEnergyLdData, - INT *noiseNrg ); - -/***************************************************************************** - - functionname: initPnsConfiguration - description: fill pnsConf with pns parameters - returns: error status - input: PNS Config struct (modified) - bitrate, samplerate, usePns, - number of sfb's, pointer to sfb offset - output: error code - -*****************************************************************************/ - -AAC_ENCODER_ERROR FDKaacEnc_InitPnsConfiguration(PNS_CONFIG *pnsConf, - INT bitRate, - INT sampleRate, - INT usePns, - INT sfbCnt, - const INT *sfbOffset, - const INT numChan, - const INT isLC) -{ - AAC_ENCODER_ERROR ErrorStatus; - - /* init noise detection */ - ErrorStatus = FDKaacEnc_GetPnsParam(&pnsConf->np, - bitRate, - sampleRate, - sfbCnt, - sfbOffset, - &usePns, - numChan, - isLC); - if (ErrorStatus != AAC_ENC_OK) - return ErrorStatus; - - pnsConf->minCorrelationEnergy = minCorrelationEnergy; - pnsConf->noiseCorrelationThresh = noiseCorrelationThresh; - - pnsConf->usePns = usePns; - - return AAC_ENC_OK; -} - - - -/***************************************************************************** - - functionname: FDKaacEnc_PnsDetect - description: do decision, if PNS shall used or not - returns: - input: pns config structure - pns data structure (modified), - lastWindowSequence (long or short blocks) - sfbActive - pointer to Sfb Energy, Threshold, Offset - pointer to mdct Spectrum - length of each group - pointer to tonality calculated in chaosmeasure - tns order and prediction gain - calculated noiseNrg at active PNS - output: pnsFlag in pns data structure - -*****************************************************************************/ -void FDKaacEnc_PnsDetect(PNS_CONFIG *pnsConf, - PNS_DATA *pnsData, - const INT lastWindowSequence, - const INT sfbActive, - const INT maxSfbPerGroup, - FIXP_DBL *sfbThresholdLdData, - const INT *sfbOffset, - FIXP_DBL *mdctSpectrum, - INT *sfbMaxScaleSpec, - FIXP_SGL *sfbtonality, - INT tnsOrder, - INT tnsPredictionGain, - INT tnsActive, - FIXP_DBL *sfbEnergyLdData, - INT *noiseNrg ) - -{ - int sfb; - int startNoiseSfb; - - if (pnsConf->np.detectionAlgorithmFlags & IS_LOW_COMLEXITY) { - if ( (!pnsConf->usePns) || /* pns enabled? */ - (lastWindowSequence == SHORT_WINDOW) ) /* currently only long blocks */ - { - FDKmemclear(pnsData->pnsFlag, MAX_GROUPED_SFB*sizeof(INT)); /* clear all pnsFlags */ - for (sfb=0; sfbusePns) - return; - - /* PNS only for long Windows */ - if (pnsConf->np.detectionAlgorithmFlags & JUST_LONG_WINDOW) { - if(lastWindowSequence != LONG_WINDOW) { - for (sfb = 0; sfb < sfbActive; sfb++) { - pnsData->pnsFlag[sfb] = 0; /* clear all pnsFlags */ - } - return; - } - } - } - /* - call noise detection - */ - FDKaacEnc_FDKaacEnc_noiseDetection( pnsConf, - pnsData, - sfbActive, - sfbOffset, - tnsOrder, - tnsPredictionGain, - tnsActive, - mdctSpectrum, - sfbMaxScaleSpec, - sfbtonality ); - - /* set startNoiseSfb (long) */ - startNoiseSfb = pnsConf->np.startSfb; - - /* Set noise substitution status */ - for(sfb = 0; sfb < sfbActive; sfb++) { - - /* No PNS below startNoiseSfb */ - if(sfb < startNoiseSfb){ - pnsData->pnsFlag[sfb] = 0; - continue; - } - - /* - do noise substitution if - fuzzy measure is high enough - sfb freq > minimum sfb freq - signal in coder band is not masked - */ - - if((pnsData->noiseFuzzyMeasure[sfb] > FL2FXCONST_SGL(0.5)) && - ( (sfbThresholdLdData[sfb] + FL2FXCONST_DBL(0.5849625f/64.0f)) /* thr * 1.5 = thrLd +ld(1.5)/64 */ - < sfbEnergyLdData[sfb] ) ) - { - /* - mark in psyout flag array that we will code - this band with PNS - */ - pnsData->pnsFlag[sfb] = 1; /* PNS_ON */ - } - else{ - pnsData->pnsFlag[sfb] = 0; /* PNS_OFF */ - } - - /* no PNS if LTP is active */ - } - - /* avoid PNS holes */ - if((pnsData->noiseFuzzyMeasure[0]>FL2FXCONST_SGL(0.5f)) && (pnsData->pnsFlag[1])) { - pnsData->pnsFlag[0] = 1; - } - - for(sfb=1; sfbnoiseFuzzyMeasure[sfb]>pnsConf->np.gapFillThr) && - (pnsData->pnsFlag[sfb-1]) && (pnsData->pnsFlag[sfb+1])) { - pnsData->pnsFlag[sfb] = 1; - } - } - - if(maxSfbPerGroup>0) { - /* avoid PNS hole */ - if((pnsData->noiseFuzzyMeasure[maxSfbPerGroup-1]>pnsConf->np.gapFillThr) && (pnsData->pnsFlag[maxSfbPerGroup-2])) { - pnsData->pnsFlag[maxSfbPerGroup-1] = 1; - } - /* avoid single PNS band */ - if(pnsData->pnsFlag[maxSfbPerGroup-2]==0) { - pnsData->pnsFlag[maxSfbPerGroup-1] = 0; - } - } - - /* avoid single PNS bands */ - if(pnsData->pnsFlag[1]==0) { - pnsData->pnsFlag[0] = 0; - } - - for(sfb=1; sfbpnsFlag[sfb-1]==0)&&(pnsData->pnsFlag[sfb+1]==0)) { - pnsData->pnsFlag[sfb] = 0; - } - } - - - /* - calculate noiseNrg's - */ - FDKaacEnc_CalcNoiseNrgs( sfbActive, - pnsData->pnsFlag, - sfbEnergyLdData, - noiseNrg ); -} - - -/***************************************************************************** - - functionname:FDKaacEnc_FDKaacEnc_noiseDetection - description: wrapper for noisedet.c - returns: - input: pns config structure - pns data structure (modified), - sfbActive - tns order and prediction gain - pointer to mdct Spectrumand Sfb Energy - pointer to Sfb tonality - output: noiseFuzzyMeasure in structure pnsData - flags tonal / nontonal - -*****************************************************************************/ -static void FDKaacEnc_FDKaacEnc_noiseDetection( PNS_CONFIG *pnsConf, - PNS_DATA *pnsData, - const INT sfbActive, - const INT *sfbOffset, - int tnsOrder, - INT tnsPredictionGain, - INT tnsActive, - FIXP_DBL *mdctSpectrum, - INT *sfbMaxScaleSpec, - FIXP_SGL *sfbtonality ) -{ - INT condition = TRUE; - if ( !(pnsConf->np.detectionAlgorithmFlags & IS_LOW_COMLEXITY) ) { - condition = (tnsOrder > 3); - } - /* - no PNS if heavy TNS activity - clear pnsData->noiseFuzzyMeasure - */ - if((pnsConf->np.detectionAlgorithmFlags & USE_TNS_GAIN_THR) && - (tnsPredictionGain >= pnsConf->np.tnsGainThreshold) && condition && - !((pnsConf->np.detectionAlgorithmFlags & USE_TNS_PNS) && (tnsPredictionGain >= pnsConf->np.tnsPNSGainThreshold) && (tnsActive)) ) - { - /* clear all noiseFuzzyMeasure */ - FDKmemclear(pnsData->noiseFuzzyMeasure, sfbActive*sizeof(FIXP_SGL)); - } - else - { - /* - call noise detection, output in pnsData->noiseFuzzyMeasure, - use real mdct spectral data - */ - FDKaacEnc_noiseDetect( mdctSpectrum, - sfbMaxScaleSpec, - sfbActive, - sfbOffset, - pnsData->noiseFuzzyMeasure, - &pnsConf->np, - sfbtonality); - } -} - - -/***************************************************************************** - - functionname:FDKaacEnc_CalcNoiseNrgs - description: Calculate the NoiseNrg's - returns: - input: sfbActive - if pnsFlag calculate NoiseNrg - pointer to sfbEnergy and groupLen - pointer to noiseNrg (modified) - output: noiseNrg's in pnsFlaged sfb's - -*****************************************************************************/ - -static void FDKaacEnc_CalcNoiseNrgs( const INT sfbActive, - INT *RESTRICT pnsFlag, - FIXP_DBL *RESTRICT sfbEnergyLdData, - INT *RESTRICT noiseNrg ) -{ - int sfb; - INT tmp = (-LOG_NORM_PCM)<<2; - - for(sfb = 0; sfb < sfbActive; sfb++) { - if(pnsFlag[sfb]) { - INT nrg = (-sfbEnergyLdData[sfb]+FL2FXCONST_DBL(0.5f/64.0f))>>(DFRACT_BITS-1-7); - noiseNrg[sfb] = tmp - nrg; - } - } -} - - -/***************************************************************************** - - functionname:FDKaacEnc_CodePnsChannel - description: Execute pns decission - returns: - input: sfbActive - pns config structure - use PNS if pnsFlag - pointer to Sfb Energy, noiseNrg, Threshold - output: set sfbThreshold high to code pe with 0, - noiseNrg marks flag for pns coding - -*****************************************************************************/ - -void FDKaacEnc_CodePnsChannel(const INT sfbActive, - PNS_CONFIG *pnsConf, - INT *RESTRICT pnsFlag, - FIXP_DBL *RESTRICT sfbEnergyLdData, - INT *RESTRICT noiseNrg, - FIXP_DBL *RESTRICT sfbThresholdLdData) -{ - INT sfb; - INT lastiNoiseEnergy = 0; - INT firstPNSband = 1; /* TRUE for first PNS-coded band */ - - /* no PNS */ - if(!pnsConf->usePns) { - for(sfb = 0; sfb < sfbActive; sfb++) { - /* no PNS coding */ - noiseNrg[sfb] = NO_NOISE_PNS; - } - return; - } - - /* code PNS */ - for(sfb = 0; sfb < sfbActive; sfb++) { - if(pnsFlag[sfb]) { - /* high sfbThreshold causes pe = 0 */ - if(noiseNrg[sfb] != NO_NOISE_PNS) - sfbThresholdLdData[sfb] = sfbEnergyLdData[sfb] + FL2FXCONST_DBL(1.0f/LD_DATA_SCALING); - - /* set noiseNrg in valid region */ - if(!firstPNSband) { - INT deltaiNoiseEnergy = noiseNrg[sfb] - lastiNoiseEnergy; - - if(deltaiNoiseEnergy > CODE_BOOK_PNS_LAV) - noiseNrg[sfb] -= deltaiNoiseEnergy - CODE_BOOK_PNS_LAV; - else if(deltaiNoiseEnergy < -CODE_BOOK_PNS_LAV) - noiseNrg[sfb] -= deltaiNoiseEnergy + CODE_BOOK_PNS_LAV; - } - else { - firstPNSband = 0; - } - lastiNoiseEnergy = noiseNrg[sfb]; - } - else { - /* no PNS coding */ - noiseNrg[sfb] = NO_NOISE_PNS; - } - } -} - - -/***************************************************************************** - - functionname:FDKaacEnc_PreProcessPnsChannelPair - description: Calculate the correlation of noise in a channel pair - - returns: - input: sfbActive - pointer to sfb energies left, right and mid channel - pns config structure - pns data structure left and right (modified) - - output: noiseEnergyCorrelation in pns data structure - -*****************************************************************************/ - -void FDKaacEnc_PreProcessPnsChannelPair(const INT sfbActive, - FIXP_DBL *RESTRICT sfbEnergyLeft, - FIXP_DBL *RESTRICT sfbEnergyRight, - FIXP_DBL *RESTRICT sfbEnergyLeftLD, - FIXP_DBL *RESTRICT sfbEnergyRightLD, - FIXP_DBL *RESTRICT sfbEnergyMid, - PNS_CONFIG *RESTRICT pnsConf, - PNS_DATA *pnsDataLeft, - PNS_DATA *pnsDataRight) -{ - INT sfb; - FIXP_DBL ccf; - - if(!pnsConf->usePns) - return; - - FIXP_DBL *RESTRICT pNoiseEnergyCorrelationL = pnsDataLeft->noiseEnergyCorrelation; - FIXP_DBL *RESTRICT pNoiseEnergyCorrelationR = pnsDataRight->noiseEnergyCorrelation; - - for(sfb=0;sfb< sfbActive;sfb++) { - FIXP_DBL quot = (sfbEnergyLeftLD[sfb]>>1) + (sfbEnergyRightLD[sfb]>>1); - - if(quot < FL2FXCONST_DBL(-32.0f/(float)LD_DATA_SCALING)) - ccf = FL2FXCONST_DBL(0.0f); - else { - FIXP_DBL accu = sfbEnergyMid[sfb]- (((sfbEnergyLeft[sfb]>>1)+(sfbEnergyRight[sfb]>>1))>>1); - INT sign = (accu < FL2FXCONST_DBL(0.0f)) ? 1 : 0 ; - accu = fixp_abs(accu); - - ccf = CalcLdData(accu) + FL2FXCONST_DBL((float)1.0f/(float)LD_DATA_SCALING) - quot; /* ld(accu*2) = ld(accu) + 1 */ - ccf = (ccf>=FL2FXCONST_DBL(0.0)) ? ((FIXP_DBL)MAXVAL_DBL) : (sign) ? -CalcInvLdData(ccf) : CalcInvLdData(ccf); - } - - pNoiseEnergyCorrelationL[sfb] = ccf; - pNoiseEnergyCorrelationR[sfb] = ccf; - } -} - - - -/***************************************************************************** - - functionname:FDKaacEnc_PostProcessPnsChannelPair - description: if PNS used at left and right channel, - use msMask to flag correlation - returns: - input: sfbActive - pns config structure - pns data structure left and right (modified) - pointer to msMask, flags correlation by pns coding (modified) - Digest of MS coding - output: pnsFlag in pns data structure, - msFlag in msMask (flags correlation) - -*****************************************************************************/ - -void FDKaacEnc_PostProcessPnsChannelPair(const INT sfbActive, - PNS_CONFIG *pnsConf, - PNS_DATA *pnsDataLeft, - PNS_DATA *pnsDataRight, - INT *RESTRICT msMask, - INT *msDigest ) -{ - INT sfb; - - if(!pnsConf->usePns) - return; - - for(sfb=0;sfbpnsFlag[sfb]) && - (pnsDataRight->pnsFlag[sfb]) ) { - /* AAC only: Standard */ - /* do this to avoid ms flags in layers that should not have it */ - if(pnsDataLeft->noiseEnergyCorrelation[sfb] <= pnsConf->noiseCorrelationThresh){ - msMask[sfb] = 0; - *msDigest = MS_SOME; - } - } - else { - /* - No PNS coding - */ - pnsDataLeft->pnsFlag[sfb] = 0; - pnsDataRight->pnsFlag[sfb] = 0; - } - } - - /* - Use MS flag to signal noise correlation if - pns is active in both channels - */ - if( (pnsDataLeft->pnsFlag[sfb]) && (pnsDataRight->pnsFlag[sfb]) ) { - if(pnsDataLeft->noiseEnergyCorrelation[sfb] > pnsConf->noiseCorrelationThresh) { - msMask[sfb] = 1; - *msDigest = MS_SOME; - } - } - } -} diff --git a/libAACenc/src/aacenc_pns.h b/libAACenc/src/aacenc_pns.h deleted file mode 100644 index 3bda9de..0000000 --- a/libAACenc/src/aacenc_pns.h +++ /dev/null @@ -1,113 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M. Lohwasser - contents/description: pns.h - -******************************************************************************/ - -#ifndef __PNS_H -#define __PNS_H - -#include "common_fix.h" - -#include "pnsparam.h" - -#define NO_NOISE_PNS FDK_INT_MIN - -typedef struct{ - NOISEPARAMS np; - FIXP_DBL minCorrelationEnergy; - FIXP_DBL noiseCorrelationThresh; - INT usePns; -} PNS_CONFIG; - -typedef struct{ - FIXP_SGL noiseFuzzyMeasure[MAX_GROUPED_SFB]; - FIXP_DBL noiseEnergyCorrelation[MAX_GROUPED_SFB]; - INT pnsFlag[MAX_GROUPED_SFB]; -} PNS_DATA; - -#endif diff --git a/libAACenc/src/aacenc_tns.cpp b/libAACenc/src/aacenc_tns.cpp deleted file mode 100644 index 5fcd309..0000000 --- a/libAACenc/src/aacenc_tns.cpp +++ /dev/null @@ -1,1341 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: Alex Groeschel, Tobias Chalupka - contents/description: Temporal noise shaping - -******************************************************************************/ - -#include "aacenc_tns.h" -#include "psy_const.h" -#include "psy_configuration.h" -#include "tns_func.h" -#include "aacEnc_rom.h" -#include "aacenc_tns.h" - -#define FILTER_DIRECTION 0 /* 0 = up, 1 = down */ - -static const FIXP_DBL acfWindowLong[12+3+1] = { - 0x7fffffff,0x7fb80000,0x7ee00000,0x7d780000,0x7b800000,0x78f80000,0x75e00000,0x72380000, - 0x6e000000,0x69380000,0x63e00000,0x5df80000,0x57800000,0x50780000,0x48e00000,0x40b80000 -}; - -static const FIXP_DBL acfWindowShort[4+3+1] = { - 0x7fffffff,0x7e000000,0x78000000,0x6e000000,0x60000000,0x4e000000,0x38000000,0x1e000000 -}; - -typedef struct{ - INT bitRateFrom[2]; /* noneSbr=0, useSbr=1 */ - INT bitRateTo[2]; /* noneSbr=0, useSbr=1 */ - TNS_PARAMETER_TABULATED paramTab[2]; /* mono=0, stereo=1 */ - -} TNS_INFO_TAB; - -#define TNS_TIMERES_SCALE (1) -#define FL2_TIMERES_FIX(a) ( FL2FXCONST_DBL(a/(float)(1<= tnsInfoTab[i].bitRateFrom[sbrLd?1:0]) && - bitRate <= tnsInfoTab[i].bitRateTo[sbrLd?1:0]) - { - tnsConfigTab = &tnsInfoTab[i].paramTab[(channels==1)?0:1]; - } - } - - return tnsConfigTab; -} - - -static INT getTnsMaxBands( - const INT sampleRate, - const INT granuleLength, - const INT isShortBlock - ) -{ - int i; - INT numBands = -1; - const TNS_MAX_TAB_ENTRY *pMaxBandsTab = NULL; - int maxBandsTabSize = 0; - - switch (granuleLength) { - case 1024: - pMaxBandsTab = tnsMaxBandsTab1024; - maxBandsTabSize = sizeof(tnsMaxBandsTab1024)/sizeof(TNS_MAX_TAB_ENTRY); - break; - case 480: - pMaxBandsTab = tnsMaxBandsTab480; - maxBandsTabSize = sizeof(tnsMaxBandsTab480)/sizeof(TNS_MAX_TAB_ENTRY); - break; - case 512: - pMaxBandsTab = tnsMaxBandsTab512; - maxBandsTabSize = sizeof(tnsMaxBandsTab512)/sizeof(TNS_MAX_TAB_ENTRY); - break; - default: - numBands = -1; - } - - if (pMaxBandsTab!=NULL) { - for (i=0; i= pMaxBandsTab[i].samplingRate) { - break; - } - } - } - - return numBands; -} - -/***************************************************************************/ -/*! - \brief FDKaacEnc_FreqToBandWithRounding - - Returns index of nearest band border - - \param frequency - \param sampling frequency - \param total number of bands - \param pointer to table of band borders - - \return band border -****************************************************************************/ - -INT FDKaacEnc_FreqToBandWithRounding( - const INT freq, - const INT fs, - const INT numOfBands, - const INT *bandStartOffset - ) -{ - INT lineNumber, band; - - /* assert(freq >= 0); */ - lineNumber = (freq*bandStartOffset[numOfBands]*4/fs+1)/2; - - /* freq > fs/2 */ - if (lineNumber >= bandStartOffset[numOfBands]) - return numOfBands; - - /* find band the line number lies in */ - for (band=0; bandlineNumber) break; - } - - /* round to nearest band border */ - if (lineNumber - bandStartOffset[band] > - bandStartOffset[band+1] - lineNumber ) - { - band++; - } - - return(band); -} - - -/***************************************************************************** - - functionname: FDKaacEnc_InitTnsConfiguration - description: fill TNS_CONFIG structure with sensible content - returns: - input: bitrate, samplerate, number of channels, - blocktype (long or short), - TNS Config struct (modified), - psy config struct, - tns active flag - output: - -*****************************************************************************/ -AAC_ENCODER_ERROR FDKaacEnc_InitTnsConfiguration(INT bitRate, - INT sampleRate, - INT channels, - INT blockType, - INT granuleLength, - INT isLowDelay, - INT ldSbrPresent, - TNS_CONFIG *tC, - PSY_CONFIGURATION *pC, - INT active, - INT useTnsPeak) -{ - int i; - //float acfTimeRes = (blockType == SHORT_WINDOW) ? 0.125f : 0.046875f; - - if (channels <= 0) - return (AAC_ENCODER_ERROR)1; - - tC->isLowDelay = isLowDelay; - - /* initialize TNS filter flag, order, and coefficient resolution (in bits per coeff) */ - tC->tnsActive = (active) ? TRUE : FALSE; - tC->maxOrder = (blockType == SHORT_WINDOW) ? 5 : 12; /* maximum: 7, 20 */ - if (bitRate < 16000) - tC->maxOrder -= 2; - tC->coefRes = (blockType == SHORT_WINDOW) ? 3 : 4; - - /* LPC stop line: highest MDCT line to be coded, but do not go beyond TNS_MAX_BANDS! */ - tC->lpcStopBand = getTnsMaxBands(sampleRate, granuleLength, (blockType == SHORT_WINDOW) ? 1 : 0); - - if (tC->lpcStopBand < 0) { - return (AAC_ENCODER_ERROR)1; - } - - tC->lpcStopBand = FDKmin(tC->lpcStopBand, pC->sfbActive); - tC->lpcStopLine = pC->sfbOffset[tC->lpcStopBand]; - - switch (granuleLength) { - case 1024: - /* TNS start line: skip lower MDCT lines to prevent artifacts due to filter mismatch */ - tC->lpcStartBand[LOFILT] = (blockType == SHORT_WINDOW) ? 0 : ((sampleRate <= 8000) ? 2 : ((sampleRate < 18783) ? 4 : 8)); - tC->lpcStartLine[LOFILT] = pC->sfbOffset[tC->lpcStartBand[LOFILT]]; - - i = tC->lpcStopBand; - while (pC->sfbOffset[i] > (tC->lpcStartLine[LOFILT] + (tC->lpcStopLine - tC->lpcStartLine[LOFILT]) / 4)) i--; - tC->lpcStartBand[HIFILT] = i; - tC->lpcStartLine[HIFILT] = pC->sfbOffset[i]; - - tC->confTab.threshOn[HIFILT] = 1437; - tC->confTab.threshOn[LOFILT] = 1500; - - tC->confTab.tnsLimitOrder[HIFILT] = tC->maxOrder; - tC->confTab.tnsLimitOrder[LOFILT] = tC->maxOrder - 7; - - tC->confTab.tnsFilterDirection[HIFILT] = FILTER_DIRECTION; - tC->confTab.tnsFilterDirection[LOFILT] = FILTER_DIRECTION; - - tC->confTab.acfSplit[HIFILT] = -1; /* signal Merged4to2QuartersAutoCorrelation in FDKaacEnc_MergedAutoCorrelation*/ - tC->confTab.acfSplit[LOFILT] = -1; /* signal Merged4to2QuartersAutoCorrelation in FDKaacEnc_MergedAutoCorrelation */ - - tC->confTab.filterEnabled[HIFILT] = 1; - tC->confTab.filterEnabled[LOFILT] = 1; - tC->confTab.seperateFiltersAllowed = 1; - - /* compute autocorrelation window based on maximum filter order for given block type */ - /* for (i = 0; i <= tC->maxOrder + 3; i++) { - float acfWinTemp = acfTimeRes * i; - acfWindow[i] = FL2FXCONST_DBL(1.0f - acfWinTemp * acfWinTemp); - } - */ - if (blockType == SHORT_WINDOW) { - FDKmemcpy(tC->acfWindow[HIFILT], acfWindowShort, FDKmin(sizeof(acfWindowShort), sizeof(tC->acfWindow[HIFILT]))); - FDKmemcpy(tC->acfWindow[LOFILT], acfWindowShort, FDKmin(sizeof(acfWindowShort), sizeof(tC->acfWindow[HIFILT]))); - } - else { - FDKmemcpy(tC->acfWindow[HIFILT], acfWindowLong, FDKmin(sizeof(acfWindowLong), sizeof(tC->acfWindow[HIFILT]))); - FDKmemcpy(tC->acfWindow[LOFILT], acfWindowLong, FDKmin(sizeof(acfWindowLong), sizeof(tC->acfWindow[HIFILT]))); - } - break; - case 480: - case 512: - { - const TNS_PARAMETER_TABULATED* pCfg = FDKaacEnc_GetTnsParam(bitRate, channels, ldSbrPresent); - - if ( pCfg != NULL ) { - - FDKmemcpy(&(tC->confTab), pCfg, sizeof(tC->confTab)); - - tC->lpcStartBand[HIFILT] = FDKaacEnc_FreqToBandWithRounding(pCfg->filterStartFreq[HIFILT], sampleRate, pC->sfbCnt, pC->sfbOffset); - tC->lpcStartLine[HIFILT] = pC->sfbOffset[tC->lpcStartBand[HIFILT]]; - tC->lpcStartBand[LOFILT] = FDKaacEnc_FreqToBandWithRounding(pCfg->filterStartFreq[LOFILT], sampleRate, pC->sfbCnt, pC->sfbOffset); - tC->lpcStartLine[LOFILT] = pC->sfbOffset[tC->lpcStartBand[LOFILT]]; - - FDKaacEnc_CalcGaussWindow(tC->acfWindow[HIFILT], tC->maxOrder+1, sampleRate, granuleLength, pCfg->tnsTimeResolution[HIFILT], TNS_TIMERES_SCALE); - FDKaacEnc_CalcGaussWindow(tC->acfWindow[LOFILT], tC->maxOrder+1, sampleRate, granuleLength, pCfg->tnsTimeResolution[LOFILT], TNS_TIMERES_SCALE); - } - else { - tC->tnsActive = FALSE; /* no configuration available, disable tns tool */ - } - } - break; - default: - tC->tnsActive = FALSE; /* no configuration available, disable tns tool */ - } - - return AAC_ENC_OK; - -} - -/***************************************************************************/ -/*! - \brief FDKaacEnc_ScaleUpSpectrum - - Scales up spectrum lines in a given frequency section - - \param scaled spectrum - \param original spectrum - \param frequency line to start scaling - \param frequency line to enc scaling - - \return scale factor - -****************************************************************************/ -static inline INT FDKaacEnc_ScaleUpSpectrum( - FIXP_DBL *dest, - const FIXP_DBL *src, - const INT startLine, - const INT stopLine - ) -{ - INT i, scale; - - FIXP_DBL maxVal = FL2FXCONST_DBL(0.f); - - /* Get highest value in given spectrum */ - for (i=startLine; i>scale); - } - } - else { - for (i=startLine; i<(stopLine-lag); i++) { - result += (fMult(spectrum[i], spectrum[i+lag])>>scale); - } - } - - return result; -} - -/***************************************************************************/ -/*! - \brief FDKaacEnc_AutoCorrNormFac - - Autocorrelation function for 1st and 2nd half of the spectrum - - \param pointer to spectrum - \param pointer to autocorrelation window - \param filter start line - -****************************************************************************/ -static inline FIXP_DBL FDKaacEnc_AutoCorrNormFac( - const FIXP_DBL value, - const INT scale, - INT *sc - ) -{ - #define HLM_MIN_NRG 0.0000000037252902984619140625f /* 2^-28 */ - #define MAX_INV_NRGFAC (1.f/HLM_MIN_NRG) - - FIXP_DBL retValue; - FIXP_DBL A, B; - - if (scale>=0) { - A = value; - B = FL2FXCONST_DBL(HLM_MIN_NRG)>>fixMin(DFRACT_BITS-1,scale); - } - else { - A = value>>fixMin(DFRACT_BITS-1,(-scale)); - B = FL2FXCONST_DBL(HLM_MIN_NRG); - } - - if (A > B) { - int shift = 0; - FIXP_DBL tmp = invSqrtNorm2(value,&shift); - - retValue = fMult(tmp,tmp); - *sc += (2*shift); - } - else { - /* MAX_INV_NRGFAC*FDKpow(2,-28) = 1/2^-28 * 2^-28 = 1.0 */ - retValue = /*FL2FXCONST_DBL(MAX_INV_NRGFAC*FDKpow(2,-28))*/ (FIXP_DBL)MAXVAL_DBL; - *sc += scale+28; - } - - return retValue; -} - -static void FDKaacEnc_MergedAutoCorrelation( - const FIXP_DBL *spectrum, - const INT isLowDelay, - const FIXP_DBL acfWindow[MAX_NUM_OF_FILTERS][TNS_MAX_ORDER+3+1], - const INT lpcStartLine[MAX_NUM_OF_FILTERS], - const INT lpcStopLine, - const INT maxOrder, - const INT acfSplit[MAX_NUM_OF_FILTERS], - FIXP_DBL *_rxx1, - FIXP_DBL *_rxx2 - ) -{ - int i, idx0, idx1, idx2, idx3, idx4, lag; - FIXP_DBL rxx1_0, rxx2_0, rxx3_0, rxx4_0; - - /* buffer for temporal spectrum */ - C_ALLOC_SCRATCH_START(pSpectrum, FIXP_DBL, (1024)); - - /* pre-initialization output */ - FDKmemclear(&_rxx1[0], sizeof(FIXP_DBL)*(maxOrder+1)); - FDKmemclear(&_rxx2[0], sizeof(FIXP_DBL)*(maxOrder+1)); - - idx0 = idx1 = idx2 = idx3 = idx4 = 0; - - /* MDCT line indices separating the 1st, 2nd, 3rd, and 4th analysis quarters */ - if ( (acfSplit[LOFILT]==-1) || (acfSplit[HIFILT]==-1) ) { - /* autocorrelation function for 1st, 2nd, 3rd, and 4th quarter of the spectrum */ - idx0 = lpcStartLine[LOFILT]; - i = lpcStopLine - lpcStartLine[LOFILT]; - idx1 = idx0 + i / 4; - idx2 = idx0 + i / 2; - idx3 = idx0 + i * 3 / 4; - idx4 = lpcStopLine; - } - else { - FDK_ASSERT(acfSplit[LOFILT]==1); - FDK_ASSERT(acfSplit[HIFILT]==3); - i = (lpcStopLine - lpcStartLine[HIFILT]) / 3; - idx0 = lpcStartLine[LOFILT]; - idx1 = lpcStartLine[HIFILT]; - idx2 = idx1 + i; - idx3 = idx2 + i; - idx4 = lpcStopLine; - } - - /* copy spectrum to temporal buffer and scale up as much as possible */ - INT sc1 = FDKaacEnc_ScaleUpSpectrum(pSpectrum, spectrum, idx0, idx1); - INT sc2 = FDKaacEnc_ScaleUpSpectrum(pSpectrum, spectrum, idx1, idx2); - INT sc3 = FDKaacEnc_ScaleUpSpectrum(pSpectrum, spectrum, idx2, idx3); - INT sc4 = FDKaacEnc_ScaleUpSpectrum(pSpectrum, spectrum, idx3, idx4); - - /* get scaling values for summation */ - INT nsc1, nsc2, nsc3, nsc4; - for (nsc1=1; (1<dataRaw.Short.subBlockInfo[subBlockNumber] - : &tnsData->dataRaw.Long.subBlockInfo; - - tnsData->filtersMerged = FALSE; - - tsbi->tnsActive[HIFILT] = FALSE; - tsbi->predictionGain[HIFILT] = 1000; - tsbi->tnsActive[LOFILT] = FALSE; - tsbi->predictionGain[LOFILT] = 1000; - - tnsInfo->numOfFilters[subBlockNumber] = 0; - tnsInfo->coefRes[subBlockNumber] = tC->coefRes; - for (i = 0; i < tC->maxOrder; i++) { - tnsInfo->coef[subBlockNumber][HIFILT][i] = tnsInfo->coef[subBlockNumber][LOFILT][i] = 0; - } - - tnsInfo->length[subBlockNumber][HIFILT] = tnsInfo->length[subBlockNumber][LOFILT] = 0; - tnsInfo->order [subBlockNumber][HIFILT] = tnsInfo->order [subBlockNumber][LOFILT] = 0; - - if ( (tC->tnsActive) && (tC->maxOrder>0) ) - { - int sumSqrCoef; - - FDKaacEnc_MergedAutoCorrelation( - spectrum, - tC->isLowDelay, - tC->acfWindow, - tC->lpcStartLine, - tC->lpcStopLine, - tC->maxOrder, - tC->confTab.acfSplit, - rxx1, - rxx2); - - /* compute higher TNS filter in lattice (ParCor) form with LeRoux-Gueguen algorithm */ - tsbi->predictionGain[HIFILT] = FDKaacEnc_AutoToParcor(rxx2, parcor_tmp, tC->confTab.tnsLimitOrder[HIFILT]); - - /* non-linear quantization of TNS lattice coefficients with given resolution */ - FDKaacEnc_Parcor2Index( - parcor_tmp, - tnsInfo->coef[subBlockNumber][HIFILT], - tC->confTab.tnsLimitOrder[HIFILT], - tC->coefRes); - - /* reduce filter order by truncating trailing zeros, compute sum(abs(coefs)) */ - for (i = tC->confTab.tnsLimitOrder[HIFILT] - 1; i >= 0; i--) { - if (tnsInfo->coef[subBlockNumber][HIFILT][i] != 0) { - break; - } - } - - tnsInfo->order[subBlockNumber][HIFILT] = i + 1; - - sumSqrCoef = 0; - for (; i >= 0; i--) { - sumSqrCoef += tnsInfo->coef[subBlockNumber][HIFILT][i] * tnsInfo->coef[subBlockNumber][HIFILT][i]; - } - - tnsInfo->direction[subBlockNumber][HIFILT] = tC->confTab.tnsFilterDirection[HIFILT]; - tnsInfo->length[subBlockNumber][HIFILT] = sfbCnt - tC->lpcStartBand[HIFILT]; - - /* disable TNS if predictionGain is less than 3dB or sumSqrCoef is too small */ - if ((tsbi->predictionGain[HIFILT] > tC->confTab.threshOn[HIFILT]) || (sumSqrCoef > (tC->confTab.tnsLimitOrder[HIFILT]/2 + 2))) - { - tsbi->tnsActive[HIFILT] = TRUE; - tnsInfo->numOfFilters[subBlockNumber]++; - - /* compute second filter for lower quarter; only allowed for long windows! */ - if ( (blockType != SHORT_WINDOW) && - (tC->confTab.filterEnabled[LOFILT]) && (tC->confTab.seperateFiltersAllowed) ) - { - /* compute second filter for lower frequencies */ - - /* compute TNS filter in lattice (ParCor) form with LeRoux-Gueguen algorithm */ - INT predGain = FDKaacEnc_AutoToParcor(rxx1, parcor_tmp, tC->confTab.tnsLimitOrder[LOFILT]); - - /* non-linear quantization of TNS lattice coefficients with given resolution */ - FDKaacEnc_Parcor2Index( - parcor_tmp, - tnsInfo->coef[subBlockNumber][LOFILT], - tC->confTab.tnsLimitOrder[LOFILT], - tC->coefRes); - - /* reduce filter order by truncating trailing zeros, compute sum(abs(coefs)) */ - for (i = tC->confTab.tnsLimitOrder[LOFILT] - 1; i >= 0; i--) { - if (tnsInfo->coef[subBlockNumber][LOFILT][i] != 0) { - break; - } - } - tnsInfo->order[subBlockNumber][LOFILT] = i + 1; - - sumSqrCoef = 0; - for (; i >= 0; i--) { - sumSqrCoef += tnsInfo->coef[subBlockNumber][LOFILT][i] * tnsInfo->coef[subBlockNumber][LOFILT][i]; - } - - tnsInfo->direction[subBlockNumber][LOFILT] = tC->confTab.tnsFilterDirection[LOFILT]; - tnsInfo->length[subBlockNumber][LOFILT] = tC->lpcStartBand[HIFILT] - tC->lpcStartBand[LOFILT]; - - /* filter lower quarter if gain is high enough, but not if it's too high */ - if ( ( (predGain > tC->confTab.threshOn[LOFILT]) && (predGain < (16000 * tC->confTab.tnsLimitOrder[LOFILT])) ) - || ( (sumSqrCoef > 9) && (sumSqrCoef < 22 * tC->confTab.tnsLimitOrder[LOFILT]) ) ) - { - /* compare lower to upper filter; if they are very similar, merge them */ - tsbi->tnsActive[LOFILT] = TRUE; - sumSqrCoef = 0; - for (i = 0; i < tC->confTab.tnsLimitOrder[LOFILT]; i++) { - sumSqrCoef += FDKabs(tnsInfo->coef[subBlockNumber][HIFILT][i] - tnsInfo->coef[subBlockNumber][LOFILT][i]); - } - if ( (sumSqrCoef < 2) && - (tnsInfo->direction[subBlockNumber][LOFILT] == tnsInfo->direction[subBlockNumber][HIFILT]) ) - { - tnsData->filtersMerged = TRUE; - tnsInfo->length[subBlockNumber][HIFILT] = sfbCnt - tC->lpcStartBand[LOFILT]; - for (; i < tnsInfo->order[subBlockNumber][HIFILT]; i++) { - if (FDKabs(tnsInfo->coef[subBlockNumber][HIFILT][i]) > 1) { - break; - } - } - for (i--; i >= 0; i--) { - if (tnsInfo->coef[subBlockNumber][HIFILT][i] != 0) { - break; - } - } - if (i < tnsInfo->order[subBlockNumber][HIFILT]) { - tnsInfo->order[subBlockNumber][HIFILT] = i + 1; - } - } - else { - tnsInfo->numOfFilters[subBlockNumber]++; - } - } /* filter lower part */ - tsbi->predictionGain[LOFILT]=predGain; - - } /* second filter allowed */ - } /* if predictionGain > 1437 ... */ - } /* maxOrder > 0 && tnsActive */ - - return 0; - -} - - -/***************************************************************************/ -/*! - \brief FDKaacLdEnc_TnsSync - - synchronize TNS parameters when TNS gain difference small (relative) - - \param pointer to TNS data structure (destination) - \param pointer to TNS data structure (source) - \param pointer to TNS config structure - \param number of sub-block - \param block type - - \return void -****************************************************************************/ -void FDKaacEnc_TnsSync( - TNS_DATA *tnsDataDest, - const TNS_DATA *tnsDataSrc, - TNS_INFO *tnsInfoDest, - TNS_INFO *tnsInfoSrc, - const INT blockTypeDest, - const INT blockTypeSrc, - const TNS_CONFIG *tC - ) -{ - int i, w, absDiff, nWindows; - TNS_SUBBLOCK_INFO *sbInfoDest; - const TNS_SUBBLOCK_INFO *sbInfoSrc; - - /* if one channel contains short blocks and the other not, do not synchronize */ - if ( (blockTypeSrc == SHORT_WINDOW && blockTypeDest != SHORT_WINDOW) || - (blockTypeDest == SHORT_WINDOW && blockTypeSrc != SHORT_WINDOW) ) - { - return; - } - - if (blockTypeDest != SHORT_WINDOW) { - sbInfoDest = &tnsDataDest->dataRaw.Long.subBlockInfo; - sbInfoSrc = &tnsDataSrc->dataRaw.Long.subBlockInfo; - nWindows = 1; - } else { - sbInfoDest = &tnsDataDest->dataRaw.Short.subBlockInfo[0]; - sbInfoSrc = &tnsDataSrc->dataRaw.Short.subBlockInfo[0]; - nWindows = 8; - } - - for (w=0; wtnsActive[HIFILT] || pSbInfoSrcW->tnsActive[HIFILT]) { - for (i = 0; i < tC->maxOrder; i++) { - absDiff = FDKabs(tnsInfoDest->coef[w][HIFILT][i] - tnsInfoSrc->coef[w][HIFILT][i]); - absDiffSum += absDiff; - /* if coefficients diverge too much between channels, do not synchronize */ - if ((absDiff > 1) || (absDiffSum > 2)) { - doSync = 0; - break; - } - } - - if (doSync) { - /* if no significant difference was detected, synchronize coefficient sets */ - if (pSbInfoSrcW->tnsActive[HIFILT]) { - /* no dest filter, or more dest than source filters: use one dest filter */ - if ((!pSbInfoDestW->tnsActive[HIFILT]) || - ((pSbInfoDestW->tnsActive[HIFILT]) && (tnsInfoDest->numOfFilters[w] > tnsInfoSrc->numOfFilters[w]))) - { - pSbInfoDestW->tnsActive[HIFILT] = tnsInfoDest->numOfFilters[w] = 1; - } - tnsDataDest->filtersMerged = tnsDataSrc->filtersMerged; - tnsInfoDest->order [w][HIFILT] = tnsInfoSrc->order [w][HIFILT]; - tnsInfoDest->length [w][HIFILT] = tnsInfoSrc->length [w][HIFILT]; - tnsInfoDest->direction [w][HIFILT] = tnsInfoSrc->direction [w][HIFILT]; - tnsInfoDest->coefCompress[w][HIFILT] = tnsInfoSrc->coefCompress[w][HIFILT]; - - for (i = 0; i < tC->maxOrder; i++) { - tnsInfoDest->coef[w][HIFILT][i] = tnsInfoSrc->coef[w][HIFILT][i]; - } - } - else - pSbInfoDestW->tnsActive[HIFILT] = tnsInfoDest->numOfFilters[w] = 0; - } - } - - } -} - -/***************************************************************************/ -/*! - \brief FDKaacEnc_TnsEncode - - perform TNS encoding - - \param pointer to TNS info structure - \param pointer to TNS data structure - \param number of sfbs - \param pointer to TNS config structure - \param low-pass line - \param pointer to spectrum - \param number of sub-block - \param block type - - \return ERROR STATUS -****************************************************************************/ -INT FDKaacEnc_TnsEncode( - TNS_INFO* tnsInfo, - TNS_DATA* tnsData, - const INT numOfSfb, - const TNS_CONFIG *tC, - const INT lowPassLine, - FIXP_DBL* spectrum, - const INT subBlockNumber, - const INT blockType - ) -{ - INT i, startLine, stopLine; - - if ( ( (blockType == SHORT_WINDOW) && (!tnsData->dataRaw.Short.subBlockInfo[subBlockNumber].tnsActive[HIFILT]) ) - || ( (blockType != SHORT_WINDOW) && (!tnsData->dataRaw.Long.subBlockInfo.tnsActive[HIFILT]) ) ) - { - return 1; - } - - startLine = (tnsData->filtersMerged) ? tC->lpcStartLine[LOFILT] : tC->lpcStartLine[HIFILT]; - stopLine = tC->lpcStopLine; - - for (i=0; inumOfFilters[subBlockNumber]; i++) { - - INT lpcGainFactor; - FIXP_DBL LpcCoeff[TNS_MAX_ORDER]; - FIXP_DBL workBuffer[TNS_MAX_ORDER]; - FIXP_DBL parcor_tmp[TNS_MAX_ORDER]; - - FDKaacEnc_Index2Parcor( - tnsInfo->coef[subBlockNumber][i], - parcor_tmp, - tnsInfo->order[subBlockNumber][i], - tC->coefRes); - - lpcGainFactor = FDKaacEnc_ParcorToLpc( - parcor_tmp, - LpcCoeff, - tnsInfo->order[subBlockNumber][i], - workBuffer); - - FDKaacEnc_AnalysisFilter( - &spectrum[startLine], - stopLine - startLine, - LpcCoeff, - tnsInfo->order[subBlockNumber][i], - lpcGainFactor); - - /* update for second filter */ - startLine = tC->lpcStartLine[LOFILT]; - stopLine = tC->lpcStartLine[HIFILT]; - } - - return(0); - -} - -static void FDKaacEnc_CalcGaussWindow( - FIXP_DBL *win, - const int winSize, - const INT samplingRate, - const INT transformResolution, - const FIXP_DBL timeResolution, - const INT timeResolution_e - ) -{ - #define PI_E (2) - #define PI_M FL2FXCONST_DBL(3.1416f/(float)(1<> (DFRACT_BITS-1)); - tmp = (FIXP_DBL)((LONG)workBuffer[0]^sign); - - if(input[0]=0; j--) { - FIXP_DBL accu1 = fMult(tmp, input[j]); - FIXP_DBL accu2 = fMult(tmp, workBuffer[j]); - workBuffer[j] += accu1; - input[j] += accu2; - } - - workBuffer++; - } - - tmp = fMult((FIXP_DBL)((LONG)TNS_PREDGAIN_SCALE<<21), fDivNorm(fAbs(autoCorr_0), fAbs(input[0]), &scale)); - if ( fMultDiv2(autoCorr_0, input[0]) FDKaacEnc_tnsCoeff3Borders[i]) - index=i; - } - return(index-4); -} - -static INT FDKaacEnc_Search4(FIXP_DBL parcor) -{ - INT i, index=0; - - for(i=0;i<16;i++){ - if(parcor > FDKaacEnc_tnsCoeff4Borders[i]) - index=i; - } - return(index-8); -} - - -/***************************************************************************** - - functionname: FDKaacEnc_Parcor2Index - -*****************************************************************************/ -static void FDKaacEnc_Parcor2Index( - const FIXP_DBL *parcor, - INT *RESTRICT index, - const INT order, - const INT bitsPerCoeff - ) -{ - INT i; - for(i=0; i, - ptr. to work buffer (required size: order) - output: LPC coefficients - -*****************************************************************************/ -static INT FDKaacEnc_ParcorToLpc( - const FIXP_DBL *reflCoeff, - FIXP_DBL *RESTRICT LpcCoeff, - const INT numOfCoeff, - FIXP_DBL *RESTRICT workBuffer - ) -{ - INT i, j; - INT shiftval, par2LpcShiftVal = 6; /* 6 should be enough, bec. max(numOfCoeff) = 20 */ - FIXP_DBL maxVal = FL2FXCONST_DBL(0.0f); - - LpcCoeff[0] = reflCoeff[0] >> par2LpcShiftVal; - for(i=1; i> par2LpcShiftVal; - } - - /* normalize LpcCoeff and calc shiftfactor */ - for(i=0; i=par2LpcShiftVal) ? par2LpcShiftVal : shiftval; - - for(i=0; i0) { - - INT idx = 0; - - /* keep filter coefficients twice and save memory copy operation in - modulo state buffer */ -#if defined(ARCH_PREFER_MULT_32x16) - FIXP_SGL coeff[2*TNS_MAX_ORDER]; - const FIXP_SGL *pCoeff; - for(i=0;i=0); - signal[j] = (tmp< 12 */ - /* for Short: length TRANS_FAC*TNS_MAX_ORDER (only 5 for short LC) is required -> 8*5=40 */ - /* Currently TRANS_FAC*TNS_MAX_ORDER = 8*12 = 96 (for LC) is used (per channel)! Memory could be saved here! */ - INT coef[TRANS_FAC][MAX_NUM_OF_FILTERS][TNS_MAX_ORDER]; -}TNS_INFO; - -INT FDKaacEnc_FreqToBandWithRounding( - const INT freq, - const INT fs, - const INT numOfBands, - const INT *bandStartOffset - ); - -#endif /* _TNS_H */ diff --git a/libAACenc/src/adj_thr.cpp b/libAACenc/src/adj_thr.cpp deleted file mode 100644 index a79a9ae..0000000 --- a/libAACenc/src/adj_thr.cpp +++ /dev/null @@ -1,2666 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M. Werner - contents/description: Threshold compensation - -******************************************************************************/ - -#include "common_fix.h" - -#include "adj_thr_data.h" -#include "adj_thr.h" -#include "qc_data.h" -#include "sf_estim.h" -#include "aacEnc_ram.h" - - - - -#define INV_INT_TAB_SIZE (8) -static const FIXP_DBL invInt[INV_INT_TAB_SIZE] = -{ - 0x7fffffff, 0x7fffffff, 0x40000000, 0x2aaaaaaa, 0x20000000, 0x19999999, 0x15555555, 0x12492492 -}; - - -#define INV_SQRT4_TAB_SIZE (8) -static const FIXP_DBL invSqrt4[INV_SQRT4_TAB_SIZE] = -{ - 0x7fffffff, 0x7fffffff, 0x6ba27e65, 0x61424bb5, 0x5a827999, 0x55994845, 0x51c8e33c, 0x4eb160d1 -}; - - -/*static const INT invRedExp = 4;*/ -static const FIXP_DBL SnrLdMin1 = (FIXP_DBL)0xfcad0ddf; /*FL2FXCONST_DBL(FDKlog(0.316)/FDKlog(2.0)/LD_DATA_SCALING);*/ -static const FIXP_DBL SnrLdMin2 = (FIXP_DBL)0x0351e1a2; /*FL2FXCONST_DBL(FDKlog(3.16) /FDKlog(2.0)/LD_DATA_SCALING);*/ -static const FIXP_DBL SnrLdFac = (FIXP_DBL)0xff5b2c3e; /*FL2FXCONST_DBL(FDKlog(0.8) /FDKlog(2.0)/LD_DATA_SCALING);*/ - -static const FIXP_DBL SnrLdMin3 = (FIXP_DBL)0xfe000000; /*FL2FXCONST_DBL(FDKlog(0.5) /FDKlog(2.0)/LD_DATA_SCALING);*/ -static const FIXP_DBL SnrLdMin4 = (FIXP_DBL)0x02000000; /*FL2FXCONST_DBL(FDKlog(2.0) /FDKlog(2.0)/LD_DATA_SCALING);*/ -static const FIXP_DBL SnrLdMin5 = (FIXP_DBL)0xfc000000; /*FL2FXCONST_DBL(FDKlog(0.25) /FDKlog(2.0)/LD_DATA_SCALING);*/ - - -/* -The bits2Pe factors are choosen for the case that some times -the crash recovery strategy will be activated once. -*/ - -typedef struct { - INT bitrate; - ULONG bits2PeFactor_mono; - ULONG bits2PeFactor_mono_slope; - ULONG bits2PeFactor_stereo; - ULONG bits2PeFactor_stereo_slope; - ULONG bits2PeFactor_mono_scfOpt; - ULONG bits2PeFactor_mono_scfOpt_slope; - ULONG bits2PeFactor_stereo_scfOpt; - ULONG bits2PeFactor_stereo_scfOpt_slope; - -} BIT_PE_SFAC; - -typedef struct { - const INT sampleRate; - const BIT_PE_SFAC * pPeTab; - const INT nEntries; - -} BITS2PE_CFG_TAB; - -static const BIT_PE_SFAC S_Bits2PeTab16000[] = { - { 10000, 0x228F5C29, 0x02FEF55D, 0x1D70A3D7, 0x09BC9D6D, 0x228F5C29, 0x02FEF55D, 0x1C28F5C3, 0x0CBB92CA}, - { 24000, 0x23D70A3D, 0x029F16B1, 0x2199999A, 0x07DD4413, 0x23D70A3D, 0x029F16B1, 0x2199999A, 0x07DD4413}, - { 32000, 0x247AE148, 0x11B1D92B, 0x23851EB8, 0x01F75105, 0x247AE148, 0x110A137F, 0x23851EB8, 0x01F75105}, - { 48000, 0x2D1EB852, 0x6833C600, 0x247AE148, 0x014F8B59, 0x2CCCCCCD, 0x68DB8BAC, 0x247AE148, 0x01F75105}, - { 64000, 0x25c28f40, 0x00000000, 0x251EB852, 0x01480000, 0x25c28f40, 0x00000000, 0x2570A3D7, 0x01480000}, - { 96000, 0x25c28f40, 0x00000000, 0x26000000, 0x01000000, 0x25c28f40, 0x00000000, 0x26000000, 0x01000000}, - {128000, 0x25c28f40, 0x00000000, 0x270a3d80, 0x01000000, 0x25c28f40, 0x00000000, 0x270a3d80, 0x01000000}, - {148000, 0x25c28f40, 0x00000000, 0x28000000, 0x00000000, 0x25c28f40, 0x00000000, 0x28000000, 0x00000000} -}; - -static const BIT_PE_SFAC S_Bits2PeTab22050[] = { - { 16000, 0x1a8f5c29, 0x1797cc3a, 0x128f5c29, 0x18e75793, 0x175c28f6, 0x221426fe, 0x00000000, 0x5a708ede}, - { 24000, 0x2051eb85, 0x092ccf6c, 0x18a3d70a, 0x13a92a30, 0x1fae147b, 0xbcbe61d, 0x16147ae1, 0x18e75793}, - { 32000, 0x228f5c29, 0x029f16b1, 0x1d70a3d7, 0x088509c0, 0x228f5c29, 0x29f16b1, 0x1c28f5c3, 0x0b242071}, - { 48000, 0x23d70a3d, 0x014f8b59, 0x2199999a, 0x03eea20a, 0x23d70a3d, 0x14f8b59, 0x2199999a, 0x03eea20a}, - { 64000, 0x247ae148, 0x08d8ec96, 0x23851eb8, 0x00fba882, 0x247ae148, 0x88509c0, 0x23851eb8, 0x00fba882}, - { 96000, 0x2d1eb852, 0x3419e300, 0x247ae148, 0x00a7c5ac, 0x2ccccccd, 0x346dc5d6, 0x247ae148, 0x00fba882}, - {128000, 0x25c28f40, 0x00000000, 0x251eb852, 0x029f16b1, 0x60000000, 0x25c28f40, 0x2570a3d7, 0x009f16b1}, - {148000, 0x25c28f40, 0x00000000, 0x26b851ec, 0x00000000, 0x60000000, 0x25c28f40, 0x270a3d71, 0x00000000} -}; - -static const BIT_PE_SFAC S_Bits2PeTab24000[] = { - { 16000, 0x19eb851f, 0x13a92a30, 0x1147ae14, 0x164840e1, 0x1999999a, 0x12599ed8, 0x00000000, 0x46c764ae}, - { 24000, 0x1eb851ec, 0x0d1b7176, 0x16b851ec, 0x18e75793, 0x1e147ae1, 0x0fba8827, 0x1147ae14, 0x2c9081c3}, - { 32000, 0x21eb851f, 0x049667b6, 0x1ccccccd, 0x07357e67, 0x21eb851f, 0x03eea20a, 0x1c28f5c3, 0x07357e67}, - { 48000, 0x2428f5c3, 0x014f8b59, 0x2051eb85, 0x053e2d62, 0x23d70a3d, 0x01f75105, 0x1fae147b, 0x07357e67}, - { 64000, 0x24cccccd, 0x05e5f30e, 0x22e147ae, 0x01a36e2f, 0x24cccccd, 0x05e5f30e, 0x23333333, 0x014f8b59}, - { 96000, 0x2a8f5c29, 0x24b33db0, 0x247ae148, 0x00fba882, 0x2a8f5c29, 0x26fe718b, 0x247ae148, 0x00fba882}, - {128000, 0x4e666666, 0x1cd5f99c, 0x2570a3d7, 0x010c6f7a, 0x50a3d70a, 0x192a7371, 0x2570a3d7, 0x010c6f7a}, - {148000, 0x25c28f40, 0x00000000, 0x26147ae1, 0x00000000, 0x25c28f40, 0x00000000, 0x26147ae1, 0x00000000} -}; - -static const BIT_PE_SFAC S_Bits2PeTab32000[] = { - { 16000, 0x247ae140, 0xFFFFAC1E, 0x270a3d80, 0xFFFE9B7C, 0x14ccccc0, 0x000110A1, 0x15c28f60, 0xFFFEEF5F}, - { 24000, 0x23333340, 0x0fba8827, 0x21999980, 0x1b866e44, 0x18f5c280, 0x0fba8827, 0x119999a0, 0x4d551d69}, - { 32000, 0x1d70a3d7, 0x07357e67, 0x17ae147b, 0x09d49518, 0x1b851eb8, 0x0a7c5ac4, 0x12e147ae, 0x110a137f}, - { 48000, 0x20f5c28f, 0x049667b6, 0x1c7ae148, 0x053e2d62, 0x20a3d70a, 0x053e2d62, 0x1b333333, 0x05e5f30e}, - { 64000, 0x23333333, 0x029f16b1, 0x1f0a3d71, 0x02f2f987, 0x23333333, 0x029f16b1, 0x1e147ae1, 0x03eea20a}, - { 96000, 0x25c28f5c, 0x2c3c9eed, 0x21eb851f, 0x01f75105, 0x25c28f5c, 0x0a7c5ac4, 0x21eb851f, 0x01a36e2f}, - {128000, 0x50f5c28f, 0x18a43bb4, 0x23d70a3d, 0x010c6f7a, 0x30000000, 0x168b5cc0, 0x23851eb8, 0x0192a737}, - {148000, 0x25c28f40, 0x00000000, 0x247ae148, 0x00dfb23b, 0x3dc28f5c, 0x300f4aaf, 0x247ae148, 0x01bf6476}, - {160000, 0x25c28f40, 0xb15b5740, 0x24cccccd, 0x053e2d62, 0x4f5c28f6, 0xbefd0072, 0x251eb852, 0x04fb1184}, - {200000, 0x25c28f40, 0x00000000, 0x2b333333, 0x0836be91, 0x25c28f40, 0x00000000, 0x2b333333, 0x0890390f}, - {320000, 0x25c28f40, 0x00000000, 0x4947ae14, 0x00000000, 0x25c28f40, 0x00000000, 0x4a8f5c29, 0x00000000} -}; - -static const BIT_PE_SFAC S_Bits2PeTab44100[] = { - { 16000, 0x10a3d70a, 0x1797cc3a, 0x00000000, 0x00000000, 0x00000000, 0x59210386, 0x00000000, 0x00000000}, - { 24000, 0x16666666, 0x1797cc3a, 0x00000000, 0x639d5e4a, 0x15c28f5c, 0x12599ed8, 0x00000000, 0x5bc01a37}, - { 32000, 0x1c28f5c3, 0x049667b6, 0x1851eb85, 0x049667b6, 0x1a3d70a4, 0x088509c0, 0x16666666, 0x053e2d62}, - { 48000, 0x1e666666, 0x05e5f30e, 0x1a8f5c29, 0x049667b6, 0x1e666666, 0x05e5f30e, 0x18f5c28f, 0x05e5f30e}, - { 64000, 0x2147ae14, 0x0346dc5d, 0x1ccccccd, 0x02f2f987, 0x2147ae14, 0x02f2f987, 0x1bd70a3d, 0x039abf34}, - { 96000, 0x247ae148, 0x068db8bb, 0x1fae147b, 0x029f16b1, 0x2428f5c3, 0x0639d5e5, 0x1f5c28f6, 0x029f16b1}, - {128000, 0x2ae147ae, 0x1b435265, 0x223d70a4, 0x0192a737, 0x2a3d70a4, 0x1040bfe4, 0x21eb851f, 0x0192a737}, - {148000, 0x3b851eb8, 0x2832069c, 0x23333333, 0x00dfb23b, 0x3428f5c3, 0x2054c288, 0x22e147ae, 0x00dfb23b}, - {160000, 0x4a3d70a4, 0xc32ebe5a, 0x23851eb8, 0x01d5c316, 0x40000000, 0xcb923a2b, 0x23333333, 0x01d5c316}, - {200000, 0x25c28f40, 0x00000000, 0x25c28f5c, 0x0713f078, 0x25c28f40, 0x00000000, 0x2570a3d7, 0x072a4f17}, - {320000, 0x25c28f40, 0x00000000, 0x3fae147b, 0x00000000, 0x25c28f40, 0x00000000, 0x3fae147b, 0x00000000} -}; - -static const BIT_PE_SFAC S_Bits2PeTab48000[] = { - { 16000, 0x0f5c28f6, 0x31ceaf25, 0x00000000, 0x00000000, 0x00000000, 0x74a771c9, 0x00000000, 0x00000000}, - { 24000, 0x1b851eb8, 0x029f16b1, 0x00000000, 0x663c74fb, 0x1c7ae148, 0xe47991bd, 0x00000000, 0x49667b5f}, - { 32000, 0x1c28f5c3, 0x029f16b1, 0x18f5c28f, 0x07357e67, 0x15c28f5c, 0x0f12c27a, 0x11eb851f, 0x13016484}, - { 48000, 0x1d70a3d7, 0x053e2d62, 0x1c7ae148, 0xfe08aefc, 0x1d1eb852, 0x068db8bb, 0x1b333333, 0xfeb074a8}, - { 64000, 0x20000000, 0x03eea20a, 0x1b851eb8, 0x0346dc5d, 0x2051eb85, 0x0346dc5d, 0x1a8f5c29, 0x039abf34}, - { 96000, 0x23d70a3d, 0x053e2d62, 0x1eb851ec, 0x029f16b1, 0x23851eb8, 0x04ea4a8c, 0x1e147ae1, 0x02f2f987}, - {128000, 0x28f5c28f, 0x14727dcc, 0x2147ae14, 0x0218def4, 0x2851eb85, 0x0e27e0f0, 0x20f5c28f, 0x0218def4}, - {148000, 0x3570a3d7, 0x1cd5f99c, 0x228f5c29, 0x01bf6476, 0x30f5c28f, 0x18777e75, 0x223d70a4, 0x01bf6476}, - {160000, 0x40000000, 0xcb923a2b, 0x23333333, 0x0192a737, 0x39eb851f, 0xd08d4bae, 0x22e147ae, 0x0192a737}, - {200000, 0x25c28f40, 0x00000000, 0x251eb852, 0x06775a1b, 0x25c28f40, 0x00000000, 0x24cccccd, 0x06a4175a}, - {320000, 0x25c28f40, 0x00000000, 0x3ccccccd, 0x00000000, 0x25c28f40, 0x00000000, 0x3d1eb852, 0x00000000} -}; - -static const BITS2PE_CFG_TAB bits2PeConfigTab[] = { - { 16000, S_Bits2PeTab16000, sizeof(S_Bits2PeTab16000)/sizeof(BIT_PE_SFAC) }, - { 22050, S_Bits2PeTab22050, sizeof(S_Bits2PeTab22050)/sizeof(BIT_PE_SFAC) }, - { 24000, S_Bits2PeTab24000, sizeof(S_Bits2PeTab24000)/sizeof(BIT_PE_SFAC) }, - { 32000, S_Bits2PeTab32000, sizeof(S_Bits2PeTab32000)/sizeof(BIT_PE_SFAC) }, - { 44100, S_Bits2PeTab44100, sizeof(S_Bits2PeTab44100)/sizeof(BIT_PE_SFAC) }, - { 48000, S_Bits2PeTab48000, sizeof(S_Bits2PeTab48000)/sizeof(BIT_PE_SFAC) } -}; - - - -/* values for avoid hole flag */ -enum _avoid_hole_state { - NO_AH =0, - AH_INACTIVE =1, - AH_ACTIVE =2 -}; - - -/* Q format definitions */ -#define Q_BITFAC (24) /* Q scaling used in FDKaacEnc_bitresCalcBitFac() calculation */ -#define Q_AVGBITS (17) /* scale bit values */ - - -/***************************************************************************** - functionname: FDKaacEnc_InitBits2PeFactor - description: retrieve bits2PeFactor from table -*****************************************************************************/ -static void FDKaacEnc_InitBits2PeFactor( - FIXP_DBL *bits2PeFactor_m, - INT *bits2PeFactor_e, - const INT bitRate, - const INT nChannels, - const INT sampleRate, - const INT advancedBitsToPe, - const INT dZoneQuantEnable, - const INT invQuant - ) -{ - /* default bits2pe factor */ - FIXP_DBL bit2PE_m = FL2FXCONST_DBL(1.18f/(1<<(1))); - INT bit2PE_e = 1; - - /* make use of advanced bits to pe factor table */ - if (advancedBitsToPe) { - - int i; - const BIT_PE_SFAC *peTab = NULL; - INT size = 0; - - - /* Get correct table entry */ - for (i=0; i<(INT)(sizeof(bits2PeConfigTab)/sizeof(BITS2PE_CFG_TAB)); i++) { - if (sampleRate >= bits2PeConfigTab[i].sampleRate) { - peTab = bits2PeConfigTab[i].pPeTab; - size = bits2PeConfigTab[i].nEntries; - } - } - - if ( (peTab!=NULL) && (size!=0) ) { - - INT startB = -1; - LONG startPF = 0; - LONG peSlope = 0; - - /* stereo or mono mode and invQuant used or not */ - for (i=0; ibitRate) || ((i==size-2)) )) - { - if (nChannels==1) - { - startPF = (!invQuant) ? peTab[i].bits2PeFactor_mono : peTab[i].bits2PeFactor_mono_scfOpt; - peSlope = (!invQuant) ? peTab[i].bits2PeFactor_mono_slope : peTab[i].bits2PeFactor_mono_scfOpt_slope; - /*endPF = (!invQuant) ? peTab[i+1].bits2PeFactor_mono : peTab[i+1].bits2PeFactor_mono_scfOpt; - endB=peTab[i+1].bitrate;*/ - startB=peTab[i].bitrate; - break; - } - else - { - startPF = (!invQuant) ? peTab[i].bits2PeFactor_stereo : peTab[i].bits2PeFactor_stereo_scfOpt; - peSlope = (!invQuant) ? peTab[i].bits2PeFactor_stereo_slope : peTab[i].bits2PeFactor_stereo_scfOpt_slope; - /*endPF = (!invQuant) ? peTab[i+1].bits2PeFactor_stereo : peTab[i+1].bits2PeFactor_stereo_scfOpt; - endB=peTab[i+1].bitrate;*/ - startB=peTab[i].bitrate; - break; - } - } - } /* for i */ - - /* if a configuration is available */ - if (startB!=-1) { - /* linear interpolate to actual PEfactor */ - FIXP_DBL peFac = fMult((FIXP_DBL)(bitRate-startB)<<14, (FIXP_DBL)peSlope) << 2; - FIXP_DBL bit2PE = peFac + (FIXP_DBL)startPF; /* startPF_float = startPF << 2 */ - - /* sanity check if bits2pe value is high enough */ - if ( bit2PE >= (FL2FXCONST_DBL(0.35f) >> 2) ) { - bit2PE_m = bit2PE; - bit2PE_e = 2; /* table is fixed scaled */ - } - } /* br */ - } /* sr */ - } /* advancedBitsToPe */ - - - if (dZoneQuantEnable) - { - if(bit2PE_m >= (FL2FXCONST_DBL(0.6f))>>bit2PE_e) - { - /* Additional headroom for addition */ - bit2PE_m >>= 1; - bit2PE_e += 1; - } - - /* the quantTendencyCompensator compensates a lower bit consumption due to increasing the tendency to quantize low spectral values to the lower quantizer border for bitrates below a certain bitrate threshold --> see also function calcSfbDistLD in quantize.c */ - if ((bitRate/nChannels > 32000) && (bitRate/nChannels <= 40000)) { - bit2PE_m += (FL2FXCONST_DBL(0.4f))>>bit2PE_e; - } - else if (bitRate/nChannels > 20000) { - bit2PE_m += (FL2FXCONST_DBL(0.3f))>>bit2PE_e; - } - else if (bitRate/nChannels >= 16000) { - bit2PE_m += (FL2FXCONST_DBL(0.3f))>>bit2PE_e; - } - else { - bit2PE_m += (FL2FXCONST_DBL(0.0f))>>bit2PE_e; - } - } - - - /***** 3.) Return bits2pe factor *****/ - *bits2PeFactor_m = bit2PE_m; - *bits2PeFactor_e = bit2PE_e; -} - - -/***************************************************************************** -functionname: FDKaacEnc_bits2pe2 -description: convert from bits to pe -*****************************************************************************/ -static INT FDKaacEnc_bits2pe2( - const INT bits, - const FIXP_DBL factor_m, - const INT factor_e - ) -{ - return (INT)(fMult(factor_m, (FIXP_DBL)(bits<> (Q_AVGBITS-factor_e)); -} - -/***************************************************************************** -functionname: FDKaacEnc_calcThreshExp -description: loudness calculation (threshold to the power of redExp) -*****************************************************************************/ -static void FDKaacEnc_calcThreshExp(FIXP_DBL thrExp[(2)][MAX_GROUPED_SFB], - QC_OUT_CHANNEL* qcOutChannel[(2)], - PSY_OUT_CHANNEL* psyOutChannel[(2)], - const INT nChannels) -{ - INT ch, sfb, sfbGrp; - FIXP_DBL thrExpLdData; - - for (ch=0; chsfbCnt;sfbGrp+= psyOutChannel[ch]->sfbPerGroup) { - for (sfb=0; sfbmaxSfbPerGroup; sfb++) { - thrExpLdData = psyOutChannel[ch]->sfbThresholdLdData[sfbGrp+sfb]>>2 ; - thrExp[ch][sfbGrp+sfb] = CalcInvLdData(thrExpLdData); - } - } - } -} - - -/***************************************************************************** - functionname: FDKaacEnc_adaptMinSnr - description: reduce minSnr requirements for bands with relative low energies -*****************************************************************************/ -static void FDKaacEnc_adaptMinSnr(QC_OUT_CHANNEL *qcOutChannel[(2)], - PSY_OUT_CHANNEL *psyOutChannel[(2)], - MINSNR_ADAPT_PARAM *msaParam, - const INT nChannels) -{ - INT ch, sfb, sfbGrp, nSfb; - FIXP_DBL avgEnLD64, dbRatio, minSnrRed; - FIXP_DBL minSnrLimitLD64 = FL2FXCONST_DBL(-0.00503012648262f); /* ld64(0.8f) */ - FIXP_DBL nSfbLD64; - FIXP_DBL accu; - - for (ch=0; chsfbCnt; sfbGrp+=psyOutChannel[ch]->sfbPerGroup) { - for (sfb=0; sfbmaxSfbPerGroup; sfb++) { - accu += psyOutChannel[ch]->sfbEnergy[sfbGrp+sfb]>>6; - nSfb++; - } - } - - if ((accu == FL2FXCONST_DBL(0.0f)) || (nSfb == 0)) { - avgEnLD64 = FL2FXCONST_DBL(-1.0f); - } - else { - nSfbLD64 = CalcLdInt(nSfb); - avgEnLD64 = CalcLdData(accu); - avgEnLD64 = avgEnLD64 + FL2FXCONST_DBL(0.09375f) - nSfbLD64; /* 0.09375f: compensate shift with 6 */ - } - - /* reduce minSnr requirement by minSnr^minSnrRed dependent on avgEn/sfbEn */ - for (sfbGrp=0; sfbGrp < psyOutChannel[ch]->sfbCnt; sfbGrp+=psyOutChannel[ch]->sfbPerGroup) { - for (sfb=0; sfbmaxSfbPerGroup; sfb++) { - if ( (msaParam->startRatio + qcOutChannel[ch]->sfbEnergyLdData[sfbGrp+sfb]) < avgEnLD64 ) { - dbRatio = fMult((avgEnLD64 - qcOutChannel[ch]->sfbEnergyLdData[sfbGrp+sfb]),FL2FXCONST_DBL(0.3010299956f)); /* scaled by (1.0f/(10.0f*64.0f)) */ - minSnrRed = msaParam->redOffs + fMult(msaParam->redRatioFac,dbRatio); /* scaled by 1.0f/64.0f*/ - minSnrRed = fixMax(minSnrRed, msaParam->maxRed); /* scaled by 1.0f/64.0f*/ - qcOutChannel[ch]->sfbMinSnrLdData[sfbGrp+sfb] = (fMult(qcOutChannel[ch]->sfbMinSnrLdData[sfbGrp+sfb],minSnrRed)) << 6; - qcOutChannel[ch]->sfbMinSnrLdData[sfbGrp+sfb] = fixMin(minSnrLimitLD64, qcOutChannel[ch]->sfbMinSnrLdData[sfbGrp+sfb]); - } - } - } - } -} - - -/***************************************************************************** -functionname: FDKaacEnc_initAvoidHoleFlag -description: determine bands where avoid hole is not necessary resp. possible -*****************************************************************************/ -static void FDKaacEnc_initAvoidHoleFlag(QC_OUT_CHANNEL *qcOutChannel[(2)], - PSY_OUT_CHANNEL *psyOutChannel[(2)], - UCHAR ahFlag[(2)][MAX_GROUPED_SFB], - struct TOOLSINFO *toolsInfo, - const INT nChannels, - const PE_DATA *peData, - AH_PARAM *ahParam) -{ - INT ch, sfb, sfbGrp; - FIXP_DBL sfbEn, sfbEnm1; - FIXP_DBL sfbEnLdData; - FIXP_DBL avgEnLdData; - - /* decrease spread energy by 3dB for long blocks, resp. 2dB for shorts - (avoid more holes in long blocks) */ - for (ch=0; chlastWindowSequence != SHORT_WINDOW) { - for (sfbGrp = 0;sfbGrp < psyOutChannel[ch]->sfbCnt;sfbGrp+= psyOutChannel[ch]->sfbPerGroup) - for (sfb=0; sfbmaxSfbPerGroup; sfb++) - qcOutChan->sfbSpreadEnergy[sfbGrp+sfb] >>= 1 ; - } - else { - for (sfbGrp = 0;sfbGrp < psyOutChannel[ch]->sfbCnt;sfbGrp+= psyOutChannel[ch]->sfbPerGroup) - for (sfb=0; sfbmaxSfbPerGroup; sfb++) - qcOutChan->sfbSpreadEnergy[sfbGrp+sfb] = - fMult(FL2FXCONST_DBL(0.63f), - qcOutChan->sfbSpreadEnergy[sfbGrp+sfb]) ; - } - } - - /* increase minSnr for local peaks, decrease it for valleys */ - if (ahParam->modifyMinSnr) { - for(ch=0; chsfbCnt;sfbGrp+= psyOutChannel[ch]->sfbPerGroup){ - for (sfb=0; sfbmaxSfbPerGroup; sfb++) { - FIXP_DBL sfbEnp1, avgEn; - if (sfb > 0) - sfbEnm1 = qcOutChan->sfbEnergy[sfbGrp+sfb-1]; - else - sfbEnm1 = qcOutChan->sfbEnergy[sfbGrp+sfb]; - - if (sfb < psyOutChannel[ch]->maxSfbPerGroup-1) - sfbEnp1 = qcOutChan->sfbEnergy[sfbGrp+sfb+1]; - else - sfbEnp1 = qcOutChan->sfbEnergy[sfbGrp+sfb]; - - avgEn = (sfbEnm1>>1) + (sfbEnp1>>1); - avgEnLdData = CalcLdData(avgEn); - sfbEn = qcOutChan->sfbEnergy[sfbGrp+sfb]; - sfbEnLdData = qcOutChan->sfbEnergyLdData[sfbGrp+sfb]; - /* peak ? */ - if (sfbEn > avgEn) { - FIXP_DBL tmpMinSnrLdData; - if (psyOutChannel[ch]->lastWindowSequence==LONG_WINDOW) - tmpMinSnrLdData = fixMax( SnrLdFac + (FIXP_DBL)(avgEnLdData - sfbEnLdData), (FIXP_DBL)SnrLdMin1 ) ; - else - tmpMinSnrLdData = fixMax( SnrLdFac + (FIXP_DBL)(avgEnLdData - sfbEnLdData), (FIXP_DBL)SnrLdMin3 ) ; - - qcOutChan->sfbMinSnrLdData[sfbGrp+sfb] = - fixMin(qcOutChan->sfbMinSnrLdData[sfbGrp+sfb], tmpMinSnrLdData); - } - /* valley ? */ - if ( ((sfbEnLdData+(FIXP_DBL)SnrLdMin4) < (FIXP_DBL)avgEnLdData) && (sfbEn > FL2FXCONST_DBL(0.0)) ) { - FIXP_DBL tmpMinSnrLdData = avgEnLdData - sfbEnLdData -(FIXP_DBL)SnrLdMin4 + qcOutChan->sfbMinSnrLdData[sfbGrp+sfb]; - tmpMinSnrLdData = fixMin((FIXP_DBL)SnrLdFac, tmpMinSnrLdData); - qcOutChan->sfbMinSnrLdData[sfbGrp+sfb] = fixMin(tmpMinSnrLdData, - (FIXP_DBL)(qcOutChan->sfbMinSnrLdData[sfbGrp+sfb] + SnrLdMin2 )); - } - } - } - } - } - - /* stereo: adapt the minimum requirements sfbMinSnr of mid and - side channels to avoid spending unnoticable bits */ - if (nChannels == 2) { - QC_OUT_CHANNEL* qcOutChanM = qcOutChannel[0]; - QC_OUT_CHANNEL* qcOutChanS = qcOutChannel[1]; - PSY_OUT_CHANNEL* psyOutChanM = psyOutChannel[0]; - for(sfbGrp = 0;sfbGrp < psyOutChanM->sfbCnt;sfbGrp+= psyOutChanM->sfbPerGroup){ - for (sfb=0; sfbmaxSfbPerGroup; sfb++) { - if (toolsInfo->msMask[sfbGrp+sfb]) { - FIXP_DBL maxSfbEnLd = fixMax(qcOutChanM->sfbEnergyLdData[sfbGrp+sfb],qcOutChanS->sfbEnergyLdData[sfbGrp+sfb]); - FIXP_DBL maxThrLd, sfbMinSnrTmpLd; - - if ( ((SnrLdMin5>>1) + (maxSfbEnLd>>1) + (qcOutChanM->sfbMinSnrLdData[sfbGrp+sfb]>>1)) <= FL2FXCONST_DBL(-0.5f)) - maxThrLd = FL2FXCONST_DBL(-1.0f) ; - else - maxThrLd = SnrLdMin5 + maxSfbEnLd + qcOutChanM->sfbMinSnrLdData[sfbGrp+sfb]; - - if (qcOutChanM->sfbEnergy[sfbGrp+sfb] > FL2FXCONST_DBL(0.0f)) - sfbMinSnrTmpLd = maxThrLd - qcOutChanM->sfbEnergyLdData[sfbGrp+sfb]; - else - sfbMinSnrTmpLd = FL2FXCONST_DBL(0.0f); - - qcOutChanM->sfbMinSnrLdData[sfbGrp+sfb] = fixMax(qcOutChanM->sfbMinSnrLdData[sfbGrp+sfb],sfbMinSnrTmpLd); - - if (qcOutChanM->sfbMinSnrLdData[sfbGrp+sfb] <= FL2FXCONST_DBL(0.0f)) - qcOutChanM->sfbMinSnrLdData[sfbGrp+sfb] = fixMin(qcOutChanM->sfbMinSnrLdData[sfbGrp+sfb], (FIXP_DBL)SnrLdFac); - - if (qcOutChanS->sfbEnergy[sfbGrp+sfb] > FL2FXCONST_DBL(0.0f)) - sfbMinSnrTmpLd = maxThrLd - qcOutChanS->sfbEnergyLdData[sfbGrp+sfb]; - else - sfbMinSnrTmpLd = FL2FXCONST_DBL(0.0f); - - qcOutChanS->sfbMinSnrLdData[sfbGrp+sfb] = fixMax(qcOutChanS->sfbMinSnrLdData[sfbGrp+sfb],sfbMinSnrTmpLd); - - if (qcOutChanS->sfbMinSnrLdData[sfbGrp+sfb] <= FL2FXCONST_DBL(0.0f)) - qcOutChanS->sfbMinSnrLdData[sfbGrp+sfb] = fixMin(qcOutChanS->sfbMinSnrLdData[sfbGrp+sfb],(FIXP_DBL)SnrLdFac); - - if (qcOutChanM->sfbEnergy[sfbGrp+sfb]>qcOutChanM->sfbSpreadEnergy[sfbGrp+sfb]) - qcOutChanS->sfbSpreadEnergy[sfbGrp+sfb] = - fMult(qcOutChanS->sfbEnergy[sfbGrp+sfb], FL2FXCONST_DBL(0.9f)); - - if (qcOutChanS->sfbEnergy[sfbGrp+sfb]>qcOutChanS->sfbSpreadEnergy[sfbGrp+sfb]) - qcOutChanM->sfbSpreadEnergy[sfbGrp+sfb] = - fMult(qcOutChanM->sfbEnergy[sfbGrp+sfb], FL2FXCONST_DBL(0.9f)); - } - } - } - } - - /* init ahFlag (0: no ah necessary, 1: ah possible, 2: ah active */ - for(ch=0; chsfbCnt;sfbGrp+= psyOutChan->sfbPerGroup){ - for (sfb=0; sfbmaxSfbPerGroup; sfb++) { - if ((qcOutChan->sfbSpreadEnergy[sfbGrp+sfb] > qcOutChan->sfbEnergy[sfbGrp+sfb]) - || (qcOutChan->sfbMinSnrLdData[sfbGrp+sfb] > FL2FXCONST_DBL(0.0f))) { - ahFlag[ch][sfbGrp+sfb] = NO_AH; - } - else { - ahFlag[ch][sfbGrp+sfb] = AH_INACTIVE; - } - } - } - } -} - - - -/** - * \brief Calculate constants that do not change during successive pe calculations. - * - * \param peData Pointer to structure containing PE data of current element. - * \param psyOutChannel Pointer to PSY_OUT_CHANNEL struct holding nChannels elements. - * \param qcOutChannel Pointer to QC_OUT_CHANNEL struct holding nChannels elements. - * \param nChannels Number of channels in element. - * \param peOffset Fixed PE offset defined while FDKaacEnc_AdjThrInit() depending on bitrate. - * - * \return void - */ -static -void FDKaacEnc_preparePe(PE_DATA *peData, - PSY_OUT_CHANNEL* psyOutChannel[(2)], - QC_OUT_CHANNEL* qcOutChannel[(2)], - const INT nChannels, - const INT peOffset) -{ - INT ch; - - for(ch=0; chpeChannelData[ch], - psyOutChan->sfbEnergyLdData, - psyOutChan->sfbThresholdLdData, - qcOutChannel[ch]->sfbFormFactorLdData, - psyOutChan->sfbOffsets, - psyOutChan->sfbCnt, - psyOutChan->sfbPerGroup, - psyOutChan->maxSfbPerGroup); - } - peData->offset = peOffset; -} - -/** - * \brief Calculate weighting factor for threshold adjustment. - * - * Calculate weighting factor to be applied at energies and thresholds in ld64 format. - * - * \param peData, Pointer to PE data in current element. - * \param psyOutChannel Pointer to PSY_OUT_CHANNEL struct holding nChannels elements. - * \param qcOutChannel Pointer to QC_OUT_CHANNEL struct holding nChannels elements. - * \param toolsInfo Pointer to tools info struct of current element. - * \param adjThrStateElement Pointer to ATS_ELEMENT holding enFacPatch states. - * \param nChannels Number of channels in element. - * \param usePatchTool Apply the weighting tool 0 (no) else (yes). - * - * \return void - */ -static -void FDKaacEnc_calcWeighting(PE_DATA *peData, - PSY_OUT_CHANNEL* psyOutChannel[(2)], - QC_OUT_CHANNEL* qcOutChannel[(2)], - struct TOOLSINFO *toolsInfo, - ATS_ELEMENT* adjThrStateElement, - const INT nChannels, - const INT usePatchTool) -{ - int ch, noShortWindowInFrame = TRUE; - INT exePatchM = 0; - - for (ch=0; chlastWindowSequence == SHORT_WINDOW) { - noShortWindowInFrame = FALSE; - } - FDKmemclear(qcOutChannel[ch]->sfbEnFacLd, MAX_GROUPED_SFB*sizeof(FIXP_DBL)); - } - - if (usePatchTool==0) { - return; /* tool is disabled */ - } - - for (ch=0; chsfbCnt; sfbGrp+=psyOutChannel[ch]->sfbPerGroup) { - for (sfb=0; sfbmaxSfbPerGroup; sfb++) { - FIXP_DBL nrgFac12 = CalcInvLdData(psyOutChan->sfbEnergyLdData[sfbGrp+sfb]>>1); /* nrg^(1/2) */ - FIXP_DBL nrgFac14 = CalcInvLdData(psyOutChan->sfbEnergyLdData[sfbGrp+sfb]>>2); /* nrg^(1/4) */ - - /* maximal number of bands is 64, results scaling factor 6 */ - nLinesSum += peData->peChannelData[ch].sfbNLines[sfbGrp+sfb]; /* relevant lines */ - nrgTotal += ( psyOutChan->sfbEnergy[sfbGrp+sfb] >> 6 ); /* sum up nrg */ - nrgSum12 += ( nrgFac12 >> 6 ); /* sum up nrg^(2/4) */ - nrgSum14 += ( nrgFac14 >> 6 ); /* sum up nrg^(1/4) */ - nrgSum34 += ( fMult(nrgFac14, nrgFac12) >> 6 ); /* sum up nrg^(3/4) */ - } - } - - nrgTotal = CalcLdData(nrgTotal); /* get ld64 of total nrg */ - - nrgFacLd_14 = CalcLdData(nrgSum14) - nrgTotal; /* ld64(nrgSum14/nrgTotal) */ - nrgFacLd_12 = CalcLdData(nrgSum12) - nrgTotal; /* ld64(nrgSum12/nrgTotal) */ - nrgFacLd_34 = CalcLdData(nrgSum34) - nrgTotal; /* ld64(nrgSum34/nrgTotal) */ - - adjThrStateElement->chaosMeasureEnFac[ch] = FDKmax( FL2FXCONST_DBL(0.1875f), fDivNorm(nLinesSum,psyOutChan->sfbOffsets[psyOutChan->sfbCnt]) ); - - usePatch = (adjThrStateElement->chaosMeasureEnFac[ch] > FL2FXCONST_DBL(0.78125f)); - exePatch = ((usePatch) && (adjThrStateElement->lastEnFacPatch[ch])); - - for (sfbGrp = 0;sfbGrp < psyOutChannel[ch]->sfbCnt; sfbGrp+=psyOutChannel[ch]->sfbPerGroup) { - for (sfb=0; sfbmaxSfbPerGroup; sfb++) { - - INT sfbExePatch; - - /* for MS coupled SFBs, also execute patch in side channel if done in mid channel */ - if ((ch == 1) && (toolsInfo->msMask[sfbGrp+sfb])) { - sfbExePatch = exePatchM; - } - else { - sfbExePatch = exePatch; - } - - if ( (sfbExePatch) && (psyOutChan->sfbEnergy[sfbGrp+sfb]>FL2FXCONST_DBL(0.f)) ) - { - /* execute patch based on spectral flatness calculated above */ - if (adjThrStateElement->chaosMeasureEnFac[ch] > FL2FXCONST_DBL(0.8125f)) { - qcOutChannel[ch]->sfbEnFacLd[sfbGrp+sfb] = ( (nrgFacLd_14 + (psyOutChan->sfbEnergyLdData[sfbGrp+sfb]+(psyOutChan->sfbEnergyLdData[sfbGrp+sfb]>>1)))>>1 ); /* sfbEnergy^(3/4) */ - } - else if (adjThrStateElement->chaosMeasureEnFac[ch] > FL2FXCONST_DBL(0.796875f)) { - qcOutChannel[ch]->sfbEnFacLd[sfbGrp+sfb] = ( (nrgFacLd_12 + psyOutChan->sfbEnergyLdData[sfbGrp+sfb])>>1 ); /* sfbEnergy^(2/4) */ - } - else { - qcOutChannel[ch]->sfbEnFacLd[sfbGrp+sfb] = ( (nrgFacLd_34 + (psyOutChan->sfbEnergyLdData[sfbGrp+sfb]>>1))>>1 ); /* sfbEnergy^(1/4) */ - } - qcOutChannel[ch]->sfbEnFacLd[sfbGrp+sfb] = fixMin(qcOutChannel[ch]->sfbEnFacLd[sfbGrp+sfb],(FIXP_DBL)0); - - } - } - } /* sfb loop */ - - adjThrStateElement->lastEnFacPatch[ch] = usePatch; - exePatchM = exePatch; - } - else { - /* !noShortWindowInFrame */ - adjThrStateElement->chaosMeasureEnFac[ch] = FL2FXCONST_DBL(0.75f); - adjThrStateElement->lastEnFacPatch[ch] = TRUE; /* allow use of sfbEnFac patch in upcoming frame */ - } - - } /* ch loop */ - -} - - - - -/***************************************************************************** -functionname: FDKaacEnc_calcPe -description: calculate pe for both channels -*****************************************************************************/ -static -void FDKaacEnc_calcPe(PSY_OUT_CHANNEL* psyOutChannel[(2)], - QC_OUT_CHANNEL* qcOutChannel[(2)], - PE_DATA *peData, - const INT nChannels) -{ - INT ch; - - peData->pe = peData->offset; - peData->constPart = 0; - peData->nActiveLines = 0; - for(ch=0; chpeChannelData[ch]; - FDKaacEnc_calcSfbPe(&peData->peChannelData[ch], - qcOutChannel[ch]->sfbWeightedEnergyLdData, - qcOutChannel[ch]->sfbThresholdLdData, - psyOutChannel[ch]->sfbCnt, - psyOutChannel[ch]->sfbPerGroup, - psyOutChannel[ch]->maxSfbPerGroup, - psyOutChannel[ch]->isBook, - psyOutChannel[ch]->isScale); - - peData->pe += peChanData->pe; - peData->constPart += peChanData->constPart; - peData->nActiveLines += peChanData->nActiveLines; - } -} - -void FDKaacEnc_peCalculation(PE_DATA *peData, - PSY_OUT_CHANNEL* psyOutChannel[(2)], - QC_OUT_CHANNEL* qcOutChannel[(2)], - struct TOOLSINFO *toolsInfo, - ATS_ELEMENT* adjThrStateElement, - const INT nChannels) -{ - /* constants that will not change during successive pe calculations */ - FDKaacEnc_preparePe(peData, psyOutChannel, qcOutChannel, nChannels, adjThrStateElement->peOffset); - - /* calculate weighting factor for threshold adjustment */ - FDKaacEnc_calcWeighting(peData, psyOutChannel, qcOutChannel, toolsInfo, adjThrStateElement, nChannels, 1); -{ - /* no weighting of threholds and energies for mlout */ - /* weight energies and thresholds */ - int ch; - for (ch=0; chsfbCnt; sfbGrp+=psyOutChannel[ch]->sfbPerGroup) { - for (sfb=0; sfbmaxSfbPerGroup; sfb++) { - pQcOutCh->sfbWeightedEnergyLdData[sfb+sfbGrp] = pQcOutCh->sfbEnergyLdData[sfb+sfbGrp] - pQcOutCh->sfbEnFacLd[sfb+sfbGrp]; - pQcOutCh->sfbThresholdLdData[sfb+sfbGrp] -= pQcOutCh->sfbEnFacLd[sfb+sfbGrp]; - } - } - } -} - - /* pe without reduction */ - FDKaacEnc_calcPe(psyOutChannel, qcOutChannel, peData, nChannels); -} - - - -/***************************************************************************** -functionname: FDKaacEnc_FDKaacEnc_calcPeNoAH -description: sum the pe data only for bands where avoid hole is inactive -*****************************************************************************/ -static void FDKaacEnc_FDKaacEnc_calcPeNoAH(INT *pe, - INT *constPart, - INT *nActiveLines, - PE_DATA *peData, - UCHAR ahFlag[(2)][MAX_GROUPED_SFB], - PSY_OUT_CHANNEL* psyOutChannel[(2)], - const INT nChannels) -{ - INT ch, sfb,sfbGrp; - - INT pe_tmp = peData->offset; - INT constPart_tmp = 0; - INT nActiveLines_tmp = 0; - for(ch=0; chpeChannelData[ch]; - for(sfbGrp = 0;sfbGrp < psyOutChannel[ch]->sfbCnt;sfbGrp+= psyOutChannel[ch]->sfbPerGroup){ - for (sfb=0; sfbmaxSfbPerGroup; sfb++) { - if(ahFlag[ch][sfbGrp+sfb] < AH_ACTIVE) { - pe_tmp += peChanData->sfbPe[sfbGrp+sfb]; - constPart_tmp += peChanData->sfbConstPart[sfbGrp+sfb]; - nActiveLines_tmp += peChanData->sfbNActiveLines[sfbGrp+sfb]; - } - } - } - } - /* correct scaled pe and constPart values */ - *pe = pe_tmp >> PE_CONSTPART_SHIFT; - *constPart = constPart_tmp >> PE_CONSTPART_SHIFT; - - *nActiveLines = nActiveLines_tmp; -} - - -/***************************************************************************** -functionname: FDKaacEnc_reduceThresholdsCBR -description: apply reduction formula -*****************************************************************************/ -static const FIXP_DBL limitThrReducedLdData = (FIXP_DBL)0x00008000; /*FL2FXCONST_DBL(FDKpow(2.0,-LD_DATA_SCALING/4.0));*/ - -static void FDKaacEnc_reduceThresholdsCBR(QC_OUT_CHANNEL* qcOutChannel[(2)], - PSY_OUT_CHANNEL* psyOutChannel[(2)], - UCHAR ahFlag[(2)][MAX_GROUPED_SFB], - FIXP_DBL thrExp[(2)][MAX_GROUPED_SFB], - const INT nChannels, - const FIXP_DBL redVal, - const SCHAR redValScaling) -{ - INT ch, sfb, sfbGrp; - FIXP_DBL sfbEnLdData, sfbThrLdData, sfbThrReducedLdData; - FIXP_DBL sfbThrExp; - - for(ch=0; chsfbCnt; sfbGrp+= psyOutChannel[ch]->sfbPerGroup){ - for (sfb=0; sfbmaxSfbPerGroup; sfb++) { - sfbEnLdData = qcOutChan->sfbWeightedEnergyLdData[sfbGrp+sfb]; - sfbThrLdData = qcOutChan->sfbThresholdLdData[sfbGrp+sfb]; - sfbThrExp = thrExp[ch][sfbGrp+sfb]; - if ((sfbEnLdData > sfbThrLdData) && (ahFlag[ch][sfbGrp+sfb] != AH_ACTIVE)) { - - /* threshold reduction formula: - float tmp = thrExp[ch][sfb]+redVal; - tmp *= tmp; - sfbThrReduced = tmp*tmp; - */ - int minScale = fixMin(CountLeadingBits(sfbThrExp), CountLeadingBits(redVal) - (DFRACT_BITS-1-redValScaling) )-1; - - /* 4*log( sfbThrExp + redVal ) */ - sfbThrReducedLdData = CalcLdData(fAbs(scaleValue(sfbThrExp, minScale) + scaleValue(redVal,(DFRACT_BITS-1-redValScaling)+minScale))) - - (FIXP_DBL)(minScale<<(DFRACT_BITS-1-LD_DATA_SHIFT)); - sfbThrReducedLdData <<= 2; - - /* avoid holes */ - if ( ((sfbThrReducedLdData - sfbEnLdData) > qcOutChan->sfbMinSnrLdData[sfbGrp+sfb] ) - && (ahFlag[ch][sfbGrp+sfb] != NO_AH) ) - { - if (qcOutChan->sfbMinSnrLdData[sfbGrp+sfb] > (FL2FXCONST_DBL(-1.0f) - sfbEnLdData) ){ - sfbThrReducedLdData = fixMax((qcOutChan->sfbMinSnrLdData[sfbGrp+sfb] + sfbEnLdData), sfbThrLdData); - } - else sfbThrReducedLdData = sfbThrLdData; - ahFlag[ch][sfbGrp+sfb] = AH_ACTIVE; - } - - /* minimum of 29 dB Ratio for Thresholds */ - if ((sfbEnLdData+(FIXP_DBL)MAXVAL_DBL) > FL2FXCONST_DBL(9.6336206/LD_DATA_SCALING)){ - sfbThrReducedLdData = fixMax(sfbThrReducedLdData, (sfbEnLdData - FL2FXCONST_DBL(9.6336206/LD_DATA_SCALING))); - } - - qcOutChan->sfbThresholdLdData[sfbGrp+sfb] = sfbThrReducedLdData; - } - } - } - } -} - -/* similar to prepareSfbPe1() */ -static FIXP_DBL FDKaacEnc_calcChaosMeasure(PSY_OUT_CHANNEL *psyOutChannel, - const FIXP_DBL *sfbFormFactorLdData) -{ - #define SCALE_FORM_FAC (4) /* (SCALE_FORM_FAC+FORM_FAC_SHIFT) >= ld(FRAME_LENGTH)*/ - #define SCALE_NRGS (8) - #define SCALE_NLINES (16) - #define SCALE_NRGS_SQRT4 (2) /* 0.25 * SCALE_NRGS */ - #define SCALE_NLINES_P34 (12) /* 0.75 * SCALE_NLINES */ - - INT sfbGrp, sfb; - FIXP_DBL chaosMeasure; - INT frameNLines = 0; - FIXP_DBL frameFormFactor = FL2FXCONST_DBL(0.f); - FIXP_DBL frameEnergy = FL2FXCONST_DBL(0.f); - - for (sfbGrp=0; sfbGrpsfbCnt; sfbGrp+=psyOutChannel->sfbPerGroup) { - for (sfb=0; sfbmaxSfbPerGroup; sfb++){ - if (psyOutChannel->sfbEnergyLdData[sfbGrp+sfb] > psyOutChannel->sfbThresholdLdData[sfbGrp+sfb]) { - frameFormFactor += (CalcInvLdData(sfbFormFactorLdData[sfbGrp+sfb])>>SCALE_FORM_FAC); - frameNLines += (psyOutChannel->sfbOffsets[sfbGrp+sfb+1] - psyOutChannel->sfbOffsets[sfbGrp+sfb]); - frameEnergy += (psyOutChannel->sfbEnergy[sfbGrp+sfb]>>SCALE_NRGS); - } - } - } - - if(frameNLines > 0){ - - /* frameNActiveLines = frameFormFactor*2^FORM_FAC_SHIFT * ((frameEnergy *2^SCALE_NRGS)/frameNLines)^-0.25 - chaosMeasure = frameNActiveLines / frameNLines */ - chaosMeasure = - CalcInvLdData( (((CalcLdData(frameFormFactor)>>1) - - (CalcLdData(frameEnergy)>>(2+1))) - - (fMultDiv2(FL2FXCONST_DBL(0.75f),CalcLdData((FIXP_DBL)frameNLines<<(DFRACT_BITS-1-SCALE_NLINES))) - - (((FIXP_DBL)(SCALE_FORM_FAC-SCALE_NRGS_SQRT4+FORM_FAC_SHIFT-(SCALE_NLINES_P34))<<(DFRACT_BITS-1-LD_DATA_SHIFT))>>1)) - )<<1 ); - } else { - - /* assuming total chaos, if no sfb is above thresholds */ - chaosMeasure = FL2FXCONST_DBL(1.f); - } - - return chaosMeasure; -} - -/* apply reduction formula for VBR-mode */ -static void FDKaacEnc_reduceThresholdsVBR(QC_OUT_CHANNEL* qcOutChannel[(2)], - PSY_OUT_CHANNEL* psyOutChannel[(2)], - UCHAR ahFlag[(2)][MAX_GROUPED_SFB], - FIXP_DBL thrExp[(2)][MAX_GROUPED_SFB], - const INT nChannels, - const FIXP_DBL vbrQualFactor, - FIXP_DBL* chaosMeasureOld) -{ - INT ch, sfbGrp, sfb; - FIXP_DBL chGroupEnergy[TRANS_FAC][2];/*energy for each group and channel*/ - FIXP_DBL chChaosMeasure[2]; - FIXP_DBL frameEnergy = FL2FXCONST_DBL(1e-10f); - FIXP_DBL chaosMeasure = FL2FXCONST_DBL(0.f); - FIXP_DBL sfbEnLdData, sfbThrLdData, sfbThrExp; - FIXP_DBL sfbThrReducedLdData; - FIXP_DBL chaosMeasureAvg; - INT groupCnt; /* loop counter */ - FIXP_DBL redVal[TRANS_FAC]; /* reduction values; in short-block case one redVal for each group */ - QC_OUT_CHANNEL *qcOutChan = NULL; - PSY_OUT_CHANNEL *psyOutChan = NULL; - -#define SCALE_GROUP_ENERGY (8) - -#define CONST_CHAOS_MEAS_AVG_FAC_0 (FL2FXCONST_DBL(0.25f)) -#define CONST_CHAOS_MEAS_AVG_FAC_1 (FL2FXCONST_DBL(1.f-0.25f)) - -#define MIN_LDTHRESH (FL2FXCONST_DBL(-0.515625f)) - - - for(ch=0; chsfbCnt; sfbGrp+=psyOutChan->sfbPerGroup, groupCnt++) { - chGroupEnergy[groupCnt][ch] = FL2FXCONST_DBL(0.f); - for (sfb=0; sfbmaxSfbPerGroup; sfb++){ - chGroupEnergy[groupCnt][ch] += (psyOutChan->sfbEnergy[sfbGrp+sfb]>>SCALE_GROUP_ENERGY); - } - chEnergy += chGroupEnergy[groupCnt][ch]; - } - frameEnergy += chEnergy; - - /* chaosMeasure */ - if (psyOutChannel[0]->lastWindowSequence == SHORT_WINDOW) { - chChaosMeasure[ch] = FL2FXCONST_DBL(0.5f); /* assume a constant chaos measure of 0.5f for short blocks */ - } else { - chChaosMeasure[ch] = FDKaacEnc_calcChaosMeasure(psyOutChannel[ch], qcOutChannel[ch]->sfbFormFactorLdData); - } - chaosMeasure += fMult(chChaosMeasure[ch], chEnergy); - } - - if(frameEnergy > chaosMeasure) { - INT scale = CntLeadingZeros(frameEnergy) - 1; - FIXP_DBL num = chaosMeasure<>2) + fMult(FL2FXCONST_DBL(0.7f/(4.f*0.3f)), (chaosMeasure - FL2FXCONST_DBL(0.2f)))); - chaosMeasure = (fixMin((FIXP_DBL)(FL2FXCONST_DBL(1.0f)>>2), fixMax((FIXP_DBL)(FL2FXCONST_DBL(0.1f)>>2), chaosMeasure)))<<2; - - /* calculation of reduction value */ - if (psyOutChannel[0]->lastWindowSequence == SHORT_WINDOW){ /* short-blocks */ - FDK_ASSERT(TRANS_FAC==8); - #define WIN_TYPE_SCALE (3) - - INT sfbGrp, groupCnt=0; - for (sfbGrp=0; sfbGrpsfbCnt; sfbGrp+=psyOutChan->sfbPerGroup,groupCnt++) { - - FIXP_DBL groupEnergy = FL2FXCONST_DBL(0.f); - - for(ch=0;chgroupLen[groupCnt]<=INV_INT_TAB_SIZE); - groupEnergy = fMult(groupEnergy,invInt[psyOutChannel[0]->groupLen[groupCnt]]); /* correction of group energy */ - groupEnergy = fixMin(groupEnergy, frameEnergy>>WIN_TYPE_SCALE); /* do not allow an higher redVal as calculated framewise */ - - groupEnergy>>=2; /* 2*WIN_TYPE_SCALE = 6 => 6+2 = 8 ==> 8/4 = int number */ - - redVal[groupCnt] = fMult(fMult(vbrQualFactor,chaosMeasure), - CalcInvLdData(CalcLdData(groupEnergy)>>2) ) - << (int)( ( 2 + (2*WIN_TYPE_SCALE) + SCALE_GROUP_ENERGY )>>2 ) ; - - } - } else { /* long-block */ - - redVal[0] = fMult( fMult(vbrQualFactor,chaosMeasure), - CalcInvLdData(CalcLdData(frameEnergy)>>2) ) - << (int)( SCALE_GROUP_ENERGY>>2 ) ; - } - - for(ch=0; chsfbCnt; sfbGrp+=psyOutChan->sfbPerGroup) { - for (sfb=0; sfbmaxSfbPerGroup; sfb++){ - - sfbEnLdData = (qcOutChan->sfbWeightedEnergyLdData[sfbGrp+sfb]); - sfbThrLdData = (qcOutChan->sfbThresholdLdData[sfbGrp+sfb]); - sfbThrExp = thrExp[ch][sfbGrp+sfb]; - - if ( (sfbThrLdData>=MIN_LDTHRESH) && (sfbEnLdData > sfbThrLdData) && (ahFlag[ch][sfbGrp+sfb] != AH_ACTIVE)) { - - /* Short-Window */ - if (psyOutChannel[ch]->lastWindowSequence == SHORT_WINDOW) { - const int groupNumber = (int) sfb/psyOutChan->sfbPerGroup; - - FDK_ASSERT(INV_SQRT4_TAB_SIZE>psyOutChan->groupLen[groupNumber]); - - sfbThrExp = fMult(sfbThrExp, fMult( FL2FXCONST_DBL(2.82f/4.f), invSqrt4[psyOutChan->groupLen[groupNumber]]))<<2 ; - - if ( sfbThrExp <= (limitThrReducedLdData-redVal[groupNumber]) ) { - sfbThrReducedLdData = FL2FXCONST_DBL(-1.0f); - } - else { - if ((FIXP_DBL)redVal[groupNumber] >= FL2FXCONST_DBL(1.0f)-sfbThrExp) - sfbThrReducedLdData = FL2FXCONST_DBL(0.0f); - else { - /* threshold reduction formula */ - sfbThrReducedLdData = CalcLdData(sfbThrExp + redVal[groupNumber]); - sfbThrReducedLdData <<= 2; - } - } - sfbThrReducedLdData += ( CalcLdInt(psyOutChan->groupLen[groupNumber]) - - ((FIXP_DBL)6<<(DFRACT_BITS-1-LD_DATA_SHIFT)) ); - } - - /* Long-Window */ - else { - if ((FIXP_DBL)redVal[0] >= FL2FXCONST_DBL(1.0f)-sfbThrExp) { - sfbThrReducedLdData = FL2FXCONST_DBL(0.0f); - } - else { - /* threshold reduction formula */ - sfbThrReducedLdData = CalcLdData(sfbThrExp + redVal[0]); - sfbThrReducedLdData <<= 2; - } - } - - /* avoid holes */ - if ( ((sfbThrReducedLdData - sfbEnLdData) > qcOutChan->sfbMinSnrLdData[sfbGrp+sfb] ) - && (ahFlag[ch][sfbGrp+sfb] != NO_AH) ) - { - if (qcOutChan->sfbMinSnrLdData[sfbGrp+sfb] > (FL2FXCONST_DBL(-1.0f) - sfbEnLdData) ){ - sfbThrReducedLdData = fixMax((qcOutChan->sfbMinSnrLdData[sfbGrp+sfb] + sfbEnLdData), sfbThrLdData); - } - else sfbThrReducedLdData = sfbThrLdData; - ahFlag[ch][sfbGrp+sfb] = AH_ACTIVE; - } - - if (sfbThrReducedLdData FL2FXCONST_DBL(9.6336206/LD_DATA_SCALING)){ - sfbThrReducedLdData = fixMax(sfbThrReducedLdData, sfbEnLdData - FL2FXCONST_DBL(9.6336206/LD_DATA_SCALING)); - } - - sfbThrReducedLdData = fixMax(MIN_LDTHRESH,sfbThrReducedLdData); - - qcOutChan->sfbThresholdLdData[sfbGrp+sfb] = sfbThrReducedLdData; - } - } - } - } -} - -/***************************************************************************** -functionname: FDKaacEnc_correctThresh -description: if pe difference deltaPe between desired pe and real pe is small enough, -the difference can be distributed among the scale factor bands. -New thresholds can be derived from this pe-difference -*****************************************************************************/ -static void FDKaacEnc_correctThresh(CHANNEL_MAPPING* cm, - QC_OUT_ELEMENT* qcElement[(8)], - PSY_OUT_ELEMENT* psyOutElement[(8)], - UCHAR ahFlag[(8)][(2)][MAX_GROUPED_SFB], - FIXP_DBL thrExp[(8)][(2)][MAX_GROUPED_SFB], - const FIXP_DBL redVal[(8)], - const SCHAR redValScaling[(8)], - const INT deltaPe, - const INT processElements, - const INT elementOffset) -{ - INT ch, sfb, sfbGrp; - QC_OUT_CHANNEL *qcOutChan; - PSY_OUT_CHANNEL *psyOutChan; - PE_CHANNEL_DATA *peChanData; - FIXP_DBL thrFactorLdData; - FIXP_DBL sfbEnLdData, sfbThrLdData, sfbThrReducedLdData; - FIXP_DBL *sfbPeFactorsLdData[(8)][(2)]; - FIXP_DBL sfbNActiveLinesLdData[(8)][(2)][MAX_GROUPED_SFB]; - INT normFactorInt; - FIXP_DBL normFactorLdData; - - INT nElements = elementOffset+processElements; - INT elementId; - - /* scratch is empty; use temporal memory from quantSpec in QC_OUT_CHANNEL */ - for(elementId=elementOffset;elementIdelInfo[elementId].nChannelsInEl; ch++) { - SHORT* ptr = qcElement[elementId]->qcOutChannel[ch]->quantSpec; - sfbPeFactorsLdData[elementId][ch] = (FIXP_DBL*)ptr; - } - } - - /* for each sfb calc relative factors for pe changes */ - normFactorInt = 0; - - for(elementId=elementOffset;elementIdelInfo[elementId].elType != ID_DSE) { - - for(ch=0; chelInfo[elementId].nChannelsInEl; ch++) { - - qcOutChan = qcElement[elementId]->qcOutChannel[ch]; - psyOutChan = psyOutElement[elementId]->psyOutChannel[ch]; - peChanData = &qcElement[elementId]->peData.peChannelData[ch]; - - for(sfbGrp = 0; sfbGrp < psyOutChan->sfbCnt; sfbGrp+= psyOutChan->sfbPerGroup){ - for (sfb=0; sfbmaxSfbPerGroup; sfb++) { - - if ( peChanData->sfbNActiveLines[sfbGrp+sfb] == 0 ) { - sfbNActiveLinesLdData[elementId][ch][sfbGrp+sfb] = FL2FXCONST_DBL(-1.0f); - } - else { - /* Both CalcLdInt and CalcLdData can be used! - * No offset has to be subtracted, because sfbNActiveLinesLdData - * is shorted while thrFactor calculation */ - sfbNActiveLinesLdData[elementId][ch][sfbGrp+sfb] = CalcLdInt(peChanData->sfbNActiveLines[sfbGrp+sfb]); - } - if ( ((ahFlag[elementId][ch][sfbGrp+sfb] < AH_ACTIVE) || (deltaPe > 0)) && - peChanData->sfbNActiveLines[sfbGrp+sfb] != 0 ) - { - if (thrExp[elementId][ch][sfbGrp+sfb] > -redVal[elementId]) { - - /* sfbPeFactors[ch][sfbGrp+sfb] = peChanData->sfbNActiveLines[sfbGrp+sfb] / - (thrExp[elementId][ch][sfbGrp+sfb] + redVal[elementId]); */ - - int minScale = fixMin(CountLeadingBits(thrExp[elementId][ch][sfbGrp+sfb]), CountLeadingBits(redVal[elementId]) - (DFRACT_BITS-1-redValScaling[elementId]) ) - 1; - - /* sumld = ld64( sfbThrExp + redVal ) */ - FIXP_DBL sumLd = CalcLdData(scaleValue(thrExp[elementId][ch][sfbGrp+sfb], minScale) + scaleValue(redVal[elementId], (DFRACT_BITS-1-redValScaling[elementId])+minScale)) - - (FIXP_DBL)(minScale<<(DFRACT_BITS-1-LD_DATA_SHIFT)); - - if (sumLd < FL2FXCONST_DBL(0.f)) { - sfbPeFactorsLdData[elementId][ch][sfbGrp+sfb] = sfbNActiveLinesLdData[elementId][ch][sfbGrp+sfb] - sumLd; - } - else { - if ( sfbNActiveLinesLdData[elementId][ch][sfbGrp+sfb] > (FL2FXCONST_DBL(-1.f) + sumLd) ) { - sfbPeFactorsLdData[elementId][ch][sfbGrp+sfb] = sfbNActiveLinesLdData[elementId][ch][sfbGrp+sfb] - sumLd; - } - else { - sfbPeFactorsLdData[elementId][ch][sfbGrp+sfb] = sfbNActiveLinesLdData[elementId][ch][sfbGrp+sfb]; - } - } - - normFactorInt += (INT)CalcInvLdData(sfbPeFactorsLdData[elementId][ch][sfbGrp+sfb]); - } - else sfbPeFactorsLdData[elementId][ch][sfbGrp+sfb] = FL2FXCONST_DBL(1.0f); - } - else sfbPeFactorsLdData[elementId][ch][sfbGrp+sfb] = FL2FXCONST_DBL(-1.0f); - } - } - } - } - } - - /* normFactorLdData = ld64(deltaPe/normFactorInt) */ - normFactorLdData = CalcLdData((FIXP_DBL)((deltaPe<0) ? (-deltaPe) : (deltaPe))) - CalcLdData((FIXP_DBL)normFactorInt); - - /* distribute the pe difference to the scalefactors - and calculate the according thresholds */ - for(elementId=elementOffset;elementIdelInfo[elementId].elType != ID_DSE) { - - for(ch=0; chelInfo[elementId].nChannelsInEl; ch++) { - qcOutChan = qcElement[elementId]->qcOutChannel[ch]; - psyOutChan = psyOutElement[elementId]->psyOutChannel[ch]; - peChanData = &qcElement[elementId]->peData.peChannelData[ch]; - - for(sfbGrp = 0;sfbGrp < psyOutChan->sfbCnt;sfbGrp+= psyOutChan->sfbPerGroup){ - for (sfb=0; sfbmaxSfbPerGroup; sfb++) { - - if (peChanData->sfbNActiveLines[sfbGrp+sfb] > 0) { - - /* pe difference for this sfb */ - if ( (sfbPeFactorsLdData[elementId][ch][sfbGrp+sfb]==FL2FXCONST_DBL(-1.0f)) || - (deltaPe==0) ) - { - thrFactorLdData = FL2FXCONST_DBL(0.f); - } - else { - /* new threshold */ - FIXP_DBL tmp = CalcInvLdData(sfbPeFactorsLdData[elementId][ch][sfbGrp+sfb] + normFactorLdData - sfbNActiveLinesLdData[elementId][ch][sfbGrp+sfb] - FL2FXCONST_DBL((float)LD_DATA_SHIFT/LD_DATA_SCALING)); - - /* limit thrFactor to 60dB */ - tmp = (deltaPe<0) ? tmp : (-tmp); - thrFactorLdData = FDKmin(tmp, FL2FXCONST_DBL(20.f/LD_DATA_SCALING)); - } - - /* new threshold */ - sfbThrLdData = qcOutChan->sfbThresholdLdData[sfbGrp+sfb]; - sfbEnLdData = qcOutChan->sfbWeightedEnergyLdData[sfbGrp+sfb]; - - if (thrFactorLdData < FL2FXCONST_DBL(0.f)) { - if( sfbThrLdData > (FL2FXCONST_DBL(-1.f)-thrFactorLdData) ) { - sfbThrReducedLdData = sfbThrLdData + thrFactorLdData; - } - else { - sfbThrReducedLdData = FL2FXCONST_DBL(-1.f); - } - } - else{ - sfbThrReducedLdData = sfbThrLdData + thrFactorLdData; - } - - /* avoid hole */ - if ( (sfbThrReducedLdData - sfbEnLdData > qcOutChan->sfbMinSnrLdData[sfbGrp+sfb]) && - (ahFlag[elementId][ch][sfbGrp+sfb] == AH_INACTIVE) ) - { - /* sfbThrReduced = max(psyOutChan[ch]->sfbMinSnr[i] * sfbEn, sfbThr); */ - if ( sfbEnLdData > (sfbThrLdData-qcOutChan->sfbMinSnrLdData[sfbGrp+sfb]) ) { - sfbThrReducedLdData = qcOutChan->sfbMinSnrLdData[sfbGrp+sfb] + sfbEnLdData; - } - else { - sfbThrReducedLdData = sfbThrLdData; - } - ahFlag[elementId][ch][sfbGrp+sfb] = AH_ACTIVE; - } - - qcOutChan->sfbThresholdLdData[sfbGrp+sfb] = sfbThrReducedLdData; - } - } - } - } - } - } -} - -/***************************************************************************** - functionname: FDKaacEnc_reduceMinSnr - description: if the desired pe can not be reached, reduce pe by - reducing minSnr -*****************************************************************************/ -void FDKaacEnc_reduceMinSnr(CHANNEL_MAPPING* cm, - QC_OUT_ELEMENT* qcElement[(8)], - PSY_OUT_ELEMENT* psyOutElement[(8)], - UCHAR ahFlag[(8)][(2)][MAX_GROUPED_SFB], - const INT desiredPe, - INT* redPeGlobal, - const INT processElements, - const INT elementOffset) - -{ - INT elementId; - INT nElements = elementOffset+processElements; - - INT newGlobalPe = *redPeGlobal; - - for(elementId=elementOffset;elementIdelInfo[elementId].elType != ID_DSE) { - INT ch; - INT maxSfbPerGroup[2]; - INT sfbCnt[2]; - INT sfbPerGroup[2]; - - for(ch=0; chelInfo[elementId].nChannelsInEl; ch++) { - maxSfbPerGroup[ch] = psyOutElement[elementId]->psyOutChannel[ch]->maxSfbPerGroup-1; - sfbCnt[ch] = psyOutElement[elementId]->psyOutChannel[ch]->sfbCnt; - sfbPerGroup[ch] = psyOutElement[elementId]->psyOutChannel[ch]->sfbPerGroup; - } - - PE_DATA *peData = &qcElement[elementId]->peData; - - do - { - for(ch=0; chelInfo[elementId].nChannelsInEl; ch++) { - - INT sfb, sfbGrp; - QC_OUT_CHANNEL *qcOutChan = qcElement[elementId]->qcOutChannel[ch]; - INT noReduction = 1; - - if (maxSfbPerGroup[ch]>=0) { /* sfb in next channel */ - INT deltaPe = 0; - sfb = maxSfbPerGroup[ch]--; - noReduction = 0; - - for (sfbGrp = 0; sfbGrp < sfbCnt[ch]; sfbGrp += sfbPerGroup[ch]) { - - if (ahFlag[elementId][ch][sfbGrp+sfb] != NO_AH && - qcOutChan->sfbMinSnrLdData[sfbGrp+sfb] < SnrLdFac) - { - /* increase threshold to new minSnr of 1dB */ - qcOutChan->sfbMinSnrLdData[sfbGrp+sfb] = SnrLdFac; - - /* sfbThrReduced = max(psyOutChan[ch]->sfbMinSnr[i] * sfbEn, sfbThr); */ - if ( qcOutChan->sfbWeightedEnergyLdData[sfbGrp+sfb] >= qcOutChan->sfbThresholdLdData[sfbGrp+sfb] - qcOutChan->sfbMinSnrLdData[sfbGrp+sfb] ) { - - qcOutChan->sfbThresholdLdData[sfbGrp+sfb] = qcOutChan->sfbWeightedEnergyLdData[sfbGrp+sfb] + qcOutChan->sfbMinSnrLdData[sfbGrp+sfb]; - - /* calc new pe */ - /* C2 + C3*ld(1/0.8) = 1.5 */ - deltaPe -= (peData->peChannelData[ch].sfbPe[sfbGrp+sfb]>>PE_CONSTPART_SHIFT); - - /* sfbPe = 1.5 * sfbNLines */ - peData->peChannelData[ch].sfbPe[sfbGrp+sfb] = (3*peData->peChannelData[ch].sfbNLines[sfbGrp+sfb]) << (PE_CONSTPART_SHIFT-1); - deltaPe += (peData->peChannelData[ch].sfbPe[sfbGrp+sfb]>>PE_CONSTPART_SHIFT); - } - } - - } /* sfbGrp loop */ - - peData->pe += deltaPe; - peData->peChannelData[ch].pe += deltaPe; - newGlobalPe += deltaPe; - - /* stop if enough has been saved */ - if (peData->pe <= desiredPe) { - goto bail; - } - - } /* sfb > 0 */ - - if ( (ch==(cm->elInfo[elementId].nChannelsInEl-1)) && noReduction ) { - goto bail; - } - - } /* ch loop */ - - } while ( peData->pe > desiredPe); - - } /* != ID_DSE */ - } /* element loop */ - - -bail: - /* update global PE */ - *redPeGlobal = newGlobalPe; -} - - -/***************************************************************************** - functionname: FDKaacEnc_allowMoreHoles - description: if the desired pe can not be reached, some more scalefactor - bands have to be quantized to zero -*****************************************************************************/ -static void FDKaacEnc_allowMoreHoles(CHANNEL_MAPPING* cm, - QC_OUT_ELEMENT* qcElement[(8)], - PSY_OUT_ELEMENT* psyOutElement[(8)], - ATS_ELEMENT* AdjThrStateElement[(8)], - UCHAR ahFlag[(8)][(2)][MAX_GROUPED_SFB], - const INT desiredPe, - const INT currentPe, - const int processElements, - const int elementOffset) -{ - INT elementId; - INT nElements = elementOffset+processElements; - INT actPe = currentPe; - - if (actPe <= desiredPe) { - return; /* nothing to do */ - } - - for (elementId = elementOffset;elementIdelInfo[elementId].elType != ID_DSE) { - - INT ch, sfb, sfbGrp; - - PE_DATA *peData = &qcElement[elementId]->peData; - const INT nChannels = cm->elInfo[elementId].nChannelsInEl; - - QC_OUT_CHANNEL* qcOutChannel[(2)] = {NULL}; - PSY_OUT_CHANNEL* psyOutChannel[(2)] = {NULL}; - - for (ch=0; chqcOutChannel[ch]; - psyOutChannel[ch] = psyOutElement[elementId]->psyOutChannel[ch]; - - for(sfbGrp=0; sfbGrp < psyOutChannel[ch]->sfbCnt; sfbGrp+= psyOutChannel[ch]->sfbPerGroup) { - for (sfb=psyOutChannel[ch]->maxSfbPerGroup; sfbsfbPerGroup; sfb++) { - peData->peChannelData[ch].sfbPe[sfbGrp+sfb] = 0; - } - } - } - - /* for MS allow hole in the channel with less energy */ - if ( nChannels==2 && psyOutChannel[0]->lastWindowSequence==psyOutChannel[1]->lastWindowSequence ) { - - for (sfb=0; sfbmaxSfbPerGroup; sfb++) { - for(sfbGrp=0; sfbGrp < psyOutChannel[0]->sfbCnt; sfbGrp+=psyOutChannel[0]->sfbPerGroup) { - if (psyOutElement[elementId]->toolsInfo.msMask[sfbGrp+sfb]) { - FIXP_DBL EnergyLd_L = qcOutChannel[0]->sfbWeightedEnergyLdData[sfbGrp+sfb]; - FIXP_DBL EnergyLd_R = qcOutChannel[1]->sfbWeightedEnergyLdData[sfbGrp+sfb]; - - /* allow hole in side channel ? */ - if ( (ahFlag[elementId][1][sfbGrp+sfb] != NO_AH) && - (((FL2FXCONST_DBL(-0.02065512648f)>>1) + (qcOutChannel[0]->sfbMinSnrLdData[sfbGrp+sfb]>>1)) - > ((EnergyLd_R>>1) - (EnergyLd_L>>1))) ) - { - ahFlag[elementId][1][sfbGrp+sfb] = NO_AH; - qcOutChannel[1]->sfbThresholdLdData[sfbGrp+sfb] = FL2FXCONST_DBL(0.015625f) + EnergyLd_R; - actPe -= peData->peChannelData[1].sfbPe[sfbGrp+sfb]>>PE_CONSTPART_SHIFT; - } - /* allow hole in mid channel ? */ - else if ( (ahFlag[elementId][0][sfbGrp+sfb] != NO_AH) && - (((FL2FXCONST_DBL(-0.02065512648f)>>1) + (qcOutChannel[1]->sfbMinSnrLdData[sfbGrp+sfb]>>1)) - > ((EnergyLd_L>>1) - (EnergyLd_R>>1))) ) - { - ahFlag[elementId][0][sfbGrp+sfb] = NO_AH; - qcOutChannel[0]->sfbThresholdLdData[sfbGrp+sfb] = FL2FXCONST_DBL(0.015625f) + EnergyLd_L; - actPe -= peData->peChannelData[0].sfbPe[sfbGrp+sfb]>>PE_CONSTPART_SHIFT; - } /* if (ahFlag) */ - } /* if MS */ - } /* sfbGrp */ - if (actPe <= desiredPe) { - return; /* stop if enough has been saved */ - } - } /* sfb */ - } /* MS possible ? */ - - /* more holes necessary? subsequently erase bands - starting with low energies */ - INT startSfb[2]; - FIXP_DBL avgEnLD64,minEnLD64; - INT ahCnt; - FIXP_DBL ahCntLD64; - INT enIdx; - FIXP_DBL enLD64[4]; - FIXP_DBL avgEn; - - /* do not go below startSfb */ - for (ch=0; chlastWindowSequence != SHORT_WINDOW) - startSfb[ch] = AdjThrStateElement[elementId]->ahParam.startSfbL; - else - startSfb[ch] = AdjThrStateElement[elementId]->ahParam.startSfbS; - } - - /* calc avg and min energies of bands that avoid holes */ - avgEn = FL2FXCONST_DBL(0.0f); - minEnLD64 = FL2FXCONST_DBL(0.0f); - ahCnt = 0; - - for (ch=0; chmaxSfbPerGroup; sfb++) { - if ((ahFlag[elementId][ch][sfbGrp+sfb]!=NO_AH) && - (qcOutChannel[ch]->sfbWeightedEnergyLdData[sfbGrp+sfb] > qcOutChannel[ch]->sfbThresholdLdData[sfbGrp+sfb])){ - minEnLD64 = fixMin(minEnLD64,qcOutChannel[ch]->sfbEnergyLdData[sfbGrp+sfb]); - avgEn += qcOutChannel[ch]->sfbEnergy[sfbGrp+sfb] >> 6; - ahCnt++; - } - } - - sfbGrp += psyOutChannel[ch]->sfbPerGroup; - sfb=0; - - } while (sfbGrp < psyOutChannel[ch]->sfbCnt); - } - - if ( (avgEn == FL2FXCONST_DBL(0.0f)) || (ahCnt == 0) ) { - avgEnLD64 = FL2FXCONST_DBL(0.0f); - } - else { - avgEnLD64 = CalcLdData(avgEn); - ahCntLD64 = CalcLdInt(ahCnt); - avgEnLD64 = avgEnLD64 + FL2FXCONST_DBL(0.09375f) - ahCntLD64; /* compensate shift with 6 */ - } - - /* calc some energy borders between minEn and avgEn */ - /* for (enIdx=0; enIdx<4; enIdx++) */ - /* en[enIdx] = minEn * (float)FDKpow(avgEn/(minEn+FLT_MIN), (2*enIdx+1)/7.0f); */ - enLD64[0] = minEnLD64 + fMult((avgEnLD64-minEnLD64),FL2FXCONST_DBL(0.14285714285f)); - enLD64[1] = minEnLD64 + fMult((avgEnLD64-minEnLD64),FL2FXCONST_DBL(0.42857142857f)); - enLD64[2] = minEnLD64 + fMult((avgEnLD64-minEnLD64),FL2FXCONST_DBL(0.71428571428f)); - enLD64[3] = minEnLD64 + (avgEnLD64-minEnLD64); - - for (enIdx=0; enIdx<4; enIdx++) { - INT noReduction = 1; - - INT maxSfbPerGroup[2]; - INT sfbCnt[2]; - INT sfbPerGroup[2]; - - for(ch=0; chelInfo[elementId].nChannelsInEl; ch++) { - maxSfbPerGroup[ch] = psyOutElement[elementId]->psyOutChannel[ch]->maxSfbPerGroup-1; - sfbCnt[ch] = psyOutElement[elementId]->psyOutChannel[ch]->sfbCnt; - sfbPerGroup[ch] = psyOutElement[elementId]->psyOutChannel[ch]->sfbPerGroup; - } - - do { - - noReduction = 1; - - for(ch=0; chelInfo[elementId].nChannelsInEl; ch++) { - - INT sfb, sfbGrp; - - /* start with lowest energy border at highest sfb */ - if (maxSfbPerGroup[ch]>=startSfb[ch]) { /* sfb in next channel */ - sfb = maxSfbPerGroup[ch]--; - noReduction = 0; - - for (sfbGrp = 0; sfbGrp < sfbCnt[ch]; sfbGrp += sfbPerGroup[ch]) { - /* sfb energy below border ? */ - if (ahFlag[elementId][ch][sfbGrp+sfb] != NO_AH && qcOutChannel[ch]->sfbEnergyLdData[sfbGrp+sfb] < enLD64[enIdx]) { - /* allow hole */ - ahFlag[elementId][ch][sfbGrp+sfb] = NO_AH; - qcOutChannel[ch]->sfbThresholdLdData[sfbGrp+sfb] = FL2FXCONST_DBL(0.015625f) + qcOutChannel[ch]->sfbWeightedEnergyLdData[sfbGrp+sfb]; - actPe -= peData->peChannelData[ch].sfbPe[sfbGrp+sfb]>>PE_CONSTPART_SHIFT; - } - } /* sfbGrp */ - - if (actPe <= desiredPe) { - return; /* stop if enough has been saved */ - } - } /* sfb > 0 */ - } /* ch loop */ - - } while( (noReduction == 0) && (actPe > desiredPe) ); - - if (actPe <= desiredPe) { - return; /* stop if enough has been saved */ - } - - } /* enIdx loop */ - - } /* EOF DSE-suppression */ - } /* EOF for all elements... */ - -} - -/* reset avoid hole flags from AH_ACTIVE to AH_INACTIVE */ -static void FDKaacEnc_resetAHFlags( UCHAR ahFlag[(2)][MAX_GROUPED_SFB], - const int nChannels, - PSY_OUT_CHANNEL *psyOutChannel[(2)]) -{ - int ch, sfb, sfbGrp; - - for(ch=0; chsfbCnt; sfbGrp+=psyOutChannel[ch]->sfbPerGroup) { - for (sfb=0; sfbmaxSfbPerGroup; sfb++) { - if ( ahFlag[ch][sfbGrp+sfb] == AH_ACTIVE) { - ahFlag[ch][sfbGrp+sfb] = AH_INACTIVE; - } - } - } - } -} - - -static FIXP_DBL CalcRedValPower(FIXP_DBL num, - FIXP_DBL denum, - INT* scaling ) -{ - FIXP_DBL value = FL2FXCONST_DBL(0.f); - - if (num>=FL2FXCONST_DBL(0.f)) { - value = fDivNorm( num, denum, scaling); - } - else { - value = -fDivNorm( -num, denum, scaling); - } - value = f2Pow(value, *scaling, scaling); - *scaling = DFRACT_BITS-1-*scaling; - - return value; -} - - -/***************************************************************************** -functionname: FDKaacEnc_adaptThresholdsToPe -description: two guesses for the reduction value and one final correction of the thresholds -*****************************************************************************/ -static void FDKaacEnc_adaptThresholdsToPe(CHANNEL_MAPPING* cm, - ATS_ELEMENT* AdjThrStateElement[(8)], - QC_OUT_ELEMENT* qcElement[(8)], - PSY_OUT_ELEMENT* psyOutElement[(8)], - const INT desiredPe, - const INT maxIter2ndGuess, - const INT processElements, - const INT elementOffset) -{ - FIXP_DBL redValue[(8)]; - SCHAR redValScaling[(8)]; - UCHAR pAhFlag[(8)][(2)][MAX_GROUPED_SFB]; - FIXP_DBL pThrExp[(8)][(2)][MAX_GROUPED_SFB]; - int iter; - - INT constPartGlobal, noRedPeGlobal, nActiveLinesGlobal, redPeGlobal; - constPartGlobal = noRedPeGlobal = nActiveLinesGlobal = redPeGlobal = 0; - - int elementId; - - int nElements = elementOffset+processElements; - if(nElements > cm->nElements) { - nElements = cm->nElements; - } - - /* ------------------------------------------------------- */ - /* Part I: Initialize data structures and variables... */ - /* ------------------------------------------------------- */ - for (elementId = elementOffset;elementIdelInfo[elementId].elType != ID_DSE) { - - INT nChannels = cm->elInfo[elementId].nChannelsInEl; - PE_DATA *peData = &qcElement[elementId]->peData; - - /* thresholds to the power of redExp */ - FDKaacEnc_calcThreshExp(pThrExp[elementId], qcElement[elementId]->qcOutChannel, psyOutElement[elementId]->psyOutChannel, nChannels); - - /* lower the minSnr requirements for low energies compared to the average - energy in this frame */ - FDKaacEnc_adaptMinSnr(qcElement[elementId]->qcOutChannel, psyOutElement[elementId]->psyOutChannel, &AdjThrStateElement[elementId]->minSnrAdaptParam, nChannels); - - /* init ahFlag (0: no ah necessary, 1: ah possible, 2: ah active */ - FDKaacEnc_initAvoidHoleFlag(qcElement[elementId]->qcOutChannel, psyOutElement[elementId]->psyOutChannel, pAhFlag[elementId], &psyOutElement[elementId]->toolsInfo, nChannels, peData, &AdjThrStateElement[elementId]->ahParam); - - /* sum up */ - constPartGlobal += peData->constPart; - noRedPeGlobal += peData->pe; - nActiveLinesGlobal += fixMax((INT)peData->nActiveLines, 1); - - } /* EOF DSE-suppression */ - } /* EOF for all elements... */ - - /* ----------------------------------------------------------------------- */ - /* Part II: Calculate bit consumption of initial bit constraints setup */ - /* ----------------------------------------------------------------------- */ - for (elementId = elementOffset;elementIdelInfo[elementId].elType != ID_DSE) { - /* - redVal = ( 2 ^ ( (constPartGlobal-desiredPe) / (invRedExp*nActiveLinesGlobal) ) - - 2 ^ ( (constPartGlobal-noRedPeGlobal) / (invRedExp*nActiveLinesGlobal) ) ) - */ - - - INT nChannels = cm->elInfo[elementId].nChannelsInEl; - PE_DATA *peData = &qcElement[elementId]->peData; - - /* first guess of reduction value */ - int scale0=0, scale1=0; - FIXP_DBL tmp0 = CalcRedValPower( constPartGlobal-desiredPe, 4*nActiveLinesGlobal, &scale0 ); - FIXP_DBL tmp1 = CalcRedValPower( constPartGlobal-noRedPeGlobal, 4*nActiveLinesGlobal, &scale1 ); - - int scalMin = FDKmin(scale0, scale1)-1; - - redValue[elementId] = scaleValue(tmp0,(scalMin-scale0)) - scaleValue(tmp1,(scalMin-scale1)); - redValScaling[elementId] = scalMin; - - /* reduce thresholds */ - FDKaacEnc_reduceThresholdsCBR(qcElement[elementId]->qcOutChannel, psyOutElement[elementId]->psyOutChannel, pAhFlag[elementId], pThrExp[elementId], nChannels, redValue[elementId], redValScaling[elementId]); - - /* pe after first guess */ - FDKaacEnc_calcPe(psyOutElement[elementId]->psyOutChannel, qcElement[elementId]->qcOutChannel, peData, nChannels); - - redPeGlobal += peData->pe; - } /* EOF DSE-suppression */ - } /* EOF for all elements... */ - - /* -------------------------------------------------- */ - /* Part III: Iterate until bit constraints are met */ - /* -------------------------------------------------- */ - iter = 0; - while ((fixp_abs(redPeGlobal - desiredPe) > fMultI(FL2FXCONST_DBL(0.05f),desiredPe)) && (iter < maxIter2ndGuess)) { - - INT desiredPeNoAHGlobal; - INT redPeNoAHGlobal = 0; - INT constPartNoAHGlobal = 0; - INT nActiveLinesNoAHGlobal = 0; - - for (elementId = elementOffset;elementIdelInfo[elementId].elType != ID_DSE) { - - INT redPeNoAH, constPartNoAH, nActiveLinesNoAH; - INT nChannels = cm->elInfo[elementId].nChannelsInEl; - PE_DATA *peData = &qcElement[elementId]->peData; - - /* pe for bands where avoid hole is inactive */ - FDKaacEnc_FDKaacEnc_calcPeNoAH(&redPeNoAH, &constPartNoAH, &nActiveLinesNoAH, - peData, pAhFlag[elementId], psyOutElement[elementId]->psyOutChannel, nChannels); - - redPeNoAHGlobal += redPeNoAH; - constPartNoAHGlobal += constPartNoAH; - nActiveLinesNoAHGlobal += nActiveLinesNoAH; - } /* EOF DSE-suppression */ - } /* EOF for all elements... */ - - /* Calculate new redVal ... */ - if(desiredPe < redPeGlobal) { - - /* new desired pe without bands where avoid hole is active */ - desiredPeNoAHGlobal = desiredPe - (redPeGlobal - redPeNoAHGlobal); - - /* limit desiredPeNoAH to positive values, as the PE can not become negative */ - desiredPeNoAHGlobal = FDKmax(0,desiredPeNoAHGlobal); - - /* second guess (only if there are bands left where avoid hole is inactive)*/ - if (nActiveLinesNoAHGlobal > 0) { - for (elementId = elementOffset;elementIdelInfo[elementId].elType != ID_DSE) { - /* - redVal += ( 2 ^ ( (constPartNoAHGlobal-desiredPeNoAHGlobal) / (invRedExp*nActiveLinesNoAHGlobal) ) - - 2 ^ ( (constPartNoAHGlobal-redPeNoAHGlobal) / (invRedExp*nActiveLinesNoAHGlobal) ) ) - */ - int scale0 = 0; - int scale1 = 0; - - FIXP_DBL tmp0 = CalcRedValPower( constPartNoAHGlobal-desiredPeNoAHGlobal, 4*nActiveLinesNoAHGlobal, &scale0 ); - FIXP_DBL tmp1 = CalcRedValPower( constPartNoAHGlobal-redPeNoAHGlobal, 4*nActiveLinesNoAHGlobal, &scale1 ); - - int scalMin = FDKmin(scale0, scale1)-1; - - tmp0 = scaleValue(tmp0,(scalMin-scale0)) - scaleValue(tmp1,(scalMin-scale1)); - scale0 = scalMin; - - /* old reduction value */ - tmp1 = redValue[elementId]; - scale1 = redValScaling[elementId]; - - scalMin = fixMin(scale0,scale1)-1; - - /* sum up old and new reduction value */ - redValue[elementId] = scaleValue(tmp0,(scalMin-scale0)) + scaleValue(tmp1,(scalMin-scale1)); - redValScaling[elementId] = scalMin; - - } /* EOF DSE-suppression */ - } /* EOF for all elements... */ - } /* nActiveLinesNoAHGlobal > 0 */ - } - else { - /* desiredPe >= redPeGlobal */ - for (elementId = elementOffset;elementIdelInfo[elementId].elType != ID_DSE) { - - INT redVal_scale = 0; - FIXP_DBL tmp = fDivNorm((FIXP_DBL)redPeGlobal, (FIXP_DBL)desiredPe, &redVal_scale); - - /* redVal *= redPeGlobal/desiredPe; */ - redValue[elementId] = fMult(redValue[elementId], tmp); - redValScaling[elementId] -= redVal_scale; - - FDKaacEnc_resetAHFlags(pAhFlag[elementId], cm->elInfo[elementId].nChannelsInEl, psyOutElement[elementId]->psyOutChannel); - } /* EOF DSE-suppression */ - } /* EOF for all elements... */ - } - - redPeGlobal = 0; - /* Calculate new redVal's PE... */ - for (elementId = elementOffset;elementIdelInfo[elementId].elType != ID_DSE) { - - INT nChannels = cm->elInfo[elementId].nChannelsInEl; - PE_DATA *peData = &qcElement[elementId]->peData; - - /* reduce thresholds */ - FDKaacEnc_reduceThresholdsCBR(qcElement[elementId]->qcOutChannel, psyOutElement[elementId]->psyOutChannel, pAhFlag[elementId], pThrExp[elementId], nChannels, redValue[elementId], redValScaling[elementId]); - - /* pe after second guess */ - FDKaacEnc_calcPe(psyOutElement[elementId]->psyOutChannel, qcElement[elementId]->qcOutChannel, peData, nChannels); - redPeGlobal += peData->pe; - - } /* EOF DSE-suppression */ - } /* EOF for all elements... */ - - iter++; - } /* EOF while */ - - - /* ------------------------------------------------------- */ - /* Part IV: if still required, further reduce constraints */ - /* ------------------------------------------------------- */ - /* 1.0* 1.15* 1.20* - * desiredPe desiredPe desiredPe - * | | | - * ...XXXXXXXXXXXXXXXXXXXXXXXXXXX| | - * | | |XXXXXXXXXXX... - * | |XXXXXXXXXXX| - * --- A --- | --- B --- | --- C --- - * - * (X): redPeGlobal - * (A): FDKaacEnc_correctThresh() - * (B): FDKaacEnc_allowMoreHoles() - * (C): FDKaacEnc_reduceMinSnr() - */ - - /* correct thresholds to get closer to the desired pe */ - if ( redPeGlobal > desiredPe ) { - FDKaacEnc_correctThresh(cm, qcElement, psyOutElement, pAhFlag, pThrExp, redValue, redValScaling, - desiredPe - redPeGlobal, processElements, elementOffset); - - /* update PE */ - redPeGlobal = 0; - for(elementId=elementOffset;elementIdelInfo[elementId].elType != ID_DSE) { - - INT nChannels = cm->elInfo[elementId].nChannelsInEl; - PE_DATA *peData = &qcElement[elementId]->peData; - - /* pe after correctThresh */ - FDKaacEnc_calcPe(psyOutElement[elementId]->psyOutChannel, qcElement[elementId]->qcOutChannel, peData, nChannels); - redPeGlobal += peData->pe; - - } /* EOF DSE-suppression */ - } /* EOF for all elements... */ - } - - if ( redPeGlobal > desiredPe ) { - /* reduce pe by reducing minSnr requirements */ - FDKaacEnc_reduceMinSnr(cm, qcElement, psyOutElement, pAhFlag, - (fMultI(FL2FXCONST_DBL(0.15f),desiredPe) + desiredPe), - &redPeGlobal, processElements, elementOffset); - - /* reduce pe by allowing additional spectral holes */ - FDKaacEnc_allowMoreHoles(cm, qcElement, psyOutElement, AdjThrStateElement, pAhFlag, - desiredPe, redPeGlobal, processElements, elementOffset); - } - -} - -/* similar to FDKaacEnc_adaptThresholdsToPe(), for VBR-mode */ -void FDKaacEnc_AdaptThresholdsVBR(QC_OUT_CHANNEL* qcOutChannel[(2)], - PSY_OUT_CHANNEL* psyOutChannel[(2)], - ATS_ELEMENT* AdjThrStateElement, - struct TOOLSINFO *toolsInfo, - PE_DATA *peData, - const INT nChannels) -{ - UCHAR (*pAhFlag)[MAX_GROUPED_SFB]; - FIXP_DBL (*pThrExp)[MAX_GROUPED_SFB]; - - /* allocate scratch memory */ - C_ALLOC_SCRATCH_START(_pAhFlag, UCHAR, (2)*MAX_GROUPED_SFB) - C_ALLOC_SCRATCH_START(_pThrExp, FIXP_DBL, (2)*MAX_GROUPED_SFB) - pAhFlag = (UCHAR(*)[MAX_GROUPED_SFB])_pAhFlag; - pThrExp = (FIXP_DBL(*)[MAX_GROUPED_SFB])_pThrExp; - - /* thresholds to the power of redExp */ - FDKaacEnc_calcThreshExp(pThrExp, qcOutChannel, psyOutChannel, nChannels); - - /* lower the minSnr requirements for low energies compared to the average - energy in this frame */ - FDKaacEnc_adaptMinSnr(qcOutChannel, psyOutChannel, &AdjThrStateElement->minSnrAdaptParam, nChannels); - - /* init ahFlag (0: no ah necessary, 1: ah possible, 2: ah active */ - FDKaacEnc_initAvoidHoleFlag(qcOutChannel, psyOutChannel, pAhFlag, toolsInfo, - nChannels, peData, &AdjThrStateElement->ahParam); - - /* reduce thresholds */ - FDKaacEnc_reduceThresholdsVBR(qcOutChannel, psyOutChannel, pAhFlag, pThrExp, nChannels, - AdjThrStateElement->vbrQualFactor, - &AdjThrStateElement->chaosMeasureOld); - - /* free scratch memory */ - C_ALLOC_SCRATCH_END(_pThrExp, FIXP_DBL, (2)*MAX_GROUPED_SFB) - C_ALLOC_SCRATCH_END(_pAhFlag, UCHAR, (2)*MAX_GROUPED_SFB) -} - - -/***************************************************************************** - - functionname: FDKaacEnc_calcBitSave - description: Calculates percentage of bit save, see figure below - returns: - input: parameters and bitres-fullness - output: percentage of bit save - -*****************************************************************************/ -/* - bitsave - maxBitSave(%)| clipLow - |---\ - | \ - | \ - | \ - | \ - |--------\--------------> bitres - | \ - minBitSave(%)| \------------ - clipHigh maxBitres -*/ -static FIXP_DBL FDKaacEnc_calcBitSave(FIXP_DBL fillLevel, - const FIXP_DBL clipLow, - const FIXP_DBL clipHigh, - const FIXP_DBL minBitSave, - const FIXP_DBL maxBitSave, - const FIXP_DBL bitsave_slope) -{ - FIXP_DBL bitsave; - - fillLevel = fixMax(fillLevel, clipLow); - fillLevel = fixMin(fillLevel, clipHigh); - - bitsave = maxBitSave - fMult((fillLevel-clipLow), bitsave_slope); - - return (bitsave); -} - -/***************************************************************************** - - functionname: FDKaacEnc_calcBitSpend - description: Calculates percentage of bit spend, see figure below - returns: - input: parameters and bitres-fullness - output: percentage of bit spend - -*****************************************************************************/ -/* - bitspend clipHigh - maxBitSpend(%)| /-----------maxBitres - | / - | / - | / - | / - | / - |----/-----------------> bitres - | / - minBitSpend(%)|--/ - clipLow -*/ -static FIXP_DBL FDKaacEnc_calcBitSpend(FIXP_DBL fillLevel, - const FIXP_DBL clipLow, - const FIXP_DBL clipHigh, - const FIXP_DBL minBitSpend, - const FIXP_DBL maxBitSpend, - const FIXP_DBL bitspend_slope) -{ - FIXP_DBL bitspend; - - fillLevel = fixMax(fillLevel, clipLow); - fillLevel = fixMin(fillLevel, clipHigh); - - bitspend = minBitSpend + fMult(fillLevel-clipLow, bitspend_slope); - - return (bitspend); -} - - -/***************************************************************************** - - functionname: FDKaacEnc_adjustPeMinMax() - description: adjusts peMin and peMax parameters over time - returns: - input: current pe, peMin, peMax, bitres size - output: adjusted peMin/peMax - -*****************************************************************************/ -static void FDKaacEnc_adjustPeMinMax(const INT currPe, - INT *peMin, - INT *peMax) -{ - FIXP_DBL minFacHi = FL2FXCONST_DBL(0.3f), maxFacHi = (FIXP_DBL)MAXVAL_DBL, minFacLo = FL2FXCONST_DBL(0.14f), maxFacLo = FL2FXCONST_DBL(0.07f); - INT diff; - - INT minDiff_fix = fMultI(FL2FXCONST_DBL(0.1666666667f), currPe); - - if (currPe > *peMax) - { - diff = (currPe-*peMax) ; - *peMin += fMultI(minFacHi,diff); - *peMax += fMultI(maxFacHi,diff); - } - else if (currPe < *peMin) - { - diff = (*peMin-currPe) ; - *peMin -= fMultI(minFacLo,diff); - *peMax -= fMultI(maxFacLo,diff); - } - else - { - *peMin += fMultI(minFacHi, (currPe - *peMin)); - *peMax -= fMultI(maxFacLo, (*peMax - currPe)); - } - - if ((*peMax - *peMin) < minDiff_fix) - { - INT peMax_fix = *peMax, peMin_fix = *peMin; - FIXP_DBL partLo_fix, partHi_fix; - - partLo_fix = (FIXP_DBL)fixMax(0, currPe - peMin_fix); - partHi_fix = (FIXP_DBL)fixMax(0, peMax_fix - currPe); - - peMax_fix = (INT)(currPe + fMultI(fDivNorm(partHi_fix, (partLo_fix+partHi_fix)), minDiff_fix)); - peMin_fix = (INT)(currPe - fMultI(fDivNorm(partLo_fix, (partLo_fix+partHi_fix)), minDiff_fix)); - peMin_fix = fixMax(0, peMin_fix); - - *peMax = peMax_fix; - *peMin = peMin_fix; - } -} - - - -/***************************************************************************** - - functionname: BitresCalcBitFac - description: calculates factor of spending bits for one frame - 1.0 : take all frame dynpart bits - >1.0 : take all frame dynpart bits + bitres - <1.0 : put bits in bitreservoir - returns: BitFac - input: bitres-fullness, pe, blockType, parameter-settings - output: - -*****************************************************************************/ -/* - bitfac(%) pemax - bitspend(%) | /-----------maxBitres - | / - | / - | / - | / - | / - |----/-----------------> pe - | / - bitsave(%) |--/ - pemin -*/ - -static FIXP_DBL FDKaacEnc_bitresCalcBitFac(const INT bitresBits, - const INT maxBitresBits, - const INT pe, - const INT lastWindowSequence, - const INT avgBits, - const FIXP_DBL maxBitFac, - ADJ_THR_STATE *AdjThr, - ATS_ELEMENT *adjThrChan) -{ - BRES_PARAM *bresParam; - INT pex; - - INT qmin, qbr, qbres, qmbr; - FIXP_DBL bitSave, bitSpend; - - FIXP_DBL bitresFac_fix, tmp_cst, tmp_fix; - FIXP_DBL pe_pers, bits_ratio, maxBrVal; - FIXP_DBL bitsave_slope, bitspend_slope, maxBitFac_tmp; - FIXP_DBL fillLevel_fix = (FIXP_DBL)0x7fffffff; - FIXP_DBL UNITY = (FIXP_DBL)0x7fffffff; - FIXP_DBL POINT7 = (FIXP_DBL)0x5999999A; - - if (maxBitresBits > bitresBits) { - fillLevel_fix = fDivNorm(bitresBits, maxBitresBits); - } - - if (lastWindowSequence != SHORT_WINDOW) - { - bresParam = &(AdjThr->bresParamLong); - bitsave_slope = (FIXP_DBL)0x3BBBBBBC; - bitspend_slope = (FIXP_DBL)0x55555555; - } - else - { - bresParam = &(AdjThr->bresParamShort); - bitsave_slope = (FIXP_DBL)0x2E8BA2E9; - bitspend_slope = (FIXP_DBL)0x7fffffff; - } - - pex = fixMax(pe, adjThrChan->peMin); - pex = fixMin(pex, adjThrChan->peMax); - - bitSave = FDKaacEnc_calcBitSave(fillLevel_fix, - bresParam->clipSaveLow, bresParam->clipSaveHigh, - bresParam->minBitSave, bresParam->maxBitSave, bitsave_slope); - - bitSpend = FDKaacEnc_calcBitSpend(fillLevel_fix, - bresParam->clipSpendLow, bresParam->clipSpendHigh, - bresParam->minBitSpend, bresParam->maxBitSpend, bitspend_slope); - - pe_pers = (pex > adjThrChan->peMin) ? fDivNorm(pex - adjThrChan->peMin, adjThrChan->peMax - adjThrChan->peMin) : 0; - tmp_fix = fMult(((FIXP_DBL)bitSpend + (FIXP_DBL)bitSave), pe_pers); - bitresFac_fix = (UNITY>>1) - ((FIXP_DBL)bitSave>>1) + (tmp_fix>>1); qbres = (DFRACT_BITS-2); - - /* (float)bitresBits/(float)avgBits */ - bits_ratio = fDivNorm(bitresBits, avgBits, &qbr); - qbr = DFRACT_BITS-1-qbr; - - /* Add 0.7 in q31 to bits_ratio in qbr */ - /* 0.7f + (float)bitresBits/(float)avgBits */ - qmin = fixMin(qbr, (DFRACT_BITS-1)); - bits_ratio = bits_ratio >> (qbr - qmin); - tmp_cst = POINT7 >> ((DFRACT_BITS-1) - qmin); - maxBrVal = (bits_ratio>>1) + (tmp_cst>>1); qmbr = qmin - 1; - - /* bitresFac_fix = fixMin(bitresFac_fix, 0.7 + bitresBits/avgBits); */ - bitresFac_fix = bitresFac_fix >> (qbres - qmbr); qbres = qmbr; - bitresFac_fix = fixMin(bitresFac_fix, maxBrVal); - - /* Compare with maxBitFac */ - qmin = fixMin(Q_BITFAC, qbres); - bitresFac_fix = bitresFac_fix >> (qbres - qmin); - maxBitFac_tmp = maxBitFac >> (Q_BITFAC - qmin); - if(maxBitFac_tmp < bitresFac_fix) - { - bitresFac_fix = maxBitFac; - } - else - { - if(qmin < Q_BITFAC) - { - bitresFac_fix = bitresFac_fix << (Q_BITFAC-qmin); - } - else - { - bitresFac_fix = bitresFac_fix >> (qmin-Q_BITFAC); - } - } - - FDKaacEnc_adjustPeMinMax(pe, &adjThrChan->peMin, &adjThrChan->peMax); - - return bitresFac_fix; -} - - -/***************************************************************************** -functionname: FDKaacEnc_AdjThrNew -description: allocate ADJ_THR_STATE -*****************************************************************************/ -INT FDKaacEnc_AdjThrNew(ADJ_THR_STATE** phAdjThr, - INT nElements) -{ - INT err = 0; - INT i; - ADJ_THR_STATE* hAdjThr = GetRam_aacEnc_AdjustThreshold(); - if (hAdjThr==NULL) { - err = 1; - goto bail; - } - - for (i=0; iadjThrStateElem[i] = GetRam_aacEnc_AdjThrStateElement(i); - if (hAdjThr->adjThrStateElem[i]==NULL) { - err = 1; - goto bail; - } - } - -bail: - *phAdjThr = hAdjThr; - return err; -} - - -/***************************************************************************** -functionname: FDKaacEnc_AdjThrInit -description: initialize ADJ_THR_STATE -*****************************************************************************/ -void FDKaacEnc_AdjThrInit( - ADJ_THR_STATE *hAdjThr, - const INT meanPe, - ELEMENT_BITS *elBits[(8)], - INT invQuant, - INT nElements, - INT nChannelsEff, - INT sampleRate, - INT advancedBitsToPe, - FIXP_DBL vbrQualFactor, - const INT dZoneQuantEnable - ) -{ - INT i; - - FIXP_DBL POINT8 = FL2FXCONST_DBL(0.8f); - FIXP_DBL POINT6 = FL2FXCONST_DBL(0.6f); - - /* Max number of iterations in second guess is 3 for lowdelay aot and for configurations with - multiple audio elements in general, otherwise iteration value is always 1. */ - hAdjThr->maxIter2ndGuess = (advancedBitsToPe!=0 || nElements>1) ? 3 : 1; - - /* common for all elements: */ - /* parameters for bitres control */ - hAdjThr->bresParamLong.clipSaveLow = (FIXP_DBL)0x1999999a; /* FL2FXCONST_DBL(0.2f); */ - hAdjThr->bresParamLong.clipSaveHigh = (FIXP_DBL)0x7999999a; /* FL2FXCONST_DBL(0.95f); */ - hAdjThr->bresParamLong.minBitSave = (FIXP_DBL)0xf999999a; /* FL2FXCONST_DBL(-0.05f); */ - hAdjThr->bresParamLong.maxBitSave = (FIXP_DBL)0x26666666; /* FL2FXCONST_DBL(0.3f); */ - hAdjThr->bresParamLong.clipSpendLow = (FIXP_DBL)0x1999999a; /* FL2FXCONST_DBL(0.2f); */ - hAdjThr->bresParamLong.clipSpendHigh = (FIXP_DBL)0x7999999a; /* FL2FXCONST_DBL(0.95f); */ - hAdjThr->bresParamLong.minBitSpend = (FIXP_DBL)0xf3333333; /* FL2FXCONST_DBL(-0.10f); */ - hAdjThr->bresParamLong.maxBitSpend = (FIXP_DBL)0x33333333; /* FL2FXCONST_DBL(0.4f); */ - - hAdjThr->bresParamShort.clipSaveLow = (FIXP_DBL)0x199999a0; /* FL2FXCONST_DBL(0.2f); */ - hAdjThr->bresParamShort.clipSaveHigh = (FIXP_DBL)0x5fffffff; /* FL2FXCONST_DBL(0.75f); */ - hAdjThr->bresParamShort.minBitSave = (FIXP_DBL)0x00000000; /* FL2FXCONST_DBL(0.0f); */ - hAdjThr->bresParamShort.maxBitSave = (FIXP_DBL)0x199999a0; /* FL2FXCONST_DBL(0.2f); */ - hAdjThr->bresParamShort.clipSpendLow = (FIXP_DBL)0x199999a0; /* FL2FXCONST_DBL(0.2f); */ - hAdjThr->bresParamShort.clipSpendHigh = (FIXP_DBL)0x5fffffff; /* FL2FXCONST_DBL(0.75f); */ - hAdjThr->bresParamShort.minBitSpend = (FIXP_DBL)0xf9999998; /* FL2FXCONST_DBL(-0.05f); */ - hAdjThr->bresParamShort.maxBitSpend = (FIXP_DBL)0x40000000; /* FL2FXCONST_DBL(0.5f); */ - - /* specific for each element: */ - for (i=0; iadjThrStateElem[i]; - MINSNR_ADAPT_PARAM *msaParam = &atsElem->minSnrAdaptParam; - INT chBitrate = elBits[i]->chBitrateEl; - - /* parameters for bitres control */ - atsElem->peMin = fMultI(POINT8, meanPe) >> 1; - atsElem->peMax = fMultI(POINT6, meanPe); - - /* for use in FDKaacEnc_reduceThresholdsVBR */ - atsElem->chaosMeasureOld = FL2FXCONST_DBL(0.3f); - - /* additional pe offset to correct pe2bits for low bitrates */ - atsElem->peOffset = 0; - - /* vbr initialisation */ - atsElem->vbrQualFactor = vbrQualFactor; - if (chBitrate < 32000) - { - atsElem->peOffset = fixMax(50, 100-fMultI((FIXP_DBL)0x666667, chBitrate)); - } - - /* avoid hole parameters */ - if (chBitrate > 20000) { - atsElem->ahParam.modifyMinSnr = TRUE; - atsElem->ahParam.startSfbL = 15; - atsElem->ahParam.startSfbS = 3; - } - else { - atsElem->ahParam.modifyMinSnr = FALSE; - atsElem->ahParam.startSfbL = 0; - atsElem->ahParam.startSfbS = 0; - } - - /* minSnr adaptation */ - msaParam->maxRed = FL2FXCONST_DBL(0.00390625f); /* 0.25f/64.0f */ - /* start adaptation of minSnr for avgEn/sfbEn > startRatio */ - msaParam->startRatio = FL2FXCONST_DBL(0.05190512648f); /* ld64(10.0f) */ - /* maximum minSnr reduction to minSnr^maxRed is reached for - avgEn/sfbEn >= maxRatio */ - /* msaParam->maxRatio = 1000.0f; */ - /*msaParam->redRatioFac = ((float)1.0f - msaParam->maxRed) / ((float)10.0f*log10(msaParam->startRatio/msaParam->maxRatio)/log10(2.0f)*(float)0.3010299956f);*/ - msaParam->redRatioFac = FL2FXCONST_DBL(-0.375f); /* -0.0375f * 10.0f */ - /*msaParam->redOffs = (float)1.0f - msaParam->redRatioFac * (float)10.0f * log10(msaParam->startRatio)/log10(2.0f) * (float)0.3010299956f;*/ - msaParam->redOffs = FL2FXCONST_DBL(0.021484375); /* 1.375f/64.0f */ - - /* init pe correction */ - atsElem->peCorrectionFactor_m = FL2FXCONST_DBL(0.5f); /* 1.0 */ - atsElem->peCorrectionFactor_e = 1; - - atsElem->dynBitsLast = -1; - atsElem->peLast = 0; - - /* init bits to pe factor */ - - /* init bits2PeFactor */ - FDKaacEnc_InitBits2PeFactor( - &atsElem->bits2PeFactor_m, - &atsElem->bits2PeFactor_e, - chBitrate*nChannelsEff, /* overall bitrate */ - nChannelsEff, /* number of channels */ - sampleRate, - advancedBitsToPe, - dZoneQuantEnable, - invQuant - ); - - } /* for nElements */ - -} - - -/***************************************************************************** - functionname: FDKaacEnc_FDKaacEnc_calcPeCorrection - description: calc desired pe -*****************************************************************************/ -static void FDKaacEnc_FDKaacEnc_calcPeCorrection( - FIXP_DBL *const correctionFac_m, - INT *const correctionFac_e, - const INT peAct, - const INT peLast, - const INT bitsLast, - const FIXP_DBL bits2PeFactor_m, - const INT bits2PeFactor_e - ) -{ - if ( (bitsLast > 0) && (peAct < 1.5f*peLast) && (peAct > 0.7f*peLast) && - (FDKaacEnc_bits2pe2(bitsLast, fMult(FL2FXCONST_DBL(1.2f/2.f), bits2PeFactor_m), bits2PeFactor_e+1) > peLast) && - (FDKaacEnc_bits2pe2(bitsLast, fMult(FL2FXCONST_DBL(0.65f), bits2PeFactor_m), bits2PeFactor_e ) < peLast) ) - { - FIXP_DBL corrFac = *correctionFac_m; - - int scaling = 0; - FIXP_DBL denum = (FIXP_DBL)FDKaacEnc_bits2pe2(bitsLast, bits2PeFactor_m, bits2PeFactor_e); - FIXP_DBL newFac = fDivNorm((FIXP_DBL)peLast, denum, &scaling); - - /* dead zone, newFac and corrFac are scaled by 0.5 */ - if ((FIXP_DBL)peLast <= denum) { /* ratio <= 1.f */ - newFac = fixMax(scaleValue(fixMin( fMult(FL2FXCONST_DBL(1.1f/2.f), newFac), scaleValue(FL2FXCONST_DBL( 1.f/2.f), -scaling)), scaling), FL2FXCONST_DBL(0.85f/2.f) ); - } - else { /* ratio < 1.f */ - newFac = fixMax( fixMin( scaleValue(fMult(FL2FXCONST_DBL(0.9f/2.f), newFac), scaling), FL2FXCONST_DBL(1.15f/2.f) ), FL2FXCONST_DBL( 1.f/2.f) ); - } - - if ( ((newFac > FL2FXCONST_DBL(1.f/2.f)) && (corrFac < FL2FXCONST_DBL(1.f/2.f))) - || ((newFac < FL2FXCONST_DBL(1.f/2.f)) && (corrFac > FL2FXCONST_DBL(1.f/2.f)))) - { - corrFac = FL2FXCONST_DBL(1.f/2.f); - } - - /* faster adaptation towards 1.0, slower in the other direction */ - if ( (corrFac < FL2FXCONST_DBL(1.f/2.f) && newFac < corrFac) - || (corrFac > FL2FXCONST_DBL(1.f/2.f) && newFac > corrFac) ) - { - corrFac = fMult(FL2FXCONST_DBL(0.85f), corrFac) + fMult(FL2FXCONST_DBL(0.15f), newFac); - } - else { - corrFac = fMult(FL2FXCONST_DBL(0.7f), corrFac) + fMult(FL2FXCONST_DBL(0.3f), newFac); - } - - corrFac = fixMax( fixMin( corrFac, FL2FXCONST_DBL(1.15f/2.f) ), FL2FXCONST_DBL(0.85/2.f) ); - - *correctionFac_m = corrFac; - *correctionFac_e = 1; - } - else { - *correctionFac_m = FL2FXCONST_DBL(1.f/2.f); - *correctionFac_e = 1; - } -} - - -static void FDKaacEnc_calcPeCorrectionLowBitRes( - FIXP_DBL *const correctionFac_m, - INT *const correctionFac_e, - const INT peLast, - const INT bitsLast, - const INT bitresLevel, - const INT nChannels, - const FIXP_DBL bits2PeFactor_m, - const INT bits2PeFactor_e - ) -{ - /* tuning params */ - const FIXP_DBL amp = FL2FXCONST_DBL(0.005); - const FIXP_DBL maxDiff = FL2FXCONST_DBL(0.25f); - - if (bitsLast > 0) { - - /* Estimate deviation of granted and used dynamic bits in previous frame, in PE units */ - const int bitsBalLast = peLast - FDKaacEnc_bits2pe2( - bitsLast, - bits2PeFactor_m, - bits2PeFactor_e); - - /* reserve n bits per channel */ - int headroom = (bitresLevel>=50*nChannels) ? 0 : (100*nChannels); - - /* in PE units */ - headroom = FDKaacEnc_bits2pe2( - headroom, - bits2PeFactor_m, - bits2PeFactor_e); - - /* - * diff = amp * ((bitsBalLast - headroom) / (bitresLevel + headroom) - * diff = max ( min ( diff, maxDiff, -maxDiff)) / 2 - */ - FIXP_DBL denominator = (FIXP_DBL)FDKaacEnc_bits2pe2(bitresLevel, bits2PeFactor_m, bits2PeFactor_e) + (FIXP_DBL)headroom; - - int scaling = 0; - FIXP_DBL diff = (bitsBalLast>=headroom) - ? fMult(amp, fDivNorm( (FIXP_DBL)(bitsBalLast - headroom), denominator, &scaling)) - : -fMult(amp, fDivNorm(-(FIXP_DBL)(bitsBalLast - headroom), denominator, &scaling)) ; - - scaling -= 1; /* divide by 2 */ - - diff = (scaling<=0) ? FDKmax( FDKmin (diff>>(-scaling), maxDiff>>1), -maxDiff>>1) - : FDKmax( FDKmin (diff, maxDiff>>(1+scaling)), -maxDiff>>(1+scaling)) << scaling; - - /* - * corrFac += diff - * corrFac = max ( min ( corrFac/2.f, 1.f/2.f, 0.75f/2.f ) ) - */ - *correctionFac_m = FDKmax(FDKmin((*correctionFac_m)+diff, FL2FXCONST_DBL(1.0f/2.f)), FL2FXCONST_DBL(0.75f/2.f)) ; - *correctionFac_e = 1; - } - else { - *correctionFac_m = FL2FXCONST_DBL(0.75/2.f); - *correctionFac_e = 1; - } -} - -void FDKaacEnc_DistributeBits(ADJ_THR_STATE *adjThrState, - ATS_ELEMENT *AdjThrStateElement, - PSY_OUT_CHANNEL *psyOutChannel[(2)], - PE_DATA *peData, - INT *grantedPe, - INT *grantedPeCorr, - const INT nChannels, - const INT commonWindow, - const INT grantedDynBits, - const INT bitresBits, - const INT maxBitresBits, - const FIXP_DBL maxBitFac, - const INT bitDistributionMode) -{ - FIXP_DBL bitFactor; - INT noRedPe = peData->pe; - - /* prefer short windows for calculation of bitFactor */ - INT curWindowSequence = LONG_WINDOW; - if (nChannels==2) { - if ((psyOutChannel[0]->lastWindowSequence == SHORT_WINDOW) || - (psyOutChannel[1]->lastWindowSequence == SHORT_WINDOW)) { - curWindowSequence = SHORT_WINDOW; - } - } - else { - curWindowSequence = psyOutChannel[0]->lastWindowSequence; - } - - if (grantedDynBits >= 1) { - if (bitDistributionMode!=0) { - *grantedPe = FDKaacEnc_bits2pe2(grantedDynBits, AdjThrStateElement->bits2PeFactor_m, AdjThrStateElement->bits2PeFactor_e); - } - else - { - /* factor dependend on current fill level and pe */ - bitFactor = FDKaacEnc_bitresCalcBitFac(bitresBits, maxBitresBits, noRedPe, - curWindowSequence, grantedDynBits, maxBitFac, - adjThrState, - AdjThrStateElement - ); - - /* desired pe for actual frame */ - /* Worst case max of grantedDynBits is = 1024 * 5.27 * 2 */ - *grantedPe = FDKaacEnc_bits2pe2(grantedDynBits, - fMult(bitFactor, AdjThrStateElement->bits2PeFactor_m), AdjThrStateElement->bits2PeFactor_e+(DFRACT_BITS-1-Q_BITFAC) - ); - } - } - else { - *grantedPe = 0; /* prevent divsion by 0 */ - } - - /* correction of pe value */ - switch (bitDistributionMode) { - case 2: - case 1: - FDKaacEnc_calcPeCorrectionLowBitRes( - &AdjThrStateElement->peCorrectionFactor_m, - &AdjThrStateElement->peCorrectionFactor_e, - AdjThrStateElement->peLast, - AdjThrStateElement->dynBitsLast, - bitresBits, - nChannels, - AdjThrStateElement->bits2PeFactor_m, - AdjThrStateElement->bits2PeFactor_e - ); - break; - case 0: - default: - FDKaacEnc_FDKaacEnc_calcPeCorrection( - &AdjThrStateElement->peCorrectionFactor_m, - &AdjThrStateElement->peCorrectionFactor_e, - fixMin(*grantedPe, noRedPe), - AdjThrStateElement->peLast, - AdjThrStateElement->dynBitsLast, - AdjThrStateElement->bits2PeFactor_m, - AdjThrStateElement->bits2PeFactor_e - ); - break; - } - - *grantedPeCorr = (INT)(fMult((FIXP_DBL)(*grantedPe<peCorrectionFactor_m) >> (Q_AVGBITS-AdjThrStateElement->peCorrectionFactor_e)); - - /* update last pe */ - AdjThrStateElement->peLast = *grantedPe; - AdjThrStateElement->dynBitsLast = -1; - -} - -/***************************************************************************** -functionname: FDKaacEnc_AdjustThresholds -description: adjust thresholds -*****************************************************************************/ -void FDKaacEnc_AdjustThresholds(ATS_ELEMENT* AdjThrStateElement[(8)], - QC_OUT_ELEMENT* qcElement[(8)], - QC_OUT* qcOut, - PSY_OUT_ELEMENT* psyOutElement[(8)], - INT CBRbitrateMode, - INT maxIter2ndGuess, - CHANNEL_MAPPING* cm) -{ - int i; - if (CBRbitrateMode) - { - /* In case, no bits must be shifted between different elements, */ - /* an element-wise execution of the pe-dependent threshold- */ - /* adaption becomes necessary... */ - for (i=0; inElements; i++) - { - ELEMENT_INFO elInfo = cm->elInfo[i]; - - if ((elInfo.elType == ID_SCE) || (elInfo.elType == ID_CPE) || - (elInfo.elType == ID_LFE)) - { - /* qcElement[i]->grantedPe = 2000; */ /* Use this only for debugging */ - //if (totalGrantedPeCorr < totalNoRedPe) { - if (qcElement[i]->grantedPe < qcElement[i]->peData.pe) - { - /* calc threshold necessary for desired pe */ - FDKaacEnc_adaptThresholdsToPe(cm, - AdjThrStateElement, - qcElement, - psyOutElement, - qcElement[i]->grantedPeCorr, - maxIter2ndGuess, - 1, /* Process only 1 element */ - i); /* Process exactly THIS element */ - - } - - } /* -end- if(ID_SCE || ID_CPE || ID_LFE) */ - - } /* -end- element loop */ - } - else { - for (i=0; inElements; i++) - { - ELEMENT_INFO elInfo = cm->elInfo[i]; - - if ((elInfo.elType == ID_SCE) || (elInfo.elType == ID_CPE) || - (elInfo.elType == ID_LFE)) - { - /* for VBR-mode */ - FDKaacEnc_AdaptThresholdsVBR(qcElement[i]->qcOutChannel, - psyOutElement[i]->psyOutChannel, - AdjThrStateElement[i], - &psyOutElement[i]->toolsInfo, - &qcElement[i]->peData, - cm->elInfo[i].nChannelsInEl); - } /* -end- if(ID_SCE || ID_CPE || ID_LFE) */ - - } /* -end- element loop */ - - } - for (i=0; inElements; i++) { - int ch,sfb,sfbGrp; - /* no weighting of threholds and energies for mlout */ - /* weight energies and thresholds */ - for (ch=0; chelInfo[i].nChannelsInEl; ch++) { - QC_OUT_CHANNEL* pQcOutCh = qcElement[i]->qcOutChannel[ch]; - for (sfbGrp = 0;sfbGrp < psyOutElement[i]->psyOutChannel[ch]->sfbCnt; sfbGrp+=psyOutElement[i]->psyOutChannel[ch]->sfbPerGroup) { - for (sfb=0; sfbpsyOutChannel[ch]->maxSfbPerGroup; sfb++) { - pQcOutCh->sfbThresholdLdData[sfb+sfbGrp] += pQcOutCh->sfbEnFacLd[sfb+sfbGrp]; - } - } - } - } -} - -void FDKaacEnc_AdjThrClose(ADJ_THR_STATE** phAdjThr) -{ - INT i; - ADJ_THR_STATE* hAdjThr = *phAdjThr; - - if (hAdjThr!=NULL) { - for (i=0; i<(8); i++) { - if (hAdjThr->adjThrStateElem[i]!=NULL) { - FreeRam_aacEnc_AdjThrStateElement(&hAdjThr->adjThrStateElem[i]); - } - } - FreeRam_aacEnc_AdjustThreshold(phAdjThr); - } -} - diff --git a/libAACenc/src/adj_thr.h b/libAACenc/src/adj_thr.h deleted file mode 100644 index be68c6e..0000000 --- a/libAACenc/src/adj_thr.h +++ /dev/null @@ -1,149 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M. Werner - contents/description: Threshold compensation - -******************************************************************************/ - -#ifndef __ADJ_THR_H -#define __ADJ_THR_H - - -#include "adj_thr_data.h" -#include "qc_data.h" -#include "line_pe.h" -#include "interface.h" - - -void FDKaacEnc_peCalculation( - PE_DATA *peData, - PSY_OUT_CHANNEL* psyOutChannel[(2)], - QC_OUT_CHANNEL* qcOutChannel[(2)], - struct TOOLSINFO *toolsInfo, - ATS_ELEMENT* adjThrStateElement, - const INT nChannels - ); - -INT FDKaacEnc_AdjThrNew(ADJ_THR_STATE** phAdjThr, - INT nElements); - -void FDKaacEnc_AdjThrInit(ADJ_THR_STATE *hAdjThr, - const INT peMean, - ELEMENT_BITS* elBits[(8)], - INT invQuant, - INT nElements, - INT nChannelsEff, - INT sampleRate, - INT advancedBitsToPe, - FIXP_DBL vbrQualFactor, - const INT dZoneQuantEnable); - - -void FDKaacEnc_DistributeBits(ADJ_THR_STATE *adjThrState, - ATS_ELEMENT *AdjThrStateElement, - PSY_OUT_CHANNEL *psyOutChannel[(2)], - PE_DATA *peData, - INT *grantedPe, - INT *grantedPeCorr, - const INT nChannels, - const INT commonWindow, - const INT avgBits, - const INT bitresBits, - const INT maxBitresBits, - const FIXP_DBL maxBitFac, - const INT bitDistributionMode); - -void FDKaacEnc_AdjustThresholds(ATS_ELEMENT* AdjThrStateElement[(8)], - QC_OUT_ELEMENT* qcElement[(8)], - QC_OUT* qcOut, - PSY_OUT_ELEMENT* psyOutElement[(8)], - INT CBRbitrateMode, - INT maxIter2ndGuess, - CHANNEL_MAPPING* cm); - -void FDKaacEnc_AdjThrClose(ADJ_THR_STATE** hAdjThr); - -#endif diff --git a/libAACenc/src/adj_thr_data.h b/libAACenc/src/adj_thr_data.h deleted file mode 100644 index 7c3a191..0000000 --- a/libAACenc/src/adj_thr_data.h +++ /dev/null @@ -1,151 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/************************* Fast MPEG AAC Audio Encoder ********************** - - Initial author: M. Schug / A. Groeschel - contents/description: threshold calculations - -******************************************************************************/ - -#ifndef __ADJ_THR_DATA_H -#define __ADJ_THR_DATA_H - - -#include "psy_const.h" - -typedef struct { - FIXP_DBL clipSaveLow, clipSaveHigh; - FIXP_DBL minBitSave, maxBitSave; - FIXP_DBL clipSpendLow, clipSpendHigh; - FIXP_DBL minBitSpend, maxBitSpend; -} BRES_PARAM; - -typedef struct { - INT modifyMinSnr; - INT startSfbL, startSfbS; -} AH_PARAM; - -typedef struct { - FIXP_DBL maxRed; - FIXP_DBL startRatio; - FIXP_DBL maxRatio; - FIXP_DBL redRatioFac; - FIXP_DBL redOffs; -} MINSNR_ADAPT_PARAM; - -typedef struct { - /* parameters for bitreservoir control */ - INT peMin, peMax; - /* constant offset to pe */ - INT peOffset; - /* constant PeFactor */ - FIXP_DBL bits2PeFactor_m; - INT bits2PeFactor_e; - /* avoid hole parameters */ - AH_PARAM ahParam; - /* values for correction of pe */ - /* paramters for adaptation of minSnr */ - MINSNR_ADAPT_PARAM minSnrAdaptParam; - INT peLast; - INT dynBitsLast; - FIXP_DBL peCorrectionFactor_m; - INT peCorrectionFactor_e; - - /* vbr encoding */ - FIXP_DBL vbrQualFactor; - FIXP_DBL chaosMeasureOld; - - /* threshold weighting */ - FIXP_DBL chaosMeasureEnFac[(2)]; - INT lastEnFacPatch[(2)]; - -} ATS_ELEMENT; - -typedef struct { - BRES_PARAM bresParamLong, bresParamShort; - ATS_ELEMENT* adjThrStateElem[(8)]; - INT maxIter2ndGuess; -} ADJ_THR_STATE; - -#endif diff --git a/libAACenc/src/band_nrg.cpp b/libAACenc/src/band_nrg.cpp deleted file mode 100644 index 861f7a8..0000000 --- a/libAACenc/src/band_nrg.cpp +++ /dev/null @@ -1,359 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/***************************** MPEG-4 AAC Encoder ************************** - - Initial author: M. Werner - contents/description: Band/Line energy calculations - -******************************************************************************/ - -#include "band_nrg.h" - - -/***************************************************************************** - functionname: FDKaacEnc_CalcSfbMaxScaleSpec - description: - input: - output: -*****************************************************************************/ -void -FDKaacEnc_CalcSfbMaxScaleSpec(const FIXP_DBL *RESTRICT mdctSpectrum, - const INT *RESTRICT bandOffset, - INT *RESTRICT sfbMaxScaleSpec, - const INT numBands) -{ - INT i,j; - FIXP_DBL maxSpc, tmp; - - for(i=0; i 0 */ - } -} - -/***************************************************************************** - functionname: FDKaacEnc_CheckBandEnergyOptim - description: - input: - output: -*****************************************************************************/ -FIXP_DBL -FDKaacEnc_CheckBandEnergyOptim(const FIXP_DBL *RESTRICT mdctSpectrum, - INT *RESTRICT sfbMaxScaleSpec, - const INT *RESTRICT bandOffset, - const INT numBands, - FIXP_DBL *RESTRICT bandEnergy, - FIXP_DBL *RESTRICT bandEnergyLdData, - INT minSpecShift) -{ - INT i, j, scale, nr = 0; - FIXP_DBL maxNrgLd = FL2FXCONST_DBL(-1.0f); - FIXP_DBL maxNrg = 0; - FIXP_DBL spec; - - for(i=0; i maxNrgLd) { - maxNrgLd = bandEnergyLdData[i]; - nr = i; - } - } - - /* return unscaled maxNrg*/ - scale = fixMax(0,sfbMaxScaleSpec[nr]-4); - scale = fixMax(2*(minSpecShift-scale),-(DFRACT_BITS-1)); - - maxNrg = scaleValue(bandEnergy[nr], scale); - - return maxNrg; -} - -/***************************************************************************** - functionname: FDKaacEnc_CalcBandEnergyOptimLong - description: - input: - output: -*****************************************************************************/ -INT -FDKaacEnc_CalcBandEnergyOptimLong(const FIXP_DBL *RESTRICT mdctSpectrum, - INT *RESTRICT sfbMaxScaleSpec, - const INT *RESTRICT bandOffset, - const INT numBands, - FIXP_DBL *RESTRICT bandEnergy, - FIXP_DBL *RESTRICT bandEnergyLdData) -{ - INT i, j, shiftBits = 0; - FIXP_DBL maxNrgLd = FL2FXCONST_DBL(0.0f); - - FIXP_DBL spec; - - for(i=0; i 7/2 = 4 (spc*spc) */ - FIXP_DBL tmp = FL2FXCONST_DBL(0.0); - /* don't use scaleValue() here, it increases workload quite sufficiently... */ - if (leadingBits>=0) { - for (j=bandOffset[i];j>shift; - tmp = fPow2AddDiv2(tmp, spec); - } - } - bandEnergy[i] = tmp<<1; - } - - /* calculate ld of bandNrg, subtract scaling */ - LdDataVector(bandEnergy, bandEnergyLdData, numBands); - for(i=numBands; i--!=0; ) { - FIXP_DBL scaleDiff = (sfbMaxScaleSpec[i]-4)*FL2FXCONST_DBL(2.0/64); - - bandEnergyLdData[i] = (bandEnergyLdData[i] >= ((FL2FXCONST_DBL(-1.f)>>1) + (scaleDiff>>1))) - ? bandEnergyLdData[i]-scaleDiff : FL2FXCONST_DBL(-1.f); - /* find maxNrgLd */ - maxNrgLd = fixMax(maxNrgLd, bandEnergyLdData[i]); - } - - if (maxNrgLd<=(FIXP_DBL)0) - { - for(i=numBands; i--!=0; ) - { - INT scale = fixMin((sfbMaxScaleSpec[i]-4)<<1,(DFRACT_BITS-1)); - bandEnergy[i] = scaleValue(bandEnergy[i], -scale); - } - return 0; - } - else - { /* scale down NRGs */ - while (maxNrgLd>FL2FXCONST_DBL(0.0f)) - { - maxNrgLd -= FL2FXCONST_DBL(2.0/64); - shiftBits++; - } - for(i=numBands; i--!=0; ) - { - INT scale = fixMin( ((sfbMaxScaleSpec[i]-4)+shiftBits)<<1, (DFRACT_BITS-1)); - bandEnergyLdData[i] -= shiftBits*FL2FXCONST_DBL(2.0/64); - bandEnergy[i] = scaleValue(bandEnergy[i], -scale); - } - return shiftBits; - } -} - - -/***************************************************************************** - functionname: FDKaacEnc_CalcBandEnergyOptimShort - description: - input: - output: -*****************************************************************************/ -void -FDKaacEnc_CalcBandEnergyOptimShort(const FIXP_DBL *RESTRICT mdctSpectrum, - INT *RESTRICT sfbMaxScaleSpec, - const INT *RESTRICT bandOffset, - const INT numBands, - FIXP_DBL *RESTRICT bandEnergy) -{ - INT i, j; - - for(i=0; i 6/2 = 3 (spc*spc) */ - FIXP_DBL tmp = FL2FXCONST_DBL(0.0); - for (j=bandOffset[i];j 6/2 = 3 (spc*spc) */ - scale = fixMax(fixMin(scale,(DFRACT_BITS-1)),-(DFRACT_BITS-1)); - bandEnergy[i] = scaleValueSaturate(bandEnergy[i], -scale); - } -} - - -/***************************************************************************** - functionname: FDKaacEnc_CalcBandNrgMSOpt - description: - input: - output: -*****************************************************************************/ -void FDKaacEnc_CalcBandNrgMSOpt(const FIXP_DBL *RESTRICT mdctSpectrumLeft, - const FIXP_DBL *RESTRICT mdctSpectrumRight, - INT *RESTRICT sfbMaxScaleSpecLeft, - INT *RESTRICT sfbMaxScaleSpecRight, - const INT *RESTRICT bandOffset, - const INT numBands, - FIXP_DBL *RESTRICT bandEnergyMid, - FIXP_DBL *RESTRICT bandEnergySide, - INT calcLdData, - FIXP_DBL *RESTRICT bandEnergyMidLdData, - FIXP_DBL *RESTRICT bandEnergySideLdData) -{ - INT i, j, minScale; - FIXP_DBL NrgMid, NrgSide, specm, specs; - - for (i=0; i 0) { - for (j=bandOffset[i];j>1; - FIXP_DBL specR = mdctSpectrumRight[j]>>1; - specm = specL + specR; - specs = specL - specR; - NrgMid = fPow2AddDiv2(NrgMid, specm); - NrgSide = fPow2AddDiv2(NrgSide, specs); - } - } - bandEnergyMid[i] = NrgMid<<1; - bandEnergySide[i] = NrgSide<<1; - } - - if(calcLdData) { - LdDataVector(bandEnergyMid, bandEnergyMidLdData, numBands); - LdDataVector(bandEnergySide, bandEnergySideLdData, numBands); - } - - for (i=0; i>= scale; - bandEnergySide[i] >>= scale; - } -} diff --git a/libAACenc/src/band_nrg.h b/libAACenc/src/band_nrg.h deleted file mode 100644 index 540a8ef..0000000 --- a/libAACenc/src/band_nrg.h +++ /dev/null @@ -1,149 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Author(s): M. Werner - Description: Band/Line energy calculation - -******************************************************************************/ - -#ifndef _BAND_NRG_H -#define _BAND_NRG_H - -#include "common_fix.h" - - -void -FDKaacEnc_CalcSfbMaxScaleSpec( - const FIXP_DBL *mdctSpectrum, - const INT *bandOffset, - INT *sfbMaxScaleSpec, - const INT numBands - ); - -FIXP_DBL -FDKaacEnc_CheckBandEnergyOptim( - const FIXP_DBL *mdctSpectrum, - INT *sfbMaxScaleSpec, - const INT *bandOffset, - const INT numBands, - FIXP_DBL *bandEnergy, - FIXP_DBL *bandEnergyLdData, - INT minSpecShift - ); - -INT -FDKaacEnc_CalcBandEnergyOptimLong( - const FIXP_DBL *mdctSpectrum, - INT *sfbMaxScaleSpec, - const INT *bandOffset, - const INT numBands, - FIXP_DBL *bandEnergy, - FIXP_DBL *bandEnergyLdData - ); - -void -FDKaacEnc_CalcBandEnergyOptimShort( - const FIXP_DBL *mdctSpectrum, - INT *sfbMaxScaleSpec, - const INT *bandOffset, - const INT numBands, - FIXP_DBL *bandEnergy - ); - - -void FDKaacEnc_CalcBandNrgMSOpt( - const FIXP_DBL *RESTRICT mdctSpectrumLeft, - const FIXP_DBL *RESTRICT mdctSpectrumRight, - INT *RESTRICT sfbMaxScaleSpecLeft, - INT *RESTRICT sfbMaxScaleSpecRight, - const INT *RESTRICT bandOffset, - const INT numBands, - FIXP_DBL *RESTRICT bandEnergyMid, - FIXP_DBL *RESTRICT bandEnergySide, - INT calcLdData, - FIXP_DBL *RESTRICT bandEnergyMidLdData, - FIXP_DBL *RESTRICT bandEnergySideLdData); - -#endif diff --git a/libAACenc/src/bandwidth.cpp b/libAACenc/src/bandwidth.cpp deleted file mode 100644 index 6937362..0000000 --- a/libAACenc/src/bandwidth.cpp +++ /dev/null @@ -1,379 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/************************* Fast MPEG AAC Audio Encoder ********************** - - Initial author: M. Schug / A. Groeschel - contents/description: bandwidth expert - -******************************************************************************/ - -#include "channel_map.h" -#include "bandwidth.h" -#include "aacEnc_ram.h" - -typedef struct{ - INT chanBitRate; - INT bandWidthMono; - INT bandWidth2AndMoreChan; - -} BANDWIDTH_TAB; - -static const BANDWIDTH_TAB bandWidthTable[] = { - {0, 3700, 5000}, - {12000, 5000, 6400}, - {20000, 6900, 9640}, - {28000, 9600, 13050}, - {40000, 12060, 14260}, - {56000, 13950, 15500}, - {72000, 14200, 16120}, - {96000, 17000, 17000}, - {576001,17000, 17000} -}; - - -static const BANDWIDTH_TAB bandWidthTable_LD_22050[] = { - { 8000, 2000, 2400}, - {12000, 2500, 2700}, - {16000, 3300, 3100}, - {24000, 6250, 7200}, - {32000, 9200, 10500}, - {40000, 16000, 16000}, - {48000, 16000, 16000}, - {360001, 16000, 16000} -}; - -static const BANDWIDTH_TAB bandWidthTable_LD_24000[] = { - { 8000, 2000, 2000}, - {12000, 2000, 2300}, - {16000, 2200, 2500}, - {24000, 5650, 7200}, - {32000, 11600, 12000}, - {40000, 12000, 16000}, - {48000, 16000, 16000}, - {64000, 16000, 16000}, - {360001, 16000, 16000} -}; - -static const BANDWIDTH_TAB bandWidthTable_LD_32000[] = { - { 8000, 2000, 2000}, - {12000, 2000, 2000}, - {24000, 4250, 7200}, - {32000, 8400, 9000}, - {40000, 9400, 11300}, - {48000, 11900, 14700}, - {64000, 14800, 16000}, - {76000, 16000, 16000}, - {360001, 16000, 16000} -}; - -static const BANDWIDTH_TAB bandWidthTable_LD_44100[] = { - { 8000, 2000, 2000}, - {24000, 2000, 2000}, - {32000, 4400, 5700}, - {40000, 7400, 8800}, - {48000, 9000, 10700}, - {56000, 11000, 12900}, - {64000, 14400, 15500}, - {80000, 16000, 16200}, - {96000, 16500, 16000}, - {128000, 16000, 16000}, - {360001, 16000, 16000} -}; - -static const BANDWIDTH_TAB bandWidthTable_LD_48000[] = { - { 8000, 2000, 2000}, - {24000, 2000, 2000}, - {32000, 4400, 5700}, - {40000, 7400, 8800}, - {48000, 9000, 10700}, - {56000, 11000, 12800}, - {64000, 14300, 15400}, - {80000, 16000, 16200}, - {96000, 16500, 16000}, - {128000, 16000, 16000}, - {360001, 16000, 16000} -}; - -typedef struct{ - AACENC_BITRATE_MODE bitrateMode; - int bandWidthMono; - int bandWidth2AndMoreChan; -} BANDWIDTH_TAB_VBR; - -static const BANDWIDTH_TAB_VBR bandWidthTableVBR[]= { - {AACENC_BR_MODE_CBR, 0, 0}, - {AACENC_BR_MODE_VBR_1, 13050, 13050}, - {AACENC_BR_MODE_VBR_2, 13050, 13050}, - {AACENC_BR_MODE_VBR_3, 14260, 14260}, - {AACENC_BR_MODE_VBR_4, 15500, 15500}, - {AACENC_BR_MODE_VBR_5, 48000, 48000}, - {AACENC_BR_MODE_SFR, 0, 0}, - {AACENC_BR_MODE_FF, 0, 0} - -}; - -static INT GetBandwidthEntry( - const INT frameLength, - const INT sampleRate, - const INT chanBitRate, - const INT entryNo) -{ - INT bandwidth = -1; - const BANDWIDTH_TAB *pBwTab = NULL; - INT bwTabSize = 0; - - switch (frameLength) { - case 1024: - pBwTab = bandWidthTable; - bwTabSize = sizeof(bandWidthTable)/sizeof(BANDWIDTH_TAB); - break; - case 480: - case 512: - switch (sampleRate) { - case 8000: - case 11025: - case 12000: - case 16000: - case 22050: - pBwTab = bandWidthTable_LD_22050; - bwTabSize = sizeof(bandWidthTable_LD_22050)/sizeof(BANDWIDTH_TAB); - break; - case 24000: - pBwTab = bandWidthTable_LD_24000; - bwTabSize = sizeof(bandWidthTable_LD_24000)/sizeof(BANDWIDTH_TAB); - break; - case 32000: - pBwTab = bandWidthTable_LD_32000; - bwTabSize = sizeof(bandWidthTable_LD_32000)/sizeof(BANDWIDTH_TAB); - break; - case (44100): - pBwTab = bandWidthTable_LD_44100; - bwTabSize = sizeof(bandWidthTable_LD_44100)/sizeof(BANDWIDTH_TAB); - break; - case 48000: - case 64000: - case 88200: - case 96000: - pBwTab = bandWidthTable_LD_48000; - bwTabSize = sizeof(bandWidthTable_LD_48000)/sizeof(BANDWIDTH_TAB); - break; - } - break; - default: - pBwTab = NULL; - bwTabSize = 0; - } - - if (pBwTab!=NULL) { - int i; - for (i=0; i= pBwTab[i].chanBitRate && - chanBitRate < pBwTab[i+1].chanBitRate) - { - switch (frameLength) { - case 1024: - bandwidth = (entryNo==0) - ? pBwTab[i].bandWidthMono - : pBwTab[i].bandWidth2AndMoreChan; - break; - case 480: - case 512: - { - INT q_res = 0; - INT startBw = (entryNo==0) ? pBwTab[i ].bandWidthMono : pBwTab[i ].bandWidth2AndMoreChan; - INT endBw = (entryNo==0) ? pBwTab[i+1].bandWidthMono : pBwTab[i+1].bandWidth2AndMoreChan; - INT startBr = pBwTab[i].chanBitRate; - INT endBr = pBwTab[i+1].chanBitRate; - - FIXP_DBL bwFac_fix = fDivNorm(chanBitRate-startBr, endBr-startBr, &q_res); - bandwidth = (INT)scaleValue(fMult(bwFac_fix, (FIXP_DBL)(endBw-startBw)),q_res) + startBw; - } - break; - default: - bandwidth = -1; - } - break; - } /* within bitrate range */ - } - } /* pBwTab!=NULL */ - - return bandwidth; -} - - -AAC_ENCODER_ERROR FDKaacEnc_DetermineBandWidth(INT* bandWidth, - INT proposedBandWidth, - INT bitrate, - AACENC_BITRATE_MODE bitrateMode, - INT sampleRate, - INT frameLength, - CHANNEL_MAPPING* cm, - CHANNEL_MODE encoderMode) -{ - AAC_ENCODER_ERROR ErrorStatus = AAC_ENC_OK; - INT chanBitRate = bitrate/cm->nChannels; - - /* vbr */ - switch(bitrateMode){ - case AACENC_BR_MODE_VBR_1: - case AACENC_BR_MODE_VBR_2: - case AACENC_BR_MODE_VBR_3: - case AACENC_BR_MODE_VBR_4: - case AACENC_BR_MODE_VBR_5: - if (proposedBandWidth != 0){ - /* use given bw */ - *bandWidth = proposedBandWidth; - } else { - /* take bw from table */ - switch(encoderMode){ - case MODE_1: - *bandWidth = bandWidthTableVBR[bitrateMode].bandWidthMono; - break; - case MODE_2: - case MODE_1_2: - case MODE_1_2_1: - case MODE_1_2_2: - case MODE_1_2_2_1: - case MODE_1_2_2_2_1: - case MODE_7_1_REAR_SURROUND: - case MODE_7_1_FRONT_CENTER: - *bandWidth = bandWidthTableVBR[bitrateMode].bandWidth2AndMoreChan; - break; - default: - return AAC_ENC_UNSUPPORTED_CHANNELCONFIG; - } - } - break; - case AACENC_BR_MODE_CBR: - case AACENC_BR_MODE_SFR: - case AACENC_BR_MODE_FF: - - /* bandwidth limiting */ - if (proposedBandWidth != 0) { - *bandWidth = FDKmin(proposedBandWidth, FDKmin(20000, sampleRate>>1)); - } - else { /* search reasonable bandwidth */ - - int entryNo = 0; - - switch(encoderMode){ - case MODE_1: /* mono */ - entryNo = 0; /* use mono bandwith settings */ - break; - - case MODE_2: /* stereo */ - case MODE_1_2: /* sce + cpe */ - case MODE_1_2_1: /* sce + cpe + sce */ - case MODE_1_2_2: /* sce + cpe + cpe */ - case MODE_1_2_2_1: /* (5.1) sce + cpe + cpe + lfe */ - case MODE_1_2_2_2_1: /* (7.1) sce + cpe + cpe + cpe + lfe */ - case MODE_7_1_REAR_SURROUND: - case MODE_7_1_FRONT_CENTER: - entryNo = 1; /* use stereo bandwith settings */ - break; - - default: - return AAC_ENC_UNSUPPORTED_CHANNELCONFIG; - } - - *bandWidth = GetBandwidthEntry( - frameLength, - sampleRate, - chanBitRate, - entryNo); - - if (*bandWidth==-1) { - ErrorStatus = AAC_ENC_INVALID_CHANNEL_BITRATE; - } - } - break; - default: - *bandWidth = 0; - return AAC_ENC_UNSUPPORTED_BITRATE_MODE; - } - - *bandWidth = FDKmin(*bandWidth, sampleRate/2); - - return ErrorStatus;; -} diff --git a/libAACenc/src/bandwidth.h b/libAACenc/src/bandwidth.h deleted file mode 100644 index 2e92453..0000000 --- a/libAACenc/src/bandwidth.h +++ /dev/null @@ -1,106 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/************************* Fast MPEG AAC Audio Encoder ********************** - - Initial author: M. Schug / A. Groeschel - contents/description: bandwidth expert - -******************************************************************************/ - -#ifndef _BANDWIDTH_H -#define _BANDWIDTH_H - - -#include "qc_data.h" - -AAC_ENCODER_ERROR FDKaacEnc_DetermineBandWidth(INT* bandWidth, - INT proposedBandwidth, - INT bitrate, - AACENC_BITRATE_MODE bitrateMode, - INT sampleRate, - INT frameLength, - CHANNEL_MAPPING* cm, - CHANNEL_MODE encoderMode); - -#endif /* BANDWIDTH_H */ diff --git a/libAACenc/src/bit_cnt.cpp b/libAACenc/src/bit_cnt.cpp deleted file mode 100644 index 926ee49..0000000 --- a/libAACenc/src/bit_cnt.cpp +++ /dev/null @@ -1,1122 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M.Werner - contents/description: Huffman Bitcounter & coder - -******************************************************************************/ - -#include "bit_cnt.h" - -#include "aacEnc_ram.h" - -#define HI_LTAB(a) (a>>16) -#define LO_LTAB(a) (a & 0xffff) - -/***************************************************************************** - - - functionname: FDKaacEnc_count1_2_3_4_5_6_7_8_9_10_11 - description: counts tables 1-11 - returns: - input: quantized spectrum - output: bitCount for tables 1-11 - -*****************************************************************************/ - -static void FDKaacEnc_count1_2_3_4_5_6_7_8_9_10_11(const SHORT *RESTRICT values, - const INT width, - INT *bitCount) -{ - - INT i; - INT bc1_2,bc3_4,bc5_6,bc7_8,bc9_10,bc11,sc; - INT t0,t1,t2,t3; - bc1_2=0; - bc3_4=0; - bc5_6=0; - bc7_8=0; - bc9_10=0; - bc11=0; - sc=0; - - for(i=0;i0)+(t1>0)+(t2>0)+(t3>0); - } - - bitCount[1]=HI_LTAB(bc1_2); - bitCount[2]=LO_LTAB(bc1_2); - bitCount[3]=HI_LTAB(bc3_4)+sc; - bitCount[4]=LO_LTAB(bc3_4)+sc; - bitCount[5]=HI_LTAB(bc5_6); - bitCount[6]=LO_LTAB(bc5_6); - bitCount[7]=HI_LTAB(bc7_8)+sc; - bitCount[8]=LO_LTAB(bc7_8)+sc; - bitCount[9]=HI_LTAB(bc9_10)+sc; - bitCount[10]=LO_LTAB(bc9_10)+sc; - bitCount[11]=bc11+sc; - -} - - -/***************************************************************************** - - functionname: FDKaacEnc_count3_4_5_6_7_8_9_10_11 - description: counts tables 3-11 - returns: - input: quantized spectrum - output: bitCount for tables 3-11 - -*****************************************************************************/ - -static void FDKaacEnc_count3_4_5_6_7_8_9_10_11(const SHORT *RESTRICT values, - const INT width, - INT *bitCount) -{ - - INT i; - INT bc3_4,bc5_6,bc7_8,bc9_10,bc11,sc; - INT t0,t1,t2,t3; - - bc3_4=0; - bc5_6=0; - bc7_8=0; - bc9_10=0; - bc11=0; - sc=0; - - for(i=0;i0)+(t1>0)+(t2>0)+(t3>0); - } - - bitCount[1]=INVALID_BITCOUNT; - bitCount[2]=INVALID_BITCOUNT; - bitCount[3]=HI_LTAB(bc3_4)+sc; - bitCount[4]=LO_LTAB(bc3_4)+sc; - bitCount[5]=HI_LTAB(bc5_6); - bitCount[6]=LO_LTAB(bc5_6); - bitCount[7]=HI_LTAB(bc7_8)+sc; - bitCount[8]=LO_LTAB(bc7_8)+sc; - bitCount[9]=HI_LTAB(bc9_10)+sc; - bitCount[10]=LO_LTAB(bc9_10)+sc; - bitCount[11]=bc11+sc; -} - - - -/***************************************************************************** - - functionname: FDKaacEnc_count5_6_7_8_9_10_11 - description: counts tables 5-11 - returns: - input: quantized spectrum - output: bitCount for tables 5-11 - -*****************************************************************************/ - - -static void FDKaacEnc_count5_6_7_8_9_10_11(const SHORT *RESTRICT values, - const INT width, - INT *bitCount) -{ - - INT i; - INT bc5_6,bc7_8,bc9_10,bc11,sc; - INT t0,t1; - bc5_6=0; - bc7_8=0; - bc9_10=0; - bc11=0; - sc=0; - - for(i=0;i0)+(t1>0); - } - bitCount[1]=INVALID_BITCOUNT; - bitCount[2]=INVALID_BITCOUNT; - bitCount[3]=INVALID_BITCOUNT; - bitCount[4]=INVALID_BITCOUNT; - bitCount[5]=HI_LTAB(bc5_6); - bitCount[6]=LO_LTAB(bc5_6); - bitCount[7]=HI_LTAB(bc7_8)+sc; - bitCount[8]=LO_LTAB(bc7_8)+sc; - bitCount[9]=HI_LTAB(bc9_10)+sc; - bitCount[10]=LO_LTAB(bc9_10)+sc; - bitCount[11]=bc11+sc; - -} - - -/***************************************************************************** - - functionname: FDKaacEnc_count7_8_9_10_11 - description: counts tables 7-11 - returns: - input: quantized spectrum - output: bitCount for tables 7-11 - -*****************************************************************************/ - -static void FDKaacEnc_count7_8_9_10_11(const SHORT *RESTRICT values, - const INT width, - INT *bitCount) -{ - - INT i; - INT bc7_8,bc9_10,bc11,sc; - INT t0,t1; - - bc7_8=0; - bc9_10=0; - bc11=0; - sc=0; - - for(i=0;i0)+(t1>0); - } - - bitCount[1]=INVALID_BITCOUNT; - bitCount[2]=INVALID_BITCOUNT; - bitCount[3]=INVALID_BITCOUNT; - bitCount[4]=INVALID_BITCOUNT; - bitCount[5]=INVALID_BITCOUNT; - bitCount[6]=INVALID_BITCOUNT; - bitCount[7]=HI_LTAB(bc7_8)+sc; - bitCount[8]=LO_LTAB(bc7_8)+sc; - bitCount[9]=HI_LTAB(bc9_10)+sc; - bitCount[10]=LO_LTAB(bc9_10)+sc; - bitCount[11]=bc11+sc; - -} - -/***************************************************************************** - - functionname: FDKaacEnc_count9_10_11 - description: counts tables 9-11 - returns: - input: quantized spectrum - output: bitCount for tables 9-11 - -*****************************************************************************/ - - - -static void FDKaacEnc_count9_10_11(const SHORT *RESTRICT values, - const INT width, - INT *bitCount) -{ - - INT i; - INT bc9_10,bc11,sc; - INT t0,t1; - - bc9_10=0; - bc11=0; - sc=0; - - for(i=0;i0)+(t1>0); - } - - bitCount[1]=INVALID_BITCOUNT; - bitCount[2]=INVALID_BITCOUNT; - bitCount[3]=INVALID_BITCOUNT; - bitCount[4]=INVALID_BITCOUNT; - bitCount[5]=INVALID_BITCOUNT; - bitCount[6]=INVALID_BITCOUNT; - bitCount[7]=INVALID_BITCOUNT; - bitCount[8]=INVALID_BITCOUNT; - bitCount[9]=HI_LTAB(bc9_10)+sc; - bitCount[10]=LO_LTAB(bc9_10)+sc; - bitCount[11]=bc11+sc; - -} - -/***************************************************************************** - - functionname: FDKaacEnc_count11 - description: counts table 11 - returns: - input: quantized spectrum - output: bitCount for table 11 - -*****************************************************************************/ - -static void FDKaacEnc_count11(const SHORT *RESTRICT values, - const INT width, - INT *bitCount) -{ - - INT i; - INT bc11,sc; - INT t0,t1; - - bc11=0; - sc=0; - for(i=0;i0)+(t1>0); - } - - bitCount[1]=INVALID_BITCOUNT; - bitCount[2]=INVALID_BITCOUNT; - bitCount[3]=INVALID_BITCOUNT; - bitCount[4]=INVALID_BITCOUNT; - bitCount[5]=INVALID_BITCOUNT; - bitCount[6]=INVALID_BITCOUNT; - bitCount[7]=INVALID_BITCOUNT; - bitCount[8]=INVALID_BITCOUNT; - bitCount[9]=INVALID_BITCOUNT; - bitCount[10]=INVALID_BITCOUNT; - bitCount[11]=bc11+sc; -} - -/***************************************************************************** - - functionname: FDKaacEnc_countEsc - description: counts table 11 (with Esc) - returns: - input: quantized spectrum - output: bitCount for tables 11 (with Esc) - -*****************************************************************************/ - -static void FDKaacEnc_countEsc(const SHORT *RESTRICT values, - const INT width, - INT *RESTRICT bitCount) -{ - - INT i; - INT bc11,ec,sc; - INT t0,t1,t00,t01; - - bc11=0; - sc=0; - ec=0; - for(i=0;i0)+(t1>0); - - t00 = fixMin(t0,16); - t01 = fixMin(t1,16); - bc11+= (INT) FDKaacEnc_huff_ltab11[t00][t01]; - - if(t0>=16){ - ec+=5; - while((t0>>=1) >= 16) - ec+=2; - } - - if(t1>=16){ - ec+=5; - while((t1>>=1) >= 16) - ec+=2; - } - } - - for (i=0; i<11; i++) - bitCount[i]=INVALID_BITCOUNT; - - bitCount[11]=bc11+sc+ec; -} - - -typedef void (*COUNT_FUNCTION)(const SHORT *RESTRICT values, - const INT width, - INT *RESTRICT bitCount); - -static const COUNT_FUNCTION countFuncTable[CODE_BOOK_ESC_LAV+1] = -{ - - FDKaacEnc_count1_2_3_4_5_6_7_8_9_10_11, /* 0 */ - FDKaacEnc_count1_2_3_4_5_6_7_8_9_10_11, /* 1 */ - FDKaacEnc_count3_4_5_6_7_8_9_10_11, /* 2 */ - FDKaacEnc_count5_6_7_8_9_10_11, /* 3 */ - FDKaacEnc_count5_6_7_8_9_10_11, /* 4 */ - FDKaacEnc_count7_8_9_10_11, /* 5 */ - FDKaacEnc_count7_8_9_10_11, /* 6 */ - FDKaacEnc_count7_8_9_10_11, /* 7 */ - FDKaacEnc_count9_10_11, /* 8 */ - FDKaacEnc_count9_10_11, /* 9 */ - FDKaacEnc_count9_10_11, /* 10 */ - FDKaacEnc_count9_10_11, /* 11 */ - FDKaacEnc_count9_10_11, /* 12 */ - FDKaacEnc_count11, /* 13 */ - FDKaacEnc_count11, /* 14 */ - FDKaacEnc_count11, /* 15 */ - FDKaacEnc_countEsc /* 16 */ -}; - - - -INT FDKaacEnc_bitCount(const SHORT *values, - const INT width, - INT maxVal, - INT *bitCount) -{ - - /* - check if we can use codebook 0 - */ - - if(maxVal == 0) - bitCount[0] = 0; - else - bitCount[0] = INVALID_BITCOUNT; - - maxVal = fixMin(maxVal,(INT)CODE_BOOK_ESC_LAV); - countFuncTable[maxVal](values,width,bitCount); - return(0); -} - - - - -/* - count difference between actual and zeroed lines -*/ -INT FDKaacEnc_countValues(SHORT *RESTRICT values, INT width, INT codeBook) -{ - - INT i,t0,t1,t2,t3,t00,t01; - INT codeLength; - INT signLength; - INT bitCnt=0; - - switch(codeBook){ - case CODE_BOOK_ZERO_NO: - break; - - case CODE_BOOK_1_NO: - for(i=0; i=16){ - INT n,p; - n=0; - p=t0; - while((p>>=1) >=16){ - bitCnt++; - n++; - } - bitCnt+=(n+5); - } - if(t1 >=16){ - INT n,p; - n=0; - p=t1; - while((p>>=1) >=16){ - bitCnt++; - n++; - } - bitCnt+=(n+5); - } - } - break; - - default: - break; - } - - return(bitCnt); -} - - - -INT FDKaacEnc_codeValues(SHORT *RESTRICT values, INT width, INT codeBook, HANDLE_FDK_BITSTREAM hBitstream) -{ - - INT i,t0,t1,t2,t3,t00,t01; - INT codeWord,codeLength; - INT sign,signLength; - - switch(codeBook){ - case CODE_BOOK_ZERO_NO: - break; - - case CODE_BOOK_1_NO: - for(i=0; i=16){ - INT n,p; - n=0; - p=t0; - while((p>>=1) >=16){ - FDKwriteBits(hBitstream,1,1); - n++; - } - FDKwriteBits(hBitstream,0,1); - FDKwriteBits(hBitstream,t0-(1<<(n+4)),n+4); - } - if(t1 >=16){ - INT n,p; - n=0; - p=t1; - while((p>>=1) >=16){ - FDKwriteBits(hBitstream,1,1); - n++; - } - FDKwriteBits(hBitstream,0,1); - FDKwriteBits(hBitstream,t1-(1<<(n+4)),n+4); - } - } - break; - - default: - break; - } - return(0); -} - -INT FDKaacEnc_codeScalefactorDelta(INT delta, HANDLE_FDK_BITSTREAM hBitstream) -{ - INT codeWord,codeLength; - - if(fixp_abs(delta) >CODE_BOOK_SCF_LAV) - return(1); - - codeWord = FDKaacEnc_huff_ctabscf[delta+CODE_BOOK_SCF_LAV]; - codeLength = (INT)FDKaacEnc_huff_ltabscf[delta+CODE_BOOK_SCF_LAV]; - FDKwriteBits(hBitstream,codeWord,codeLength); - return(0); -} - - - diff --git a/libAACenc/src/bit_cnt.h b/libAACenc/src/bit_cnt.h deleted file mode 100644 index 7c4b59e..0000000 --- a/libAACenc/src/bit_cnt.h +++ /dev/null @@ -1,187 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M.Werner - contents/description: Huffman Bitcounter & coder - -******************************************************************************/ - -#ifndef __BITCOUNT_H -#define __BITCOUNT_H - - -#include "common_fix.h" -#include "FDK_bitstream.h" -#include "aacEnc_rom.h" - -#define INVALID_BITCOUNT (FDK_INT_MAX/4) - -/* - code book number table -*/ - -enum codeBookNo{ - CODE_BOOK_ZERO_NO= 0, - CODE_BOOK_1_NO= 1, - CODE_BOOK_2_NO= 2, - CODE_BOOK_3_NO= 3, - CODE_BOOK_4_NO= 4, - CODE_BOOK_5_NO= 5, - CODE_BOOK_6_NO= 6, - CODE_BOOK_7_NO= 7, - CODE_BOOK_8_NO= 8, - CODE_BOOK_9_NO= 9, - CODE_BOOK_10_NO= 10, - CODE_BOOK_ESC_NO= 11, - CODE_BOOK_RES_NO= 12, - CODE_BOOK_PNS_NO= 13, - CODE_BOOK_IS_OUT_OF_PHASE_NO= 14, - CODE_BOOK_IS_IN_PHASE_NO= 15 - -}; - -/* - code book index table -*/ - -enum codeBookNdx{ - CODE_BOOK_ZERO_NDX, - CODE_BOOK_1_NDX, - CODE_BOOK_2_NDX, - CODE_BOOK_3_NDX, - CODE_BOOK_4_NDX, - CODE_BOOK_5_NDX, - CODE_BOOK_6_NDX, - CODE_BOOK_7_NDX, - CODE_BOOK_8_NDX, - CODE_BOOK_9_NDX, - CODE_BOOK_10_NDX, - CODE_BOOK_ESC_NDX, - CODE_BOOK_RES_NDX, - CODE_BOOK_PNS_NDX, - CODE_BOOK_IS_OUT_OF_PHASE_NDX, - CODE_BOOK_IS_IN_PHASE_NDX, - NUMBER_OF_CODE_BOOKS -}; - -/* - code book lav table -*/ - -enum codeBookLav{ - CODE_BOOK_ZERO_LAV=0, - CODE_BOOK_1_LAV=1, - CODE_BOOK_2_LAV=1, - CODE_BOOK_3_LAV=2, - CODE_BOOK_4_LAV=2, - CODE_BOOK_5_LAV=4, - CODE_BOOK_6_LAV=4, - CODE_BOOK_7_LAV=7, - CODE_BOOK_8_LAV=7, - CODE_BOOK_9_LAV=12, - CODE_BOOK_10_LAV=12, - CODE_BOOK_ESC_LAV=16, - CODE_BOOK_SCF_LAV=60, - CODE_BOOK_PNS_LAV=60 - }; - -INT FDKaacEnc_bitCount(const SHORT *aQuantSpectrum, - const INT noOfSpecLines, - INT maxVal, - INT *bitCountLut); - -INT FDKaacEnc_countValues(SHORT *values, INT width, INT codeBook); - -INT FDKaacEnc_codeValues(SHORT *values, INT width, INT codeBook, HANDLE_FDK_BITSTREAM hBitstream); - -INT FDKaacEnc_codeScalefactorDelta(INT scalefactor, HANDLE_FDK_BITSTREAM hBitstream); - -inline INT FDKaacEnc_bitCountScalefactorDelta(const INT delta) -{ - FDK_ASSERT( (0 <= (delta+CODE_BOOK_SCF_LAV)) && ((delta+CODE_BOOK_SCF_LAV)<(int)(sizeof(FDKaacEnc_huff_ltabscf)/sizeof((FDKaacEnc_huff_ltabscf[0])))) ); - return((INT)FDKaacEnc_huff_ltabscf[delta+CODE_BOOK_SCF_LAV]); -} - -#endif diff --git a/libAACenc/src/bitenc.cpp b/libAACenc/src/bitenc.cpp deleted file mode 100644 index 8e477aa..0000000 --- a/libAACenc/src/bitenc.cpp +++ /dev/null @@ -1,1474 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - - /******************************** MPEG Audio Encoder ************************** - - Initial author: M. Werner - contents/description: Bitstream encoder - -******************************************************************************/ - -#include "bitenc.h" -#include "bit_cnt.h" -#include "dyn_bits.h" -#include "qc_data.h" -#include "interface.h" -#include "aacEnc_ram.h" - - -#include "tpenc_lib.h" - -#include "FDK_tools_rom.h" /* needed for the bitstream syntax tables */ - -static const int globalGainOffset = 100; -static const int icsReservedBit = 0; -static const int noiseOffset = 90; - -/***************************************************************************** - - functionname: FDKaacEnc_encodeSpectralData - description: encode spectral data - returns: the number of written bits - input: - output: - -*****************************************************************************/ -static INT FDKaacEnc_encodeSpectralData(INT *sfbOffset, - SECTION_DATA *sectionData, - SHORT *quantSpectrum, - HANDLE_FDK_BITSTREAM hBitStream) -{ - INT i,sfb; - INT dbgVal = FDKgetValidBits(hBitStream); - - for(i=0;inoOfSections;i++) - { - if(sectionData->huffsection[i].codeBook != CODE_BOOK_PNS_NO) - { - /* huffencode spectral data for this huffsection */ - INT tmp = sectionData->huffsection[i].sfbStart+sectionData->huffsection[i].sfbCnt; - for(sfb=sectionData->huffsection[i].sfbStart; sfbhuffsection[i].codeBook, - hBitStream); - } - } - } - return(FDKgetValidBits(hBitStream)-dbgVal); -} - -/***************************************************************************** - - functionname:FDKaacEnc_encodeGlobalGain - description: encodes Global Gain (common scale factor) - returns: the number of static bits - input: - output: - -*****************************************************************************/ -static INT FDKaacEnc_encodeGlobalGain(INT globalGain, - INT scalefac, - HANDLE_FDK_BITSTREAM hBitStream, - INT mdctScale) -{ - if (hBitStream != NULL) { - FDKwriteBits(hBitStream,globalGain - scalefac + globalGainOffset-4*(LOG_NORM_PCM-mdctScale),8); - } - return (8); -} - - -/***************************************************************************** - - functionname:FDKaacEnc_encodeIcsInfo - description: encodes Ics Info - returns: the number of static bits - input: - output: - -*****************************************************************************/ - -static INT FDKaacEnc_encodeIcsInfo(INT blockType, - INT windowShape, - INT groupingMask, - INT maxSfbPerGroup, - HANDLE_FDK_BITSTREAM hBitStream, - UINT syntaxFlags) -{ - INT statBits; - - if (blockType == SHORT_WINDOW) { - statBits = 8 + TRANS_FAC - 1; - } else { - if (syntaxFlags & AC_ELD) { - statBits = 6; - } else - { - statBits = (!(syntaxFlags & AC_SCALABLE)) ? 11 : 10; - } - } - - if (hBitStream != NULL) { - - if (!(syntaxFlags & AC_ELD)){ - FDKwriteBits(hBitStream,icsReservedBit,1); - FDKwriteBits(hBitStream,blockType,2); - FDKwriteBits(hBitStream, (windowShape == LOL_WINDOW) ? KBD_WINDOW : windowShape,1); - } - - switch(blockType){ - case LONG_WINDOW: - case START_WINDOW: - case STOP_WINDOW: - FDKwriteBits(hBitStream,maxSfbPerGroup,6); - - if (!(syntaxFlags & (AC_SCALABLE|AC_ELD)) ) { /* If not scalable syntax then ... */ - /* No predictor data present */ - FDKwriteBits(hBitStream, 0, 1); - } - break; - - case SHORT_WINDOW: - FDKwriteBits(hBitStream,maxSfbPerGroup,4); - - /* Write grouping bits */ - FDKwriteBits(hBitStream,groupingMask,TRANS_FAC-1); - break; - } - } - - return (statBits); -} - -/***************************************************************************** - - functionname: FDKaacEnc_encodeSectionData - description: encode section data (common Huffman codebooks for adjacent - SFB's) - returns: none - input: - output: - -*****************************************************************************/ -static INT FDKaacEnc_encodeSectionData(SECTION_DATA *sectionData, - HANDLE_FDK_BITSTREAM hBitStream, - UINT useVCB11) -{ - if (hBitStream != NULL) { - INT sectEscapeVal=0,sectLenBits=0; - INT sectLen; - INT i; - INT dbgVal=FDKgetValidBits(hBitStream); - INT sectCbBits = 4; - - switch(sectionData->blockType) - { - case LONG_WINDOW: - case START_WINDOW: - case STOP_WINDOW: - sectEscapeVal = SECT_ESC_VAL_LONG; - sectLenBits = SECT_BITS_LONG; - break; - - case SHORT_WINDOW: - sectEscapeVal = SECT_ESC_VAL_SHORT; - sectLenBits = SECT_BITS_SHORT; - break; - } - - for(i=0;inoOfSections;i++) - { - INT codeBook = sectionData->huffsection[i].codeBook; - - FDKwriteBits(hBitStream,codeBook,sectCbBits); - - { - sectLen = sectionData->huffsection[i].sfbCnt; - - while(sectLen >= sectEscapeVal) - { - FDKwriteBits(hBitStream,sectEscapeVal,sectLenBits); - sectLen-=sectEscapeVal; - } - FDKwriteBits(hBitStream,sectLen,sectLenBits); - } - } - return(FDKgetValidBits(hBitStream)-dbgVal); - } - return (0); -} - -/***************************************************************************** - - functionname: FDKaacEnc_encodeScaleFactorData - description: encode DPCM coded scale factors - returns: none - input: - output: - -*****************************************************************************/ -static INT FDKaacEnc_encodeScaleFactorData(UINT *maxValueInSfb, - SECTION_DATA *sectionData, - INT *scalefac, - HANDLE_FDK_BITSTREAM hBitStream, - INT *RESTRICT noiseNrg, - const INT *isScale, - INT globalGain) -{ - if (hBitStream != NULL) { - INT i,j,lastValScf,deltaScf; - INT deltaPns; - INT lastValPns = 0; - INT noisePCMFlag = TRUE; - INT lastValIs; - - INT dbgVal = FDKgetValidBits(hBitStream); - - lastValScf=scalefac[sectionData->firstScf]; - lastValPns = globalGain-scalefac[sectionData->firstScf]+globalGainOffset-4*LOG_NORM_PCM-noiseOffset; - lastValIs = 0; - - for(i=0; inoOfSections; i++){ - if (sectionData->huffsection[i].codeBook != CODE_BOOK_ZERO_NO) { - - if ((sectionData->huffsection[i].codeBook == CODE_BOOK_IS_OUT_OF_PHASE_NO) || - (sectionData->huffsection[i].codeBook == CODE_BOOK_IS_IN_PHASE_NO)) - { - INT sfbStart = sectionData->huffsection[i].sfbStart; - INT tmp = sfbStart + sectionData->huffsection[i].sfbCnt; - for(j=sfbStart; jhuffsection[i].codeBook == CODE_BOOK_PNS_NO) { - INT sfbStart = sectionData->huffsection[i].sfbStart; - INT tmp = sfbStart + sectionData->huffsection[i].sfbCnt; - for(j=sfbStart; jhuffsection[i].sfbStart+sectionData->huffsection[i].sfbCnt; - for(j=sectionData->huffsection[i].sfbStart; jhuffsection[i].codeBook != CODE_BOOK_ZERO_NO */ - } /* section loop */ - - return(FDKgetValidBits(hBitStream)-dbgVal); - } /* if (hBitStream != NULL) */ - - return (0); -} - -/***************************************************************************** - - functionname:encodeMsInfo - description: encodes MS-Stereo Info - returns: the number of static bits - input: - output: - -*****************************************************************************/ -static INT FDKaacEnc_encodeMSInfo(INT sfbCnt, - INT grpSfb, - INT maxSfb, - INT msDigest, - INT *jsFlags, - HANDLE_FDK_BITSTREAM hBitStream) -{ - INT sfb, sfbOff, msBits = 0; - - if (hBitStream != NULL) - { - switch(msDigest) - { - case MS_NONE: - FDKwriteBits(hBitStream,SI_MS_MASK_NONE,2); - msBits += 2; - break; - - case MS_ALL: - FDKwriteBits(hBitStream,SI_MS_MASK_ALL,2); - msBits += 2; - break; - - case MS_SOME: - FDKwriteBits(hBitStream,SI_MS_MASK_SOME,2); - msBits += 2; - for(sfbOff = 0; sfbOff < sfbCnt; sfbOff+=grpSfb) - { - for(sfb=0; sfbnumOfFilters[i]!=0) { - tnsPresent=1; - break; - } - } - - if (tnsPresent==0) { - FDKwriteBits(hBitStream,0,1); - } else { - FDKwriteBits(hBitStream,1,1); - } - } - return (1); -} - -/***************************************************************************** - - functionname: FDKaacEnc_encodeTnsData - description: encode TNS data (filter order, coeffs, ..) - returns: the number of static bits - input: - output: - -*****************************************************************************/ -static INT FDKaacEnc_encodeTnsData(TNS_INFO *tnsInfo, - INT blockType, - HANDLE_FDK_BITSTREAM hBitStream) -{ - INT tnsBits = 0; - - if (tnsInfo!=NULL) { - - INT i,j,k; - INT tnsPresent = 0; - INT coefBits; - INT numOfWindows=(blockType==SHORT_WINDOW?TRANS_FAC:1); - - for (i=0; inumOfFilters[i]!=0) { - tnsPresent=1; - } - } - - if (hBitStream != NULL) - { - if (tnsPresent==1) { /* there is data to be written*/ - for (i=0; inumOfFilters[i],(blockType==SHORT_WINDOW?1:2)); - tnsBits += (blockType==SHORT_WINDOW?1:2); - if (tnsInfo->numOfFilters[i]) { - FDKwriteBits(hBitStream,(tnsInfo->coefRes[i]==4?1:0),1); - tnsBits += 1; - } - for (j=0; jnumOfFilters[i]; j++) { - FDKwriteBits(hBitStream,tnsInfo->length[i][j],(blockType==SHORT_WINDOW?4:6)); - tnsBits += (blockType==SHORT_WINDOW?4:6); - FDK_ASSERT(tnsInfo->order[i][j] <= 12); - FDKwriteBits(hBitStream,tnsInfo->order[i][j],(blockType==SHORT_WINDOW?3:5)); - tnsBits += (blockType==SHORT_WINDOW?3:5); - if (tnsInfo->order[i][j]){ - FDKwriteBits(hBitStream,tnsInfo->direction[i][j],1); - tnsBits +=1; /*direction*/ - if(tnsInfo->coefRes[i] == 4) { - coefBits = 3; - for(k=0; korder[i][j]; k++) { - if (tnsInfo->coef[i][j][k]> 3 || - tnsInfo->coef[i][j][k]< -4) { - coefBits = 4; - break; - } - } - } else { - coefBits = 2; - for(k=0; korder[i][j]; k++) { - if ( tnsInfo->coef[i][j][k]> 1 - || tnsInfo->coef[i][j][k]< -2) { - coefBits = 3; - break; - } - } - } - FDKwriteBits(hBitStream,-(coefBits - tnsInfo->coefRes[i]),1); /*coef_compres*/ - tnsBits +=1; /*coef_compression */ - for (k=0; korder[i][j]; k++ ) { - static const INT rmask[] = {0,1,3,7,15}; - FDKwriteBits(hBitStream,tnsInfo->coef[i][j][k] & rmask[coefBits],coefBits); - tnsBits += coefBits; - } - } - } - } - } - } - else { - if (tnsPresent != 0) { - for (i=0; inumOfFilters[i]) { - tnsBits += 1; - for (j=0; jnumOfFilters[i]; j++) { - tnsBits += (blockType==SHORT_WINDOW?4:6); - tnsBits += (blockType==SHORT_WINDOW?3:5); - if (tnsInfo->order[i][j]) { - tnsBits +=1; /*direction*/ - tnsBits +=1; /*coef_compression */ - if (tnsInfo->coefRes[i] == 4) { - coefBits=3; - for (k=0; korder[i][j]; k++) { - if (tnsInfo->coef[i][j][k]> 3 || tnsInfo->coef[i][j][k]< -4) { - coefBits = 4; - break; - } - } - } - else { - coefBits = 2; - for (k=0; korder[i][j]; k++) { - if (tnsInfo->coef[i][j][k]> 1 || tnsInfo->coef[i][j][k]< -2) { - coefBits = 3; - break; - } - } - } - for (k=0; korder[i][j]; k++) { - tnsBits += coefBits; - } - } - } - } - } - } - } - } /* (tnsInfo!=NULL) */ - - return (tnsBits); -} - -/***************************************************************************** - - functionname: FDKaacEnc_encodeGainControlData - description: unsupported - returns: none - input: - output: - -*****************************************************************************/ -static INT FDKaacEnc_encodeGainControlData(HANDLE_FDK_BITSTREAM hBitStream) -{ - if (hBitStream != NULL) { - FDKwriteBits(hBitStream,0,1); - } - return (1); -} - -/***************************************************************************** - - functionname: FDKaacEnc_encodePulseData - description: not supported yet (dummy) - returns: none - input: - output: - -*****************************************************************************/ -static INT FDKaacEnc_encodePulseData(HANDLE_FDK_BITSTREAM hBitStream) -{ - if (hBitStream != NULL) { - FDKwriteBits(hBitStream,0,1); - } - return (1); -} - - -/***************************************************************************** - - functionname: FDKaacEnc_writeExtensionPayload - description: write extension payload to bitstream - returns: number of written bits - input: - output: - -*****************************************************************************/ -static INT FDKaacEnc_writeExtensionPayload( HANDLE_FDK_BITSTREAM hBitStream, - EXT_PAYLOAD_TYPE extPayloadType, - const UCHAR *extPayloadData, - INT extPayloadBits - ) -{ - #define EXT_TYPE_BITS ( 4 ) - #define DATA_EL_VERSION_BITS ( 4 ) - #define FILL_NIBBLE_BITS ( 4 ) - - INT extBitsUsed = 0; - - if (extPayloadBits >= EXT_TYPE_BITS) - { - UCHAR fillByte = 0x00; /* for EXT_FIL and EXT_FILL_DATA */ - - if (hBitStream != NULL) { - FDKwriteBits(hBitStream, extPayloadType, EXT_TYPE_BITS); - } - extBitsUsed += EXT_TYPE_BITS; - - switch (extPayloadType) { - case EXT_DYNAMIC_RANGE: - /* case EXT_SAC_DATA: */ - case EXT_SBR_DATA: - case EXT_SBR_DATA_CRC: - if (hBitStream != NULL) { - int i, writeBits = extPayloadBits; - for (i=0; writeBits >= 8; i++) { - FDKwriteBits(hBitStream, extPayloadData[i], 8); - writeBits -= 8; - } - if (writeBits > 0) { - FDKwriteBits(hBitStream, extPayloadData[i]>>(8-writeBits), writeBits); - } - } - extBitsUsed += extPayloadBits; - break; - - case EXT_DATA_ELEMENT: - { - INT dataElementLength = (extPayloadBits+7)>>3; - INT cnt = dataElementLength; - int loopCounter = 1; - - while (dataElementLength >= 255) { - loopCounter++; - dataElementLength -= 255; - } - - if (hBitStream != NULL) { - int i; - FDKwriteBits(hBitStream, 0x00, DATA_EL_VERSION_BITS); /* data_element_version = ANC_DATA */ - - for (i=1; i= 8) { - FDKwriteBits(hBitStream, fillByte, 8); - writeBits -= 8; - } - } - extBitsUsed += FILL_NIBBLE_BITS + (extPayloadBits & ~0x7) - 8; - break; - } - } - - return (extBitsUsed); -} - - -/***************************************************************************** - - functionname: FDKaacEnc_writeDataStreamElement - description: write data stream elements like ancillary data ... - returns: the amount of used bits - input: - output: - -******************************************************************************/ -static INT FDKaacEnc_writeDataStreamElement( HANDLE_TRANSPORTENC hTpEnc, - INT elementInstanceTag, - INT dataPayloadBytes, - UCHAR *dataBuffer, - UINT alignAnchor ) -{ - #define DATA_BYTE_ALIGN_FLAG ( 0 ) - - #define EL_INSTANCE_TAG_BITS ( 4 ) - #define DATA_BYTE_ALIGN_FLAG_BITS ( 1 ) - #define DATA_LEN_COUNT_BITS ( 8 ) - #define DATA_LEN_ESC_COUNT_BITS ( 8 ) - - #define MAX_DATA_ALIGN_BITS ( 7 ) - #define MAX_DSE_DATA_BYTES ( 510 ) - - INT dseBitsUsed = 0; - - while (dataPayloadBytes > 0) - { - int esc_count = -1; - int cnt = 0; - INT crcReg = -1; - - dseBitsUsed += EL_ID_BITS + EL_INSTANCE_TAG_BITS - + DATA_BYTE_ALIGN_FLAG_BITS + DATA_LEN_COUNT_BITS; - - if (DATA_BYTE_ALIGN_FLAG) { - dseBitsUsed += MAX_DATA_ALIGN_BITS; - } - - cnt = fixMin(MAX_DSE_DATA_BYTES, dataPayloadBytes); - if ( cnt >= 255 ) { - esc_count = cnt - 255; - dseBitsUsed += DATA_LEN_ESC_COUNT_BITS; - } - - dataPayloadBytes -= cnt; - dseBitsUsed += cnt * 8; - - if (hTpEnc != NULL) { - HANDLE_FDK_BITSTREAM hBitStream = transportEnc_GetBitstream(hTpEnc); - int i; - - FDKwriteBits(hBitStream, ID_DSE, EL_ID_BITS); - - crcReg = transportEnc_CrcStartReg(hTpEnc, 0); - - FDKwriteBits(hBitStream, elementInstanceTag, EL_INSTANCE_TAG_BITS); - FDKwriteBits(hBitStream, DATA_BYTE_ALIGN_FLAG, DATA_BYTE_ALIGN_FLAG_BITS); - - /* write length field(s) */ - if ( esc_count >= 0 ) { - FDKwriteBits(hBitStream, 255, DATA_LEN_COUNT_BITS); - FDKwriteBits(hBitStream, esc_count, DATA_LEN_ESC_COUNT_BITS); - } else { - FDKwriteBits(hBitStream, cnt, DATA_LEN_COUNT_BITS); - } - - if (DATA_BYTE_ALIGN_FLAG) { - INT tmp = (INT)FDKgetValidBits(hBitStream); - FDKbyteAlign(hBitStream, alignAnchor); - /* count actual bits */ - dseBitsUsed += (INT)FDKgetValidBits(hBitStream) - tmp - MAX_DATA_ALIGN_BITS; - } - - /* write payload */ - for (i=0; inPayloadBits; - INT extBitsUsed = 0; - - if (hTpEnc != NULL) { - hBitStream = transportEnc_GetBitstream(hTpEnc); - } - - if (syntaxFlags & (AC_SCALABLE|AC_ER)) - { - if ( syntaxFlags & AC_DRM ) - { /* CAUTION: The caller has to assure that fill - data is written before the SBR payload. */ - UCHAR *extPayloadData = pExtension->pPayload; - - switch (pExtension->type) - { - case EXT_SBR_DATA: - case EXT_SBR_DATA_CRC: - /* SBR payload is written in reverse */ - if (hBitStream != NULL) { - int i, writeBits = payloadBits; - - FDKpushFor(hBitStream, payloadBits-1); /* Does a cache sync internally */ - - for (i=0; writeBits >= 8; i++) { - FDKwriteBitsBwd(hBitStream, extPayloadData[i], 8); - writeBits -= 8; - } - if (writeBits > 0) { - FDKwriteBitsBwd(hBitStream, extPayloadData[i]>>(8-writeBits), writeBits); - } - - FDKsyncCacheBwd (hBitStream); - FDKpushFor (hBitStream, payloadBits+1); - } - extBitsUsed += payloadBits; - break; - - case EXT_FILL_DATA: - case EXT_FIL: - default: - if (hBitStream != NULL) { - int writeBits = payloadBits; - while (writeBits >= 8) { - FDKwriteBits(hBitStream, 0x00, 8); - writeBits -= 8; - } - FDKwriteBits(hBitStream, 0x00, writeBits); - } - extBitsUsed += payloadBits; - break; - } - } - else { - if ( (syntaxFlags & AC_ELD) && ((pExtension->type==EXT_SBR_DATA) || (pExtension->type==EXT_SBR_DATA_CRC)) ) { - - if (hBitStream != NULL) { - int i, writeBits = payloadBits; - UCHAR *extPayloadData = pExtension->pPayload; - - for (i=0; writeBits >= 8; i++) { - FDKwriteBits(hBitStream, extPayloadData[i], 8); - writeBits -= 8; - } - if (writeBits > 0) { - FDKwriteBits(hBitStream, extPayloadData[i]>>(8-writeBits), writeBits); - } - } - extBitsUsed += payloadBits; - } - else - { - /* ER or scalable syntax -> write extension en bloc */ - extBitsUsed += FDKaacEnc_writeExtensionPayload( hBitStream, - pExtension->type, - pExtension->pPayload, - payloadBits ); - } - } - } - else { - /* We have normal GA bitstream payload (AOT 2,5,29) so pack - the data into a fill elements or DSEs */ - - if ( pExtension->type == EXT_DATA_ELEMENT ) - { - extBitsUsed += FDKaacEnc_writeDataStreamElement( hTpEnc, - elInstanceTag, - pExtension->nPayloadBits>>3, - pExtension->pPayload, - alignAnchor ); - } - else { - while (payloadBits >= (EL_ID_BITS + FILL_EL_COUNT_BITS)) { - INT cnt, esc_count=-1, alignBits=7; - - if ( (pExtension->type == EXT_FILL_DATA) || (pExtension->type == EXT_FIL) ) - { - payloadBits -= EL_ID_BITS + FILL_EL_COUNT_BITS; - if (payloadBits >= 15*8) { - payloadBits -= FILL_EL_ESC_COUNT_BITS; - esc_count = 0; /* write esc_count even if cnt becomes smaller 15 */ - } - alignBits = 0; - } - - cnt = fixMin(MAX_FILL_DATA_BYTES, (payloadBits+alignBits)>>3); - - if (cnt >= 15) { - esc_count = cnt - 15 + 1; - } - - if (hBitStream != NULL) { - /* write bitstream */ - FDKwriteBits(hBitStream, ID_FIL, EL_ID_BITS); - if (esc_count >= 0) { - FDKwriteBits(hBitStream, 15, FILL_EL_COUNT_BITS); - FDKwriteBits(hBitStream, esc_count, FILL_EL_ESC_COUNT_BITS); - } else { - FDKwriteBits(hBitStream, cnt, FILL_EL_COUNT_BITS); - } - } - - extBitsUsed += EL_ID_BITS + FILL_EL_COUNT_BITS + ((esc_count>=0) ? FILL_EL_ESC_COUNT_BITS : 0); - - cnt = fixMin(cnt*8, payloadBits); /* convert back to bits */ - extBitsUsed += FDKaacEnc_writeExtensionPayload( hBitStream, - pExtension->type, - pExtension->pPayload, - cnt ); - payloadBits -= cnt; - } - } - } - - return (extBitsUsed); -} - - -/***************************************************************************** - - functionname: FDKaacEnc_ByteAlignment - description: - returns: - input: - output: - -*****************************************************************************/ -static void FDKaacEnc_ByteAlignment(HANDLE_FDK_BITSTREAM hBitStream, int alignBits) -{ - FDKwriteBits(hBitStream, 0, alignBits); -} - -AAC_ENCODER_ERROR FDKaacEnc_ChannelElementWrite( HANDLE_TRANSPORTENC hTpEnc, - ELEMENT_INFO *pElInfo, - QC_OUT_CHANNEL *qcOutChannel[(2)], - PSY_OUT_ELEMENT *psyOutElement, - PSY_OUT_CHANNEL *psyOutChannel[(2)], - UINT syntaxFlags, - AUDIO_OBJECT_TYPE aot, - SCHAR epConfig, - INT *pBitDemand, - UCHAR minCnt - ) -{ - AAC_ENCODER_ERROR error = AAC_ENC_OK; - HANDLE_FDK_BITSTREAM hBitStream = NULL; - INT bitDemand = 0; - const element_list_t *list; - int i, ch, decision_bit; - INT crcReg1 = -1, crcReg2 = -1; - UCHAR numberOfChannels; - - if (hTpEnc != NULL) { - /* Get bitstream handle */ - hBitStream = transportEnc_GetBitstream(hTpEnc); - } - - if ( (pElInfo->elType==ID_SCE) || (pElInfo->elType==ID_LFE) ) { - numberOfChannels = 1; - } else { - numberOfChannels = 2; - } - - /* Get channel element sequence table */ - list = getBitstreamElementList(aot, epConfig, numberOfChannels, 0); - if (list == NULL) { - error = AAC_ENC_UNSUPPORTED_AOT; - goto bail; - } - - if (!(syntaxFlags & (AC_SCALABLE|AC_ER))) { - if (hBitStream != NULL) { - FDKwriteBits(hBitStream, pElInfo->elType, EL_ID_BITS); - } - bitDemand += EL_ID_BITS; - } - - /* Iterate through sequence table */ - i = 0; - ch = 0; - decision_bit = 0; - do { - /* some tmp values */ - SECTION_DATA *pChSectionData = NULL; - INT *pChScf = NULL; - UINT *pChMaxValueInSfb = NULL; - TNS_INFO *pTnsInfo = NULL; - INT chGlobalGain = 0; - INT chBlockType = 0; - INT chMaxSfbPerGrp = 0; - INT chSfbPerGrp = 0; - INT chSfbCnt = 0; - INT chFirstScf = 0; - - if (minCnt==0) { - if ( qcOutChannel!=NULL ) { - pChSectionData = &(qcOutChannel[ch]->sectionData); - pChScf = qcOutChannel[ch]->scf; - chGlobalGain = qcOutChannel[ch]->globalGain; - pChMaxValueInSfb = qcOutChannel[ch]->maxValueInSfb; - chBlockType = pChSectionData->blockType; - chMaxSfbPerGrp = pChSectionData->maxSfbPerGroup; - chSfbPerGrp = pChSectionData->sfbPerGroup; - chSfbCnt = pChSectionData->sfbCnt; - chFirstScf = pChScf[pChSectionData->firstScf]; - } - else { - /* get values from PSY */ - chSfbCnt = psyOutChannel[ch]->sfbCnt; - chSfbPerGrp = psyOutChannel[ch]->sfbPerGroup; - chMaxSfbPerGrp = psyOutChannel[ch]->maxSfbPerGroup; - } - pTnsInfo = &psyOutChannel[ch]->tnsInfo; - } /* minCnt==0 */ - - if ( qcOutChannel==NULL ) { - chBlockType = psyOutChannel[ch]->lastWindowSequence; - } - - switch (list->id[i]) - { - case element_instance_tag: - /* Write element instance tag */ - if (hBitStream != NULL) { - FDKwriteBits(hBitStream, pElInfo->instanceTag, 4); - } - bitDemand += 4; - break; - - case common_window: - /* Write common window flag */ - decision_bit = psyOutElement->commonWindow; - if (hBitStream != NULL) { - FDKwriteBits(hBitStream, psyOutElement->commonWindow, 1); - } - bitDemand += 1; - break; - - case ics_info: - /* Write individual channel info */ - bitDemand += FDKaacEnc_encodeIcsInfo( chBlockType, - psyOutChannel[ch]->windowShape, - psyOutChannel[ch]->groupingMask, - chMaxSfbPerGrp, - hBitStream, - syntaxFlags); - break; - - case ltp_data_present: - /* Write LTP data present flag */ - if (hBitStream != NULL) { - FDKwriteBits(hBitStream, 0, 1); - } - bitDemand += 1; - break; - - case ltp_data: - /* Predictor data not supported. - Nothing to do here. */ - break; - - case ms: - /* Write MS info */ - bitDemand += FDKaacEnc_encodeMSInfo( chSfbCnt, - chSfbPerGrp, - chMaxSfbPerGrp, - (minCnt==0) ? psyOutElement->toolsInfo.msDigest : MS_NONE, - psyOutElement->toolsInfo.msMask, - hBitStream); - break; - - case global_gain: - bitDemand += FDKaacEnc_encodeGlobalGain( chGlobalGain, - chFirstScf, - hBitStream, - psyOutChannel[ch]->mdctScale ); - break; - - case section_data: - { - INT siBits = FDKaacEnc_encodeSectionData(pChSectionData, hBitStream, (syntaxFlags & AC_ER_VCB11)?1:0); - if (hBitStream != NULL) { - if (siBits != qcOutChannel[ch]->sectionData.sideInfoBits) { - error = AAC_ENC_WRITE_SEC_ERROR; - } - } - bitDemand += siBits; - } - break; - - case scale_factor_data: - { - INT sfDataBits = FDKaacEnc_encodeScaleFactorData( pChMaxValueInSfb, - pChSectionData, - pChScf, - hBitStream, - psyOutChannel[ch]->noiseNrg, - psyOutChannel[ch]->isScale, - chGlobalGain ); - if ( (hBitStream != NULL) - && (sfDataBits != (qcOutChannel[ch]->sectionData.scalefacBits + qcOutChannel[ch]->sectionData.noiseNrgBits)) ) { - error = AAC_ENC_WRITE_SCAL_ERROR; - } - bitDemand += sfDataBits; - } - break; - - case esc2_rvlc: - if (syntaxFlags & AC_ER_RVLC) { - /* write RVLC data into bitstream (error sens. cat. 2) */ - error = AAC_ENC_UNSUPPORTED_AOT; - } - break; - - case pulse: - /* Write pulse data */ - bitDemand += FDKaacEnc_encodePulseData(hBitStream); - break; - - case tns_data_present: - /* Write TNS data present flag */ - bitDemand += FDKaacEnc_encodeTnsDataPresent(pTnsInfo, - chBlockType, - hBitStream); - break; - case tns_data: - /* Write TNS data */ - bitDemand += FDKaacEnc_encodeTnsData(pTnsInfo, - chBlockType, - hBitStream); - break; - - case gain_control_data: - /* Nothing to do here */ - break; - - case gain_control_data_present: - bitDemand += FDKaacEnc_encodeGainControlData(hBitStream); - break; - - - case esc1_hcr: - if (syntaxFlags & AC_ER_HCR) - { - error = AAC_ENC_UNKNOWN; - } - break; - - case spectral_data: - if (hBitStream != NULL) - { - INT spectralBits = 0; - - spectralBits = FDKaacEnc_encodeSpectralData( psyOutChannel[ch]->sfbOffsets, - pChSectionData, - qcOutChannel[ch]->quantSpec, - hBitStream ); - - if (spectralBits != qcOutChannel[ch]->sectionData.huffmanBits) { - return AAC_ENC_WRITE_SPEC_ERROR; - } - bitDemand += spectralBits; - } - break; - - /* Non data cases */ - case adtscrc_start_reg1: - if (hTpEnc != NULL) { - crcReg1 = transportEnc_CrcStartReg(hTpEnc, 192); - } - break; - case adtscrc_start_reg2: - if (hTpEnc != NULL) { - crcReg2 = transportEnc_CrcStartReg(hTpEnc, 128); - } - break; - case adtscrc_end_reg1: - case drmcrc_end_reg: - if (hTpEnc != NULL) { - transportEnc_CrcEndReg(hTpEnc, crcReg1); - } - break; - case adtscrc_end_reg2: - if (hTpEnc != NULL) { - transportEnc_CrcEndReg(hTpEnc, crcReg2); - } - break; - case drmcrc_start_reg: - if (hTpEnc != NULL) { - crcReg1 = transportEnc_CrcStartReg(hTpEnc, 0); - } - break; - case next_channel: - ch = (ch + 1) % numberOfChannels; - break; - case link_sequence: - list = list->next[decision_bit]; - i=-1; - break; - - default: - error = AAC_ENC_UNKNOWN; - break; - } - - if (error != AAC_ENC_OK) { - return error; - } - - i++; - - } while (list->id[i] != end_of_sequence); - -bail: - if (pBitDemand != NULL) { - *pBitDemand = bitDemand; - } - - return error; -} - - -//----------------------------------------------------------------------------------------------- - -AAC_ENCODER_ERROR FDKaacEnc_WriteBitstream(HANDLE_TRANSPORTENC hTpEnc, - CHANNEL_MAPPING *channelMapping, - QC_OUT *qcOut, - PSY_OUT* psyOut, - QC_STATE *qcKernel, - AUDIO_OBJECT_TYPE aot, - UINT syntaxFlags, - SCHAR epConfig - ) -{ - HANDLE_FDK_BITSTREAM hBs = transportEnc_GetBitstream(hTpEnc); - AAC_ENCODER_ERROR ErrorStatus = AAC_ENC_OK; - int i, n, doByteAlign = 1; - INT bitMarkUp; - INT frameBits; - /* Get first bit of raw data block. - In case of ADTS+PCE, AU would start at PCE. - This is okay because PCE assures alignment. */ - UINT alignAnchor = FDKgetValidBits(hBs); - - frameBits = bitMarkUp = alignAnchor; - - - /* Channel element loop */ - for (i=0; inElements; i++) { - - ELEMENT_INFO elInfo = channelMapping->elInfo[i]; - INT elementUsedBits = 0; - - switch (elInfo.elType) - { - case ID_SCE: /* single channel */ - case ID_CPE: /* channel pair */ - case ID_LFE: /* low freq effects channel */ - { - if ( AAC_ENC_OK != (ErrorStatus = FDKaacEnc_ChannelElementWrite( hTpEnc, - &elInfo, - qcOut->qcElement[i]->qcOutChannel, - psyOut->psyOutElement[i], - psyOut->psyOutElement[i]->psyOutChannel, - syntaxFlags, /* syntaxFlags (ER tools ...) */ - aot, /* aot: AOT_AAC_LC, AOT_SBR, AOT_PS */ - epConfig, /* epConfig -1, 0, 1 */ - NULL, - 0 )) ) - { - return ErrorStatus; - } - - if ( !(syntaxFlags & AC_ER) ) - { - /* Write associated extension payload */ - for (n = 0; n < qcOut->qcElement[i]->nExtensions; n++) { - FDKaacEnc_writeExtensionData( hTpEnc, - &qcOut->qcElement[i]->extension[n], - 0, - alignAnchor, - syntaxFlags, - aot, - epConfig ); - } - } - } - break; - - /* In FDK, DSE signalling explicit done in elDSE. See channel_map.cpp */ - default: - return AAC_ENC_INVALID_ELEMENTINFO_TYPE; - - } /* switch */ - - if(elInfo.elType != ID_DSE) { - elementUsedBits -= bitMarkUp; - bitMarkUp = FDKgetValidBits(hBs); - elementUsedBits += bitMarkUp; - frameBits += elementUsedBits; - } - - } /* for (i=0; inElements; i++) { - for (n = 0; n < qcOut->qcElement[i]->nExtensions; n++) { - - if ( (qcOut->qcElement[i]->extension[n].type==EXT_SBR_DATA) - || (qcOut->qcElement[i]->extension[n].type==EXT_SBR_DATA_CRC) ) - { - /* Write sbr extension payload */ - FDKaacEnc_writeExtensionData( hTpEnc, - &qcOut->qcElement[i]->extension[n], - 0, - alignAnchor, - syntaxFlags, - aot, - epConfig ); - - channelElementExtensionWritten[i][n] = 1; - } /* SBR */ - } /* n */ - } /* i */ - } /* AC_ELD */ - - for (i=0; inElements; i++) { - for (n = 0; n < qcOut->qcElement[i]->nExtensions; n++) { - - if (channelElementExtensionWritten[i][n]==0) - { - /* Write all ramaining extension payloads in element */ - FDKaacEnc_writeExtensionData( hTpEnc, - &qcOut->qcElement[i]->extension[n], - 0, - alignAnchor, - syntaxFlags, - aot, - epConfig ); - } - } /* n */ - } /* i */ - } /* if AC_ER */ - - /* Extend global extension payload table with fill bits */ - if ( syntaxFlags & AC_DRM ) - { - /* Exception for Drm */ - for (n = 0; n < qcOut->nExtensions; n++) { - if ( (qcOut->extension[n].type == EXT_SBR_DATA) - || (qcOut->extension[n].type == EXT_SBR_DATA_CRC) ) { - /* SBR data must be the last extension! */ - FDKmemcpy(&qcOut->extension[qcOut->nExtensions], &qcOut->extension[n], sizeof(QC_OUT_EXTENSION)); - break; - } - } - /* Do byte alignment after AAC (+ MPS) payload. - Assure that MPS has been written as channel assigned extension payload! */ - if (((FDKgetValidBits(hBs)-alignAnchor+(UINT)qcOut->totFillBits)&0x7)!=(UINT)qcOut->alignBits) { - return AAC_ENC_WRITTEN_BITS_ERROR; - } - FDKaacEnc_ByteAlignment(hBs, qcOut->alignBits); - doByteAlign = 0; - - } /* AC_DRM */ - - /* Add fill data / stuffing bits */ - n = qcOut->nExtensions; - qcOut->extension[n].type = EXT_FILL_DATA; - qcOut->extension[n].nPayloadBits = qcOut->totFillBits; - qcOut->nExtensions++; - - /* Write global extension payload and fill data */ - for (n = 0; (n < qcOut->nExtensions) && (n < (2+2)); n++) - { - FDKaacEnc_writeExtensionData( hTpEnc, - &qcOut->extension[n], - 0, - alignAnchor, - syntaxFlags, - aot, - epConfig ); - - /* For EXT_FIL or EXT_FILL_DATA we could do an additional sanity check here */ - } - - if (!(syntaxFlags & (AC_SCALABLE|AC_ER))) { - FDKwriteBits(hBs, ID_END, EL_ID_BITS); - } - - if (doByteAlign) { - /* Assure byte alignment*/ - if (((alignAnchor-FDKgetValidBits(hBs))&0x7)!=(UINT)qcOut->alignBits) { - return AAC_ENC_WRITTEN_BITS_ERROR; - } - - FDKaacEnc_ByteAlignment(hBs, qcOut->alignBits); - } - - frameBits -= bitMarkUp; - frameBits += FDKgetValidBits(hBs); - - transportEnc_EndAccessUnit(hTpEnc, &frameBits); - - if (frameBits != qcOut->totalBits + qcKernel->globHdrBits){ - return AAC_ENC_WRITTEN_BITS_ERROR; - } - - return ErrorStatus; -} - diff --git a/libAACenc/src/bitenc.h b/libAACenc/src/bitenc.h deleted file mode 100644 index 498be7c..0000000 --- a/libAACenc/src/bitenc.h +++ /dev/null @@ -1,183 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M. Werner - contents/description: Bitstream encoder - -******************************************************************************/ - -#ifndef _BITENC_H -#define _BITENC_H - - -#include "qc_data.h" -#include "aacenc_tns.h" -#include "channel_map.h" -#include "interface.h" /* obsolete, when PSY_OUT is thrown out of the WritBS-call! */ -#include "FDK_audio.h" -#include "aacenc.h" - -#include "tpenc_lib.h" - -typedef enum{ - MAX_ENCODER_CHANNELS = 9, - MAX_BLOCK_TYPES = 4, - MAX_AAC_LAYERS = 9, - MAX_LAYERS = MAX_AAC_LAYERS , /* only one core layer if present */ - FIRST_LAY = 1 /* default layer number for AAC nonscalable */ -} _MAX_CONST; - -#define BUFFER_MX_HUFFCB_SIZE (32*sizeof(INT)) /* our FDK_bitbuffer needs size of power 2 */ - -#define EL_ID_BITS ( 3 ) - - -/** - * \brief Arbitrary order bitstream writer. This function can either assemble a bit stream - * and write into the bit buffer of hTpEnc or calculate the number of static bits (signal independent) - * TpEnc handle must be NULL in this case. Or also Calculate the minimum possible number of - * static bits which by disabling all tools e.g. MS, TNS and sbfCnt=0. The minCnt parameter - * has to be 1 in this latter case. - * \param hTpEnc Transport encoder handle. If NULL, the number of static bits will be returned into - * *pBitDemand. - * \param pElInfo - * \param qcOutChannel - * \param hReorderInfo - * \param psyOutElement - * \param psyOutChannel - * \param syntaxFlags Bit stream syntax flags as defined in FDK_audio.h (Audio Codec flags). - * \param aot - * \param epConfig - * \param pBitDemand Pointer to an int where the amount of bits is returned into. The returned value - * depends on if hTpEnc is NULL and minCnt. - * \param minCnt If non-zero the value returned into *pBitDemand is the absolute minimum required amount of - * static bits in order to write a valid bit stream. - * \return AAC_ENCODER_ERROR error code - */ -AAC_ENCODER_ERROR FDKaacEnc_ChannelElementWrite( HANDLE_TRANSPORTENC hTpEnc, - ELEMENT_INFO *pElInfo, - QC_OUT_CHANNEL *qcOutChannel[(2)], - PSY_OUT_ELEMENT *psyOutElement, - PSY_OUT_CHANNEL *psyOutChannel[(2)], - UINT syntaxFlags, - AUDIO_OBJECT_TYPE aot, - SCHAR epConfig, - INT *pBitDemand, - UCHAR minCnt - ); -/** - * \brief Write bit stream or account static bits - * \param hTpEnc transport encoder handle. If NULL, the function will - * not write any bit stream data but only count the amount - * of static (signal independent) bits - * \param channelMapping Channel mapping info - * \param qcOut - * \param psyOut - * \param qcKernel - * \param hBSE - * \param aot Audio Object Type being encoded - * \param syntaxFlags Flags indicating format specific detail - * \param epConfig Error protection config - */ -AAC_ENCODER_ERROR FDKaacEnc_WriteBitstream (HANDLE_TRANSPORTENC hTpEnc, - CHANNEL_MAPPING *channelMapping, - QC_OUT* qcOut, - PSY_OUT* psyOut, - QC_STATE* qcKernel, - AUDIO_OBJECT_TYPE aot, - UINT syntaxFlags, - SCHAR epConfig - ); - -INT FDKaacEnc_writeExtensionData( HANDLE_TRANSPORTENC hTpEnc, - QC_OUT_EXTENSION *pExtension, - INT elInstanceTag, - UINT alignAnchor, - UINT syntaxFlags, - AUDIO_OBJECT_TYPE aot, - SCHAR epConfig - ); - -#endif /* _BITENC_H */ diff --git a/libAACenc/src/block_switch.cpp b/libAACenc/src/block_switch.cpp deleted file mode 100644 index 7b3e275..0000000 --- a/libAACenc/src/block_switch.cpp +++ /dev/null @@ -1,545 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/***************************** MPEG-4 AAC Encoder ************************** - - Author(s): M. Werner, Tobias Chalupka - Description: Block switching - -******************************************************************************/ - -/****************** Includes *****************************/ - -#include "block_switch.h" -#include "genericStds.h" - - -#define LOWOV_WINDOW _LOWOV_WINDOW - -/**************** internal function prototypes ***********/ - -static FIXP_DBL FDKaacEnc_GetWindowEnergy(const FIXP_DBL in[], const INT blSwWndIdx); - -static void FDKaacEnc_CalcWindowEnergy( - BLOCK_SWITCHING_CONTROL *RESTRICT blockSwitchingControl, - INT windowLen, - const INT_PCM *pTimeSignal - ); - -/****************** Constants *****************************/ -/* LONG START SHORT STOP LOWOV */ -static const INT blockType2windowShape[2][5] = { {SINE_WINDOW, KBD_WINDOW, WRONG_WINDOW, SINE_WINDOW, KBD_WINDOW}, /* LD */ - {KBD_WINDOW, SINE_WINDOW, SINE_WINDOW, KBD_WINDOW, WRONG_WINDOW} }; /* LC */ - -/* IIR high pass coeffs */ - -#ifndef SINETABLE_16BIT - -static const FIXP_DBL hiPassCoeff[BLOCK_SWITCHING_IIR_LEN]= -{ - FL2FXCONST_DBL(-0.5095),FL2FXCONST_DBL(0.7548) -}; - -static const FIXP_DBL accWindowNrgFac = FL2FXCONST_DBL(0.3f); /* factor for accumulating filtered window energies */ -static const FIXP_DBL oneMinusAccWindowNrgFac = FL2FXCONST_DBL(0.7f); -/* static const float attackRatio = 10.0; */ /* lower ratio limit for attacks */ -static const FIXP_DBL invAttackRatio = FL2FXCONST_DBL(0.1f); /* inverted lower ratio limit for attacks */ - -/* The next constants are scaled, because they are used for comparison with scaled values*/ -/* minimum energy for attacks */ -static const FIXP_DBL minAttackNrg = (FL2FXCONST_DBL(1e+6f*NORM_PCM_ENERGY)>>BLOCK_SWITCH_ENERGY_SHIFT); /* minimum energy for attacks */ - -#else - -static const FIXP_SGL hiPassCoeff[BLOCK_SWITCHING_IIR_LEN]= -{ - FL2FXCONST_SGL(-0.5095),FL2FXCONST_SGL(0.7548) -}; - -static const FIXP_DBL accWindowNrgFac = FL2FXCONST_DBL(0.3f); /* factor for accumulating filtered window energies */ -static const FIXP_SGL oneMinusAccWindowNrgFac = FL2FXCONST_SGL(0.7f); -/* static const float attackRatio = 10.0; */ /* lower ratio limit for attacks */ -static const FIXP_SGL invAttackRatio = FL2FXCONST_SGL(0.1f); /* inverted lower ratio limit for attacks */ -/* minimum energy for attacks */ -static const FIXP_DBL minAttackNrg = (FL2FXCONST_DBL(1e+6f*NORM_PCM_ENERGY)>>BLOCK_SWITCH_ENERGY_SHIFT); /* minimum energy for attacks */ - -#endif - -/**************** internal function prototypes ***********/ - -/****************** Routines ****************************/ -void FDKaacEnc_InitBlockSwitching(BLOCK_SWITCHING_CONTROL *blockSwitchingControl, INT isLowDelay) -{ - FDKmemclear (blockSwitchingControl, sizeof(BLOCK_SWITCHING_CONTROL)); - - if (isLowDelay) - { - blockSwitchingControl->nBlockSwitchWindows = 4; - blockSwitchingControl->allowShortFrames = 0; - blockSwitchingControl->allowLookAhead = 0; - } - else - { - blockSwitchingControl->nBlockSwitchWindows = 8; - blockSwitchingControl->allowShortFrames = 1; - blockSwitchingControl->allowLookAhead = 1; - } - - blockSwitchingControl->noOfGroups = MAX_NO_OF_GROUPS; - - /* Initialize startvalue for blocktype */ - blockSwitchingControl->lastWindowSequence = LONG_WINDOW; - blockSwitchingControl->windowShape = blockType2windowShape[blockSwitchingControl->allowShortFrames][blockSwitchingControl->lastWindowSequence]; - -} - -static const INT suggestedGroupingTable[TRANS_FAC][MAX_NO_OF_GROUPS] = -{ - /* Attack in Window 0 */ {1, 3, 3, 1}, - /* Attack in Window 1 */ {1, 1, 3, 3}, - /* Attack in Window 2 */ {2, 1, 3, 2}, - /* Attack in Window 3 */ {3, 1, 3, 1}, - /* Attack in Window 4 */ {3, 1, 1, 3}, - /* Attack in Window 5 */ {3, 2, 1, 2}, - /* Attack in Window 6 */ {3, 3, 1, 1}, - /* Attack in Window 7 */ {3, 3, 1, 1} -}; - -/* change block type depending on current blocktype and whether there's an attack */ -/* assume no look-ahead */ -static const INT chgWndSq[2][N_BLOCKTYPES] = -{ - /* LONG WINDOW START_WINDOW SHORT_WINDOW STOP_WINDOW, LOWOV_WINDOW, WRONG_WINDOW */ - /*no attack*/ {LONG_WINDOW, STOP_WINDOW, WRONG_WINDOW, LONG_WINDOW, STOP_WINDOW , WRONG_WINDOW }, - /*attack */ {START_WINDOW, LOWOV_WINDOW, WRONG_WINDOW, START_WINDOW, LOWOV_WINDOW, WRONG_WINDOW } -}; - -/* change block type depending on current blocktype and whether there's an attack */ -/* assume look-ahead */ -static const INT chgWndSqLkAhd[2][2][N_BLOCKTYPES] = -{ - /*attack LONG WINDOW START_WINDOW SHORT_WINDOW STOP_WINDOW LOWOV_WINDOW, WRONG_WINDOW */ /* last attack */ - /*no attack*/ { {LONG_WINDOW, SHORT_WINDOW, STOP_WINDOW, LONG_WINDOW, WRONG_WINDOW, WRONG_WINDOW}, /* no attack */ - /*attack */ {START_WINDOW, SHORT_WINDOW, SHORT_WINDOW, START_WINDOW, WRONG_WINDOW, WRONG_WINDOW} }, /* no attack */ - /*no attack*/ { {LONG_WINDOW, SHORT_WINDOW, SHORT_WINDOW, LONG_WINDOW, WRONG_WINDOW, WRONG_WINDOW}, /* attack */ - /*attack */ {START_WINDOW, SHORT_WINDOW, SHORT_WINDOW, START_WINDOW, WRONG_WINDOW, WRONG_WINDOW} } /* attack */ -}; - -int FDKaacEnc_BlockSwitching(BLOCK_SWITCHING_CONTROL *blockSwitchingControl, const INT granuleLength, const int isLFE, const INT_PCM *pTimeSignal) -{ - UINT i; - FIXP_DBL enM1, enMax; - - UINT nBlockSwitchWindows = blockSwitchingControl->nBlockSwitchWindows; - - /* for LFE : only LONG window allowed */ - if (isLFE) { - - /* case LFE: */ - /* only long blocks, always use sine windows (MPEG2 AAC, MPEG4 AAC) */ - blockSwitchingControl->lastWindowSequence = LONG_WINDOW; - blockSwitchingControl->windowShape = SINE_WINDOW; - blockSwitchingControl->noOfGroups = 1; - blockSwitchingControl->groupLen[0] = 1; - - return(0); - }; - - /* Save current attack index as last attack index */ - blockSwitchingControl->lastattack = blockSwitchingControl->attack; - blockSwitchingControl->lastAttackIndex = blockSwitchingControl->attackIndex; - - /* Save current window energy as last window energy */ - FDKmemcpy(blockSwitchingControl->windowNrg[0], blockSwitchingControl->windowNrg[1], sizeof(blockSwitchingControl->windowNrg[0])); - FDKmemcpy(blockSwitchingControl->windowNrgF[0], blockSwitchingControl->windowNrgF[1], sizeof(blockSwitchingControl->windowNrgF[0])); - - if (blockSwitchingControl->allowShortFrames) - { - /* Calculate suggested grouping info for the last frame */ - - /* Reset grouping info */ - FDKmemclear (blockSwitchingControl->groupLen, sizeof(blockSwitchingControl->groupLen)); - - /* Set grouping info */ - blockSwitchingControl->noOfGroups = MAX_NO_OF_GROUPS; - - FDKmemcpy(blockSwitchingControl->groupLen, suggestedGroupingTable[blockSwitchingControl->lastAttackIndex], sizeof(blockSwitchingControl->groupLen)); - - if (blockSwitchingControl->attack == TRUE) - blockSwitchingControl->maxWindowNrg = FDKaacEnc_GetWindowEnergy(blockSwitchingControl->windowNrg[0], blockSwitchingControl->lastAttackIndex); - else - blockSwitchingControl->maxWindowNrg = FL2FXCONST_DBL(0.0); - - } - - - /* Calculate unfiltered and filtered energies in subwindows and combine to segments */ - FDKaacEnc_CalcWindowEnergy(blockSwitchingControl, granuleLength>>(nBlockSwitchWindows==4? 2:3 ), pTimeSignal); - - /* now calculate if there is an attack */ - - /* reset attack */ - blockSwitchingControl->attack = FALSE; - - /* look for attack */ - enMax = FL2FXCONST_DBL(0.0f); - enM1 = blockSwitchingControl->windowNrgF[0][nBlockSwitchWindows-1]; - - for (i=0; iaccWindowNrg); - blockSwitchingControl->accWindowNrg = fMultAdd(tmp, accWindowNrgFac, enM1) ; - - if (fMult(blockSwitchingControl->windowNrgF[1][i],invAttackRatio) > blockSwitchingControl->accWindowNrg ) { - blockSwitchingControl->attack = TRUE; - blockSwitchingControl->attackIndex = i; - } - enM1 = blockSwitchingControl->windowNrgF[1][i]; - enMax = fixMax(enMax, enM1); - } - - - if (enMax < minAttackNrg) blockSwitchingControl->attack = FALSE; - - /* Check if attack spreads over frame border */ - if((blockSwitchingControl->attack == FALSE) && (blockSwitchingControl->lastattack == TRUE)) { - /* if attack is in last window repeat SHORT_WINDOW */ - if ( ((blockSwitchingControl->windowNrgF[0][nBlockSwitchWindows-1]>>4) > fMult((FIXP_DBL)(10<<(DFRACT_BITS-1-4)), blockSwitchingControl->windowNrgF[1][1])) - && (blockSwitchingControl->lastAttackIndex == (INT)nBlockSwitchWindows-1) - ) - { - blockSwitchingControl->attack = TRUE; - blockSwitchingControl->attackIndex = 0; - } - } - - - if(blockSwitchingControl->allowLookAhead) - { - - - blockSwitchingControl->lastWindowSequence = - chgWndSqLkAhd[blockSwitchingControl->lastattack][blockSwitchingControl->attack][blockSwitchingControl->lastWindowSequence]; - } - else - { - /* Low Delay */ - blockSwitchingControl->lastWindowSequence = - chgWndSq[blockSwitchingControl->attack][blockSwitchingControl->lastWindowSequence]; - } - - - /* update window shape */ - blockSwitchingControl->windowShape = blockType2windowShape[blockSwitchingControl->allowShortFrames][blockSwitchingControl->lastWindowSequence]; - - return(0); -} - - - -static FIXP_DBL FDKaacEnc_GetWindowEnergy(const FIXP_DBL in[], const INT blSwWndIdx) -{ -/* For coherency, change FDKaacEnc_GetWindowEnergy() to calcluate the energy for a block switching analysis windows, - not for a short block. The same is done FDKaacEnc_CalcWindowEnergy(). The result of FDKaacEnc_GetWindowEnergy() - is used for a comparision of the max energy of left/right channel. */ - - return in[blSwWndIdx]; - -} - -static void FDKaacEnc_CalcWindowEnergy(BLOCK_SWITCHING_CONTROL *RESTRICT blockSwitchingControl, INT windowLen, const INT_PCM *pTimeSignal) -{ - INT i; - UINT w; - - FIXP_SGL hiPassCoeff0 = hiPassCoeff[0]; - FIXP_SGL hiPassCoeff1 = hiPassCoeff[1]; - - /* sum up scalarproduct of timesignal as windowed Energies */ - for (w=0; w < blockSwitchingControl->nBlockSwitchWindows; w++) { - - FIXP_DBL temp_windowNrg = FL2FXCONST_DBL(0.0f); - FIXP_DBL temp_windowNrgF = FL2FXCONST_DBL(0.0f); - FIXP_DBL temp_iirState0 = blockSwitchingControl->iirStates[0]; - FIXP_DBL temp_iirState1 = blockSwitchingControl->iirStates[1]; - - /* windowNrg = sum(timesample^2) */ - for(i=0;i> 1; -#else - tempUnfiltered = (FIXP_DBL) *pTimeSignal++ << (DFRACT_BITS-SAMPLE_BITS-1); -#endif - t1 = fMultDiv2(hiPassCoeff1, tempUnfiltered-temp_iirState0); - t2 = fMultDiv2(hiPassCoeff0, temp_iirState1); - tempFiltred = (t1 - t2) << 1; - - temp_iirState0 = tempUnfiltered; - temp_iirState1 = tempFiltred; - - /* subtract 2 from overallscaling (BLOCK_SWITCH_ENERGY_SHIFT) - * because tempUnfiltered was already scaled with 1 (is 2 after squaring) - * subtract 1 from overallscaling (BLOCK_SWITCH_ENERGY_SHIFT) - * because of fMultDiv2 is doing a scaling by one */ - temp_windowNrg += fPow2Div2(tempUnfiltered) >> (BLOCK_SWITCH_ENERGY_SHIFT - 1 - 2); - temp_windowNrgF += fPow2Div2(tempFiltred) >> (BLOCK_SWITCH_ENERGY_SHIFT - 1 - 2); - } - blockSwitchingControl->windowNrg[1][w] = temp_windowNrg; - blockSwitchingControl->windowNrgF[1][w] = temp_windowNrgF; - blockSwitchingControl->iirStates[0] = temp_iirState0; - blockSwitchingControl->iirStates[1] = temp_iirState1; - } -} - - -static const UCHAR synchronizedBlockTypeTable[5][5] = -{ - /* LONG_WINDOW START_WINDOW SHORT_WINDOW STOP_WINDOW LOWOV_WINDOW*/ - /* LONG_WINDOW */ {LONG_WINDOW, START_WINDOW, SHORT_WINDOW, STOP_WINDOW, LOWOV_WINDOW}, - /* START_WINDOW */ {START_WINDOW, START_WINDOW, SHORT_WINDOW, SHORT_WINDOW, LOWOV_WINDOW}, - /* SHORT_WINDOW */ {SHORT_WINDOW, SHORT_WINDOW, SHORT_WINDOW, SHORT_WINDOW, WRONG_WINDOW}, - /* STOP_WINDOW */ {STOP_WINDOW, SHORT_WINDOW, SHORT_WINDOW, STOP_WINDOW, LOWOV_WINDOW}, - /* LOWOV_WINDOW */ {LOWOV_WINDOW, LOWOV_WINDOW, WRONG_WINDOW, LOWOV_WINDOW, LOWOV_WINDOW}, -}; - -int FDKaacEnc_SyncBlockSwitching ( - BLOCK_SWITCHING_CONTROL *blockSwitchingControlLeft, - BLOCK_SWITCHING_CONTROL *blockSwitchingControlRight, - const INT nChannels, - const INT commonWindow ) -{ - UCHAR patchType = LONG_WINDOW; - - if( nChannels == 2 && commonWindow == TRUE) - { - /* could be better with a channel loop (need a handle to psy_data) */ - /* get suggested Block Types and synchronize */ - patchType = synchronizedBlockTypeTable[patchType][blockSwitchingControlLeft->lastWindowSequence]; - patchType = synchronizedBlockTypeTable[patchType][blockSwitchingControlRight->lastWindowSequence]; - - /* sanity check (no change from low overlap window to short winow and vice versa) */ - if (patchType == WRONG_WINDOW) - return -1; /* mixed up AAC-LC and AAC-LD */ - - /* Set synchronized Blocktype */ - blockSwitchingControlLeft->lastWindowSequence = patchType; - blockSwitchingControlRight->lastWindowSequence = patchType; - - /* update window shape */ - blockSwitchingControlLeft->windowShape = blockType2windowShape[blockSwitchingControlLeft->allowShortFrames][blockSwitchingControlLeft->lastWindowSequence]; - blockSwitchingControlRight->windowShape = blockType2windowShape[blockSwitchingControlLeft->allowShortFrames][blockSwitchingControlRight->lastWindowSequence]; - } - - if (blockSwitchingControlLeft->allowShortFrames) - { - int i; - - if( nChannels == 2 ) - { - if (commonWindow == TRUE) - { - /* Synchronize grouping info */ - int windowSequenceLeftOld = blockSwitchingControlLeft->lastWindowSequence; - int windowSequenceRightOld = blockSwitchingControlRight->lastWindowSequence; - - /* Long Blocks */ - if(patchType != SHORT_WINDOW) { - /* Set grouping info */ - blockSwitchingControlLeft->noOfGroups = 1; - blockSwitchingControlRight->noOfGroups = 1; - blockSwitchingControlLeft->groupLen[0] = 1; - blockSwitchingControlRight->groupLen[0] = 1; - - for (i = 1; i < MAX_NO_OF_GROUPS; i++) - { - blockSwitchingControlLeft->groupLen[i] = 0; - blockSwitchingControlRight->groupLen[i] = 0; - } - } - - /* Short Blocks */ - else { - /* in case all two channels were detected as short-blocks before syncing, use the grouping of channel with higher maxWindowNrg */ - if( (windowSequenceLeftOld == SHORT_WINDOW) && - (windowSequenceRightOld == SHORT_WINDOW) ) - { - if(blockSwitchingControlLeft->maxWindowNrg > blockSwitchingControlRight->maxWindowNrg) { - /* Left Channel wins */ - blockSwitchingControlRight->noOfGroups = blockSwitchingControlLeft->noOfGroups; - for (i = 0; i < MAX_NO_OF_GROUPS; i++){ - blockSwitchingControlRight->groupLen[i] = blockSwitchingControlLeft->groupLen[i]; - } - } - else { - /* Right Channel wins */ - blockSwitchingControlLeft->noOfGroups = blockSwitchingControlRight->noOfGroups; - for (i = 0; i < MAX_NO_OF_GROUPS; i++){ - blockSwitchingControlLeft->groupLen[i] = blockSwitchingControlRight->groupLen[i]; - } - } - } - else if ( (windowSequenceLeftOld == SHORT_WINDOW) && - (windowSequenceRightOld != SHORT_WINDOW) ) - { - /* else use grouping of short-block channel */ - blockSwitchingControlRight->noOfGroups = blockSwitchingControlLeft->noOfGroups; - for (i = 0; i < MAX_NO_OF_GROUPS; i++){ - blockSwitchingControlRight->groupLen[i] = blockSwitchingControlLeft->groupLen[i]; - } - } - else if ( (windowSequenceRightOld == SHORT_WINDOW) && - (windowSequenceLeftOld != SHORT_WINDOW) ) - { - blockSwitchingControlLeft->noOfGroups = blockSwitchingControlRight->noOfGroups; - for (i = 0; i < MAX_NO_OF_GROUPS; i++){ - blockSwitchingControlLeft->groupLen[i] = blockSwitchingControlRight->groupLen[i]; - } - } else { - /* syncing a start and stop window ... */ - blockSwitchingControlLeft->noOfGroups = blockSwitchingControlRight->noOfGroups = 2; - blockSwitchingControlLeft->groupLen[0] = blockSwitchingControlRight->groupLen[0] = 4; - blockSwitchingControlLeft->groupLen[1] = blockSwitchingControlRight->groupLen[1] = 4; - } - } /* Short Blocks */ - } - else { - /* stereo, no common window */ - if (blockSwitchingControlLeft->lastWindowSequence!=SHORT_WINDOW){ - blockSwitchingControlLeft->noOfGroups = 1; - blockSwitchingControlLeft->groupLen[0] = 1; - for (i = 1; i < MAX_NO_OF_GROUPS; i++) - { - blockSwitchingControlLeft->groupLen[i] = 0; - } - } - if (blockSwitchingControlRight->lastWindowSequence!=SHORT_WINDOW){ - blockSwitchingControlRight->noOfGroups = 1; - blockSwitchingControlRight->groupLen[0] = 1; - for (i = 1; i < MAX_NO_OF_GROUPS; i++) - { - blockSwitchingControlRight->groupLen[i] = 0; - } - } - } /* common window */ - } else { - /* Mono */ - if (blockSwitchingControlLeft->lastWindowSequence!=SHORT_WINDOW){ - blockSwitchingControlLeft->noOfGroups = 1; - blockSwitchingControlLeft->groupLen[0] = 1; - - for (i = 1; i < MAX_NO_OF_GROUPS; i++) - { - blockSwitchingControlLeft->groupLen[i] = 0; - } - } - } - } /* allowShortFrames */ - - - /* Translate LOWOV_WINDOW block type to a meaningful window shape. */ - if ( ! blockSwitchingControlLeft->allowShortFrames ) { - if ( blockSwitchingControlLeft->lastWindowSequence != LONG_WINDOW - && blockSwitchingControlLeft->lastWindowSequence != STOP_WINDOW ) - { - blockSwitchingControlLeft->lastWindowSequence = LONG_WINDOW; - blockSwitchingControlLeft->windowShape = LOL_WINDOW; - } - } - if (nChannels == 2) { - if ( ! blockSwitchingControlRight->allowShortFrames ) { - if ( blockSwitchingControlRight->lastWindowSequence != LONG_WINDOW - && blockSwitchingControlRight->lastWindowSequence != STOP_WINDOW ) - { - blockSwitchingControlRight->lastWindowSequence = LONG_WINDOW; - blockSwitchingControlRight->windowShape = LOL_WINDOW; - } - } - } - - return 0; -} - - diff --git a/libAACenc/src/block_switch.h b/libAACenc/src/block_switch.h deleted file mode 100644 index e94b6f5..0000000 --- a/libAACenc/src/block_switch.h +++ /dev/null @@ -1,146 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/***************************** MPEG-4 AAC Encoder ************************** - - Author(s): M. Werner - Description: Block switching - -******************************************************************************/ - -#ifndef _BLOCK_SWITCH_H -#define _BLOCK_SWITCH_H - -#include "common_fix.h" - -#include "psy_const.h" - -/****************** Defines ******************************/ - #define BLOCK_SWITCH_WINDOWS 8 /* number of windows for energy calculation */ - -#define BLOCK_SWITCHING_IIR_LEN 2 /* Length of HighPass-IIR-Filter for Attack-Detection */ -#define BLOCK_SWITCH_ENERGY_SHIFT 7 /* should be logDualis(BLOCK_SWITCH_WINDOW_LEN) to avoid overflow in windowNrgs. */ - -#define LAST_WINDOW 0 -#define THIS_WINDOW 1 - - -/****************** Structures ***************************/ -typedef struct{ - INT lastWindowSequence; - INT windowShape; - INT lastWindowShape; - UINT nBlockSwitchWindows; /* number of windows for energy calculation */ - INT attack; - INT lastattack; - INT attackIndex; - INT lastAttackIndex; - INT allowShortFrames; /* for Low Delay, don't allow short frames */ - INT allowLookAhead; /* for Low Delay, don't do look-ahead */ - INT noOfGroups; - INT groupLen[MAX_NO_OF_GROUPS]; - FIXP_DBL maxWindowNrg; /* max energy in subwindows */ - - FIXP_DBL windowNrg[2][BLOCK_SWITCH_WINDOWS]; /* time signal energy in Subwindows (last and current) */ - FIXP_DBL windowNrgF[2][BLOCK_SWITCH_WINDOWS]; /* filtered time signal energy in segments (last and current) */ - FIXP_DBL accWindowNrg; /* recursively accumulated windowNrgF */ - - FIXP_DBL iirStates[BLOCK_SWITCHING_IIR_LEN]; /* filter delay-line */ - -} BLOCK_SWITCHING_CONTROL; - - - - - -void FDKaacEnc_InitBlockSwitching(BLOCK_SWITCHING_CONTROL *blockSwitchingControl, INT isLowDelay); - -int FDKaacEnc_BlockSwitching(BLOCK_SWITCHING_CONTROL *blockSwitchingControl, const INT granuleLength, const int isLFE, const INT_PCM *pTimeSignal); - -int FDKaacEnc_SyncBlockSwitching( - BLOCK_SWITCHING_CONTROL *blockSwitchingControlLeft, - BLOCK_SWITCHING_CONTROL *blockSwitchingControlRight, - const INT noOfChannels, - const INT commonWindow); - -#endif /* #ifndef _BLOCK_SWITCH_H */ diff --git a/libAACenc/src/channel_map.cpp b/libAACenc/src/channel_map.cpp deleted file mode 100644 index 99ed2b5..0000000 --- a/libAACenc/src/channel_map.cpp +++ /dev/null @@ -1,566 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/************************* Fast MPEG AAC Audio Encoder ********************** - - Initial author: A. Groeschel - contents/description: channel mapping functionality - -******************************************************************************/ - -#include "channel_map.h" -#include "bitenc.h" -#include "psy_const.h" -#include "qc_data.h" -#include "aacEnc_ram.h" - - -/* channel_assignment treats the relationship of Input file channels - to the encoder channels. - This is necessary because the usual order in RIFF files (.wav) - is different from the elements order in the coder given - by Table 8.1 (implicit speaker mapping) of the AAC standard. - - In mono and stereo case, this is trivial. - In mc case, it looks like this: - - Channel Input file coder chan -5ch: - front center 2 0 (SCE channel) - left center 0 1 (1st of 1st CPE) - right center 1 2 (2nd of 1st CPE) - left surround 3 3 (1st of 2nd CPE) - right surround 4 4 (2nd of 2nd CPE) - -5.1ch: - front center 2 0 (SCE channel) - left center 0 1 (1st of 1st CPE) - right center 1 2 (2nd of 1st CPE) - left surround 4 3 (1st of 2nd CPE) - right surround 5 4 (2nd of 2nd CPE) - LFE 3 5 (LFE) -*/ - -typedef struct { - - CHANNEL_MODE encoderMode; - INT channel_assignment[/*(8)*/12]; - -} CHANNEL_ASSIGNMENT_INFO_TAB; - - -static const CHANNEL_ASSIGNMENT_INFO_TAB assignmentInfoTabMpeg[] = -{ - { MODE_INVALID, {-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1} }, /* invalid */ - { MODE_1, { 0,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1} }, /* mono */ - { MODE_2, { 0, 1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1} }, /* stereo */ - { MODE_1_2, { 0, 1, 2,-1,-1,-1,-1,-1,-1,-1,-1,-1} }, /* 3ch */ - { MODE_1_2_1, { 0, 1, 2, 3,-1,-1,-1,-1,-1,-1,-1,-1} }, /* 4ch */ - { MODE_1_2_2, { 0, 1, 2, 3, 4,-1,-1,-1,-1,-1,-1,-1} }, /* 5ch */ - { MODE_1_2_2_1, { 0, 1, 2, 3, 4, 5,-1,-1,-1,-1,-1,-1} }, /* 5.1ch */ - { MODE_1_2_2_2_1, { 0, 1, 2, 3, 4, 5, 6, 7,-1,-1,-1,-1} }, /* 7.1ch */ - { MODE_7_1_REAR_SURROUND, { 0, 1, 2, 3, 4, 5, 6, 7,-1,-1,-1,-1} }, /* 7.1ch */ - { MODE_7_1_FRONT_CENTER, { 0, 1, 2, 3, 4, 5, 6, 7,-1,-1,-1,-1} } /* 7.1ch */ -}; - -static const CHANNEL_ASSIGNMENT_INFO_TAB assignmentInfoTabWav[] = -{ - { MODE_INVALID, {-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1} }, /* invalid */ - { MODE_1, { 0,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1} }, /* mono */ - { MODE_2, { 0, 1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1} }, /* stereo */ - { MODE_1_2, { 2, 0, 1,-1,-1,-1,-1,-1,-1,-1,-1,-1} }, /* 3ch */ - { MODE_1_2_1, { 2, 0, 1, 3,-1,-1,-1,-1,-1,-1,-1,-1} }, /* 4ch */ - { MODE_1_2_2, { 2, 0, 1, 3, 4,-1,-1,-1,-1,-1,-1,-1} }, /* 5ch */ - { MODE_1_2_2_1, { 2, 0, 1, 4, 5, 3,-1,-1,-1,-1,-1,-1} }, /* 5.1ch */ - { MODE_1_2_2_2_1, { 2, 6, 7, 0, 1, 4, 5, 3,-1,-1,-1,-1} }, /* 7.1ch */ - { MODE_7_1_REAR_SURROUND, { 2, 0, 1, 6, 7, 4, 5, 3,-1,-1,-1,-1} }, /* 7.1ch */ - { MODE_7_1_FRONT_CENTER, { 2, 6, 7, 0, 1, 4, 5, 3,-1,-1,-1,-1} }, /* 7.1ch */ -}; - -static const CHANNEL_ASSIGNMENT_INFO_TAB assignmentInfoTabWg4[] = -{ - { MODE_INVALID, {-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1} }, /* invalid */ - { MODE_1, { 0,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1} }, /* mono */ - { MODE_2, { 0, 1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1} }, /* stereo */ - { MODE_1_2, { 2, 0, 1,-1,-1,-1,-1,-1,-1,-1,-1,-1} }, /* 3ch */ - { MODE_1_2_1, { 3, 0, 1, 2,-1,-1,-1,-1,-1,-1,-1,-1} }, /* 4ch */ - { MODE_1_2_2, { 4, 0, 1, 2, 3,-1,-1,-1,-1,-1,-1,-1} }, /* 5ch */ - { MODE_1_2_2_1, { 4, 0, 1, 2, 3, 5,-1,-1,-1,-1,-1,-1} }, /* 5.1ch */ - { MODE_1_2_2_2_1, { 6, 0, 1, 2, 3, 4, 5, 7,-1,-1,-1,-1} }, /* 7.1ch */ -}; - -/* Channel mode configuration tab provides, - corresponding number of channels and elements -*/ -static const CHANNEL_MODE_CONFIG_TAB channelModeConfig[] = -{ - { MODE_1, 1, 1, 1 }, /* SCE */ - { MODE_2, 2, 2, 1 }, /* CPE */ - { MODE_1_2, 3, 3, 2 }, /* SCE,CPE */ - { MODE_1_2_1, 4, 4, 3 }, /* SCE,CPE,SCE */ - { MODE_1_2_2, 5, 5, 3 }, /* SCE,CPE,CPE */ - { MODE_1_2_2_1, 6, 5, 4 }, /* SCE,CPE,CPE,LFE */ - { MODE_1_2_2_2_1, 8, 7, 5 }, /* SCE,CPE,CPE,CPE,LFE */ - { MODE_7_1_REAR_SURROUND, 8, 7, 5 }, - { MODE_7_1_FRONT_CENTER, 8, 7, 5 }, -}; - -#define MAX_MODES (sizeof(assignmentInfoTabWav)/sizeof(CHANNEL_ASSIGNMENT_INFO_TAB)) - -const INT* FDKaacEnc_getChannelAssignment(CHANNEL_MODE encMode, CHANNEL_ORDER co) -{ - const CHANNEL_ASSIGNMENT_INFO_TAB *pTab; - int i; - - if (co == CH_ORDER_MPEG) - pTab = assignmentInfoTabMpeg; - else if (co == CH_ORDER_WAV) - pTab = assignmentInfoTabWav; - else - pTab = assignmentInfoTabWg4; - - for(i=MAX_MODES-1; i>0; i--) { - if (encMode== pTab[i].encoderMode) { - break; - } - } - return (pTab[i].channel_assignment); -} - -AAC_ENCODER_ERROR FDKaacEnc_DetermineEncoderMode(CHANNEL_MODE* mode, INT nChannels) -{ - INT i; - CHANNEL_MODE encMode = MODE_INVALID; - - if (*mode==MODE_UNKNOWN) { - for (i=0; i<(INT)sizeof(channelModeConfig)/(INT)sizeof(CHANNEL_MODE_CONFIG_TAB); i++) { - if (channelModeConfig[i].nChannels==nChannels) { - encMode = channelModeConfig[i].encMode; - break; - } - } - *mode = encMode; - } - else { - /* check if valid channel configuration */ - if (FDKaacEnc_GetChannelModeConfiguration(*mode)->nChannels==nChannels) { - encMode = *mode; - } - } - - if (encMode==MODE_INVALID) { - return AAC_ENC_UNSUPPORTED_CHANNELCONFIG; - } - - return AAC_ENC_OK; -} - -static INT FDKaacEnc_initElement (ELEMENT_INFO* elInfo, MP4_ELEMENT_ID elType, INT* cnt, CHANNEL_MODE mode, CHANNEL_ORDER co, INT* it_cnt, const FIXP_DBL relBits) { - - INT error=0; - INT counter =*cnt; - - const INT *assign = FDKaacEnc_getChannelAssignment(mode, co); - - elInfo->elType=elType; - elInfo->relativeBits = relBits; - - switch(elInfo->elType) { - case ID_SCE: case ID_LFE: case ID_CCE: - elInfo->nChannelsInEl=1; - elInfo->ChannelIndex[0]=assign[counter++]; - elInfo->instanceTag=it_cnt[elType]++; - - break; - case ID_CPE: - elInfo->nChannelsInEl=2; - elInfo->ChannelIndex[0]=assign[counter++]; - elInfo->ChannelIndex[1]=assign[counter++]; - elInfo->instanceTag=it_cnt[elType]++; - break; - case ID_DSE: - elInfo->nChannelsInEl=0; - elInfo->ChannelIndex[0]=0; - elInfo->ChannelIndex[1]=0; - elInfo->instanceTag=it_cnt[elType]++; - break; - default: error=1; - }; - *cnt = counter; - return error; - -} - -AAC_ENCODER_ERROR FDKaacEnc_InitChannelMapping(CHANNEL_MODE mode, CHANNEL_ORDER co, CHANNEL_MAPPING* cm) -{ - INT count=0; /* count through coder channels */ - INT it_cnt[ID_END+1]; - INT i; - - for (i=0; iencMode = channelModeConfig[i].encMode; - cm->nChannels = channelModeConfig[i].nChannels; - cm->nChannelsEff = channelModeConfig[i].nChannelsEff; - cm->nElements = channelModeConfig[i].nElements; - - break; - } - } - - /* init element info struct */ - switch(mode) { - case MODE_1: - /* (mono) sce */ - FDKaacEnc_initElement(&cm->elInfo[0], ID_SCE, &count, mode, co, it_cnt, (FIXP_DBL)MAXVAL_DBL); - break; - case MODE_2: - /* (stereo) cpe */ - FDKaacEnc_initElement(&cm->elInfo[0], ID_CPE, &count, mode, co, it_cnt, (FIXP_DBL)MAXVAL_DBL); - break; - - case MODE_1_2: - /* sce + cpe */ - FDKaacEnc_initElement(&cm->elInfo[0], ID_SCE, &count, mode, co, it_cnt, FL2FXCONST_DBL(0.4f)); - FDKaacEnc_initElement(&cm->elInfo[1], ID_CPE, &count, mode, co, it_cnt, FL2FXCONST_DBL(0.6f)); - break; - - case MODE_1_2_1: - /* sce + cpe + sce */ - FDKaacEnc_initElement(&cm->elInfo[0], ID_SCE, &count, mode, co, it_cnt, FL2FXCONST_DBL(0.3f)); - FDKaacEnc_initElement(&cm->elInfo[1], ID_CPE, &count, mode, co, it_cnt, FL2FXCONST_DBL(0.4f)); - FDKaacEnc_initElement(&cm->elInfo[2], ID_SCE, &count, mode, co, it_cnt, FL2FXCONST_DBL(0.3f)); - break; - - case MODE_1_2_2: - /* sce + cpe + cpe */ - FDKaacEnc_initElement(&cm->elInfo[0], ID_SCE, &count, mode, co, it_cnt, FL2FXCONST_DBL(0.26f)); - FDKaacEnc_initElement(&cm->elInfo[1], ID_CPE, &count, mode, co, it_cnt, FL2FXCONST_DBL(0.37f)); - FDKaacEnc_initElement(&cm->elInfo[2], ID_CPE, &count, mode, co, it_cnt, FL2FXCONST_DBL(0.37f)); - break; - - case MODE_1_2_2_1: - /* (5.1) sce + cpe + cpe + lfe */ - FDKaacEnc_initElement(&cm->elInfo[0], ID_SCE, &count, mode, co, it_cnt, FL2FXCONST_DBL(0.24f)); - FDKaacEnc_initElement(&cm->elInfo[1], ID_CPE, &count, mode, co, it_cnt, FL2FXCONST_DBL(0.35f)); - FDKaacEnc_initElement(&cm->elInfo[2], ID_CPE, &count, mode, co, it_cnt, FL2FXCONST_DBL(0.35f)); - FDKaacEnc_initElement(&cm->elInfo[3], ID_LFE, &count, mode, co, it_cnt, FL2FXCONST_DBL(0.06f)); - break; - - case MODE_1_2_2_2_1: - case MODE_7_1_REAR_SURROUND: - case MODE_7_1_FRONT_CENTER: - /* (7.1) sce + cpe + cpe + cpe + lfe */ - FDKaacEnc_initElement(&cm->elInfo[0], ID_SCE, &count, mode, co, it_cnt, FL2FXCONST_DBL(0.18f)); - FDKaacEnc_initElement(&cm->elInfo[1], ID_CPE, &count, mode, co, it_cnt, FL2FXCONST_DBL(0.26f)); - FDKaacEnc_initElement(&cm->elInfo[2], ID_CPE, &count, mode, co, it_cnt, FL2FXCONST_DBL(0.26f)); - FDKaacEnc_initElement(&cm->elInfo[3], ID_CPE, &count, mode, co, it_cnt, FL2FXCONST_DBL(0.26f)); - FDKaacEnc_initElement(&cm->elInfo[4], ID_LFE, &count, mode, co, it_cnt, FL2FXCONST_DBL(0.04f)); - break; - default: - //*chMap=0; - return AAC_ENC_UNSUPPORTED_CHANNELCONFIG; - }; - - - FDK_ASSERT(cm->nElements<=(8)); - - - return AAC_ENC_OK; -} - -AAC_ENCODER_ERROR FDKaacEnc_InitElementBits(QC_STATE *hQC, - CHANNEL_MAPPING *cm, - INT bitrateTot, - INT averageBitsTot, - INT maxChannelBits) -{ - int sc_brTot = CountLeadingBits(bitrateTot); - - switch(cm->encMode) { - case MODE_1: - hQC->elementBits[0]->chBitrateEl = bitrateTot; - - hQC->elementBits[0]->maxBitsEl = maxChannelBits; - - hQC->elementBits[0]->relativeBitsEl = cm->elInfo[0].relativeBits; - break; - - case MODE_2: - hQC->elementBits[0]->chBitrateEl = bitrateTot>>1; - - hQC->elementBits[0]->maxBitsEl = 2*maxChannelBits; - - hQC->elementBits[0]->relativeBitsEl = cm->elInfo[0].relativeBits; - break; - case MODE_1_2: { - hQC->elementBits[0]->relativeBitsEl = cm->elInfo[0].relativeBits; - hQC->elementBits[1]->relativeBitsEl = cm->elInfo[1].relativeBits; - FIXP_DBL sceRate = cm->elInfo[0].relativeBits; - FIXP_DBL cpeRate = cm->elInfo[1].relativeBits; - - hQC->elementBits[0]->chBitrateEl = fMult(sceRate, (FIXP_DBL)(bitrateTot<>sc_brTot; - hQC->elementBits[1]->chBitrateEl = fMult(cpeRate, (FIXP_DBL)(bitrateTot<>(sc_brTot+1); - - hQC->elementBits[0]->maxBitsEl = maxChannelBits; - hQC->elementBits[1]->maxBitsEl = 2*maxChannelBits; - break; - } - case MODE_1_2_1: { - /* sce + cpe + sce */ - hQC->elementBits[0]->relativeBitsEl = cm->elInfo[0].relativeBits; - hQC->elementBits[1]->relativeBitsEl = cm->elInfo[1].relativeBits; - hQC->elementBits[2]->relativeBitsEl = cm->elInfo[2].relativeBits; - FIXP_DBL sce1Rate = cm->elInfo[0].relativeBits; - FIXP_DBL cpeRate = cm->elInfo[1].relativeBits; - FIXP_DBL sce2Rate = cm->elInfo[2].relativeBits; - - hQC->elementBits[0]->chBitrateEl = fMult(sce1Rate, (FIXP_DBL)(bitrateTot<>sc_brTot; - hQC->elementBits[1]->chBitrateEl = fMult(cpeRate, (FIXP_DBL)(bitrateTot<>(sc_brTot+1); - hQC->elementBits[2]->chBitrateEl = fMult(sce2Rate, (FIXP_DBL)(bitrateTot<>sc_brTot; - - hQC->elementBits[0]->maxBitsEl = maxChannelBits; - hQC->elementBits[1]->maxBitsEl = 2*maxChannelBits; - hQC->elementBits[2]->maxBitsEl = maxChannelBits; - break; - } - case MODE_1_2_2: { - /* sce + cpe + cpe */ - hQC->elementBits[0]->relativeBitsEl = cm->elInfo[0].relativeBits; - hQC->elementBits[1]->relativeBitsEl = cm->elInfo[1].relativeBits; - hQC->elementBits[2]->relativeBitsEl = cm->elInfo[2].relativeBits; - FIXP_DBL sceRate = cm->elInfo[0].relativeBits; - FIXP_DBL cpe1Rate = cm->elInfo[1].relativeBits; - FIXP_DBL cpe2Rate = cm->elInfo[2].relativeBits; - - hQC->elementBits[0]->chBitrateEl = fMult(sceRate, (FIXP_DBL)(bitrateTot<>sc_brTot; - hQC->elementBits[1]->chBitrateEl = fMult(cpe1Rate, (FIXP_DBL)(bitrateTot<>(sc_brTot+1); - hQC->elementBits[2]->chBitrateEl = fMult(cpe2Rate, (FIXP_DBL)(bitrateTot<>(sc_brTot+1); - - hQC->elementBits[0]->maxBitsEl = maxChannelBits; - hQC->elementBits[1]->maxBitsEl = 2*maxChannelBits; - hQC->elementBits[2]->maxBitsEl = 2*maxChannelBits; - break; - } - - case MODE_1_2_2_1: { - /* (5.1) sce + cpe + cpe + lfe */ - hQC->elementBits[0]->relativeBitsEl = cm->elInfo[0].relativeBits; - hQC->elementBits[1]->relativeBitsEl = cm->elInfo[1].relativeBits; - hQC->elementBits[2]->relativeBitsEl = cm->elInfo[2].relativeBits; - hQC->elementBits[3]->relativeBitsEl = cm->elInfo[3].relativeBits; - FIXP_DBL sceRate = cm->elInfo[0].relativeBits; - FIXP_DBL cpe1Rate = cm->elInfo[1].relativeBits; - FIXP_DBL cpe2Rate = cm->elInfo[2].relativeBits; - FIXP_DBL lfeRate = cm->elInfo[3].relativeBits; - - int maxBitsTot = maxChannelBits * 5; /* LFE does not add to bit reservoir */ - int sc = CountLeadingBits(fixMax(maxChannelBits,averageBitsTot)); - int maxLfeBits = (int) FDKmax ( (INT)((fMult(lfeRate,(FIXP_DBL)(maxChannelBits<>sc)<<1), - (INT)((fMult(FL2FXCONST_DBL(1.1f/2.f),fMult(lfeRate,(FIXP_DBL)(averageBitsTot<>sc) ); - - maxChannelBits = (maxBitsTot - maxLfeBits); - sc = CountLeadingBits(maxChannelBits); - - maxChannelBits = fMult((FIXP_DBL)maxChannelBits<>sc; - - hQC->elementBits[0]->chBitrateEl = fMult(sceRate, (FIXP_DBL)(bitrateTot<>sc_brTot; - hQC->elementBits[1]->chBitrateEl = fMult(cpe1Rate, (FIXP_DBL)(bitrateTot<>(sc_brTot+1); - hQC->elementBits[2]->chBitrateEl = fMult(cpe2Rate, (FIXP_DBL)(bitrateTot<>(sc_brTot+1); - hQC->elementBits[3]->chBitrateEl = fMult(lfeRate, (FIXP_DBL)(bitrateTot<>sc_brTot; - - hQC->elementBits[0]->maxBitsEl = maxChannelBits; - hQC->elementBits[1]->maxBitsEl = 2*maxChannelBits; - hQC->elementBits[2]->maxBitsEl = 2*maxChannelBits; - hQC->elementBits[3]->maxBitsEl = maxLfeBits; - - break; - } - case MODE_7_1_REAR_SURROUND: - case MODE_7_1_FRONT_CENTER: - case MODE_1_2_2_2_1: { - int cpe3Idx = 3; - int lfeIdx = 4; - - /* (7.1) sce + cpe + cpe + cpe + lfe */ - FIXP_DBL sceRate = hQC->elementBits[0]->relativeBitsEl = cm->elInfo[0].relativeBits; - FIXP_DBL cpe1Rate = hQC->elementBits[1]->relativeBitsEl = cm->elInfo[1].relativeBits; - FIXP_DBL cpe2Rate = hQC->elementBits[2]->relativeBitsEl = cm->elInfo[2].relativeBits; - FIXP_DBL cpe3Rate = hQC->elementBits[cpe3Idx]->relativeBitsEl = cm->elInfo[cpe3Idx].relativeBits; - FIXP_DBL lfeRate = hQC->elementBits[lfeIdx]->relativeBitsEl = cm->elInfo[lfeIdx].relativeBits; - - int maxBitsTot = maxChannelBits * 7; /* LFE does not add to bit reservoir */ - int sc = CountLeadingBits(fixMax(maxChannelBits,averageBitsTot)); - int maxLfeBits = (int) FDKmax ( (INT)((fMult(lfeRate,(FIXP_DBL)(maxChannelBits<>sc)<<1), - (INT)((fMult(FL2FXCONST_DBL(1.1f/2.f),fMult(lfeRate,(FIXP_DBL)(averageBitsTot<>sc) ); - - maxChannelBits = (maxBitsTot - maxLfeBits) / 7; - - hQC->elementBits[0]->chBitrateEl = fMult(sceRate, (FIXP_DBL)(bitrateTot<>sc_brTot; - hQC->elementBits[1]->chBitrateEl = fMult(cpe1Rate, (FIXP_DBL)(bitrateTot<>(sc_brTot+1); - hQC->elementBits[2]->chBitrateEl = fMult(cpe2Rate, (FIXP_DBL)(bitrateTot<>(sc_brTot+1); - hQC->elementBits[cpe3Idx]->chBitrateEl = fMult(cpe3Rate, (FIXP_DBL)(bitrateTot<>(sc_brTot+1); - hQC->elementBits[lfeIdx]->chBitrateEl = fMult(lfeRate, (FIXP_DBL)(bitrateTot<>sc_brTot; - - hQC->elementBits[0]->maxBitsEl = maxChannelBits; - hQC->elementBits[1]->maxBitsEl = 2*maxChannelBits; - hQC->elementBits[2]->maxBitsEl = 2*maxChannelBits; - hQC->elementBits[cpe3Idx]->maxBitsEl = 2*maxChannelBits; - hQC->elementBits[lfeIdx]->maxBitsEl = maxLfeBits; - break; - } - default: - return AAC_ENC_UNSUPPORTED_CHANNELCONFIG; - } - - return AAC_ENC_OK; -} - -/********************************************************************************/ -/* */ -/* function: GetMonoStereoMODE(const CHANNEL_MODE mode) */ -/* */ -/* description: Determines encoder setting from channel mode. */ -/* Multichannel modes are mapped to mono or stereo modes */ -/* returns MODE_MONO in case of mono, */ -/* MODE_STEREO in case of stereo */ -/* MODE_INVALID in case of error */ -/* */ -/* input: CHANNEL_MODE mode: Encoder mode (see qc_data.h). */ -/* output: return: CM_STEREO_MODE monoStereoSetting */ -/* (MODE_INVALID: error, */ -/* MODE_MONO: mono */ -/* MODE_STEREO: stereo). */ -/* */ -/* misc: No memory is allocated. */ -/* */ -/********************************************************************************/ - -ELEMENT_MODE FDKaacEnc_GetMonoStereoMode(const CHANNEL_MODE mode){ - - ELEMENT_MODE monoStereoSetting = EL_MODE_INVALID; - - switch(mode){ - case MODE_1: /* mono setups */ - monoStereoSetting = EL_MODE_MONO; - break; - case MODE_2: /* stereo setups */ - case MODE_1_2: - case MODE_1_2_1: - case MODE_1_2_2: - case MODE_1_2_2_1: - case MODE_1_2_2_2_1: - case MODE_7_1_REAR_SURROUND: - case MODE_7_1_FRONT_CENTER: - monoStereoSetting = EL_MODE_STEREO; - break; - default: /* error */ - monoStereoSetting = EL_MODE_INVALID; - break; - } - - return monoStereoSetting; -} - -const CHANNEL_MODE_CONFIG_TAB* FDKaacEnc_GetChannelModeConfiguration(const CHANNEL_MODE mode) -{ - INT i; - const CHANNEL_MODE_CONFIG_TAB *cm_config = NULL; - - /* get channel mode config */ - for (i=0; i<(INT)sizeof(channelModeConfig)/(INT)sizeof(CHANNEL_MODE_CONFIG_TAB); i++) { - if (channelModeConfig[i].encMode==mode) - { - cm_config = &channelModeConfig[i]; - break; - } - } - return cm_config; -} diff --git a/libAACenc/src/channel_map.h b/libAACenc/src/channel_map.h deleted file mode 100644 index 2cfb486..0000000 --- a/libAACenc/src/channel_map.h +++ /dev/null @@ -1,132 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/************************* Fast MPEG AAC Audio Encoder ********************** - - Initial author: A. Groeschel - contents/description: channel mapping functionality - -******************************************************************************/ - -#ifndef _CHANNEL_MAP_H -#define _CHANNEL_MAP_H - - -#include "aacenc.h" -#include "psy_const.h" -#include "qc_data.h" - -typedef struct { - CHANNEL_MODE encMode; - INT nChannels; - INT nChannelsEff; - INT nElements; -} CHANNEL_MODE_CONFIG_TAB; - - -/* Element mode */ -typedef enum { - EL_MODE_INVALID = 0, - EL_MODE_MONO, - EL_MODE_STEREO -} ELEMENT_MODE; - - -AAC_ENCODER_ERROR FDKaacEnc_DetermineEncoderMode(CHANNEL_MODE* mode, - INT nChannels); - -AAC_ENCODER_ERROR FDKaacEnc_InitChannelMapping(CHANNEL_MODE mode, - CHANNEL_ORDER co, - CHANNEL_MAPPING* chMap); - -AAC_ENCODER_ERROR FDKaacEnc_InitElementBits(QC_STATE *hQC, - CHANNEL_MAPPING *cm, - INT bitrateTot, - INT averageBitsTot, - INT maxChannelBits); - -ELEMENT_MODE FDKaacEnc_GetMonoStereoMode(const CHANNEL_MODE mode); - -const CHANNEL_MODE_CONFIG_TAB* FDKaacEnc_GetChannelModeConfiguration(const CHANNEL_MODE mode); - -#endif /* CHANNEL_MAP_H */ diff --git a/libAACenc/src/chaosmeasure.cpp b/libAACenc/src/chaosmeasure.cpp deleted file mode 100644 index 4e56e9e..0000000 --- a/libAACenc/src/chaosmeasure.cpp +++ /dev/null @@ -1,161 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M. Werner - contents/description: Chaos measure calculation - -******************************************************************************/ - -#include "chaosmeasure.h" - -/***************************************************************************** - functionname: FDKaacEnc_FDKaacEnc_CalculateChaosMeasurePeakFast - description: Eberlein method of chaos measure calculation by high-pass - filtering amplitude spectrum - A special case of FDKaacEnc_CalculateChaosMeasureTonalGeneric -- - highly optimized -*****************************************************************************/ -static void -FDKaacEnc_FDKaacEnc_CalculateChaosMeasurePeakFast( FIXP_DBL *RESTRICT paMDCTDataNM0, - INT numberOfLines, - FIXP_DBL *RESTRICT chaosMeasure ) -{ - INT i, j; - - /* calculate chaos measure by "peak filter" */ - for (i=0; i<2; i++) { - /* make even and odd pass through data */ - FIXP_DBL left,center; /* left, center tap of filter */ - - left = (FIXP_DBL)((LONG)paMDCTDataNM0[i]^((LONG)paMDCTDataNM0[i]>>(DFRACT_BITS-1))); - center = (FIXP_DBL)((LONG)paMDCTDataNM0[i+2]^((LONG)paMDCTDataNM0[i+2]>>(DFRACT_BITS-1))); - - for (j = i+2; j < numberOfLines - 2; j+=2) { - FIXP_DBL right = (FIXP_DBL)((LONG)paMDCTDataNM0[j+2]^((LONG)paMDCTDataNM0[j+2]>>(DFRACT_BITS-1))); - FIXP_DBL tmp = (left>>1)+(right>>1); - - if (tmp < center ) { - INT leadingBits = CntLeadingZeros(center)-1; - tmp = schur_div(tmp<codeBook == 11) || (huffsection->codeBook >= 16)) ) { - sideInfoBits = 5; - } - else { - sideInfoBits = sideInfoTab[huffsection->sfbCnt]; - } - - return (sideInfoBits); -} - -/* count bits using all possible tables */ -static void FDKaacEnc_buildBitLookUp( - const SHORT* const quantSpectrum, - const INT maxSfb, - const INT* const sfbOffset, - const UINT* const sfbMax, - INT bitLookUp[MAX_SFB_LONG][CODE_BOOK_ESC_NDX + 1], - SECTION_INFO* const huffsection - ) -{ - INT i, sfbWidth; - - for (i = 0; i < maxSfb; i++) - { - huffsection[i].sfbCnt = 1; - huffsection[i].sfbStart = i; - huffsection[i].sectionBits = INVALID_BITCOUNT; - huffsection[i].codeBook = -1; - sfbWidth = sfbOffset[i + 1] - sfbOffset[i]; - FDKaacEnc_bitCount(quantSpectrum + sfbOffset[i], sfbWidth, sfbMax[i], bitLookUp[i]); - } -} - -/* essential helper functions */ -static INT FDKaacEnc_findBestBook( - const INT* const bc, - INT* const book, - const INT useVCB11 - ) -{ - INT minBits = INVALID_BITCOUNT, j; - - int end = CODE_BOOK_ESC_NDX; - - - for (j = 0; j <= end; j++) - { - if (bc[j] < minBits) - { - minBits = bc[j]; - *book = j; - } - } - return (minBits); -} - -static INT FDKaacEnc_findMinMergeBits( - const INT* const bc1, - const INT* const bc2, - const INT useVCB11 - ) -{ - INT minBits = INVALID_BITCOUNT, j; - - int end = CODE_BOOK_ESC_NDX; - - - for (j = 0; j <= end; j++) - { - if (bc1[j] + bc2[j] < minBits) - { - minBits = bc1[j] + bc2[j]; - } - } - return (minBits); -} - -static void FDKaacEnc_mergeBitLookUp( - INT* const bc1, - const INT* const bc2 - ) -{ - int j; - - for (j = 0; j <= CODE_BOOK_ESC_NDX; j++) - { - bc1[j] = fixMin(bc1[j] + bc2[j], INVALID_BITCOUNT); - } -} - -static INT FDKaacEnc_findMaxMerge( - const INT* const mergeGainLookUp, - const SECTION_INFO* const huffsection, - const INT maxSfb, - INT* const maxNdx - ) -{ - INT i, maxMergeGain = 0; - - for (i = 0; i + huffsection[i].sfbCnt < maxSfb; i += huffsection[i].sfbCnt) - { - if (mergeGainLookUp[i] > maxMergeGain) - { - maxMergeGain = mergeGainLookUp[i]; - *maxNdx = i; - } - } - return (maxMergeGain); -} - -static INT FDKaacEnc_CalcMergeGain( - const SECTION_INFO* const huffsection, - const INT bitLookUp[MAX_SFB_LONG][CODE_BOOK_ESC_NDX + 1], - const SHORT* const sideInfoTab, - const INT ndx1, - const INT ndx2, - const INT useVCB11 - ) -{ - INT MergeGain, MergeBits, SplitBits; - - MergeBits = sideInfoTab[huffsection[ndx1].sfbCnt + huffsection[ndx2].sfbCnt] + FDKaacEnc_findMinMergeBits(bitLookUp[ndx1], bitLookUp[ndx2], useVCB11); - SplitBits = huffsection[ndx1].sectionBits + huffsection[ndx2].sectionBits; /* Bit amount for splitted huffsections */ - MergeGain = SplitBits - MergeBits; - - if ( (huffsection[ndx1].codeBook==CODE_BOOK_PNS_NO)||(huffsection[ndx2].codeBook==CODE_BOOK_PNS_NO) - || (huffsection[ndx1].codeBook==CODE_BOOK_IS_OUT_OF_PHASE_NO)||(huffsection[ndx2].codeBook==CODE_BOOK_IS_OUT_OF_PHASE_NO) - || (huffsection[ndx1].codeBook==CODE_BOOK_IS_IN_PHASE_NO)||(huffsection[ndx2].codeBook==CODE_BOOK_IS_IN_PHASE_NO) - ) - { - MergeGain = -1; - } - - return (MergeGain); -} - - -/* sectioning Stage 0:find minimum codbooks */ -static void FDKaacEnc_gmStage0( - SECTION_INFO* const RESTRICT huffsection, - const INT bitLookUp[MAX_SFB_LONG][CODE_BOOK_ESC_NDX + 1], - const INT maxSfb, - const INT* const noiseNrg, - const INT* const isBook - ) -{ - INT i; - - for (i = 0; i < maxSfb; i++) - { - /* Side-Info bits will be calculated in Stage 1! */ - if (huffsection[i].sectionBits == INVALID_BITCOUNT) - { - /* intensity and pns codebooks are already allocated in bitcount.c */ - if(noiseNrg[i] != NO_NOISE_PNS){ - huffsection[i].codeBook=CODE_BOOK_PNS_NO; - huffsection[i].sectionBits = 0; - } - else if( isBook[i] ) { - huffsection[i].codeBook=isBook[i]; - huffsection[i].sectionBits = 0; - } - else { - huffsection[i].sectionBits = FDKaacEnc_findBestBook(bitLookUp[i], &(huffsection[i].codeBook), 0); /* useVCB11 must be 0!!! */ - } - } - } -} - -/* - sectioning Stage 1:merge all connected regions with the same code book and - calculate side info - */ -static void FDKaacEnc_gmStage1( - SECTION_INFO* const RESTRICT huffsection, - INT bitLookUp[MAX_SFB_LONG][CODE_BOOK_ESC_NDX + 1], - const INT maxSfb, - const SHORT* const sideInfoTab, - const INT useVCB11 - ) -{ - INT mergeStart = 0, mergeEnd; - - do - { - for (mergeEnd = mergeStart + 1; mergeEnd < maxSfb; mergeEnd++) - { - if (huffsection[mergeStart].codeBook != huffsection[mergeEnd].codeBook) - break; - - - /* we can merge. update tables, side info bits will be updated outside of this loop */ - huffsection[mergeStart].sfbCnt++; - huffsection[mergeStart].sectionBits += huffsection[mergeEnd].sectionBits; - - /* update bit look up for all code books */ - FDKaacEnc_mergeBitLookUp(bitLookUp[mergeStart], bitLookUp[mergeEnd]); - } - - /* add side info info bits */ - huffsection[mergeStart].sectionBits += FDKaacEnc_getSideInfoBits(&huffsection[mergeStart], sideInfoTab, useVCB11); - huffsection[mergeEnd - 1].sfbStart = huffsection[mergeStart].sfbStart; /* speed up prev search */ - - mergeStart = mergeEnd; - - } while (mergeStart < maxSfb); -} - -/* - sectioning Stage 2:greedy merge algorithm, merge connected sections with - maximum bit gain until no more gain is possible - */ -static void -FDKaacEnc_gmStage2( - SECTION_INFO* const RESTRICT huffsection, - INT* const RESTRICT mergeGainLookUp, - INT bitLookUp[MAX_SFB_LONG][CODE_BOOK_ESC_NDX + 1], - const INT maxSfb, - const SHORT* const sideInfoTab, - const INT useVCB11 - ) -{ - INT i; - - for (i = 0; i + huffsection[i].sfbCnt < maxSfb; i += huffsection[i].sfbCnt) - { - mergeGainLookUp[i] = FDKaacEnc_CalcMergeGain(huffsection, - bitLookUp, - sideInfoTab, - i, - i + huffsection[i].sfbCnt, - useVCB11); - } - - while (TRUE) - { - INT maxMergeGain, maxNdx = 0, maxNdxNext, maxNdxLast; - - maxMergeGain = FDKaacEnc_findMaxMerge(mergeGainLookUp, huffsection, maxSfb, &maxNdx); - - /* exit while loop if no more gain is possible */ - if (maxMergeGain <= 0) - break; - - maxNdxNext = maxNdx + huffsection[maxNdx].sfbCnt; - - /* merge sections with maximum bit gain */ - huffsection[maxNdx].sfbCnt += huffsection[maxNdxNext].sfbCnt; - huffsection[maxNdx].sectionBits += huffsection[maxNdxNext].sectionBits - maxMergeGain; - - /* update bit look up table for merged huffsection */ - FDKaacEnc_mergeBitLookUp(bitLookUp[maxNdx], bitLookUp[maxNdxNext]); - - /* update mergeLookUpTable */ - if (maxNdx != 0) - { - maxNdxLast = huffsection[maxNdx - 1].sfbStart; - mergeGainLookUp[maxNdxLast] = FDKaacEnc_CalcMergeGain(huffsection, - bitLookUp, - sideInfoTab, - maxNdxLast, - maxNdx, - useVCB11); - - } - maxNdxNext = maxNdx + huffsection[maxNdx].sfbCnt; - - huffsection[maxNdxNext - 1].sfbStart = huffsection[maxNdx].sfbStart; - - if (maxNdxNext < maxSfb) - mergeGainLookUp[maxNdx] = FDKaacEnc_CalcMergeGain(huffsection, - bitLookUp, - sideInfoTab, - maxNdx, - maxNdxNext, - useVCB11); - - } -} - -/* count bits used by the noiseless coder */ -static void FDKaacEnc_noiselessCounter( - SECTION_DATA* const RESTRICT sectionData, - INT mergeGainLookUp[MAX_SFB_LONG], - INT bitLookUp[MAX_SFB_LONG][CODE_BOOK_ESC_NDX + 1], - const SHORT* const quantSpectrum, - const UINT* const maxValueInSfb, - const INT* const sfbOffset, - const INT blockType, - const INT* const noiseNrg, - const INT* const isBook, - const INT useVCB11 - ) -{ - INT grpNdx, i; - const SHORT *sideInfoTab = NULL; - SECTION_INFO *huffsection; - - /* use appropriate side info table */ - switch (blockType) - { - case LONG_WINDOW: - case START_WINDOW: - case STOP_WINDOW: - sideInfoTab = FDKaacEnc_sideInfoTabLong; - break; - case SHORT_WINDOW: - sideInfoTab = FDKaacEnc_sideInfoTabShort; - break; - } - - sectionData->noOfSections = 0; - sectionData->huffmanBits = 0; - sectionData->sideInfoBits = 0; - - - if (sectionData->maxSfbPerGroup == 0) - return; - - /* loop trough groups */ - for (grpNdx = 0; grpNdx < sectionData->sfbCnt; grpNdx += sectionData->sfbPerGroup) - { - huffsection = sectionData->huffsection + sectionData->noOfSections; - - /* count bits in this group */ - FDKaacEnc_buildBitLookUp(quantSpectrum, - sectionData->maxSfbPerGroup, - sfbOffset + grpNdx, - maxValueInSfb + grpNdx, - bitLookUp, - huffsection); - - /* 0.Stage :Find minimum Codebooks */ - FDKaacEnc_gmStage0(huffsection, bitLookUp, sectionData->maxSfbPerGroup, noiseNrg+grpNdx, isBook+grpNdx); - - /* 1.Stage :Merge all connected regions with the same code book */ - FDKaacEnc_gmStage1(huffsection, bitLookUp, sectionData->maxSfbPerGroup, sideInfoTab, useVCB11); - - - /* - 2.Stage - greedy merge algorithm, merge connected huffsections with maximum bit - gain until no more gain is possible - */ - - FDKaacEnc_gmStage2(huffsection, - mergeGainLookUp, - bitLookUp, - sectionData->maxSfbPerGroup, - sideInfoTab, - useVCB11); - - - - /* - compress output, calculate total huff and side bits - since we did not update the actual codebook in stage 2 - to save time, we must set it here for later use in bitenc - */ - - for (i = 0; i < sectionData->maxSfbPerGroup; i += huffsection[i].sfbCnt) - { - if ((huffsection[i].codeBook==CODE_BOOK_PNS_NO) || - (huffsection[i].codeBook==CODE_BOOK_IS_OUT_OF_PHASE_NO) || - (huffsection[i].codeBook==CODE_BOOK_IS_IN_PHASE_NO)) - { - huffsection[i].sectionBits=0; - } else { - /* the sections in the sectionData are now marked with the optimal code book */ - - FDKaacEnc_findBestBook(bitLookUp[i], &(huffsection[i].codeBook), useVCB11); - - sectionData->huffmanBits += huffsection[i].sectionBits - FDKaacEnc_getSideInfoBits(&huffsection[i], sideInfoTab, useVCB11); - } - - huffsection[i].sfbStart += grpNdx; - - /* sum up side info bits (section data bits) */ - sectionData->sideInfoBits += FDKaacEnc_getSideInfoBits(&huffsection[i], sideInfoTab, useVCB11); - sectionData->huffsection[sectionData->noOfSections++] = huffsection[i]; - } - } -} - - -/******************************************************************************* - - functionname: FDKaacEnc_scfCount - returns : --- - description : count bits used by scalefactors. - - not in all cases if maxValueInSfb[] == 0 we set deltaScf - to zero. only if the difference of the last and future - scalefacGain is not greater then CODE_BOOK_SCF_LAV (60). - - example: - ^ - scalefacGain | - | - | last 75 - | | - | | - | | - | | current 50 - | | | - | | | - | | | - | | | - | | | future 5 - | | | | - --- ... ---------------------------- ... ---------> - sfb - - - if maxValueInSfb[] of current is zero because of a - notfallstrategie, we do not save bits and transmit a - deltaScf of 25. otherwise the deltaScf between the last - scalfacGain (75) and the future scalefacGain (5) is 70. - -********************************************************************************/ -static void FDKaacEnc_scfCount( - const INT* const scalefacGain, - const UINT* const maxValueInSfb, - SECTION_DATA* const RESTRICT sectionData, - const INT* const isScale - ) -{ - INT i, j, k, m, n; - - INT lastValScf = 0; - INT deltaScf = 0; - INT found = 0; - INT scfSkipCounter = 0; - INT lastValIs = 0; - - sectionData->scalefacBits = 0; - - if (scalefacGain == NULL) - return; - - sectionData->firstScf = 0; - - for (i=0; inoOfSections; i++) - { - if (sectionData->huffsection[i].codeBook != CODE_BOOK_ZERO_NO) - { - sectionData->firstScf = sectionData->huffsection[i].sfbStart; - lastValScf = scalefacGain[sectionData->firstScf]; - break; - } - } - - for (i=0; inoOfSections; i++) - { - if ((sectionData->huffsection[i].codeBook == CODE_BOOK_IS_OUT_OF_PHASE_NO) || - (sectionData->huffsection[i].codeBook == CODE_BOOK_IS_IN_PHASE_NO)) - { - for (j = sectionData->huffsection[i].sfbStart; - j < sectionData->huffsection[i].sfbStart + sectionData->huffsection[i].sfbCnt; - j++) - { - INT deltaIs = isScale[j]-lastValIs; - lastValIs = isScale[j]; - sectionData->scalefacBits+=FDKaacEnc_bitCountScalefactorDelta(deltaIs); - } - } /* Intensity */ - else if ((sectionData->huffsection[i].codeBook != CODE_BOOK_ZERO_NO) && - (sectionData->huffsection[i].codeBook != CODE_BOOK_PNS_NO)) - { - INT tmp = sectionData->huffsection[i].sfbStart + sectionData->huffsection[i].sfbCnt; - for (j = sectionData->huffsection[i].sfbStart; jnoOfSections) && (found == 0); m++) - { - if ((sectionData->huffsection[m].codeBook != CODE_BOOK_ZERO_NO) && (sectionData->huffsection[m].codeBook != CODE_BOOK_PNS_NO)) - { - INT end = sectionData->huffsection[m].sfbStart + sectionData->huffsection[m].sfbCnt; - for (n = sectionData->huffsection[m].sfbStart; nscalefacBits += FDKaacEnc_bitCountScalefactorDelta(deltaScf); - } - } - } /* for (i=0; inoOfSections; i++) */ -} - -#ifdef PNS_PRECOUNT_ENABLE -/* - preCount bits used pns -*/ -/* estimate bits used by pns for correction of static bits */ -/* no codebook switch estimation, see AAC LD FASTENC */ -INT noisePreCount(const INT *noiseNrg, INT maxSfb) -{ - INT noisePCMFlag = TRUE; - INT lastValPns = 0, deltaPns; - int i, bits=0; - - for (i = 0; i < maxSfb; i++) { - if (noiseNrg[i] != NO_NOISE_PNS) { - - if (noisePCMFlag) { - bits+=PNS_PCM_BITS; - lastValPns = noiseNrg[i]; - noisePCMFlag = FALSE; - }else { - deltaPns = noiseNrg[i]-lastValPns; - lastValPns = noiseNrg[i]; - bits+=FDKaacEnc_bitCountScalefactorDelta(deltaPns); - } - } - } - return ( bits ); -} -#endif /* PNS_PRECOUNT_ENABLE */ - -/* count bits used by pns */ -static void FDKaacEnc_noiseCount( - SECTION_DATA* const RESTRICT sectionData, - const INT* const noiseNrg - ) -{ - INT noisePCMFlag = TRUE; - INT lastValPns = 0, deltaPns; - int i, j; - - sectionData->noiseNrgBits = 0; - - for (i = 0; i < sectionData->noOfSections; i++) { - if (sectionData->huffsection[i].codeBook == CODE_BOOK_PNS_NO) { - int sfbStart = sectionData->huffsection[i].sfbStart; - int sfbEnd = sfbStart + sectionData->huffsection[i].sfbCnt; - for (j=sfbStart; jnoiseNrgBits+=PNS_PCM_BITS; - lastValPns = noiseNrg[j]; - noisePCMFlag = FALSE; - } else { - deltaPns = noiseNrg[j]-lastValPns; - lastValPns = noiseNrg[j]; - sectionData->noiseNrgBits+=FDKaacEnc_bitCountScalefactorDelta(deltaPns); - } - } - } - } -} - -INT FDKaacEnc_dynBitCount( - BITCNTR_STATE* const hBC, - const SHORT* const quantSpectrum, - const UINT* const maxValueInSfb, - const INT* const scalefac, - const INT blockType, - const INT sfbCnt, - const INT maxSfbPerGroup, - const INT sfbPerGroup, - const INT* const sfbOffset, - SECTION_DATA* const RESTRICT sectionData, - const INT* const noiseNrg, - const INT* const isBook, - const INT* const isScale, - const UINT syntaxFlags - ) -{ - sectionData->blockType = blockType; - sectionData->sfbCnt = sfbCnt; - sectionData->sfbPerGroup = sfbPerGroup; - sectionData->noOfGroups = sfbCnt / sfbPerGroup; - sectionData->maxSfbPerGroup = maxSfbPerGroup; - - FDKaacEnc_noiselessCounter( - sectionData, - hBC->mergeGainLookUp, - (lookUpTable)hBC->bitLookUp, - quantSpectrum, - maxValueInSfb, - sfbOffset, - blockType, - noiseNrg, - isBook, - (syntaxFlags & AC_ER_VCB11)?1:0); - - FDKaacEnc_scfCount( - scalefac, - maxValueInSfb, - sectionData, - isScale); - - FDKaacEnc_noiseCount(sectionData, - noiseNrg); - - return (sectionData->huffmanBits + - sectionData->sideInfoBits + - sectionData->scalefacBits + - sectionData->noiseNrgBits); -} - -INT FDKaacEnc_BCNew(BITCNTR_STATE **phBC - ,UCHAR* dynamic_RAM - ) -{ - BITCNTR_STATE *hBC = GetRam_aacEnc_BitCntrState(); - - if (hBC) - { - *phBC = hBC; - hBC->bitLookUp = GetRam_aacEnc_BitLookUp(0,dynamic_RAM); - hBC->mergeGainLookUp = GetRam_aacEnc_MergeGainLookUp(0,dynamic_RAM); - if (hBC->bitLookUp == 0 || - hBC->mergeGainLookUp == 0) - { - return 1; - } - } - return (hBC == 0) ? 1 : 0; -} - -void FDKaacEnc_BCClose(BITCNTR_STATE **phBC) -{ - if (*phBC!=NULL) { - - FreeRam_aacEnc_BitCntrState(phBC); - } -} - - - diff --git a/libAACenc/src/dyn_bits.h b/libAACenc/src/dyn_bits.h deleted file mode 100644 index ae78a4c..0000000 --- a/libAACenc/src/dyn_bits.h +++ /dev/null @@ -1,167 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M.Werner - contents/description: Noiseless coder module - -******************************************************************************/ - -#ifndef __DYN_BITS_H -#define __DYN_BITS_H - -#include "common_fix.h" - -#include "psy_const.h" -#include "aacenc_tns.h" - -#define MAX_SECTIONS MAX_GROUPED_SFB -#define SECT_ESC_VAL_LONG 31 -#define SECT_ESC_VAL_SHORT 7 -#define CODE_BOOK_BITS 4 -#define SECT_BITS_LONG 5 -#define SECT_BITS_SHORT 3 -#define PNS_PCM_BITS 9 - -typedef struct -{ - INT codeBook; - INT sfbStart; - INT sfbCnt; - INT sectionBits; /* huff + si ! */ -} SECTION_INFO; - - -typedef struct -{ - INT blockType; - INT noOfGroups; - INT sfbCnt; - INT maxSfbPerGroup; - INT sfbPerGroup; - INT noOfSections; - SECTION_INFO huffsection[MAX_SECTIONS]; - INT sideInfoBits; /* sectioning bits */ - INT huffmanBits; /* huffman coded bits */ - INT scalefacBits; /* scalefac coded bits */ - INT noiseNrgBits; /* noiseEnergy coded bits */ - INT firstScf; /* first scf to be coded */ -} SECTION_DATA; - - -struct BITCNTR_STATE -{ - INT *bitLookUp; - INT *mergeGainLookUp; -}; - - -INT FDKaacEnc_BCNew(BITCNTR_STATE **phBC - ,UCHAR* dynamic_RAM - ); - -void FDKaacEnc_BCClose(BITCNTR_STATE **phBC); - -#if defined(PNS_PRECOUNT_ENABLE) -INT noisePreCount(const INT *noiseNrg, INT maxSfb); -#endif - -INT FDKaacEnc_dynBitCount( - BITCNTR_STATE* const hBC, - const SHORT* const quantSpectrum, - const UINT* const maxValueInSfb, - const INT* const scalefac, - const INT blockType, - const INT sfbCnt, - const INT maxSfbPerGroup, - const INT sfbPerGroup, - const INT* const sfbOffset, - SECTION_DATA* const RESTRICT sectionData, - const INT* const noiseNrg, - const INT* const isBook, - const INT* const isScale, - const UINT syntaxFlags - ); - -#endif diff --git a/libAACenc/src/grp_data.cpp b/libAACenc/src/grp_data.cpp deleted file mode 100644 index 465865f..0000000 --- a/libAACenc/src/grp_data.cpp +++ /dev/null @@ -1,272 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M.Werner - contents/description: Short block grouping - -******************************************************************************/ -#include "psy_const.h" -#include "interface.h" - -/* -* this routine does not work in-place -*/ - -static inline FIXP_DBL nrgAddSaturate(const FIXP_DBL a, const FIXP_DBL b) { - return ( (a>=(FIXP_DBL)MAXVAL_DBL-b) ? (FIXP_DBL)MAXVAL_DBL : (a + b) ); -} - -void -FDKaacEnc_groupShortData(FIXP_DBL *mdctSpectrum, /* in-out */ - SFB_THRESHOLD *sfbThreshold, /* in-out */ - SFB_ENERGY *sfbEnergy, /* in-out */ - SFB_ENERGY *sfbEnergyMS, /* in-out */ - SFB_ENERGY *sfbSpreadEnergy, - const INT sfbCnt, - const INT sfbActive, - const INT *sfbOffset, - const FIXP_DBL *sfbMinSnrLdData, - INT *groupedSfbOffset, /* out */ - INT *maxSfbPerGroup, /* out */ - FIXP_DBL *groupedSfbMinSnrLdData, - const INT noOfGroups, - const INT *groupLen, - const INT granuleLength) -{ - INT i,j; - INT line; /* counts through lines */ - INT sfb; /* counts through scalefactor bands */ - INT grp; /* counts through groups */ - INT wnd; /* counts through windows in a group */ - INT offset; /* needed in sfbOffset grouping */ - INT highestSfb; - - INT granuleLength_short = granuleLength/TRANS_FAC; - - /* for short blocks: regroup spectrum and */ - /* group energies and thresholds according to grouping */ - C_ALLOC_SCRATCH_START(tmpSpectrum, FIXP_DBL, (1024)); - - /* calculate maxSfbPerGroup */ - highestSfb = 0; - for (wnd = 0; wnd < TRANS_FAC; wnd++) - { - for (sfb = sfbActive-1; sfb >= highestSfb; sfb--) - { - for (line = sfbOffset[sfb+1]-1; line >= sfbOffset[sfb]; line--) - { - if ( mdctSpectrum[wnd*granuleLength_short+line] != FL2FXCONST_SPC(0.0) ) break; /* this band is not completely zero */ - } - if (line >= sfbOffset[sfb]) break; /* this band was not completely zero */ - } - highestSfb = fixMax(highestSfb, sfb); - } - highestSfb = highestSfb > 0 ? highestSfb : 0; - *maxSfbPerGroup = highestSfb+1; - - /* calculate groupedSfbOffset */ - i = 0; - offset = 0; - for (grp = 0; grp < noOfGroups; grp++) - { - for (sfb = 0; sfb < sfbActive+1; sfb++) - { - groupedSfbOffset[i++] = offset + sfbOffset[sfb] * groupLen[grp]; - } - i += sfbCnt-sfb; - offset += groupLen[grp] * granuleLength_short; - } - groupedSfbOffset[i++] = granuleLength; - - /* calculate groupedSfbMinSnr */ - i = 0; - for (grp = 0; grp < noOfGroups; grp++) - { - for (sfb = 0; sfb < sfbActive; sfb++) - { - groupedSfbMinSnrLdData[i++] = sfbMinSnrLdData[sfb]; - } - i += sfbCnt-sfb; - } - - /* sum up sfbThresholds */ - wnd = 0; - i = 0; - for (grp = 0; grp < noOfGroups; grp++) - { - for (sfb = 0; sfb < sfbActive; sfb++) - { - FIXP_DBL thresh = sfbThreshold->Short[wnd][sfb]; - for (j=1; jShort[wnd+j][sfb]); - } - sfbThreshold->Long[i++] = thresh; - } - i += sfbCnt-sfb; - wnd += groupLen[grp]; - } - - /* sum up sfbEnergies left/right */ - wnd = 0; - i = 0; - for (grp = 0; grp < noOfGroups; grp++) - { - for (sfb = 0; sfb < sfbActive; sfb++) - { - FIXP_DBL energy = sfbEnergy->Short[wnd][sfb]; - for (j=1; jShort[wnd+j][sfb]); - } - sfbEnergy->Long[i++] = energy; - } - i += sfbCnt-sfb; - wnd += groupLen[grp]; - } - - /* sum up sfbEnergies mid/side */ - wnd = 0; - i = 0; - for (grp = 0; grp < noOfGroups; grp++) - { - for (sfb = 0; sfb < sfbActive; sfb++) - { - FIXP_DBL energy = sfbEnergyMS->Short[wnd][sfb]; - for (j=1; jShort[wnd+j][sfb]); - } - sfbEnergyMS->Long[i++] = energy; - } - i += sfbCnt-sfb; - wnd += groupLen[grp]; - } - - /* sum up sfbSpreadEnergies */ - wnd = 0; - i = 0; - for (grp = 0; grp < noOfGroups; grp++) - { - for (sfb = 0; sfb < sfbActive; sfb++) - { - FIXP_DBL energy = sfbSpreadEnergy->Short[wnd][sfb]; - for (j=1; jShort[wnd+j][sfb]); - } - sfbSpreadEnergy->Long[i++] = energy; - } - i += sfbCnt-sfb; - wnd += groupLen[grp]; - } - - /* re-group spectrum */ - wnd = 0; - i = 0; - for (grp = 0; grp < noOfGroups; grp++) - { - for (sfb = 0; sfb < sfbActive; sfb++) - { - int width = sfbOffset[sfb+1]-sfbOffset[sfb]; - FIXP_DBL *pMdctSpectrum = &mdctSpectrum[sfbOffset[sfb]] + wnd*granuleLength_short; - for (j = 0; j < groupLen[grp]; j++) - { - FIXP_DBL *pTmp = pMdctSpectrum; - for (line = width; line > 0; line--) - { - tmpSpectrum[i++] = *pTmp++; - } - pMdctSpectrum += granuleLength_short; - } - } - i += (groupLen[grp]*(sfbOffset[sfbCnt]-sfbOffset[sfb])); - wnd += groupLen[grp]; - } - - FDKmemcpy(mdctSpectrum, tmpSpectrum, granuleLength*sizeof(FIXP_DBL)); - - C_ALLOC_SCRATCH_END(tmpSpectrum, FIXP_DBL, (1024)) -} diff --git a/libAACenc/src/grp_data.h b/libAACenc/src/grp_data.h deleted file mode 100644 index f061855..0000000 --- a/libAACenc/src/grp_data.h +++ /dev/null @@ -1,115 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M.Werner - contents/description: Short block grouping - -******************************************************************************/ -#ifndef __GRP_DATA_H__ -#define __GRP_DATA_H__ - -#include "common_fix.h" - -#include "psy_data.h" - - -void -FDKaacEnc_groupShortData(FIXP_DBL *mdctSpectrum, /* in-out */ - SFB_THRESHOLD *sfbThreshold, /* in-out */ - SFB_ENERGY *sfbEnergy, /* in-out */ - SFB_ENERGY *sfbEnergyMS, /* in-out */ - SFB_ENERGY *sfbSpreadEnergy, - const INT sfbCnt, - const INT sfbActive, - const INT *sfbOffset, - const FIXP_DBL *sfbMinSnrLdData, - INT *groupedSfbOffset, /* out */ - INT *maxSfbPerGroup, - FIXP_DBL *groupedSfbMinSnrLdData, - const INT noOfGroups, - const INT *groupLen, - const INT granuleLength); - -#endif /* _INTERFACE_H */ diff --git a/libAACenc/src/intensity.cpp b/libAACenc/src/intensity.cpp deleted file mode 100644 index b45b27b..0000000 --- a/libAACenc/src/intensity.cpp +++ /dev/null @@ -1,760 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: A. Horndasch (code originally from lwr) / Josef Hoepfl (FDK) - contents/description: intensity stereo processing - -******************************************************************************/ - -#include "intensity.h" -#include "interface.h" -#include "psy_configuration.h" -#include "psy_const.h" -#include "qc_main.h" -#include "bit_cnt.h" - -/* only set an IS seed it left/right channel correlation is above IS_CORR_THRESH */ -#define IS_CORR_THRESH FL2FXCONST_DBL(0.95f) - -/* when expanding the IS region to more SFBs only accept an error that is - * not more than IS_TOTAL_ERROR_THRESH overall and - * not more than IS_LOCAL_ERROR_THRESH for the current SFB */ -#define IS_TOTAL_ERROR_THRESH FL2FXCONST_DBL(0.04f) -#define IS_LOCAL_ERROR_THRESH FL2FXCONST_DBL(0.01f) - -/* the maximum allowed change of the intensity direction (unit: IS scale) - scaled with factor 0.25 - */ -#define IS_DIRECTION_DEVIATION_THRESH_SF 2 -#define IS_DIRECTION_DEVIATION_THRESH FL2FXCONST_DBL(2.0f/(1< no IS if the panning angle is not far from the middle, MS will do */ -/* this is equivalent to a scale of +/-1.02914634566 */ -#define IS_LEFT_RIGHT_RATIO_THRESH FL2FXCONST_DBL(0.7f) - -/* scalefactor of realScale */ -#define REAL_SCALE_SF 1 - -/* scalefactor overallLoudness */ -#define OVERALL_LOUDNESS_SF 6 - -/* scalefactor for sum over max samples per goup */ -#define MAX_SFB_PER_GROUP_SF 6 - -/* scalefactor for sum of mdct spectrum */ -#define MDCT_SPEC_SF 6 - - -typedef struct -{ - - FIXP_DBL corr_thresh; /*!< Only set an IS seed it left/right channel correlation is above corr_thresh */ - - FIXP_DBL total_error_thresh; /*!< When expanding the IS region to more SFBs only accept an error that is - not more than 'total_error_thresh' overall. */ - - FIXP_DBL local_error_thresh; /*!< When expanding the IS region to more SFBs only accept an error that is - not more than 'local_error_thresh' for the current SFB. */ - - FIXP_DBL direction_deviation_thresh; /*!< The maximum allowed change of the intensity direction (unit: IS scale) */ - - FIXP_DBL is_region_min_loudness; /*!< IS regions need to have a minimal percentage of the overall loudness, e.g. 0.06 == 6% */ - - INT min_is_sfbs; /*!< Only perform IS if 'min_is_sfbs' neighboring SFBs can be processed */ - - FIXP_DBL left_right_ratio_threshold; /*!< No IS if the panning angle is not far from the middle, MS will do */ - -} INTENSITY_PARAMETERS; - - -/***************************************************************************** - - functionname: calcSfbMaxScale - - description: Calc max value in scalefactor band - - input: *mdctSpectrum - l1 - l2 - - output: none - - returns: scalefactor - -*****************************************************************************/ -static INT -calcSfbMaxScale(const FIXP_DBL *mdctSpectrum, - const INT l1, - const INT l2) -{ - INT i; - INT sfbMaxScale; - FIXP_DBL maxSpc; - - maxSpc = FL2FXCONST_DBL(0.0); - for (i=l1; icorr_thresh = IS_CORR_THRESH; - isParams->total_error_thresh = IS_TOTAL_ERROR_THRESH; - isParams->local_error_thresh = IS_LOCAL_ERROR_THRESH; - isParams->direction_deviation_thresh = IS_DIRECTION_DEVIATION_THRESH; - isParams->is_region_min_loudness = IS_REGION_MIN_LOUDNESS; - isParams->min_is_sfbs = IS_MIN_SFBS; - isParams->left_right_ratio_threshold = IS_LEFT_RIGHT_RATIO_THRESH; -} - - -/***************************************************************************** - - functionname: FDKaacEnc_prepareIntensityDecision - - description: Prepares intensity decision - - input: sfbEnergyLeft - sfbEnergyRight - sfbEnergyLdDataLeft - sfbEnergyLdDataRight - mdctSpectrumLeft - sfbEnergyLdDataRight - isParams - - output: hrrErr scale: none - isMask scale: none - realScale scale: LD_DATA_SHIFT + REAL_SCALE_SF - normSfbLoudness scale: none - - returns: none - -*****************************************************************************/ -static void -FDKaacEnc_prepareIntensityDecision(const FIXP_DBL *sfbEnergyLeft, - const FIXP_DBL *sfbEnergyRight, - const FIXP_DBL *sfbEnergyLdDataLeft, - const FIXP_DBL *sfbEnergyLdDataRight, - const FIXP_DBL *mdctSpectrumLeft, - const FIXP_DBL *mdctSpectrumRight, - const INTENSITY_PARAMETERS *isParams, - FIXP_DBL *hrrErr, - INT *isMask, - FIXP_DBL *realScale, - FIXP_DBL *normSfbLoudness, - const INT sfbCnt, - const INT sfbPerGroup, - const INT maxSfbPerGroup, - const INT *sfbOffset) -{ - INT j,sfb,sfboffs; - INT grpCounter; - - /* temporary variables to compute loudness */ - FIXP_DBL overallLoudness[MAX_NO_OF_GROUPS]; - - /* temporary variables to compute correlation */ - FIXP_DBL channelCorr[MAX_GROUPED_SFB]; - FIXP_DBL ml, mr; - FIXP_DBL prod_lr; - FIXP_DBL square_l, square_r; - FIXP_DBL tmp_l, tmp_r; - FIXP_DBL inv_n; - - FDKmemclear(channelCorr, MAX_GROUPED_SFB*sizeof(FIXP_DBL)); - FDKmemclear(normSfbLoudness, MAX_GROUPED_SFB*sizeof(FIXP_DBL)); - FDKmemclear(overallLoudness, MAX_NO_OF_GROUPS*sizeof(FIXP_DBL)); - FDKmemclear(realScale, MAX_GROUPED_SFB*sizeof(FIXP_DBL)); - - for (grpCounter = 0, sfboffs = 0; sfboffs < sfbCnt; sfboffs += sfbPerGroup, grpCounter++) { - overallLoudness[grpCounter] = FL2FXCONST_DBL(0.0f); - for (sfb = 0; sfb < maxSfbPerGroup; sfb++) { - INT sL,sR,s; - FIXP_DBL isValue = sfbEnergyLdDataLeft[sfb+sfboffs]-sfbEnergyLdDataRight[sfb+sfboffs]; - - /* delimitate intensity scale value to representable range */ - realScale[sfb + sfboffs] = fixMin(FL2FXCONST_DBL(60.f/(1<<(REAL_SCALE_SF+LD_DATA_SHIFT))), fixMax(FL2FXCONST_DBL(-60.f/(1<<(REAL_SCALE_SF+LD_DATA_SHIFT))), isValue)); - - sL = fixMax(0,(CntLeadingZeros(sfbEnergyLeft[sfb + sfboffs])-1)); - sR = fixMax(0,(CntLeadingZeros(sfbEnergyRight[sfb + sfboffs])-1)); - s = (fixMin(sL,sR)>>2)<<2; - normSfbLoudness[sfb + sfboffs] = sqrtFixp(sqrtFixp(((sfbEnergyLeft[sfb + sfboffs]<> 1) + ((sfbEnergyRight[sfb + sfboffs]<> 1))) >> (s>>2); - - overallLoudness[grpCounter] += normSfbLoudness[sfb + sfboffs] >> OVERALL_LOUDNESS_SF; - /* don't do intensity if - * - panning angle is too close to the middle or - * - one channel is non-existent or - * - if it is dual mono */ - if( (sfbEnergyLeft[sfb + sfboffs] >= fMult(isParams->left_right_ratio_threshold,sfbEnergyRight[sfb + sfboffs])) - && (fMult(isParams->left_right_ratio_threshold,sfbEnergyLeft[sfb + sfboffs]) <= sfbEnergyRight[sfb + sfboffs]) ) { - - /* this will prevent post processing from considering this SFB for merging */ - hrrErr[sfb + sfboffs] = FL2FXCONST_DBL(1.0/8.0); - } - } - } - - for (grpCounter = 0, sfboffs = 0; sfboffs < sfbCnt; sfboffs += sfbPerGroup, grpCounter++) { - INT invOverallLoudnessSF; - FIXP_DBL invOverallLoudness; - - if (overallLoudness[grpCounter] == FL2FXCONST_DBL(0.0)) { - invOverallLoudness = FL2FXCONST_DBL(0.0); - invOverallLoudnessSF = 0; - } - else { - invOverallLoudness = fDivNorm((FIXP_DBL)MAXVAL_DBL, overallLoudness[grpCounter],&invOverallLoudnessSF); - invOverallLoudnessSF = invOverallLoudnessSF - OVERALL_LOUDNESS_SF + 1; /* +1: compensate fMultDiv2() in subsequent loop */ - } - invOverallLoudnessSF = fixMin(fixMax(invOverallLoudnessSF,-(DFRACT_BITS-1)),DFRACT_BITS-1); - - for (sfb = 0; sfb < maxSfbPerGroup; sfb++) { - FIXP_DBL tmp; - - tmp = fMultDiv2((normSfbLoudness[sfb + sfboffs]>>OVERALL_LOUDNESS_SF)<>1); - - if (inv_n > FL2FXCONST_DBL(0.0f)) { - INT s,sL,sR; - - /* correlation := Pearson's product-moment coefficient */ - /* compute correlation between channels and check if it is over threshold */ - ml = FL2FXCONST_DBL(0.0f); - mr = FL2FXCONST_DBL(0.0f); - prod_lr = FL2FXCONST_DBL(0.0f); - square_l = FL2FXCONST_DBL(0.0f); - square_r = FL2FXCONST_DBL(0.0f); - - sL = calcSfbMaxScale(mdctSpectrumLeft,sfbOffset[sfb+sfboffs],sfbOffset[sfb+sfboffs+1]); - sR = calcSfbMaxScale(mdctSpectrumRight,sfbOffset[sfb+sfboffs],sfbOffset[sfb+sfboffs+1]); - s = fixMin(sL,sR); - - for (j = sfbOffset[sfb + sfboffs]; j < sfbOffset[sfb + sfboffs + 1]; j++) { - ml += fMultDiv2((mdctSpectrumLeft[j] << s),inv_n); // scaled with mdctScale - s + inv_n - mr += fMultDiv2((mdctSpectrumRight[j] << s),inv_n); // scaled with mdctScale - s + inv_n - } - ml = fMultDiv2(ml,inv_n); // scaled with mdctScale - s + inv_n - mr = fMultDiv2(mr,inv_n); // scaled with mdctScale - s + inv_n - - for (j = sfbOffset[sfb + sfboffs]; j < sfbOffset[sfb + sfboffs + 1]; j++) { - tmp_l = fMultDiv2((mdctSpectrumLeft[j] << s),inv_n) - ml; // scaled with mdctScale - s + inv_n - tmp_r = fMultDiv2((mdctSpectrumRight[j] << s),inv_n) - mr; // scaled with mdctScale - s + inv_n - - prod_lr += fMultDiv2(tmp_l,tmp_r); // scaled with 2*(mdctScale - s + inv_n) + 1 - square_l += fPow2Div2(tmp_l); // scaled with 2*(mdctScale - s + inv_n) + 1 - square_r += fPow2Div2(tmp_r); // scaled with 2*(mdctScale - s + inv_n) + 1 - } - prod_lr = prod_lr << 1; // scaled with 2*(mdctScale - s + inv_n) - square_l = square_l << 1; // scaled with 2*(mdctScale - s + inv_n) - square_r = square_r << 1; // scaled with 2*(mdctScale - s + inv_n) - - if (square_l > FL2FXCONST_DBL(0.0f) && square_r > FL2FXCONST_DBL(0.0f)) { - INT channelCorrSF = 0; - - /* local scaling of square_l and square_r is compensated after sqrt calculation */ - sL = fixMax(0,(CntLeadingZeros(square_l)-1)); - sR = fixMax(0,(CntLeadingZeros(square_r)-1)); - s = ((sL + sR)>>1)<<1; - sL = fixMin(sL,s); - sR = s-sL; - tmp = fMult(square_l< FL2FXCONST_DBL(0.0f)); - - /* numerator and denominator have the same scaling */ - if (prod_lr < FL2FXCONST_DBL(0.0f) ) { - channelCorr[sfb + sfboffs] = -(fDivNorm(-prod_lr,tmp,&channelCorrSF)); - - } - else { - channelCorr[sfb + sfboffs] = (fDivNorm( prod_lr,tmp,&channelCorrSF)); - } - channelCorrSF = fixMin(fixMax(( channelCorrSF + ((sL+sR)>>1)),-(DFRACT_BITS-1)),DFRACT_BITS-1); - - if (channelCorrSF < 0) { - channelCorr[sfb + sfboffs] = channelCorr[sfb + sfboffs] >> (-channelCorrSF); - } - else { - /* avoid overflows due to limited computational accuracy */ - if ( fAbs(channelCorr[sfb + sfboffs]) > (((FIXP_DBL)MAXVAL_DBL)>>channelCorrSF) ) { - if (channelCorr[sfb + sfboffs] < FL2FXCONST_DBL(0.0f)) - channelCorr[sfb + sfboffs] = -(FIXP_DBL) MAXVAL_DBL; - else - channelCorr[sfb + sfboffs] = (FIXP_DBL) MAXVAL_DBL; - } - else { - channelCorr[sfb + sfboffs] = channelCorr[sfb + sfboffs] << channelCorrSF; - } - } - } - } - - /* for post processing: hrrErr is the error in terms of (too little) correlation - * weighted with the loudness of the SFB; SFBs with small hrrErr can be merged */ - if (hrrErr[sfb + sfboffs] == FL2FXCONST_DBL(1.0/8.0)) { - continue; - } - - hrrErr[sfb + sfboffs] = fMultDiv2((FL2FXCONST_DBL(0.25f)-(channelCorr[sfb + sfboffs]>>2)),normSfbLoudness[sfb + sfboffs]); - - /* set IS mask/vector to 1, if correlation is high enough */ - if (fAbs(channelCorr[sfb + sfboffs]) >= isParams->corr_thresh) { - isMask[sfb + sfboffs] = 1; - } - } - } -} - - -/***************************************************************************** - - functionname: FDKaacEnc_finalizeIntensityDecision - - description: Finalizes intensity decision - - input: isParams scale: none - hrrErr scale: none - realIsScale scale: LD_DATA_SHIFT + REAL_SCALE_SF - normSfbLoudness scale: none - - output: isMask scale: none - - returns: none - -*****************************************************************************/ -static void -FDKaacEnc_finalizeIntensityDecision(const FIXP_DBL *hrrErr, - INT *isMask, - const FIXP_DBL *realIsScale, - const FIXP_DBL *normSfbLoudness, - const INTENSITY_PARAMETERS *isParams, - const INT sfbCnt, - const INT sfbPerGroup, - const INT maxSfbPerGroup) -{ - INT sfb,sfboffs, j; - FIXP_DBL isScaleLast = FL2FXCONST_DBL(0.0f); - INT isStartValueFound = 0; - - for (sfboffs = 0; sfboffs < sfbCnt; sfboffs += sfbPerGroup) { - INT startIsSfb = 0; - INT inIsBlock = 0; - INT currentIsSfbCount = 0; - FIXP_DBL overallHrrError = FL2FXCONST_DBL(0.0f); - FIXP_DBL isRegionLoudness = FL2FXCONST_DBL(0.0f); - - for (sfb = 0; sfb < maxSfbPerGroup; sfb++) { - if (isMask[sfboffs + sfb] == 1) { - if (currentIsSfbCount == 0) { - startIsSfb = sfboffs + sfb; - } - if (isStartValueFound==0) { - isScaleLast = realIsScale[sfboffs + sfb]; - isStartValueFound = 1; - } - inIsBlock = 1; - currentIsSfbCount++; - overallHrrError += hrrErr[sfboffs + sfb] >> (MAX_SFB_PER_GROUP_SF-3); - isRegionLoudness += normSfbLoudness[sfboffs + sfb] >> MAX_SFB_PER_GROUP_SF; - } - else { - /* based on correlation, IS should not be used - * -> use it anyway, if overall error is below threshold - * and if local error does not exceed threshold - * otherwise: check if there are enough IS SFBs - */ - if (inIsBlock) { - overallHrrError += hrrErr[sfboffs + sfb] >> (MAX_SFB_PER_GROUP_SF-3); - isRegionLoudness += normSfbLoudness[sfboffs + sfb] >> MAX_SFB_PER_GROUP_SF; - - if ( (hrrErr[sfboffs + sfb] < (isParams->local_error_thresh>>3)) && (overallHrrError < (isParams->total_error_thresh>>MAX_SFB_PER_GROUP_SF)) ) { - currentIsSfbCount++; - /* overwrite correlation based decision */ - isMask[sfboffs + sfb] = 1; - } else { - inIsBlock = 0; - } - } - } - /* check for large direction deviation */ - if (inIsBlock) { - if( fAbs(isScaleLast-realIsScale[sfboffs + sfb]) < (isParams->direction_deviation_thresh>>(REAL_SCALE_SF+LD_DATA_SHIFT-IS_DIRECTION_DEVIATION_THRESH_SF)) ) { - isScaleLast = realIsScale[sfboffs + sfb]; - } - else{ - isMask[sfboffs + sfb] = 0; - inIsBlock = 0; - currentIsSfbCount--; - } - } - - if (currentIsSfbCount > 0 && (!inIsBlock || sfb == maxSfbPerGroup - 1)) { - /* not enough SFBs -> do not use IS */ - if (currentIsSfbCount < isParams->min_is_sfbs || (isRegionLoudness < isParams->is_region_min_loudness>>MAX_SFB_PER_GROUP_SF)) { - for(j = startIsSfb; j <= sfboffs + sfb; j++) { - isMask[j] = 0; - } - isScaleLast = FL2FXCONST_DBL(0.0f); - isStartValueFound = 0; - for (j=0; j < startIsSfb; j++) { - if (isMask[j]!=0) { - isScaleLast = realIsScale[j]; - isStartValueFound = 1; - } - } - } - currentIsSfbCount = 0; - overallHrrError = FL2FXCONST_DBL(0.0f); - isRegionLoudness = FL2FXCONST_DBL(0.0f); - } - } - } -} - - -/***************************************************************************** - - functionname: FDKaacEnc_IntensityStereoProcessing - - description: Intensity stereo processing tool - - input: sfbEnergyLeft - sfbEnergyRight - mdctSpectrumLeft - mdctSpectrumRight - sfbThresholdLeft - sfbThresholdRight - sfbSpreadEnLeft - sfbSpreadEnRight - sfbEnergyLdDataLeft - sfbEnergyLdDataRight - - output: isBook - isScale - pnsData->pnsFlag - msDigest zeroed from start to sfbCnt - msMask zeroed from start to sfbCnt - mdctSpectrumRight zeroed where isBook!=0 - sfbEnergyRight zeroed where isBook!=0 - sfbSpreadEnRight zeroed where isBook!=0 - sfbThresholdRight zeroed where isBook!=0 - sfbEnergyLdDataRight FL2FXCONST_DBL(-1.0) where isBook!=0 - sfbThresholdLdDataRight FL2FXCONST_DBL(-0.515625f) where isBook!=0 - - returns: none - -*****************************************************************************/ -void FDKaacEnc_IntensityStereoProcessing( - FIXP_DBL *sfbEnergyLeft, - FIXP_DBL *sfbEnergyRight, - FIXP_DBL *mdctSpectrumLeft, - FIXP_DBL *mdctSpectrumRight, - FIXP_DBL *sfbThresholdLeft, - FIXP_DBL *sfbThresholdRight, - FIXP_DBL *sfbThresholdLdDataRight, - FIXP_DBL *sfbSpreadEnLeft, - FIXP_DBL *sfbSpreadEnRight, - FIXP_DBL *sfbEnergyLdDataLeft, - FIXP_DBL *sfbEnergyLdDataRight, - INT *msDigest, - INT *msMask, - const INT sfbCnt, - const INT sfbPerGroup, - const INT maxSfbPerGroup, - const INT *sfbOffset, - const INT allowIS, - INT *isBook, - INT *isScale, - PNS_DATA *RESTRICT pnsData[2] - ) -{ - INT sfb,sfboffs, j; - FIXP_DBL scale; - FIXP_DBL lr; - FIXP_DBL hrrErr[MAX_GROUPED_SFB]; - FIXP_DBL normSfbLoudness[MAX_GROUPED_SFB]; - FIXP_DBL realIsScale[MAX_GROUPED_SFB]; - INTENSITY_PARAMETERS isParams; - INT isMask[MAX_GROUPED_SFB]; - - FDKmemclear((void*)isBook,sfbCnt*sizeof(INT)); - FDKmemclear((void*)isMask,sfbCnt*sizeof(INT)); - FDKmemclear((void*)realIsScale,sfbCnt*sizeof(FIXP_DBL)); - FDKmemclear((void*)isScale,sfbCnt*sizeof(INT)); - FDKmemclear((void*)hrrErr,sfbCnt*sizeof(FIXP_DBL)); - - if (!allowIS) - return; - - FDKaacEnc_initIsParams(&isParams); - - /* compute / set the following values per SFB: - * - left/right ratio between channels - * - normalized loudness - * + loudness == average of energy in channels to 0.25 - * + normalization: division by sum of all SFB loudnesses - * - isMask (is set to 0 if channels are the same or one is 0) - */ - FDKaacEnc_prepareIntensityDecision(sfbEnergyLeft, - sfbEnergyRight, - sfbEnergyLdDataLeft, - sfbEnergyLdDataRight, - mdctSpectrumLeft, - mdctSpectrumRight, - &isParams, - hrrErr, - isMask, - realIsScale, - normSfbLoudness, - sfbCnt, - sfbPerGroup, - maxSfbPerGroup, - sfbOffset); - - FDKaacEnc_finalizeIntensityDecision(hrrErr, - isMask, - realIsScale, - normSfbLoudness, - &isParams, - sfbCnt, - sfbPerGroup, - maxSfbPerGroup); - - for (sfb=0; sfb sfbThresholdRight[sfb+sfboffs]) ) { - continue; - } - /* NEW: if there is a big-enough IS region, switch off PNS */ - if (pnsData[0]) { - if(pnsData[0]->pnsFlag[sfb+sfboffs]) { - pnsData[0]->pnsFlag[sfb+sfboffs] = 0; - } - if(pnsData[1]->pnsFlag[sfb+sfboffs]) { - pnsData[1]->pnsFlag[sfb+sfboffs] = 0; - } - } - - inv_n = GetInvInt((sfbOffset[sfb + sfboffs + 1] - sfbOffset[sfb + sfboffs])>>1); // scaled with 2 to compensate fMultDiv2() in subsequent loop - sL = calcSfbMaxScale(mdctSpectrumLeft,sfbOffset[sfb+sfboffs],sfbOffset[sfb+sfboffs+1]); - sR = calcSfbMaxScale(mdctSpectrumRight,sfbOffset[sfb+sfboffs],sfbOffset[sfb+sfboffs+1]); - - lr = FL2FXCONST_DBL(0.0f); - for (j=sfbOffset[sfb+sfboffs]; j>1) - ((mdctSpectrumRight[j]<>1); - ed += fMultDiv2(d,d)>>(MDCT_SPEC_SF-1); - } - msMask[sfb+sfboffs] = 1; - tmp = fDivNorm(sfbEnergyLeft[sfb+sfboffs],ed,&s1); - s2 = (s1) + (2*s0) - 2 - MDCT_SPEC_SF; - if (s2 & 1) { - tmp = tmp>>1; - s2 = s2+1; - } - s2 = (s2>>1) + 1; // +1 compensate fMultDiv2() in subsequent loop - s2 = fixMin(fixMax(s2,-(DFRACT_BITS-1)),(DFRACT_BITS-1)); - scale = sqrtFixp(tmp); - if (s2 < 0) { - s2 = -s2; - for (j=sfbOffset[sfb+sfboffs]; j> s2; - mdctSpectrumRight[j] = FL2FXCONST_DBL(0.0f); - } - } - else { - for (j=sfbOffset[sfb+sfboffs]; j>1) + ((mdctSpectrumRight[j]<>1); - es += fMultDiv2(s,s)>>(MDCT_SPEC_SF-1); // scaled 2*(mdctScale - s0 + 1) + MDCT_SPEC_SF - } - msMask[sfb+sfboffs] = 0; - tmp = fDivNorm(sfbEnergyLeft[sfb+sfboffs],es,&s1); - s2 = (s1) + (2*s0) - 2 - MDCT_SPEC_SF; - if (s2 & 1) { - tmp = tmp>>1; - s2 = s2 + 1; - } - s2 = (s2>>1) + 1; // +1 compensate fMultDiv2() in subsequent loop - s2 = fixMin(fixMax(s2,-(DFRACT_BITS-1)),(DFRACT_BITS-1)); - scale = sqrtFixp(tmp); - if (s2 < 0) { - s2 = -s2; - for (j=sfbOffset[sfb+sfboffs]; j> s2; - mdctSpectrumRight[j] = FL2FXCONST_DBL(0.0f); - } - } - else { - for (j=sfbOffset[sfb+sfboffs]; j>1)-FL2FXCONST_DBL(0.5f/(1<<(REAL_SCALE_SF+LD_DATA_SHIFT+1))))>>(DFRACT_BITS-1-REAL_SCALE_SF-LD_DATA_SHIFT-1)) + 1; - } - else { - isScale[sfb+sfboffs] = (INT)(((realIsScale[sfb+sfboffs]>>1)+FL2FXCONST_DBL(0.5f/(1<<(REAL_SCALE_SF+LD_DATA_SHIFT+1))))>>(DFRACT_BITS-1-REAL_SCALE_SF-LD_DATA_SHIFT-1)); - } - - sfbEnergyRight[sfb+sfboffs] = FL2FXCONST_DBL(0.0f); - sfbEnergyLdDataRight[sfb+sfboffs] = FL2FXCONST_DBL(-1.0f); - sfbThresholdRight[sfb+sfboffs] = FL2FXCONST_DBL(0.0f); - sfbThresholdLdDataRight[sfb+sfboffs] = FL2FXCONST_DBL(-0.515625f); - sfbSpreadEnRight[sfb+sfboffs] = FL2FXCONST_DBL(0.0f); - - *msDigest = MS_SOME; - } - } -} - diff --git a/libAACenc/src/intensity.h b/libAACenc/src/intensity.h deleted file mode 100644 index 2acc292..0000000 --- a/libAACenc/src/intensity.h +++ /dev/null @@ -1,122 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: A. Horndasch (code originally from lwr and rtb) / Josef Höpfl (FDK) - contents/description: intensity stereo prototype - -******************************************************************************/ - -#ifndef _INTENSITY_H -#define _INTENSITY_H - -#include "aacenc_pns.h" - - -void FDKaacEnc_IntensityStereoProcessing( - FIXP_DBL *sfbEnergyLeft, - FIXP_DBL *sfbEnergyRight, - FIXP_DBL *mdctSpectrumLeft, - FIXP_DBL *mdctSpectrumRight, - FIXP_DBL *sfbThresholdLeft, - FIXP_DBL *sfbThresholdRight, - FIXP_DBL *sfbThresholdLdDataRight, - FIXP_DBL *sfbSpreadEnLeft, - FIXP_DBL *sfbSpreadEnRight, - FIXP_DBL *sfbEnergyLdDataLeft, - FIXP_DBL *sfbEnergyLdDataRight, - INT *msDigest, - INT *msMask, - const INT sfbCnt, - const INT sfbPerGroup, - const INT maxSfbPerGroup, - const INT *sfbOffset, - const INT allowIS, - INT *isBook, - INT *isScale, - PNS_DATA *RESTRICT pnsData[2] - ); - -#endif /* _INTENSITY_H */ - diff --git a/libAACenc/src/interface.h b/libAACenc/src/interface.h deleted file mode 100644 index 51fb72a..0000000 --- a/libAACenc/src/interface.h +++ /dev/null @@ -1,169 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M.Werner - contents/description: Interface psychoaccoustic/quantizer - -******************************************************************************/ - -#ifndef _INTERFACE_H -#define _INTERFACE_H - -#include "common_fix.h" -#include "FDK_audio.h" - -#include "psy_data.h" -#include "aacenc_tns.h" - -enum -{ - MS_NONE = 0, - MS_SOME = 1, - MS_ALL = 2 -}; - -enum -{ - MS_ON = 1 -}; - -struct TOOLSINFO { - INT msDigest; /* 0 = no MS; 1 = some MS, 2 = all MS */ - INT msMask[MAX_GROUPED_SFB]; -}; - - -typedef struct { - INT sfbCnt; - INT sfbPerGroup; - INT maxSfbPerGroup; - INT lastWindowSequence; - INT windowShape; - INT groupingMask; - INT sfbOffsets[MAX_GROUPED_SFB+1]; - - INT mdctScale; /* number of transform shifts */ - INT groupLen[MAX_NO_OF_GROUPS]; - - TNS_INFO tnsInfo; - INT noiseNrg[MAX_GROUPED_SFB]; - INT isBook[MAX_GROUPED_SFB]; - INT isScale[MAX_GROUPED_SFB]; - - /* memory located in QC_OUT_CHANNEL */ - FIXP_DBL *mdctSpectrum; - FIXP_DBL *sfbEnergy; - FIXP_DBL *sfbSpreadEnergy; - FIXP_DBL *sfbThresholdLdData; - FIXP_DBL *sfbMinSnrLdData; - FIXP_DBL *sfbEnergyLdData; - - - }PSY_OUT_CHANNEL; - -typedef struct { - - /* information specific to each channel */ - PSY_OUT_CHANNEL* psyOutChannel[(2)]; - - /* information shared by both channels */ - INT commonWindow; - struct TOOLSINFO toolsInfo; - -} PSY_OUT_ELEMENT; - -typedef struct { - - PSY_OUT_ELEMENT* psyOutElement[(8)]; - PSY_OUT_CHANNEL* pPsyOutChannels[(8)]; - -}PSY_OUT; - -inline int isLowDelay( AUDIO_OBJECT_TYPE aot ) -{ - return (aot==AOT_ER_AAC_LD || aot==AOT_ER_AAC_ELD); -} - -#endif /* _INTERFACE_H */ diff --git a/libAACenc/src/line_pe.cpp b/libAACenc/src/line_pe.cpp deleted file mode 100644 index f3c0dab..0000000 --- a/libAACenc/src/line_pe.cpp +++ /dev/null @@ -1,209 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M. Werner - contents/description: Perceptual entropie module - -******************************************************************************/ - -#include "line_pe.h" -#include "sf_estim.h" -#include "bit_cnt.h" - -#include "genericStds.h" - -static const FIXP_DBL C1LdData = FL2FXCONST_DBL(3.0/LD_DATA_SCALING); /* C1 = 3.0 = log(8.0)/log(2) */ -static const FIXP_DBL C2LdData = FL2FXCONST_DBL(1.3219281/LD_DATA_SCALING); /* C2 = 1.3219281 = log(2.5)/log(2) */ -static const FIXP_DBL C3LdData = FL2FXCONST_DBL(0.5593573); /* 1-C2/C1 */ - - -/* constants that do not change during successive pe calculations */ -void FDKaacEnc_prepareSfbPe(PE_CHANNEL_DATA *peChanData, - const FIXP_DBL *sfbEnergyLdData, - const FIXP_DBL *sfbThresholdLdData, - const FIXP_DBL *sfbFormFactorLdData, - const INT *sfbOffset, - const INT sfbCnt, - const INT sfbPerGroup, - const INT maxSfbPerGroup) -{ - INT sfbGrp,sfb; - INT sfbWidth; - FIXP_DBL avgFormFactorLdData; - const FIXP_DBL formFacScaling = FL2FXCONST_DBL((float)FORM_FAC_SHIFT/LD_DATA_SCALING); - - for (sfbGrp = 0;sfbGrp < sfbCnt;sfbGrp+=sfbPerGroup) { - for (sfb=0; sfb (FIXP_DBL)sfbThresholdLdData[sfbGrp+sfb]) { - sfbWidth = sfbOffset[sfbGrp+sfb+1] - sfbOffset[sfbGrp+sfb]; - /* estimate number of active lines */ - avgFormFactorLdData = ((-sfbEnergyLdData[sfbGrp+sfb]>>1) + (CalcLdInt(sfbWidth)>>1))>>1; - peChanData->sfbNLines[sfbGrp+sfb] = - (INT)CalcInvLdData( (sfbFormFactorLdData[sfbGrp+sfb] + formFacScaling) + avgFormFactorLdData); - /* Make sure sfbNLines is never greater than sfbWidth due to unaccuracies (e.g. sfbEnergyLdData[sfbGrp+sfb] = 0x80000000) */ - peChanData->sfbNLines[sfbGrp+sfb] = fMin(sfbWidth, peChanData->sfbNLines[sfbGrp+sfb]); - } - else { - peChanData->sfbNLines[sfbGrp+sfb] = 0; - } - } - } -} - -/* - formula for one sfb: - pe = n * ld(en/thr), if ld(en/thr) >= C1 - pe = n * (C2 + C3 * ld(en/thr)), if ld(en/thr) < C1 - n: estimated number of lines in sfb, - ld(x) = log(x)/log(2) - - constPart is sfbPe without the threshold part n*ld(thr) or n*C3*ld(thr) -*/ -void FDKaacEnc_calcSfbPe(PE_CHANNEL_DATA *RESTRICT peChanData, - const FIXP_DBL *RESTRICT sfbEnergyLdData, - const FIXP_DBL *RESTRICT sfbThresholdLdData, - const INT sfbCnt, - const INT sfbPerGroup, - const INT maxSfbPerGroup, - const INT *isBook, - const INT *isScale) -{ - INT sfbGrp,sfb; - INT nLines; - FIXP_DBL logDataRatio; - INT lastValIs = 0; - - peChanData->pe = 0; - peChanData->constPart = 0; - peChanData->nActiveLines = 0; - - for(sfbGrp = 0;sfbGrp < sfbCnt;sfbGrp+=sfbPerGroup){ - for (sfb=0; sfb (FIXP_DBL)sfbThresholdLdData[sfbGrp+sfb]) { - logDataRatio = (FIXP_DBL)(sfbEnergyLdData[sfbGrp+sfb] - sfbThresholdLdData[sfbGrp+sfb]); - nLines = peChanData->sfbNLines[sfbGrp+sfb]; - if (logDataRatio >= C1LdData) { - /* scale sfbPe and sfbConstPart with PE_CONSTPART_SHIFT */ - peChanData->sfbPe[sfbGrp+sfb] = fMultDiv2(logDataRatio, (FIXP_DBL)(nLines<<(LD_DATA_SHIFT+PE_CONSTPART_SHIFT+1))); - peChanData->sfbConstPart[sfbGrp+sfb] = - fMultDiv2(sfbEnergyLdData[sfbGrp+sfb], (FIXP_DBL)(nLines<<(LD_DATA_SHIFT+PE_CONSTPART_SHIFT+1))); ; - - } - else { - /* scale sfbPe and sfbConstPart with PE_CONSTPART_SHIFT */ - peChanData->sfbPe[sfbGrp+sfb] = - fMultDiv2(((FIXP_DBL)C2LdData + fMult(C3LdData,logDataRatio)), (FIXP_DBL)(nLines<<(LD_DATA_SHIFT+PE_CONSTPART_SHIFT+1))); - - peChanData->sfbConstPart[sfbGrp+sfb] = - fMultDiv2(((FIXP_DBL)C2LdData + fMult(C3LdData,sfbEnergyLdData[sfbGrp+sfb])), - (FIXP_DBL)(nLines<<(LD_DATA_SHIFT+PE_CONSTPART_SHIFT+1))) ; - - nLines = fMultI(C3LdData, nLines); - } - peChanData->sfbNActiveLines[sfbGrp+sfb] = nLines; - } - else if( isBook[sfbGrp+sfb] ) { - /* provide for cost of scale factor for Intensity */ - INT delta = isScale[sfbGrp+sfb] - lastValIs; - lastValIs = isScale[sfbGrp+sfb]; - peChanData->sfbPe[sfbGrp+sfb] = FDKaacEnc_bitCountScalefactorDelta(delta)<sfbConstPart[sfbGrp+sfb] = 0; - peChanData->sfbNActiveLines[sfbGrp+sfb] = 0; - } - else { - peChanData->sfbPe[sfbGrp+sfb] = 0; - peChanData->sfbConstPart[sfbGrp+sfb] = 0; - peChanData->sfbNActiveLines[sfbGrp+sfb] = 0; - } - /* sum up peChanData values */ - peChanData->pe += peChanData->sfbPe[sfbGrp+sfb]; - peChanData->constPart += peChanData->sfbConstPart[sfbGrp+sfb]; - peChanData->nActiveLines += peChanData->sfbNActiveLines[sfbGrp+sfb]; - } - } - /* correct scaled pe and constPart values */ - peChanData->pe>>=PE_CONSTPART_SHIFT; - peChanData->constPart>>=PE_CONSTPART_SHIFT; -} diff --git a/libAACenc/src/line_pe.h b/libAACenc/src/line_pe.h deleted file mode 100644 index 3d5cfd5..0000000 --- a/libAACenc/src/line_pe.h +++ /dev/null @@ -1,139 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M. Werner - contents/description: Perceptual entropie module - -******************************************************************************/ -#ifndef __LINE_PE_H -#define __LINE_PE_H - - -#include "common_fix.h" - -#include "psy_const.h" - -#define PE_CONSTPART_SHIFT FRACT_BITS - -typedef struct { - /* calculated by FDKaacEnc_prepareSfbPe */ - INT sfbNLines[MAX_GROUPED_SFB]; /* number of relevant lines in sfb */ - /* the rest is calculated by FDKaacEnc_calcSfbPe */ - INT sfbPe[MAX_GROUPED_SFB]; /* pe for each sfb */ - INT sfbConstPart[MAX_GROUPED_SFB]; /* constant part for each sfb */ - INT sfbNActiveLines[MAX_GROUPED_SFB]; /* number of active lines in sfb */ - INT pe; /* sum of sfbPe */ - INT constPart; /* sum of sfbConstPart */ - INT nActiveLines; /* sum of sfbNActiveLines */ -} PE_CHANNEL_DATA; - -typedef struct { - PE_CHANNEL_DATA peChannelData[(2)]; - INT pe; - INT constPart; - INT nActiveLines; - INT offset; -} PE_DATA; - - -void FDKaacEnc_prepareSfbPe(PE_CHANNEL_DATA *peChanData, - const FIXP_DBL *sfbEnergyLdData, - const FIXP_DBL *sfbThresholdLdData, - const FIXP_DBL *sfbFormFactorLdData, - const INT *sfbOffset, - const INT sfbCnt, - const INT sfbPerGroup, - const INT maxSfbPerGroup); - -void FDKaacEnc_calcSfbPe(PE_CHANNEL_DATA *RESTRICT peChanData, - const FIXP_DBL *RESTRICT sfbEnergyLdData, - const FIXP_DBL *RESTRICT sfbThresholdLdData, - const INT sfbCnt, - const INT sfbPerGroup, - const INT maxSfbPerGroup, - const INT *isBook, - const INT *isScale); - -#endif diff --git a/libAACenc/src/metadata_compressor.cpp b/libAACenc/src/metadata_compressor.cpp deleted file mode 100644 index 68a64ae..0000000 --- a/libAACenc/src/metadata_compressor.cpp +++ /dev/null @@ -1,1038 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/********************** Fraunhofer IIS FDK AAC Encoder lib ****************** - - Author(s): M. Neusinger - Description: Compressor for AAC Metadata Generator - -******************************************************************************/ - - -#include "metadata_compressor.h" -#include "channel_map.h" - - -#define LOG2 0.69314718056f /* natural logarithm of 2 */ -#define ILOG2 1.442695041f /* 1/LOG2 */ -#define FIXP_ILOG2_DIV2 (FL2FXCONST_DBL(ILOG2/2)) - -/*----------------- defines ----------------------*/ - -#define MAX_DRC_CHANNELS (8) /*!< Max number of audio input channels. */ -#define DOWNMIX_SHIFT (3) /*!< Max 8 channel. */ -#define WEIGHTING_FILTER_SHIFT (2) /*!< Scaling used in weighting filter. */ - -#define METADATA_INT_BITS 10 -#define METADATA_LINT_BITS 20 -#define METADATA_INT_SCALE (INT64(1)<<(METADATA_INT_BITS)) -#define METADATA_FRACT_BITS (DFRACT_BITS-1-METADATA_INT_BITS) -#define METADATA_FRACT_SCALE (INT64(1)<<(METADATA_FRACT_BITS)) - -/** - * Enum for channel assignment. - */ -enum { - L = 0, - R = 1, - C = 2, - LFE = 3, - LS = 4, - RS = 5, - S = 6, - LS2 = 7, - RS2 = 8 -}; - -/*--------------- structure definitions --------------------*/ - -/** - * Structure holds weighting filter filter states. - */ -struct WEIGHTING_STATES { - FIXP_DBL x1; - FIXP_DBL x2; - FIXP_DBL y1; - FIXP_DBL y2; -}; - -/** - * Dynamic Range Control compressor structure. - */ -struct DRC_COMP { - - FIXP_DBL maxBoostThr[2]; /*!< Max boost threshold. */ - FIXP_DBL boostThr[2]; /*!< Boost threshold. */ - FIXP_DBL earlyCutThr[2]; /*!< Early cut threshold. */ - FIXP_DBL cutThr[2]; /*!< Cut threshold. */ - FIXP_DBL maxCutThr[2]; /*!< Max cut threshold. */ - - FIXP_DBL boostFac[2]; /*!< Precalculated factor for boost compression. */ - FIXP_DBL earlyCutFac[2]; /*!< Precalculated factor for early cut compression. */ - FIXP_DBL cutFac[2]; /*!< Precalculated factor for cut compression. */ - - FIXP_DBL maxBoost[2]; /*!< Maximum boost. */ - FIXP_DBL maxCut[2]; /*!< Maximum cut. */ - FIXP_DBL maxEarlyCut[2]; /*!< Maximum early cut. */ - - FIXP_DBL fastAttack[2]; /*!< Fast attack coefficient. */ - FIXP_DBL fastDecay[2]; /*!< Fast release coefficient. */ - FIXP_DBL slowAttack[2]; /*!< Slow attack coefficient. */ - FIXP_DBL slowDecay[2]; /*!< Slow release coefficient. */ - UINT holdOff[2]; /*!< Hold time in blocks. */ - - FIXP_DBL attackThr[2]; /*!< Slow/fast attack threshold. */ - FIXP_DBL decayThr[2]; /*!< Slow/fast release threshold. */ - - DRC_PROFILE profile[2]; /*!< DRC profile. */ - INT blockLength; /*!< Block length in samples. */ - UINT sampleRate; /*!< Sample rate. */ - CHANNEL_MODE chanConfig; /*!< Channel configuration. */ - - UCHAR useWeighting; /*!< Use weighting filter. */ - - UINT channels; /*!< Number of channels. */ - UINT fullChannels; /*!< Number of full range channels. */ - INT channelIdx[9]; /*!< Offsets of interleaved channel samples (L, R, C, LFE, Ls, Rs, S, Ls2, Rs2). */ - - FIXP_DBL smoothLevel[2]; /*!< level smoothing states */ - FIXP_DBL smoothGain[2]; /*!< gain smoothing states */ - UINT holdCnt[2]; /*!< hold counter */ - - FIXP_DBL limGain[2]; /*!< limiter gain */ - FIXP_DBL limDecay; /*!< limiter decay (linear) */ - FIXP_DBL prevPeak[2]; /*!< max peak of previous block (stereo/mono)*/ - - WEIGHTING_STATES filter[MAX_DRC_CHANNELS]; /*!< array holds weighting filter states */ - -}; - -/*---------------- constants -----------------------*/ - -/** - * Profile tables. - */ -static const FIXP_DBL tabMaxBoostThr[] = { - (FIXP_DBL)(int)((unsigned)-43<limDecay = FL2FXCONST_DBL( ((0.006f / 256) * blockLength) / METADATA_INT_SCALE ); - - /* Save parameters. */ - drcComp->blockLength = blockLength; - drcComp->sampleRate = sampleRate; - drcComp->chanConfig = channelMode; - drcComp->useWeighting = useWeighting; - - if (FDK_DRC_Generator_setDrcProfile(drcComp, profileLine, profileRF)!=0) { /* expects initialized blockLength and sampleRate */ - return (-1); - } - - /* Set number of channels and channel offsets. */ - if (FDKaacEnc_InitChannelMapping(channelMode, channelOrder, &channelMapping)!=AAC_ENC_OK) { - return (-2); - } - - for (i = 0; i < 9; i++) drcComp->channelIdx[i] = -1; - - switch (channelMode) { - case MODE_1: /* mono */ - drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0]; - break; - case MODE_2: /* stereo */ - drcComp->channelIdx[L] = channelMapping.elInfo[0].ChannelIndex[0]; - drcComp->channelIdx[R] = channelMapping.elInfo[0].ChannelIndex[1]; - break; - case MODE_1_2: /* 3ch */ - drcComp->channelIdx[L] = channelMapping.elInfo[1].ChannelIndex[0]; - drcComp->channelIdx[R] = channelMapping.elInfo[1].ChannelIndex[1]; - drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0]; - break; - case MODE_1_2_1: /* 4ch */ - drcComp->channelIdx[L] = channelMapping.elInfo[1].ChannelIndex[0]; - drcComp->channelIdx[R] = channelMapping.elInfo[1].ChannelIndex[1]; - drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0]; - drcComp->channelIdx[S] = channelMapping.elInfo[2].ChannelIndex[0]; - break; - case MODE_1_2_2: /* 5ch */ - drcComp->channelIdx[L] = channelMapping.elInfo[1].ChannelIndex[0]; - drcComp->channelIdx[R] = channelMapping.elInfo[1].ChannelIndex[1]; - drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0]; - drcComp->channelIdx[LS] = channelMapping.elInfo[2].ChannelIndex[0]; - drcComp->channelIdx[RS] = channelMapping.elInfo[2].ChannelIndex[1]; - break; - case MODE_1_2_2_1: /* 5.1 ch */ - drcComp->channelIdx[L] = channelMapping.elInfo[1].ChannelIndex[0]; - drcComp->channelIdx[R] = channelMapping.elInfo[1].ChannelIndex[1]; - drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0]; - drcComp->channelIdx[LFE] = channelMapping.elInfo[3].ChannelIndex[0]; - drcComp->channelIdx[LS] = channelMapping.elInfo[2].ChannelIndex[0]; - drcComp->channelIdx[RS] = channelMapping.elInfo[2].ChannelIndex[1]; - break; - case MODE_1_2_2_2_1: /* 7.1 ch */ - case MODE_7_1_FRONT_CENTER: - drcComp->channelIdx[L] = channelMapping.elInfo[2].ChannelIndex[0]; /* l */ - drcComp->channelIdx[R] = channelMapping.elInfo[2].ChannelIndex[1]; /* r */ - drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0]; /* c */ - drcComp->channelIdx[LFE] = channelMapping.elInfo[4].ChannelIndex[0]; /* lfe */ - drcComp->channelIdx[LS] = channelMapping.elInfo[3].ChannelIndex[0]; /* ls */ - drcComp->channelIdx[RS] = channelMapping.elInfo[3].ChannelIndex[1]; /* rs */ - drcComp->channelIdx[LS2] = channelMapping.elInfo[1].ChannelIndex[0]; /* lc */ - drcComp->channelIdx[RS2] = channelMapping.elInfo[1].ChannelIndex[1]; /* rc */ - break; - case MODE_7_1_REAR_SURROUND: - drcComp->channelIdx[L] = channelMapping.elInfo[1].ChannelIndex[0]; /* l */ - drcComp->channelIdx[R] = channelMapping.elInfo[1].ChannelIndex[1]; /* r */ - drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0]; /* c */ - drcComp->channelIdx[LFE] = channelMapping.elInfo[4].ChannelIndex[0]; /* lfe */ - drcComp->channelIdx[LS] = channelMapping.elInfo[3].ChannelIndex[0]; /* lrear */ - drcComp->channelIdx[RS] = channelMapping.elInfo[3].ChannelIndex[1]; /* rrear */ - drcComp->channelIdx[LS2] = channelMapping.elInfo[2].ChannelIndex[0]; /* ls */ - drcComp->channelIdx[RS2] = channelMapping.elInfo[2].ChannelIndex[1]; /* rs */ - break; - case MODE_1_1: - case MODE_1_1_1_1: - case MODE_1_1_1_1_1_1: - case MODE_1_1_1_1_1_1_1_1: - case MODE_1_1_1_1_1_1_1_1_1_1_1_1: - case MODE_2_2: - case MODE_2_2_2: - case MODE_2_2_2_2: - case MODE_2_2_2_2_2_2: - default: - return (-1); - } - - drcComp->fullChannels = channelMapping.nChannelsEff; - drcComp->channels = channelMapping.nChannels; - - /* Init states. */ - drcComp->smoothLevel[0] = drcComp->smoothLevel[1] = (FIXP_DBL)(int)((unsigned)-135<smoothGain, sizeof(drcComp->smoothGain)); - FDKmemclear(drcComp->holdCnt, sizeof(drcComp->holdCnt)); - FDKmemclear(drcComp->limGain, sizeof(drcComp->limGain)); - FDKmemclear(drcComp->prevPeak, sizeof(drcComp->prevPeak)); - FDKmemclear(drcComp->filter, sizeof(drcComp->filter)); - - return (0); -} - - -INT FDK_DRC_Generator_setDrcProfile( - HDRC_COMP drcComp, - const DRC_PROFILE profileLine, - const DRC_PROFILE profileRF - ) -{ - int profileIdx, i; - - drcComp->profile[0] = profileLine; - drcComp->profile[1] = profileRF; - - for (i = 0; i < 2; i++) { - /* get profile index */ - switch (drcComp->profile[i]) { - case DRC_NONE: - case DRC_FILMSTANDARD: profileIdx = 0; break; - case DRC_FILMLIGHT: profileIdx = 1; break; - case DRC_MUSICSTANDARD: profileIdx = 2; break; - case DRC_MUSICLIGHT: profileIdx = 3; break; - case DRC_SPEECH: profileIdx = 4; break; - case DRC_DELAY_TEST: profileIdx = 5; break; - default: return (-1); - } - - /* get parameters for selected profile */ - if (profileIdx >= 0) { - drcComp->maxBoostThr[i] = tabMaxBoostThr[profileIdx]; - drcComp->boostThr[i] = tabBoostThr[profileIdx]; - drcComp->earlyCutThr[i] = tabEarlyCutThr[profileIdx]; - drcComp->cutThr[i] = tabCutThr[profileIdx]; - drcComp->maxCutThr[i] = tabMaxCutThr[profileIdx]; - - drcComp->boostFac[i] = tabBoostRatio[profileIdx]; - drcComp->earlyCutFac[i] = tabEarlyCutRatio[profileIdx]; - drcComp->cutFac[i] = tabCutRatio[profileIdx]; - - drcComp->maxBoost[i] = tabMaxBoost[profileIdx]; - drcComp->maxCut[i] = tabMaxCut[profileIdx]; - drcComp->maxEarlyCut[i] = - fMult((drcComp->cutThr[i] - drcComp->earlyCutThr[i]), drcComp->earlyCutFac[i]); /* no scaling after mult needed, earlyCutFac is in FIXP_DBL */ - - drcComp->fastAttack[i] = tc2Coeff(tabFastAttack[profileIdx], drcComp->sampleRate, drcComp->blockLength); - drcComp->fastDecay[i] = tc2Coeff(tabFastDecay[profileIdx], drcComp->sampleRate, drcComp->blockLength); - drcComp->slowAttack[i] = tc2Coeff(tabSlowAttack[profileIdx], drcComp->sampleRate, drcComp->blockLength); - drcComp->slowDecay[i] = tc2Coeff(tabSlowDecay[profileIdx], drcComp->sampleRate, drcComp->blockLength); - drcComp->holdOff[i] = tabHoldOff[profileIdx] * 256 / drcComp->blockLength; - - drcComp->attackThr[i] = tabAttackThr[profileIdx]; - drcComp->decayThr[i] = tabDecayThr[profileIdx]; - } - - drcComp->smoothGain[i] = FL2FXCONST_DBL(0.f); - } - return (0); -} - - -INT FDK_DRC_Generator_Calc( - HDRC_COMP drcComp, - const INT_PCM * const inSamples, - const INT dialnorm, - const INT drc_TargetRefLevel, - const INT comp_TargetRefLevel, - FIXP_DBL clev, - FIXP_DBL slev, - INT * const pDynrng, - INT * const pCompr - ) -{ - int i, c; - FIXP_DBL peak[2]; - - - /************************************************************************** - * compressor - **************************************************************************/ - if ((drcComp->profile[0] != DRC_NONE) || (drcComp->profile[1] != DRC_NONE)) { - /* Calc loudness level */ - FIXP_DBL level_b = FL2FXCONST_DBL(0.f); - int level_e = DFRACT_BITS-1; - - /* Increase energy time resolution with shorter processing blocks. 32 is an empiric value. */ - const int granuleLength = fixMin(32, drcComp->blockLength); - - if (drcComp->useWeighting) { - FIXP_DBL x1, x2, y, y1, y2; - /* sum of filter coefficients about 2.5 -> squared value is 6.25 - WEIGHTING_FILTER_SHIFT is 2 -> scaling about 16, therefore reduce granuleShift by 1. - */ - const int granuleShift = getShiftFactor(granuleLength)-1; - - for (c = 0; c < (int)drcComp->channels; c++) { - const INT_PCM* pSamples = &inSamples[c]; - - if (c == drcComp->channelIdx[LFE]) { - continue; /* skip LFE */ - } - - /* get filter states */ - x1 = drcComp->filter[c].x1; - x2 = drcComp->filter[c].x2; - y1 = drcComp->filter[c].y1; - y2 = drcComp->filter[c].y2; - - i = 0; - - do { - - int offset = i; - FIXP_DBL accu = FL2FXCONST_DBL(0.f); - - for (i=offset; i < fixMin(offset+granuleLength,drcComp->blockLength); i++) { - /* apply weighting filter */ - FIXP_DBL x = FX_PCM2FX_DBL((FIXP_PCM)pSamples[i*drcComp->channels]) >> WEIGHTING_FILTER_SHIFT; - - /* y = b0 * (x - x2) - a1 * y1 - a2 * y2; */ - y = fMult(b0,x-x2) - fMult(a1,y1) - fMult(a2,y2); - - x2 = x1; - x1 = x; - y2 = y1; - y1 = y; - - accu += fPow2Div2(y)>>(granuleShift-1); /* partial energy */ - } /* i */ - - fixpAdd(accu, granuleShift+2*WEIGHTING_FILTER_SHIFT, &level_b, &level_e); /* sup up partial energies */ - - } while ( i < drcComp->blockLength ); - - - /* save filter states */ - drcComp->filter[c].x1 = x1; - drcComp->filter[c].x2 = x2; - drcComp->filter[c].y1 = y1; - drcComp->filter[c].y2 = y2; - } /* c */ - } /* weighting */ - else { - const int granuleShift = getShiftFactor(granuleLength); - - for (c = 0; c < (int)drcComp->channels; c++) { - const INT_PCM* pSamples = &inSamples[c]; - - if ((int)c == drcComp->channelIdx[LFE]) { - continue; /* skip LFE */ - } - - i = 0; - - do { - int offset = i; - FIXP_DBL accu = FL2FXCONST_DBL(0.f); - - for (i=offset; i < fixMin(offset+granuleLength,drcComp->blockLength); i++) { - /* partial energy */ - accu += fPow2Div2((FIXP_PCM)pSamples[i*drcComp->channels])>>(granuleShift-1); - } /* i */ - - fixpAdd(accu, granuleShift, &level_b, &level_e); /* sup up partial energies */ - - } while ( i < drcComp->blockLength ); - } - } /* weighting */ - - /* - * Convert to dBFS, apply dialnorm - */ - /* level scaling */ - - /* descaled level in ld64 representation */ - FIXP_DBL ldLevel = CalcLdData(level_b) + (FIXP_DBL)((level_e-12)<<(DFRACT_BITS-1-LD_DATA_SHIFT)) - CalcLdData((FIXP_DBL)(drcComp->blockLength<<(DFRACT_BITS-1-12))); - - /* if (level < 1e-10) level = 1e-10f; */ - ldLevel = FDKmax(ldLevel, FL2FXCONST_DBL(-0.51905126482615036685473741085772f)); - - /* level = 10 * log(level)/log(10) + 3; - * = 10*log(2)/log(10) * ld(level) + 3; - * = 10 * 0.30102999566398119521373889472449 * ld(level) + 3 - * = 10 * (0.30102999566398119521373889472449 * ld(level) + 0.3) - * = 10 * (0.30102999566398119521373889472449 * ld64(level) + 0.3/64) * 64 - * - * additional scaling with METADATA_FRACT_BITS: - * = 10 * (0.30102999566398119521373889472449 * ld64(level) + 0.3/64) * 64 * 2^(METADATA_FRACT_BITS) - * = 10 * (0.30102999566398119521373889472449 * ld64(level) + 0.3/64) * 2^(METADATA_FRACT_BITS+LD_DATA_SHIFT) - * = 10*2^(METADATA_FRACT_BITS+LD_DATA_SHIFT) * ( 0.30102999566398119521373889472449 * ld64(level) + 0.3/64 ) - * */ - FIXP_DBL level = fMult((FIXP_DBL)(10<<(METADATA_FRACT_BITS+LD_DATA_SHIFT)), fMult( FL2FXCONST_DBL(0.30102999566398119521373889472449f), ldLevel) + (FIXP_DBL)(FL2FXCONST_DBL(0.3f)>>LD_DATA_SHIFT) ); - - /* level -= dialnorm + 31 */ /* this is fixed to Dolby-ReferenceLevel as compressor profiles are defined relative to this */ - level -= ((FIXP_DBL)(dialnorm<<(METADATA_FRACT_BITS-16)) + (FIXP_DBL)(31<profile[i] == DRC_NONE) { - /* no compression */ - drcComp->smoothGain[i] = FL2FXCONST_DBL(0.f); - } - else { - FIXP_DBL gain, alpha, lvl2smthlvl; - - /* calc static gain */ - if (level <= drcComp->maxBoostThr[i]) { - /* max boost */ - gain = drcComp->maxBoost[i]; - } - else if (level < drcComp->boostThr[i]) { - /* boost range */ - gain = fMult((level - drcComp->boostThr[i]),drcComp->boostFac[i]); - } - else if (level <= drcComp->earlyCutThr[i]) { - /* null band */ - gain = FL2FXCONST_DBL(0.f); - } - else if (level <= drcComp->cutThr[i]) { - /* early cut range */ - gain = fMult((level - drcComp->earlyCutThr[i]), drcComp->earlyCutFac[i]); - } - else if (level < drcComp->maxCutThr[i]) { - /* cut range */ - gain = fMult((level - drcComp->cutThr[i]), drcComp->cutFac[i]) - drcComp->maxEarlyCut[i]; - } - else { - /* max cut */ - gain = -drcComp->maxCut[i]; - } - - /* choose time constant */ - lvl2smthlvl = level - drcComp->smoothLevel[i]; - if (gain < drcComp->smoothGain[i]) { - /* attack */ - if (lvl2smthlvl > drcComp->attackThr[i]) { - /* fast attack */ - alpha = drcComp->fastAttack[i]; - } - else { - /* slow attack */ - alpha = drcComp->slowAttack[i]; - } - } - else { - /* release */ - if (lvl2smthlvl < -drcComp->decayThr[i]) { - /* fast release */ - alpha = drcComp->fastDecay[i]; - } - else { - /* slow release */ - alpha = drcComp->slowDecay[i]; - } - } - - /* smooth gain & level */ - if ((gain < drcComp->smoothGain[i]) || (drcComp->holdCnt[i] == 0)) { /* hold gain unless we have an attack or hold period is over */ - FIXP_DBL accu; - - /* drcComp->smoothLevel[i] = (1-alpha) * drcComp->smoothLevel[i] + alpha * level; */ - accu = fMult(((FIXP_DBL)MAXVAL_DBL-alpha), drcComp->smoothLevel[i]); - accu += fMult(alpha,level); - drcComp->smoothLevel[i] = accu; - - /* drcComp->smoothGain[i] = (1-alpha) * drcComp->smoothGain[i] + alpha * gain; */ - accu = fMult(((FIXP_DBL)MAXVAL_DBL-alpha), drcComp->smoothGain[i]); - accu += fMult(alpha,gain); - drcComp->smoothGain[i] = accu; - } - - /* hold counter */ - if (drcComp->holdCnt[i]) { - drcComp->holdCnt[i]--; - } - if (gain < drcComp->smoothGain[i]) { - drcComp->holdCnt[i] = drcComp->holdOff[i]; - } - } /* profile != DRC_NONE */ - } /* for i=1..2 */ - } else { - /* no compression */ - drcComp->smoothGain[0] = FL2FXCONST_DBL(0.f); - drcComp->smoothGain[1] = FL2FXCONST_DBL(0.f); - } - - /************************************************************************** - * limiter - **************************************************************************/ - - /* find peak level */ - peak[0] = peak[1] = FL2FXCONST_DBL(0.f); - for (i = 0; i < drcComp->blockLength; i++) { - FIXP_DBL tmp; - const INT_PCM* pSamples = &inSamples[i*drcComp->channels]; - INT_PCM maxSample = 0; - - /* single channels */ - for (c = 0; c < (int)drcComp->channels; c++) { - maxSample = FDKmax(maxSample, fAbs(pSamples[c])); - } - peak[0] = fixMax(peak[0], FX_PCM2FX_DBL(maxSample)>>DOWNMIX_SHIFT); - - /* Lt/Rt downmix */ - if (drcComp->fullChannels > 2) { - /* Lt */ - tmp = FL2FXCONST_DBL(0.f); - - if (drcComp->channelIdx[LS] >= 0) tmp -= fMultDiv2(FL2FXCONST_DBL(0.707f), (FIXP_PCM)pSamples[drcComp->channelIdx[LS]])>>(DOWNMIX_SHIFT-1); /* Ls */ - if (drcComp->channelIdx[LS2] >= 0) tmp -= fMultDiv2(FL2FXCONST_DBL(0.707f), (FIXP_PCM)pSamples[drcComp->channelIdx[LS2]])>>(DOWNMIX_SHIFT-1); /* Ls2 */ - if (drcComp->channelIdx[RS] >= 0) tmp -= fMultDiv2(FL2FXCONST_DBL(0.707f), (FIXP_PCM)pSamples[drcComp->channelIdx[RS]])>>(DOWNMIX_SHIFT-1); /* Rs */ - if (drcComp->channelIdx[RS2] >= 0) tmp -= fMultDiv2(FL2FXCONST_DBL(0.707f), (FIXP_PCM)pSamples[drcComp->channelIdx[RS2]])>>(DOWNMIX_SHIFT-1); /* Rs2 */ - if ((drcComp->channelIdx[LS] >= 0) && (drcComp->channelIdx[LS2] >= 0)) tmp = fMult(FL2FXCONST_DBL(0.707f), tmp); /* 7.1ch */ - if (drcComp->channelIdx[S] >= 0) tmp -= fMultDiv2(FL2FXCONST_DBL(0.707f), (FIXP_PCM)pSamples[drcComp->channelIdx[S]])>>(DOWNMIX_SHIFT-1); /* S */ - if (drcComp->channelIdx[C] >= 0) tmp += fMultDiv2(FL2FXCONST_DBL(0.707f), (FIXP_PCM)pSamples[drcComp->channelIdx[C]])>>(DOWNMIX_SHIFT-1); /* C */ - tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[L]])>>DOWNMIX_SHIFT); /* L */ - - peak[0] = fixMax(peak[0], fixp_abs(tmp)); - - /* Rt */ - tmp = FL2FXCONST_DBL(0.f); - if (drcComp->channelIdx[LS] >= 0) tmp += fMultDiv2(FL2FXCONST_DBL(0.707f), (FIXP_PCM)pSamples[drcComp->channelIdx[LS]])>>(DOWNMIX_SHIFT-1); /* Ls */ - if (drcComp->channelIdx[LS2] >= 0) tmp += fMultDiv2(FL2FXCONST_DBL(0.707f), (FIXP_PCM)pSamples[drcComp->channelIdx[LS2]])>>(DOWNMIX_SHIFT-1); /* Ls2 */ - if (drcComp->channelIdx[RS] >= 0) tmp += fMultDiv2(FL2FXCONST_DBL(0.707f), (FIXP_PCM)pSamples[drcComp->channelIdx[RS]])>>(DOWNMIX_SHIFT-1); /* Rs */ - if (drcComp->channelIdx[RS2] >= 0) tmp += fMultDiv2(FL2FXCONST_DBL(0.707f), (FIXP_PCM)pSamples[drcComp->channelIdx[RS2]])>>(DOWNMIX_SHIFT-1); /* Rs2 */ - if ((drcComp->channelIdx[RS] >= 0) && (drcComp->channelIdx[RS2] >= 0)) tmp = fMult(FL2FXCONST_DBL(0.707f), tmp); /* 7.1ch */ - if (drcComp->channelIdx[S] >= 0) tmp += fMultDiv2(FL2FXCONST_DBL(0.707f), (FIXP_PCM)pSamples[drcComp->channelIdx[S]])>>(DOWNMIX_SHIFT-1); /* S */ - if (drcComp->channelIdx[C] >= 0) tmp += fMultDiv2(FL2FXCONST_DBL(0.707f), (FIXP_PCM)pSamples[drcComp->channelIdx[C]])>>(DOWNMIX_SHIFT-1); /* C */ - tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[R]])>>DOWNMIX_SHIFT); /* R */ - - peak[0] = fixMax(peak[0], fixp_abs(tmp)); - } - - /* Lo/Ro downmix */ - if (drcComp->fullChannels > 2) { - /* Lo */ - tmp = FL2FXCONST_DBL(0.f); - if (drcComp->channelIdx[LS] >= 0) tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[LS]])>>(DOWNMIX_SHIFT-1); /* Ls */ - if (drcComp->channelIdx[LS2] >= 0) tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[LS2]])>>(DOWNMIX_SHIFT-1); /* Ls2 */ - if ((drcComp->channelIdx[LS] >= 0) && (drcComp->channelIdx[LS2] >= 0)) tmp = fMult(FL2FXCONST_DBL(0.707f), tmp); /* 7.1ch */ - if (drcComp->channelIdx[S] >= 0) tmp += fMultDiv2(slev, fMult(FL2FXCONST_DBL(0.7f), (FIXP_PCM)pSamples[drcComp->channelIdx[S]]))>>(DOWNMIX_SHIFT-1); /* S */ - if (drcComp->channelIdx[C] >= 0) tmp += fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[C]])>>(DOWNMIX_SHIFT-1); /* C */ - tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[L]])>>DOWNMIX_SHIFT); /* L */ - - peak[0] = fixMax(peak[0], fixp_abs(tmp)); - - /* Ro */ - tmp = FL2FXCONST_DBL(0.f); - if (drcComp->channelIdx[RS] >= 0) tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[RS]])>>(DOWNMIX_SHIFT-1); /* Rs */ - if (drcComp->channelIdx[RS2] >= 0) tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[RS2]])>>(DOWNMIX_SHIFT-1); /* Rs2 */ - if ((drcComp->channelIdx[RS] >= 0) && (drcComp->channelIdx[RS2] >= 0)) tmp = fMult(FL2FXCONST_DBL(0.707f), tmp); /* 7.1ch */ - if (drcComp->channelIdx[S] >= 0) tmp += fMultDiv2(slev, fMult(FL2FXCONST_DBL(0.7f), (FIXP_PCM)pSamples[drcComp->channelIdx[S]]))>>(DOWNMIX_SHIFT-1); /* S */ - if (drcComp->channelIdx[C] >= 0) tmp += fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[C]])>>(DOWNMIX_SHIFT-1); /* C */ - tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[R]])>>DOWNMIX_SHIFT); /* R */ - - peak[0] = fixMax(peak[0], fixp_abs(tmp)); - } - - peak[1] = fixMax(peak[0], peak[1]); - - /* Mono Downmix - for comp_val only */ - if (drcComp->fullChannels > 1) { - tmp = FL2FXCONST_DBL(0.f); - if (drcComp->channelIdx[LS] >= 0) tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[LS]])>>(DOWNMIX_SHIFT-1); /* Ls */ - if (drcComp->channelIdx[LS2] >= 0) tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[LS2]])>>(DOWNMIX_SHIFT-1); /* Ls2 */ - if (drcComp->channelIdx[RS] >= 0) tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[RS]])>>(DOWNMIX_SHIFT-1); /* Rs */ - if (drcComp->channelIdx[RS2] >= 0) tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[RS2]])>>(DOWNMIX_SHIFT-1); /* Rs2 */ - if ((drcComp->channelIdx[LS] >= 0) && (drcComp->channelIdx[LS2] >= 0)) tmp = fMult(FL2FXCONST_DBL(0.707f), tmp); /* 7.1ch */ - /*if ((drcComp->channelIdx[RS] >= 0) && (drcComp->channelIdx[RS2] >= 0)) tmp *=0.707f;*/ /* 7.1ch */ - if (drcComp->channelIdx[S] >= 0) tmp += fMultDiv2(slev, fMult(FL2FXCONST_DBL(0.7f), (FIXP_PCM)pSamples[drcComp->channelIdx[S]]))>>(DOWNMIX_SHIFT-1); /* S */ - if (drcComp->channelIdx[C] >= 0) tmp += fMult(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[C]])>>(DOWNMIX_SHIFT-1); /* C (2*clev) */ - tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[L]])>>DOWNMIX_SHIFT); /* L */ - tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[R]])>>DOWNMIX_SHIFT); /* R */ - - peak[1] = fixMax(peak[1], fixp_abs(tmp)); - } - } - - for (i=0; i<2; i++) { - FIXP_DBL tmp = drcComp->prevPeak[i]; - drcComp->prevPeak[i] = peak[i]; - peak[i] = fixMax(peak[i], tmp); - - /* - * Convert to dBFS, apply dialnorm - */ - /* descaled peak in ld64 representation */ - FIXP_DBL ld_peak = CalcLdData(peak[i]) + (FIXP_DBL)((LONG)DOWNMIX_SHIFT<<(DFRACT_BITS-1-LD_DATA_SHIFT)); - - /* if (peak < 1e-6) level = 1e-6f; */ - ld_peak = FDKmax(ld_peak, FL2FXCONST_DBL(-0.31143075889569022011284244651463f)); - - /* peak[i] = 20 * log(peak[i])/log(10) + 0.2f + (drcComp->smoothGain[i]*2^METADATA_FRACT_BITS) - * peak[i] = 20 * log(2)/log(10) * ld(peak[i]) + 0.2f + (drcComp->smoothGain[i]*2^METADATA_FRACT_BITS) - * peak[i] = 10 * 2*0.30102999566398119521373889472449 * ld(peak[i]) + 0.2f + (drcComp->smoothGain[i]*2^METADATA_FRACT_BITS) - * - * additional scaling with METADATA_FRACT_BITS: - * peak[i] = (10 * 2*0.30102999566398119521373889472449 * ld64(peak[i]) * 64 + 0.2f + (drcComp->smoothGain[i]*2^METADATA_FRACT_BITS))*2^(-METADATA_FRACT_BITS) - * peak[i] = 10*2^(METADATA_FRACT_BITS+LD_DATA_SHIFT) * 2*0.30102999566398119521373889472449 * ld64(peak[i]) - * + 0.2f*2^(-METADATA_FRACT_BITS) + drcComp->smoothGain[i] - */ - peak[i] = fMult((FIXP_DBL)(10<<(METADATA_FRACT_BITS+LD_DATA_SHIFT)), fMult( FL2FX_DBL(2*0.30102999566398119521373889472449f), ld_peak)); - peak[i] += (FL2FX_DBL(0.5f)>>METADATA_INT_BITS); /* add a little bit headroom */ - peak[i] += drcComp->smoothGain[i]; - } - - /* peak -= dialnorm + 31; */ /* this is Dolby style only */ - peak[0] -= (FIXP_DBL)((dialnorm-drc_TargetRefLevel)<<(METADATA_FRACT_BITS-16)); /* peak[0] -= dialnorm - drc_TargetRefLevel */ - - /* peak += 11; */ /* this is Dolby style only */ /* RF mode output is 11dB higher */ - /*peak += comp_TargetRefLevel - drc_TargetRefLevel;*/ - peak[1] -= (FIXP_DBL)((dialnorm-comp_TargetRefLevel)<<(METADATA_FRACT_BITS-16)); /* peak[1] -= dialnorm - comp_TargetRefLevel */ - - /* limiter gain */ - drcComp->limGain[0] += drcComp->limDecay; /* linear limiter release */ - drcComp->limGain[0] = fixMin(drcComp->limGain[0], -peak[0]); - - drcComp->limGain[1] += 2*drcComp->limDecay; /* linear limiter release */ - drcComp->limGain[1] = fixMin(drcComp->limGain[1], -peak[1]); - - /*************************************************************************/ - - /* apply limiting, return DRC gains*/ - { - FIXP_DBL tmp; - - tmp = drcComp->smoothGain[0]; - if (drcComp->limGain[0] < FL2FXCONST_DBL(0.f)) { - tmp += drcComp->limGain[0]; - } - *pDynrng = (LONG) scaleValue(tmp, -(METADATA_FRACT_BITS-16)); - - tmp = drcComp->smoothGain[1]; - if (drcComp->limGain[1] < FL2FXCONST_DBL(0.f)) { - tmp += drcComp->limGain[1]; - } - *pCompr = (LONG) scaleValue(tmp, -(METADATA_FRACT_BITS-16)); - } - - return 0; -} - - -DRC_PROFILE FDK_DRC_Generator_getDrcProfile(const HDRC_COMP drcComp) -{ - return drcComp->profile[0]; -} - -DRC_PROFILE FDK_DRC_Generator_getCompProfile(const HDRC_COMP drcComp) -{ - return drcComp->profile[1]; -} - - diff --git a/libAACenc/src/metadata_compressor.h b/libAACenc/src/metadata_compressor.h deleted file mode 100644 index ff639b5..0000000 --- a/libAACenc/src/metadata_compressor.h +++ /dev/null @@ -1,252 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/********************** Fraunhofer IIS FDK AAC Encoder lib ****************** - - Author(s): M. Neusinger - Description: Compressor for AAC Metadata Generator - -******************************************************************************/ - -#ifndef _METADATA_COMPRESSOR_H -#define _METADATA_COMPRESSOR_H - - -#include "FDK_audio.h" -#include "common_fix.h" - -#include "aacenc.h" - - -/** - * DRC compression profiles. - */ -typedef enum DRC_PROFILE { - DRC_NONE = 0, - DRC_FILMSTANDARD = 1, - DRC_FILMLIGHT = 2, - DRC_MUSICSTANDARD = 3, - DRC_MUSICLIGHT = 4, - DRC_SPEECH = 5, - DRC_DELAY_TEST = 6 - -} DRC_PROFILE; - - -/** - * DRC Compressor handle. - */ -typedef struct DRC_COMP DRC_COMP, *HDRC_COMP; - -/** - * \brief Open a DRC Compressor instance. - * - * Allocate memory for a compressor instance. - * - * \param phDrcComp A pointer to a compressor handle. Initialized on return. - * - * \return - * - 0, on succes. - * - unequal 0, on failure. - */ -INT FDK_DRC_Generator_Open( - HDRC_COMP *phDrcComp - ); - - -/** - * \brief Close the DRC Compressor instance. - * - * Deallocate instance and free whole memory. - * - * \param phDrcComp Pointer to the compressor handle to be deallocated. - * - * \return - * - 0, on succes. - * - unequal 0, on failure. - */ -INT FDK_DRC_Generator_Close( - HDRC_COMP *phDrcComp - ); - -/** - * \brief Configure DRC Compressor. - * - * \param drcComp Compressor handle. - * \param profileLine DRC profile for line mode. - * \param profileRF DRC profile for RF mode. - * \param blockLength Length of processing block in samples per channel. - * \param sampleRate Sampling rate in Hz. - * \param channelMode Channel configuration. - * \param channelOrder Channel order, MPEG or WAV. - * \param useWeighting Use weighting filter for loudness calculation - * - * \return - * - 0, on success, - * - unequal 0, on failure - */ -INT FDK_DRC_Generator_Initialize( - HDRC_COMP drcComp, - const DRC_PROFILE profileLine, - const DRC_PROFILE profileRF, - const INT blockLength, - const UINT sampleRate, - const CHANNEL_MODE channelMode, - const CHANNEL_ORDER channelOrder, - const UCHAR useWeighting - ); - -/** - * \brief Calculate DRC Compressor Gain. - * - * \param drcComp Compressor handle. - * \param inSamples Pointer to interleaved input audio samples. - * \param dialnorm Dialog Level in dB (typically -31...-1). - * \param drc_TargetRefLevel - * \param comp_TargetRefLevel - * \param clev Downmix center mix factor (typically 0.707, 0.595 or 0.5) - * \param slev Downmix surround mix factor (typically 0.707, 0.5, or 0) - * \param dynrng Pointer to variable receiving line mode DRC gain in dB - * \param compr Pointer to variable receiving RF mode DRC gain in dB - * - * \return - * - 0, on success, - * - unequal 0, on failure - */ -INT FDK_DRC_Generator_Calc( - HDRC_COMP drcComp, - const INT_PCM * const inSamples, - const INT dialnorm, - const INT drc_TargetRefLevel, - const INT comp_TargetRefLevel, - FIXP_DBL clev, - FIXP_DBL slev, - INT * const dynrng, - INT * const compr - ); - - -/** - * \brief Configure DRC Compressor Profile. - * - * \param drcComp Compressor handle. - * \param profileLine DRC profile for line mode. - * \param profileRF DRC profile for RF mode. - * - * \return - * - 0, on success, - * - unequal 0, on failure - */ -INT FDK_DRC_Generator_setDrcProfile( - HDRC_COMP drcComp, - const DRC_PROFILE profileLine, - const DRC_PROFILE profileRF - ); - - -/** - * \brief Get DRC profile for line mode. - * - * \param drcComp Compressor handle. - * - * \return Current Profile. - */ -DRC_PROFILE FDK_DRC_Generator_getDrcProfile( - const HDRC_COMP drcComp - ); - - -/** - * \brief Get DRC profile for RF mode. - * - * \param drcComp Compressor handle. - * - * \return Current Profile. - */ -DRC_PROFILE FDK_DRC_Generator_getCompProfile( - const HDRC_COMP drcComp - ); - - -#endif /* _METADATA_COMPRESSOR_H */ - diff --git a/libAACenc/src/metadata_main.cpp b/libAACenc/src/metadata_main.cpp deleted file mode 100644 index 90f8f4e..0000000 --- a/libAACenc/src/metadata_main.cpp +++ /dev/null @@ -1,869 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/********************** Fraunhofer IIS FDK AAC Encoder lib ****************** - - Author(s): V. Bacigalupo - Description: Metadata Encoder library interface functions - -******************************************************************************/ - - -#include "metadata_main.h" -#include "metadata_compressor.h" -#include "FDK_bitstream.h" -#include "FDK_audio.h" -#include "genericStds.h" - -/*----------------- defines ----------------------*/ -#define MAX_DRC_BANDS (1<<4) -#define MAX_DRC_CHANNELS (8) -#define MAX_DRC_FRAMELEN (2*1024) - -/*--------------- structure definitions --------------------*/ - -typedef struct AAC_METADATA -{ - /* MPEG: Dynamic Range Control */ - struct { - UCHAR prog_ref_level_present; - SCHAR prog_ref_level; - - UCHAR dyn_rng_sgn[MAX_DRC_BANDS]; - UCHAR dyn_rng_ctl[MAX_DRC_BANDS]; - - UCHAR drc_bands_present; - UCHAR drc_band_incr; - UCHAR drc_band_top[MAX_DRC_BANDS]; - UCHAR drc_interpolation_scheme; - AACENC_METADATA_DRC_PROFILE drc_profile; - INT drc_TargetRefLevel; /* used for Limiter */ - - /* excluded channels */ - UCHAR excluded_chns_present; - UCHAR exclude_mask[2]; /* MAX_NUMBER_CHANNELS/8 */ - } mpegDrc; - - /* ETSI: addtl ancillary data */ - struct { - /* Heavy Compression */ - UCHAR compression_on; /* flag, if compression value should be written */ - UCHAR compression_value; /* compression value */ - AACENC_METADATA_DRC_PROFILE comp_profile; - INT comp_TargetRefLevel; /* used for Limiter */ - INT timecode_coarse_status; - INT timecode_fine_status; - } etsiAncData; - - SCHAR centerMixLevel; /* center downmix level (0...7, according to table) */ - SCHAR surroundMixLevel; /* surround downmix level (0...7, according to table) */ - UCHAR WritePCEMixDwnIdx; /* flag */ - UCHAR DmxLvl_On; /* flag */ - - UCHAR dolbySurroundMode; - - UCHAR metadataMode; /* indicate meta data mode in current frame (delay line) */ - -} AAC_METADATA; - -struct FDK_METADATA_ENCODER -{ - INT metadataMode; - HDRC_COMP hDrcComp; - AACENC_MetaData submittedMetaData; - - INT nAudioDataDelay; - INT nMetaDataDelay; - INT nChannels; - - INT_PCM audioDelayBuffer[MAX_DRC_CHANNELS*MAX_DRC_FRAMELEN]; - int audioDelayIdx; - - AAC_METADATA metaDataBuffer[3]; - int metaDataDelayIdx; - - UCHAR drcInfoPayload[12]; - UCHAR drcDsePayload[8]; - - INT matrix_mixdown_idx; - AACENC_EXT_PAYLOAD exPayload[2]; - INT nExtensions; - - INT finalizeMetaData; /* Delay switch off by one frame and write default configuration to - finalize the metadata setup. */ -}; - - -/*---------------- constants -----------------------*/ -static const AACENC_MetaData defaultMetaDataSetup = { - AACENC_METADATA_DRC_NONE, - AACENC_METADATA_DRC_NONE, - -(31<<16), - -(31<<16), - 0, - -(31<<16), - 0, - 0, - 0, - 0, - 0 -}; - -static const FIXP_DBL dmxTable[8] = { - ((FIXP_DBL)MAXVAL_DBL), FL2FXCONST_DBL(0.841f), FL2FXCONST_DBL(0.707f), FL2FXCONST_DBL(0.596f), - FL2FXCONST_DBL(0.500f), FL2FXCONST_DBL(0.422f), FL2FXCONST_DBL(0.355f), FL2FXCONST_DBL(0.000f) -}; - -static const UCHAR surmix2matrix_mixdown_idx[8] = { - 0, 0, 0, 1, 1, 2, 2, 3 -}; - - -/*--------------- function declarations --------------------*/ -static FDK_METADATA_ERROR WriteMetadataPayload( - const HANDLE_FDK_METADATA_ENCODER hMetaData, - const AAC_METADATA * const pMetadata - ); - -static INT WriteDynamicRangeInfoPayload( - const AAC_METADATA* const pMetadata, - UCHAR* const pExtensionPayload - ); - -static INT WriteEtsiAncillaryDataPayload( - const AAC_METADATA* const pMetadata, - UCHAR* const pExtensionPayload - ); - -static FDK_METADATA_ERROR CompensateAudioDelay( - HANDLE_FDK_METADATA_ENCODER hMetaDataEnc, - INT_PCM * const pAudioSamples, - const INT nAudioSamples - ); - -static FDK_METADATA_ERROR LoadSubmittedMetadata( - const AACENC_MetaData * const hMetadata, - const INT nChannels, - const INT metadataMode, - AAC_METADATA * const pAacMetaData - ); - -static FDK_METADATA_ERROR ProcessCompressor( - AAC_METADATA *pMetadata, - HDRC_COMP hDrcComp, - const INT_PCM * const pSamples, - const INT nSamples - ); - -/*------------- function definitions ----------------*/ - -static DRC_PROFILE convertProfile(AACENC_METADATA_DRC_PROFILE aacProfile) -{ - DRC_PROFILE drcProfile = DRC_NONE; - - switch(aacProfile) { - case AACENC_METADATA_DRC_NONE: drcProfile = DRC_NONE; break; - case AACENC_METADATA_DRC_FILMSTANDARD: drcProfile = DRC_FILMSTANDARD; break; - case AACENC_METADATA_DRC_FILMLIGHT: drcProfile = DRC_FILMLIGHT; break; - case AACENC_METADATA_DRC_MUSICSTANDARD: drcProfile = DRC_MUSICSTANDARD; break; - case AACENC_METADATA_DRC_MUSICLIGHT: drcProfile = DRC_MUSICLIGHT; break; - case AACENC_METADATA_DRC_SPEECH: drcProfile = DRC_SPEECH; break; - default: drcProfile = DRC_NONE; break; - } - return drcProfile; -} - - -/* convert dialog normalization to program reference level */ -/* NOTE: this only is correct, if the decoder target level is set to -31dB for line mode / -20dB for RF mode */ -static UCHAR dialnorm2progreflvl(const INT d) -{ - return ((UCHAR)FDKmax(0, FDKmin((-d + (1<<13)) >> 14, 127))); -} - -/* convert program reference level to dialog normalization */ -static INT progreflvl2dialnorm(const UCHAR p) -{ - return -((INT)(p<<(16-2))); -} - -/* encode downmix levels to Downmixing_levels_MPEG4 */ -static SCHAR encodeDmxLvls(const SCHAR cmixlev, const SCHAR surmixlev) -{ - SCHAR dmxLvls = 0; - dmxLvls |= 0x80 | (cmixlev << 4); /* center_mix_level_on */ - dmxLvls |= 0x08 | surmixlev; /* surround_mix_level_on */ - - return dmxLvls; -} - -/* encode AAC DRC gain (ISO/IEC 14496-3:2005 4.5.2.7) */ -static void encodeDynrng(INT gain, UCHAR* const dyn_rng_ctl, UCHAR* const dyn_rng_sgn ) -{ - if(gain < 0) - { - *dyn_rng_sgn = 1; - gain = -gain; - } - else - { - *dyn_rng_sgn = 0; - } - gain = FDKmin(gain,(127<<14)); - - *dyn_rng_ctl = (UCHAR)((gain + (1<<13)) >> 14); -} - -/* decode AAC DRC gain (ISO/IEC 14496-3:2005 4.5.2.7) */ -static INT decodeDynrng(const UCHAR dyn_rng_ctl, const UCHAR dyn_rng_sgn) -{ - INT tmp = ((INT)dyn_rng_ctl << (16-2)); - if (dyn_rng_sgn) tmp = -tmp; - - return tmp; -} - -/* encode AAC compression value (ETSI TS 101 154 page 99) */ -static UCHAR encodeCompr(INT gain) -{ - UCHAR x, y; - INT tmp; - - /* tmp = (int)((48.164f - gain) / 6.0206f * 15 + 0.5f); */ - tmp = ((3156476 - gain) * 15 + 197283) / 394566; - - if (tmp >= 240) { - return 0xFF; - } - else if (tmp < 0) { - return 0; - } - else { - x = tmp / 15; - y = tmp % 15; - } - - return (x << 4) | y; -} - -/* decode AAC compression value (ETSI TS 101 154 page 99) */ -static INT decodeCompr(const UCHAR compr) -{ - INT gain; - SCHAR x = compr >> 4; /* 4 MSB of compr */ - UCHAR y = (compr & 0x0F); /* 4 LSB of compr */ - - /* gain = (INT)((48.164f - 6.0206f * x - 0.4014f * y) ); */ - gain = (INT)( scaleValue(((LONG)FL2FXCONST_DBL(6.0206f/128.f)*(8-x) - (LONG)FL2FXCONST_DBL(0.4014f/128.f)*y), -(DFRACT_BITS-1-7-16)) ); - - return gain; -} - - -FDK_METADATA_ERROR FDK_MetadataEnc_Open( - HANDLE_FDK_METADATA_ENCODER *phMetaData - ) -{ - FDK_METADATA_ERROR err = METADATA_OK; - HANDLE_FDK_METADATA_ENCODER hMetaData = NULL; - - if (phMetaData == NULL) { - err = METADATA_INVALID_HANDLE; - goto bail; - } - - /* allocate memory */ - hMetaData = (HANDLE_FDK_METADATA_ENCODER) FDKcalloc(1, sizeof(FDK_METADATA_ENCODER) ); - - if (hMetaData == NULL) { - err = METADATA_MEMORY_ERROR; - goto bail; - } - - FDKmemclear(hMetaData, sizeof(FDK_METADATA_ENCODER)); - - /* Allocate DRC Compressor. */ - if (FDK_DRC_Generator_Open(&hMetaData->hDrcComp)!=0) { - err = METADATA_MEMORY_ERROR; - goto bail; - } - - /* Return metadata instance */ - *phMetaData = hMetaData; - - return err; - -bail: - FDK_MetadataEnc_Close(&hMetaData); - return err; -} - -FDK_METADATA_ERROR FDK_MetadataEnc_Close( - HANDLE_FDK_METADATA_ENCODER *phMetaData - ) -{ - FDK_METADATA_ERROR err = METADATA_OK; - - if (phMetaData == NULL) { - err = METADATA_INVALID_HANDLE; - goto bail; - } - - if (*phMetaData != NULL) { - FDK_DRC_Generator_Close(&(*phMetaData)->hDrcComp); - FDKfree(*phMetaData); - *phMetaData = NULL; - } -bail: - return err; -} - -FDK_METADATA_ERROR FDK_MetadataEnc_Init( - HANDLE_FDK_METADATA_ENCODER hMetaData, - const INT resetStates, - const INT metadataMode, - const INT audioDelay, - const UINT frameLength, - const UINT sampleRate, - const UINT nChannels, - const CHANNEL_MODE channelMode, - const CHANNEL_ORDER channelOrder - ) -{ - FDK_METADATA_ERROR err = METADATA_OK; - int i, nFrames, delay; - - if (hMetaData==NULL) { - err = METADATA_INVALID_HANDLE; - goto bail; - } - - /* Determine values for delay compensation. */ - for (nFrames=0, delay=audioDelay-frameLength; delay>0; delay-=frameLength, nFrames++); - - if ( (hMetaData->nChannels>MAX_DRC_CHANNELS) || ((-delay)>MAX_DRC_FRAMELEN) ) { - err = METADATA_INIT_ERROR; - goto bail; - } - - /* Initialize with default setup. */ - FDKmemcpy(&hMetaData->submittedMetaData, &defaultMetaDataSetup, sizeof(AACENC_MetaData)); - - hMetaData->finalizeMetaData = 0; /* finalize meta data only while on/off switching, else disabled */ - - /* Reset delay lines. */ - if ( resetStates || (hMetaData->nAudioDataDelay!=-delay) || (hMetaData->nChannels!=(INT)nChannels) ) - { - FDKmemclear(hMetaData->audioDelayBuffer, sizeof(hMetaData->audioDelayBuffer)); - FDKmemclear(hMetaData->metaDataBuffer, sizeof(hMetaData->metaDataBuffer)); - hMetaData->audioDelayIdx = 0; - hMetaData->metaDataDelayIdx = 0; - } - else { - /* Enable meta data. */ - if ( (hMetaData->metadataMode==0) && (metadataMode!=0) ) { - /* disable meta data in all delay lines */ - for (i=0; i<(int)(sizeof(hMetaData->metaDataBuffer)/sizeof(AAC_METADATA)); i++) { - LoadSubmittedMetadata(&hMetaData->submittedMetaData, nChannels, 0, &hMetaData->metaDataBuffer[i]); - } - } - - /* Disable meta data.*/ - if ( (hMetaData->metadataMode!=0) && (metadataMode==0) ) { - hMetaData->finalizeMetaData = hMetaData->metadataMode; - } - } - - /* Initialize delay. */ - hMetaData->nAudioDataDelay = -delay; - hMetaData->nMetaDataDelay = nFrames; - hMetaData->nChannels = nChannels; - hMetaData->metadataMode = metadataMode; - - /* Initialize compressor. */ - if (metadataMode != 0) { - if ( FDK_DRC_Generator_Initialize( - hMetaData->hDrcComp, - DRC_NONE, - DRC_NONE, - frameLength, - sampleRate, - channelMode, - channelOrder, - 1) != 0) - { - err = METADATA_INIT_ERROR; - } - } -bail: - return err; -} - -static FDK_METADATA_ERROR ProcessCompressor( - AAC_METADATA *pMetadata, - HDRC_COMP hDrcComp, - const INT_PCM * const pSamples, - const INT nSamples - ) -{ - FDK_METADATA_ERROR err = METADATA_OK; - - if ( (pMetadata==NULL) || (hDrcComp==NULL) ) { - err = METADATA_INVALID_HANDLE; - return err; - } - DRC_PROFILE profileDrc = convertProfile(pMetadata->mpegDrc.drc_profile); - DRC_PROFILE profileComp = convertProfile(pMetadata->etsiAncData.comp_profile); - - /* first, check if profile is same as last frame - * otherwise, update setup */ - if ( (profileDrc != FDK_DRC_Generator_getDrcProfile(hDrcComp)) - || (profileComp != FDK_DRC_Generator_getCompProfile(hDrcComp)) ) - { - FDK_DRC_Generator_setDrcProfile(hDrcComp, profileDrc, profileComp); - } - - /* Sanity check */ - if (profileComp == DRC_NONE) { - pMetadata->etsiAncData.compression_value = 0x80; /* to ensure no external values will be written if not configured */ - } - - /* in case of embedding external values, copy this now (limiter may overwrite them) */ - INT dynrng = decodeDynrng(pMetadata->mpegDrc.dyn_rng_ctl[0], pMetadata->mpegDrc.dyn_rng_sgn[0]); - INT compr = decodeCompr(pMetadata->etsiAncData.compression_value); - - /* Call compressor */ - if (FDK_DRC_Generator_Calc(hDrcComp, - pSamples, - progreflvl2dialnorm(pMetadata->mpegDrc.prog_ref_level), - pMetadata->mpegDrc.drc_TargetRefLevel, - pMetadata->etsiAncData.comp_TargetRefLevel, - dmxTable[pMetadata->centerMixLevel], - dmxTable[pMetadata->surroundMixLevel], - &dynrng, - &compr) != 0) - { - err = METADATA_ENCODE_ERROR; - goto bail; - } - - /* Write DRC values */ - pMetadata->mpegDrc.drc_band_incr = 0; - encodeDynrng(dynrng, pMetadata->mpegDrc.dyn_rng_ctl, pMetadata->mpegDrc.dyn_rng_sgn); - pMetadata->etsiAncData.compression_value = encodeCompr(compr); - -bail: - return err; -} - -FDK_METADATA_ERROR FDK_MetadataEnc_Process( - HANDLE_FDK_METADATA_ENCODER hMetaDataEnc, - INT_PCM * const pAudioSamples, - const INT nAudioSamples, - const AACENC_MetaData * const pMetadata, - AACENC_EXT_PAYLOAD ** ppMetaDataExtPayload, - UINT * nMetaDataExtensions, - INT * matrix_mixdown_idx - ) -{ - FDK_METADATA_ERROR err = METADATA_OK; - int metaDataDelayWriteIdx, metaDataDelayReadIdx, metadataMode; - - /* Where to write new meta data info */ - metaDataDelayWriteIdx = hMetaDataEnc->metaDataDelayIdx; - - /* How to write the data */ - metadataMode = hMetaDataEnc->metadataMode; - - /* Compensate meta data delay. */ - hMetaDataEnc->metaDataDelayIdx++; - if (hMetaDataEnc->metaDataDelayIdx > hMetaDataEnc->nMetaDataDelay) hMetaDataEnc->metaDataDelayIdx = 0; - - /* Where to read pending meta data info from. */ - metaDataDelayReadIdx = hMetaDataEnc->metaDataDelayIdx; - - /* Submit new data if available. */ - if (pMetadata!=NULL) { - FDKmemcpy(&hMetaDataEnc->submittedMetaData, pMetadata, sizeof(AACENC_MetaData)); - } - - /* Write one additional frame with default configuration of meta data. Ensure defined behaviour on decoder side. */ - if ( (hMetaDataEnc->finalizeMetaData!=0) && (hMetaDataEnc->metadataMode==0)) { - FDKmemcpy(&hMetaDataEnc->submittedMetaData, &defaultMetaDataSetup, sizeof(AACENC_MetaData)); - metadataMode = hMetaDataEnc->finalizeMetaData; - hMetaDataEnc->finalizeMetaData = 0; - } - - /* Get last submitted data. */ - if ( (err = LoadSubmittedMetadata( - &hMetaDataEnc->submittedMetaData, - hMetaDataEnc->nChannels, - metadataMode, - &hMetaDataEnc->metaDataBuffer[metaDataDelayWriteIdx])) != METADATA_OK ) - { - goto bail; - } - - /* Calculate compressor if necessary and updata meta data info */ - if (hMetaDataEnc->metaDataBuffer[metaDataDelayWriteIdx].metadataMode != 0) { - if ( (err = ProcessCompressor( - &hMetaDataEnc->metaDataBuffer[metaDataDelayWriteIdx], - hMetaDataEnc->hDrcComp, - pAudioSamples, - nAudioSamples)) != METADATA_OK) - { - /* Get last submitted data again. */ - LoadSubmittedMetadata( - &hMetaDataEnc->submittedMetaData, - hMetaDataEnc->nChannels, - metadataMode, - &hMetaDataEnc->metaDataBuffer[metaDataDelayWriteIdx]); - } - } - - /* Convert Meta Data side info to bitstream data. */ - if ( (err = WriteMetadataPayload(hMetaDataEnc, &hMetaDataEnc->metaDataBuffer[metaDataDelayReadIdx])) != METADATA_OK ) { - goto bail; - } - - /* Assign meta data to output */ - *ppMetaDataExtPayload = hMetaDataEnc->exPayload; - *nMetaDataExtensions = hMetaDataEnc->nExtensions; - *matrix_mixdown_idx = hMetaDataEnc->matrix_mixdown_idx; - -bail: - /* Compensate audio delay, reset err status. */ - err = CompensateAudioDelay(hMetaDataEnc, pAudioSamples, nAudioSamples); - - return err; -} - - -static FDK_METADATA_ERROR CompensateAudioDelay( - HANDLE_FDK_METADATA_ENCODER hMetaDataEnc, - INT_PCM * const pAudioSamples, - const INT nAudioSamples - ) -{ - FDK_METADATA_ERROR err = METADATA_OK; - - if (hMetaDataEnc->nAudioDataDelay) { - int i, delaySamples = hMetaDataEnc->nAudioDataDelay*hMetaDataEnc->nChannels; - - for (i = 0; i < nAudioSamples; i++) { - INT_PCM tmp = pAudioSamples[i]; - pAudioSamples[i] = hMetaDataEnc->audioDelayBuffer[hMetaDataEnc->audioDelayIdx]; - hMetaDataEnc->audioDelayBuffer[hMetaDataEnc->audioDelayIdx] = tmp; - - hMetaDataEnc->audioDelayIdx++; - if (hMetaDataEnc->audioDelayIdx >= delaySamples) hMetaDataEnc->audioDelayIdx = 0; - } - } - - return err; -} - -/*----------------------------------------------------------------------------- - - functionname: WriteMetadataPayload - description: fills anc data and extension payload - returns: Error status - - ------------------------------------------------------------------------------*/ -static FDK_METADATA_ERROR WriteMetadataPayload( - const HANDLE_FDK_METADATA_ENCODER hMetaData, - const AAC_METADATA * const pMetadata - ) -{ - FDK_METADATA_ERROR err = METADATA_OK; - - if ( (hMetaData==NULL) || (pMetadata==NULL) ) { - err = METADATA_INVALID_HANDLE; - goto bail; - } - - hMetaData->nExtensions = 0; - hMetaData->matrix_mixdown_idx = -1; - - /* AAC-DRC */ - if (pMetadata->metadataMode != 0) - { - hMetaData->exPayload[hMetaData->nExtensions].pData = hMetaData->drcInfoPayload; - hMetaData->exPayload[hMetaData->nExtensions].dataType = EXT_DYNAMIC_RANGE; - hMetaData->exPayload[hMetaData->nExtensions].associatedChElement = -1; - - hMetaData->exPayload[hMetaData->nExtensions].dataSize = - WriteDynamicRangeInfoPayload(pMetadata, hMetaData->exPayload[hMetaData->nExtensions].pData); - - hMetaData->nExtensions++; - - /* Matrix Mixdown Coefficient in PCE */ - if (pMetadata->WritePCEMixDwnIdx) { - hMetaData->matrix_mixdown_idx = surmix2matrix_mixdown_idx[pMetadata->surroundMixLevel]; - } - - /* ETSI TS 101 154 (DVB) - MPEG4 ancillary_data() */ - if (pMetadata->metadataMode == 2) /* MP4_METADATA_MPEG_ETSI */ - { - hMetaData->exPayload[hMetaData->nExtensions].pData = hMetaData->drcDsePayload; - hMetaData->exPayload[hMetaData->nExtensions].dataType = EXT_DATA_ELEMENT; - hMetaData->exPayload[hMetaData->nExtensions].associatedChElement = -1; - - hMetaData->exPayload[hMetaData->nExtensions].dataSize = - WriteEtsiAncillaryDataPayload(pMetadata,hMetaData->exPayload[hMetaData->nExtensions].pData); - - hMetaData->nExtensions++; - } /* metadataMode == 2 */ - - } /* metadataMode != 0 */ - -bail: - return err; -} - -static INT WriteDynamicRangeInfoPayload( - const AAC_METADATA* const pMetadata, - UCHAR* const pExtensionPayload - ) -{ - const INT pce_tag_present = 0; /* yet fixed setting! */ - const INT prog_ref_lev_res_bits = 0; - INT i, drc_num_bands = 1; - - FDK_BITSTREAM bsWriter; - FDKinitBitStream(&bsWriter, pExtensionPayload, 16, 0, BS_WRITER); - - /* dynamic_range_info() */ - FDKwriteBits(&bsWriter, pce_tag_present, 1); /* pce_tag_present */ - if (pce_tag_present) { - FDKwriteBits(&bsWriter, 0x0, 4); /* pce_instance_tag */ - FDKwriteBits(&bsWriter, 0x0, 4); /* drc_tag_reserved_bits */ - } - - /* Exclude channels */ - FDKwriteBits(&bsWriter, (pMetadata->mpegDrc.excluded_chns_present) ? 1 : 0, 1); /* excluded_chns_present*/ - - /* Multiband DRC */ - FDKwriteBits(&bsWriter, (pMetadata->mpegDrc.drc_bands_present) ? 1 : 0, 1); /* drc_bands_present */ - if (pMetadata->mpegDrc.drc_bands_present) - { - FDKwriteBits(&bsWriter, pMetadata->mpegDrc.drc_band_incr, 4); /* drc_band_incr */ - FDKwriteBits(&bsWriter, pMetadata->mpegDrc.drc_interpolation_scheme, 4); /* drc_interpolation_scheme */ - drc_num_bands += pMetadata->mpegDrc.drc_band_incr; - for (i=0; impegDrc.drc_band_top[i], 8); /* drc_band_top */ - } - } - - /* Program Reference Level */ - FDKwriteBits(&bsWriter, pMetadata->mpegDrc.prog_ref_level_present, 1); /* prog_ref_level_present */ - if (pMetadata->mpegDrc.prog_ref_level_present) - { - FDKwriteBits(&bsWriter, pMetadata->mpegDrc.prog_ref_level, 7); /* prog_ref_level */ - FDKwriteBits(&bsWriter, prog_ref_lev_res_bits, 1); /* prog_ref_level_reserved_bits */ - } - - /* DRC Values */ - for (i=0; impegDrc.dyn_rng_sgn[i]) ? 1 : 0, 1); /* dyn_rng_sgn[ */ - FDKwriteBits(&bsWriter, pMetadata->mpegDrc.dyn_rng_ctl[i], 7); /* dyn_rng_ctl */ - } - - /* return number of valid bits in extension payload. */ - return FDKgetValidBits(&bsWriter); -} - -static INT WriteEtsiAncillaryDataPayload( - const AAC_METADATA* const pMetadata, - UCHAR* const pExtensionPayload - ) -{ - FDK_BITSTREAM bsWriter; - FDKinitBitStream(&bsWriter, pExtensionPayload, 16, 0, BS_WRITER); - - /* ancillary_data_sync */ - FDKwriteBits(&bsWriter, 0xBC, 8); - - /* bs_info */ - FDKwriteBits(&bsWriter, 0x3, 2); /* mpeg_audio_type */ - FDKwriteBits(&bsWriter, pMetadata->dolbySurroundMode, 2); /* dolby_surround_mode */ - FDKwriteBits(&bsWriter, 0x0, 4); /* reserved */ - - /* ancillary_data_status */ - FDKwriteBits(&bsWriter, 0, 3); /* 3 bit Reserved, set to "0" */ - FDKwriteBits(&bsWriter, (pMetadata->DmxLvl_On) ? 1 : 0, 1); /* downmixing_levels_MPEG4_status */ - FDKwriteBits(&bsWriter, 0, 1); /* Reserved, set to "0" */ - FDKwriteBits(&bsWriter, (pMetadata->etsiAncData.compression_on) ? 1 : 0, 1); /* audio_coding_mode_and_compression status */ - FDKwriteBits(&bsWriter, (pMetadata->etsiAncData.timecode_coarse_status) ? 1 : 0, 1); /* coarse_grain_timecode_status */ - FDKwriteBits(&bsWriter, (pMetadata->etsiAncData.timecode_fine_status) ? 1 : 0, 1); /* fine_grain_timecode_status */ - - /* downmixing_levels_MPEG4_status */ - if (pMetadata->DmxLvl_On) { - FDKwriteBits(&bsWriter, encodeDmxLvls(pMetadata->centerMixLevel, pMetadata->surroundMixLevel), 8); - } - - /* audio_coding_mode_and_compression_status */ - if (pMetadata->etsiAncData.compression_on) { - FDKwriteBits(&bsWriter, 0x01, 8); /* audio coding mode */ - FDKwriteBits(&bsWriter, pMetadata->etsiAncData.compression_value, 8); /* compression value */ - } - - /* grain-timecode coarse/fine */ - if (pMetadata->etsiAncData.timecode_coarse_status) { - FDKwriteBits(&bsWriter, 0x0, 16); /* not yet supported */ - } - - if (pMetadata->etsiAncData.timecode_fine_status) { - FDKwriteBits(&bsWriter, 0x0, 16); /* not yet supported */ - } - - return FDKgetValidBits(&bsWriter); -} - - -static FDK_METADATA_ERROR LoadSubmittedMetadata( - const AACENC_MetaData * const hMetadata, - const INT nChannels, - const INT metadataMode, - AAC_METADATA * const pAacMetaData - ) -{ - FDK_METADATA_ERROR err = METADATA_OK; - - if (pAacMetaData==NULL) { - err = METADATA_INVALID_HANDLE; - } - else { - /* init struct */ - FDKmemclear(pAacMetaData, sizeof(AAC_METADATA)); - - if (hMetadata!=NULL) { - /* convert data */ - pAacMetaData->mpegDrc.drc_profile = hMetadata->drc_profile; - pAacMetaData->etsiAncData.comp_profile = hMetadata->comp_profile; - pAacMetaData->mpegDrc.drc_TargetRefLevel = hMetadata->drc_TargetRefLevel; - pAacMetaData->etsiAncData.comp_TargetRefLevel= hMetadata->comp_TargetRefLevel; - pAacMetaData->mpegDrc.prog_ref_level_present = hMetadata->prog_ref_level_present; - pAacMetaData->mpegDrc.prog_ref_level = dialnorm2progreflvl(hMetadata->prog_ref_level); - - pAacMetaData->centerMixLevel = hMetadata->centerMixLevel; - pAacMetaData->surroundMixLevel = hMetadata->surroundMixLevel; - pAacMetaData->WritePCEMixDwnIdx = hMetadata->PCE_mixdown_idx_present; - pAacMetaData->DmxLvl_On = hMetadata->ETSI_DmxLvl_present; - - pAacMetaData->etsiAncData.compression_on = 1; - - - if (nChannels == 2) { - pAacMetaData->dolbySurroundMode = hMetadata->dolbySurroundMode; /* dolby_surround_mode */ - } else { - pAacMetaData->dolbySurroundMode = 0; - } - - pAacMetaData->etsiAncData.timecode_coarse_status = 0; /* not yet supported - attention: Update GetEstMetadataBytesPerFrame() if enable this! */ - pAacMetaData->etsiAncData.timecode_fine_status = 0; /* not yet supported - attention: Update GetEstMetadataBytesPerFrame() if enable this! */ - - pAacMetaData->metadataMode = metadataMode; - } - else { - pAacMetaData->metadataMode = 0; /* there is no configuration available */ - } - } - - return err; -} - -INT FDK_MetadataEnc_GetDelay( - HANDLE_FDK_METADATA_ENCODER hMetadataEnc - ) -{ - INT delay = 0; - - if (hMetadataEnc!=NULL) { - delay = hMetadataEnc->nAudioDataDelay; - } - - return delay; -} - - diff --git a/libAACenc/src/metadata_main.h b/libAACenc/src/metadata_main.h deleted file mode 100644 index bfc8ae1..0000000 --- a/libAACenc/src/metadata_main.h +++ /dev/null @@ -1,224 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/********************** Fraunhofer IIS FDK AAC Encoder lib ****************** - - Author(s): V. Bacigalupo - Description: Metadata Encoder library interface functions - -******************************************************************************/ - -#ifndef _METADATA_MAIN_H -#define _METADATA_MAIN_H - - -/* Includes ******************************************************************/ -#include "aacenc_lib.h" -#include "aacenc.h" - - -/* Defines *******************************************************************/ - -/* Data Types ****************************************************************/ - -typedef enum { - METADATA_OK = 0x0000, /*!< No error happened. All fine. */ - METADATA_INVALID_HANDLE = 0x0020, /*!< Handle passed to function call was invalid. */ - METADATA_MEMORY_ERROR = 0x0021, /*!< Memory allocation failed. */ - METADATA_INIT_ERROR = 0x0040, /*!< General initialization error. */ - METADATA_ENCODE_ERROR = 0x0060 /*!< The encoding process was interrupted by an unexpected error. */ - -} FDK_METADATA_ERROR; - -/** - * Meta Data handle. - */ -typedef struct FDK_METADATA_ENCODER *HANDLE_FDK_METADATA_ENCODER; - - -/** - * \brief Open a Meta Data instance. - * - * \param phMetadataEnc A pointer to a Meta Data handle to be allocated. Initialized on return. - * - * \return - * - METADATA_OK, on succes. - * - METADATA_INVALID_HANDLE, METADATA_MEMORY_ERROR, on failure. - */ -FDK_METADATA_ERROR FDK_MetadataEnc_Open( - HANDLE_FDK_METADATA_ENCODER *phMetadataEnc - ); - - -/** - * \brief Initialize a Meta Data instance. - * - * \param hMetadataEnc Meta Data handle. - * \param resetStates Indication for full reset of all states. - * \param metadataMode Configures metat data output format (0,1,2). - * \param audioDelay Delay cause by the audio encoder. - * \param frameLength Number of samples to be processes within one frame. - * \param sampleRate Sampling rat in Hz of audio input signal. - * \param nChannels Number of audio input channels. - * \param channelMode Channel configuration which is used by the encoder. - * \param channelOrder Channel order of the input data. (WAV, MPEG) - * - * \return - * - METADATA_OK, on succes. - * - METADATA_INVALID_HANDLE, METADATA_INIT_ERROR, on failure. - */ -FDK_METADATA_ERROR FDK_MetadataEnc_Init( - HANDLE_FDK_METADATA_ENCODER hMetadataEnc, - const INT resetStates, - const INT metadataMode, - const INT audioDelay, - const UINT frameLength, - const UINT sampleRate, - const UINT nChannels, - const CHANNEL_MODE channelMode, - const CHANNEL_ORDER channelOrder - ); - - -/** - * \brief Calculate Meta Data processing. - * - * This function treats all step necessary for meta data processing. - * - Receive new meta data and make usable. - * - Calculate DRC compressor and extract meta data info. - * - Make meta data available for extern use. - * - Apply audio data and meta data delay compensation. - * - * \param hMetadataEnc Meta Data handle. - * \param pAudioSamples Pointer to audio input data. Existing function overwrites audio data with delayed audio samples. - * \param nAudioSamples Number of input audio samples to be prcessed. - * \param pMetadata Pointer to Metat Data input. - * \param ppMetaDataExtPayload Pointer to extension payload array. Filled on return. - * \param nMetaDataExtensions Pointer to variable to describe number of available extension payloads. Filled on return. - * \param matrix_mixdown_idx Pointer to variable for matrix mixdown coefficient. Filled on return. - * - * \return - * - METADATA_OK, on succes. - * - METADATA_INVALID_HANDLE, METADATA_ENCODE_ERROR, on failure. - */ -FDK_METADATA_ERROR FDK_MetadataEnc_Process( - HANDLE_FDK_METADATA_ENCODER hMetadataEnc, - INT_PCM * const pAudioSamples, - const INT nAudioSamples, - const AACENC_MetaData * const pMetadata, - AACENC_EXT_PAYLOAD ** ppMetaDataExtPayload, - UINT * nMetaDataExtensions, - INT * matrix_mixdown_idx - ); - - -/** - * \brief Close the Meta Data instance. - * - * Deallocate instance and free whole memory. - * - * \param phMetaData Pointer to the Meta Data handle to be deallocated. - * - * \return - * - METADATA_OK, on succes. - * - METADATA_INVALID_HANDLE, on failure. - */ -FDK_METADATA_ERROR FDK_MetadataEnc_Close( - HANDLE_FDK_METADATA_ENCODER *phMetaData - ); - - -/** - * \brief Get Meta Data Encoder delay. - * - * \param hMetadataEnc Meta Data Encoder handle. - * - * \return Delay caused by Meta Data module. - */ -INT FDK_MetadataEnc_GetDelay( - HANDLE_FDK_METADATA_ENCODER hMetadataEnc - ); - - -#endif /* _METADATA_MAIN_H */ - diff --git a/libAACenc/src/ms_stereo.cpp b/libAACenc/src/ms_stereo.cpp deleted file mode 100644 index 306d490..0000000 --- a/libAACenc/src/ms_stereo.cpp +++ /dev/null @@ -1,251 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M.Werner - contents/description: MS stereo processing - -******************************************************************************/ -#include "ms_stereo.h" - -#include "psy_const.h" - -/* static const float scaleMinThres = 1.0f; */ /* 0.75f for 3db boost */ - -void FDKaacEnc_MsStereoProcessing(PSY_DATA *RESTRICT psyData[(2)], - PSY_OUT_CHANNEL* psyOutChannel[2], - const INT *isBook, - INT *msDigest, /* output */ - INT *msMask, /* output */ - const INT sfbCnt, - const INT sfbPerGroup, - const INT maxSfbPerGroup, - const INT *sfbOffset) -{ - FIXP_DBL *sfbEnergyLeft = psyData[0]->sfbEnergy.Long; /* modified where msMask==1 */ - FIXP_DBL *sfbEnergyRight = psyData[1]->sfbEnergy.Long; /* modified where msMask==1 */ - const FIXP_DBL *sfbEnergyMid = psyData[0]->sfbEnergyMS.Long; - const FIXP_DBL *sfbEnergySide = psyData[1]->sfbEnergyMS.Long; - FIXP_DBL *sfbThresholdLeft = psyData[0]->sfbThreshold.Long; /* modified where msMask==1 */ - FIXP_DBL *sfbThresholdRight = psyData[1]->sfbThreshold.Long; /* modified where msMask==1 */ - - FIXP_DBL *sfbSpreadEnLeft = psyData[0]->sfbSpreadEnergy.Long; - FIXP_DBL *sfbSpreadEnRight = psyData[1]->sfbSpreadEnergy.Long; - - FIXP_DBL *sfbEnergyLeftLdData = psyOutChannel[0]->sfbEnergyLdData; /* modified where msMask==1 */ - FIXP_DBL *sfbEnergyRightLdData = psyOutChannel[1]->sfbEnergyLdData; /* modified where msMask==1 */ - FIXP_DBL *sfbEnergyMidLdData = psyData[0]->sfbEnergyMSLdData; - FIXP_DBL *sfbEnergySideLdData = psyData[1]->sfbEnergyMSLdData; - FIXP_DBL *sfbThresholdLeftLdData = psyOutChannel[0]->sfbThresholdLdData; /* modified where msMask==1 */ - FIXP_DBL *sfbThresholdRightLdData = psyOutChannel[1]->sfbThresholdLdData; /* modified where msMask==1 */ - - FIXP_DBL *mdctSpectrumLeft = psyData[0]->mdctSpectrum; /* modified where msMask==1 */ - FIXP_DBL *mdctSpectrumRight = psyData[1]->mdctSpectrum; /* modified where msMask==1 */ - - INT sfb,sfboffs, j; /* loop counters */ - FIXP_DBL pnlrLdData, pnmsLdData; - FIXP_DBL minThresholdLdData; - FIXP_DBL minThreshold; - INT useMS; - - INT msMaskTrueSomewhere = 0; /* to determine msDigest */ - INT numMsMaskFalse = 0; /* number of non-intensity bands where L/R coding is used */ - - for(sfb=0; sfb pnlr); -*/ - - /* we assume that scaleMinThres == 1.0f and we can drop it */ - minThresholdLdData = fixMin(sfbThresholdLeftLdData[sfb+sfboffs], sfbThresholdRightLdData[sfb+sfboffs]); - - /* pnlrLdData = sfbThresholdLeftLdData[sfb+sfboffs] - - max(sfbEnergyLeftLdData[sfb+sfboffs], sfbThresholdLeftLdData[sfb+sfboffs]) + - sfbThresholdRightLdData[sfb+sfboffs] - - max(sfbEnergyRightLdData[sfb+sfboffs], sfbThresholdRightLdData[sfb+sfboffs]); */ - tmp = fixMax(sfbEnergyLeftLdData[sfb+sfboffs], sfbThresholdLeftLdData[sfb+sfboffs]); - pnlrLdData = (sfbThresholdLeftLdData[sfb+sfboffs]>>1) - (tmp>>1); - pnlrLdData = pnlrLdData + (sfbThresholdRightLdData[sfb+sfboffs]>>1); - tmp = fixMax(sfbEnergyRightLdData[sfb+sfboffs], sfbThresholdRightLdData[sfb+sfboffs]); - pnlrLdData = pnlrLdData - (tmp>>1); - - /* pnmsLdData = minThresholdLdData - max(sfbEnergyMidLdData[sfb+sfboffs], minThresholdLdData) + - minThresholdLdData - max(sfbEnergySideLdData[sfb+sfboffs], minThresholdLdData); */ - tmp = fixMax(sfbEnergyMidLdData[sfb+sfboffs], minThresholdLdData); - pnmsLdData = minThresholdLdData - (tmp>>1); - tmp = fixMax(sfbEnergySideLdData[sfb+sfboffs], minThresholdLdData); - pnmsLdData = pnmsLdData - (tmp>>1); - useMS = (pnmsLdData > (pnlrLdData)); - - - if (useMS) { - msMask[sfb+sfboffs] = 1; - msMaskTrueSomewhere = 1; - for(j=sfbOffset[sfb+sfboffs]; j>1; - specR = mdctSpectrumRight[j]>>1; - mdctSpectrumLeft[j] = specL + specR; - mdctSpectrumRight[j] = specL - specR; - } - minThreshold = fixMin(sfbThresholdLeft[sfb+sfboffs], sfbThresholdRight[sfb+sfboffs]); - sfbThresholdLeft[sfb+sfboffs] = sfbThresholdRight[sfb+sfboffs] = minThreshold; - sfbThresholdLeftLdData[sfb+sfboffs] = sfbThresholdRightLdData[sfb+sfboffs] = minThresholdLdData; - sfbEnergyLeft[sfb+sfboffs] = sfbEnergyMid[sfb+sfboffs]; - sfbEnergyRight[sfb+sfboffs] = sfbEnergySide[sfb+sfboffs]; - sfbEnergyLeftLdData[sfb+sfboffs] = sfbEnergyMidLdData[sfb+sfboffs]; - sfbEnergyRightLdData[sfb+sfboffs] = sfbEnergySideLdData[sfb+sfboffs]; - - sfbSpreadEnLeft[sfb+sfboffs] = sfbSpreadEnRight[sfb+sfboffs] = - fixMin( sfbSpreadEnLeft[sfb+sfboffs], - sfbSpreadEnRight[sfb+sfboffs] ) >> 1; - - } - else { - msMask[sfb+sfboffs] = 0; - numMsMaskFalse++; - } /* useMS */ - } /* isBook */ - else { - /* keep mDigest from IS module */ - if (msMask[sfb+sfboffs]) { - msMaskTrueSomewhere = 1; - } - /* prohibit MS_MASK_ALL in combination with IS */ - numMsMaskFalse = 9; - } /* isBook */ - } /* sfboffs */ - } /* sfb */ - - - if(msMaskTrueSomewhere == 1) { - if ((numMsMaskFalse == 0) || ((numMsMaskFalse < maxSfbPerGroup) && (numMsMaskFalse < 9))) { - *msDigest = SI_MS_MASK_ALL; - /* loop through M/S bands; if msMask==0, set it to 1 and apply M/S */ - for (sfb = 0; sfb < sfbCnt; sfb += sfbPerGroup) { - for (sfboffs = 0; sfboffs < maxSfbPerGroup; sfboffs++) { - if (( (isBook == NULL) ? 1 : (isBook[sfb+sfboffs] == 0) ) && (msMask[sfb+sfboffs] == 0)) { - msMask[sfb+sfboffs] = 1; - /* apply M/S coding */ - for(j=sfbOffset[sfb+sfboffs]; j>1; - specR = mdctSpectrumRight[j]>>1; - mdctSpectrumLeft[j] = specL + specR; - mdctSpectrumRight[j] = specL - specR; - } - minThreshold = fixMin(sfbThresholdLeft[sfb+sfboffs], sfbThresholdRight[sfb+sfboffs]); - sfbThresholdLeft[sfb+sfboffs] = sfbThresholdRight[sfb+sfboffs] = minThreshold; - minThresholdLdData = fixMin(sfbThresholdLeftLdData[sfb+sfboffs], sfbThresholdRightLdData[sfb+sfboffs]); - sfbThresholdLeftLdData[sfb+sfboffs] = sfbThresholdRightLdData[sfb+sfboffs] = minThresholdLdData; - sfbEnergyLeft[sfb+sfboffs] = sfbEnergyMid[sfb+sfboffs]; - sfbEnergyRight[sfb+sfboffs] = sfbEnergySide[sfb+sfboffs]; - sfbEnergyLeftLdData[sfb+sfboffs] = sfbEnergyMidLdData[sfb+sfboffs]; - sfbEnergyRightLdData[sfb+sfboffs] = sfbEnergySideLdData[sfb+sfboffs]; - - sfbSpreadEnLeft[sfb+sfboffs] = sfbSpreadEnRight[sfb+sfboffs] = - fixMin( sfbSpreadEnLeft[sfb+sfboffs], - sfbSpreadEnRight[sfb+sfboffs] ) >> 1; - } - } - } - } else { - *msDigest = SI_MS_MASK_SOME; - } - } else { - *msDigest = SI_MS_MASK_NONE; - } -} diff --git a/libAACenc/src/ms_stereo.h b/libAACenc/src/ms_stereo.h deleted file mode 100644 index 2f3addb..0000000 --- a/libAACenc/src/ms_stereo.h +++ /dev/null @@ -1,107 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M.Werner - contents/description: MS stereo processing - -******************************************************************************/ - -#ifndef __MS_STEREO_H__ -#define __MS_STEREO_H__ - - -#include "interface.h" - -void FDKaacEnc_MsStereoProcessing(PSY_DATA *RESTRICT psyData[(2)], - PSY_OUT_CHANNEL* psyOutChannel[2], - const INT *isBook, - INT *msDigest, /* output */ - INT *msMask, /* output */ - const INT sfbCnt, - const INT sfbPerGroup, - const INT maxSfbPerGroup, - const INT *sfbOffset); - -#endif diff --git a/libAACenc/src/noisedet.cpp b/libAACenc/src/noisedet.cpp deleted file mode 100644 index f3c51de..0000000 --- a/libAACenc/src/noisedet.cpp +++ /dev/null @@ -1,228 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M. Lohwasser - contents/description: noisedet.c - Routines for Noise Detection - -******************************************************************************/ - -#include "noisedet.h" - -#include "aacenc_pns.h" -#include "pnsparam.h" - - -/***************************************************************************** - - functionname: FDKaacEnc_fuzzyIsSmaller - description: Fuzzy value calculation for "testVal is smaller than refVal" - returns: fuzzy value - input: test and ref Value, - low and high Lim - output: return fuzzy value - -*****************************************************************************/ -static FIXP_SGL FDKaacEnc_fuzzyIsSmaller( FIXP_DBL testVal, - FIXP_DBL refVal, - FIXP_DBL loLim, - FIXP_DBL hiLim ) -{ - if (refVal <= FL2FXCONST_DBL(0.0)) - return( FL2FXCONST_SGL(0.0f) ); - else if (testVal >= fMult((hiLim>>1)+(loLim>>1), refVal)) - return( FL2FXCONST_SGL(0.0f) ); - else return( (FIXP_SGL)MAXVAL_SGL ); -} - - - -/***************************************************************************** - - functionname: FDKaacEnc_noiseDetect - description: detect tonal sfb's; two tests - Powerdistribution: - sfb splittet in four regions, - compare the energy in all sections - PsychTonality: - compare tonality from chaosmeasure with reftonality - returns: - input: spectrum of one large mdct - number of sfb's - pointer to offset of sfb's - pointer to noiseFuzzyMeasure (modified) - noiseparams struct - pointer to sfb energies - pointer to tonality calculated in chaosmeasure - output: noiseFuzzy Measure - -*****************************************************************************/ - -void FDKaacEnc_noiseDetect(FIXP_DBL *RESTRICT mdctSpectrum, - INT *RESTRICT sfbMaxScaleSpec, - INT sfbActive, - const INT *RESTRICT sfbOffset, - FIXP_SGL *RESTRICT noiseFuzzyMeasure, - NOISEPARAMS *np, - FIXP_SGL *RESTRICT sfbtonality ) - -{ - int i, k, sfb, sfbWidth; - FIXP_SGL fuzzy, fuzzyTotal; - FIXP_DBL refVal, testVal; - - /***** Start detection phase *****/ - /* Start noise detection for each band based on a number of checks */ - for (sfb=0; sfbstartSfb || sfbWidth < np->minSfbWidth) { - noiseFuzzyMeasure[sfb] = FL2FXCONST_SGL(0.0f); - continue; - } - - if ( (np->detectionAlgorithmFlags & USE_POWER_DISTRIBUTION) && (fuzzyTotal > FL2FXCONST_SGL(0.5f)) ) { - FIXP_DBL fhelp1, fhelp2, fhelp3, fhelp4, maxVal, minVal; - INT leadingBits = fixMax(0,(sfbMaxScaleSpec[sfb] - 3)); /* max sfbWidth = 96/4 ; 2^5=32 => 5/2 = 3 (spc*spc) */ - - /* check power distribution in four regions */ - fhelp1 = fhelp2 = fhelp3 = fhelp4 = FL2FXCONST_DBL(0.0f); - k = sfbWidth >>2; /* Width of a quarter band */ - - for (i=sfbOffset[sfb]; ipowDistPSDcurve[sfb]); - - fuzzy = FDKaacEnc_fuzzyIsSmaller(testVal, /* 1/2 * maxValue * PSDcurve */ - refVal, /* 1 * minValue */ - FL2FXCONST_DBL(0.495), /* 1/2 * loLim (0.99f/2) */ - FL2FXCONST_DBL(0.505)); /* 1/2 * hiLim (1.01f/2) */ - - fuzzyTotal = fixMin(fuzzyTotal, fuzzy); - } - - if ( (np->detectionAlgorithmFlags & USE_PSYCH_TONALITY) && (fuzzyTotal > FL2FXCONST_SGL(0.5f)) ) { - /* Detection with tonality-value of psych. acoustic (here: 1 is tonal!)*/ - - testVal = FX_SGL2FX_DBL(sfbtonality[sfb])>>1; /* 1/2 * sfbTonality */ - refVal = np->refTonality; - - fuzzy = FDKaacEnc_fuzzyIsSmaller(testVal, - refVal, - FL2FXCONST_DBL(0.45f), /* 1/2 * loLim (0.9f/2) */ - FL2FXCONST_DBL(0.55f)); /* 1/2 * hiLim (1.1f/2) */ - - fuzzyTotal = fixMin(fuzzyTotal, fuzzy); - } - - - /* Output of final result */ - noiseFuzzyMeasure[sfb] = fuzzyTotal; - } -} diff --git a/libAACenc/src/noisedet.h b/libAACenc/src/noisedet.h deleted file mode 100644 index 8d5e365..0000000 --- a/libAACenc/src/noisedet.h +++ /dev/null @@ -1,108 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M. Lohwasser - contents/description: noisedet.h - -******************************************************************************/ - -#ifndef __NOISEDET_H -#define __NOISEDET_H - -#include "common_fix.h" - -#include "pnsparam.h" -#include "psy_data.h" - - -void FDKaacEnc_noiseDetect( FIXP_DBL *mdctSpectrum, - INT *sfbMaxScaleSpec, - INT sfbActive, - const INT *sfbOffset, - FIXP_SGL noiseFuzzyMeasure[], - NOISEPARAMS *np, - FIXP_SGL *sfbtonality ); - -#endif diff --git a/libAACenc/src/pns_func.h b/libAACenc/src/pns_func.h deleted file mode 100644 index efa44ef..0000000 --- a/libAACenc/src/pns_func.h +++ /dev/null @@ -1,150 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M. Lohwasser - contents/description: pns_func.h - -******************************************************************************/ - -#ifndef _PNS_FUNC_H -#define _PNS_FUNC_H - -#include "common_fix.h" - -#include "aacenc_pns.h" -#include "psy_data.h" - - - -AAC_ENCODER_ERROR FDKaacEnc_InitPnsConfiguration(PNS_CONFIG *pnsConf, - INT bitRate, - INT sampleRate, - INT usePns, - INT sfbCnt, - const INT *sfbOffset, - const INT numChan, - const INT isLC ); - -void FDKaacEnc_PnsDetect( PNS_CONFIG *pnsConf, - PNS_DATA *pnsData, - const INT lastWindowSequence, - const INT sfbActive, - const INT maxSfbPerGroup, - FIXP_DBL *sfbThresholdLdData, - const INT *sfbOffset, - FIXP_DBL *mdctSpectrum, - INT *sfbMaxScaleSpec, - FIXP_SGL *sfbtonality, - int tnsOrder, - INT tnsPredictionGain, - INT tnsActive, - FIXP_DBL *sfbEnergyLdData, - INT *noiseNrg ); - -void FDKaacEnc_CodePnsChannel( const INT sfbActive, - PNS_CONFIG *pnsConf, - INT *pnsFlag, - FIXP_DBL *sfbEnergy, - INT *noiseNrg, - FIXP_DBL *sfbThreshold ); - -void FDKaacEnc_PreProcessPnsChannelPair( const INT sfbActive, - FIXP_DBL *sfbEnergyLeft, - FIXP_DBL *sfbEnergyRight, - FIXP_DBL *sfbEnergyLeftLD, - FIXP_DBL *sfbEnergyRightLD, - FIXP_DBL *sfbEnergyMid, - PNS_CONFIG *pnsConfLeft, - PNS_DATA *pnsDataLeft, - PNS_DATA *pnsDataRight ); - -void FDKaacEnc_PostProcessPnsChannelPair( const INT sfbActive, - PNS_CONFIG *pnsConf, - PNS_DATA *pnsDataLeft, - PNS_DATA *pnsDataRight, - INT *msMask, - INT *msDigest ); - -#endif /* _PNS_FUNC_H */ diff --git a/libAACenc/src/pnsparam.cpp b/libAACenc/src/pnsparam.cpp deleted file mode 100644 index 9d59ddc..0000000 --- a/libAACenc/src/pnsparam.cpp +++ /dev/null @@ -1,311 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M.Lohwasser - contents/description: PNS parameters depending on bitrate and bandwidth - -******************************************************************************/ - -#include "pnsparam.h" -#include "psy_configuration.h" - -typedef struct { - SHORT startFreq; - /* Parameters for detection */ - FIXP_SGL refPower; - FIXP_SGL refTonality; - SHORT tnsGainThreshold; /* scaled by TNS_PREDGAIN_SCALE (=1000) */ - SHORT tnsPNSGainThreshold; /* scaled by TNS_PREDGAIN_SCALE (=1000) */ - FIXP_SGL gapFillThr; - SHORT minSfbWidth; - USHORT detectionAlgorithmFlags; -} PNS_INFO_TAB; - - -typedef struct { - ULONG brFrom; - ULONG brTo; - UCHAR S16000; - UCHAR S22050; - UCHAR S24000; - UCHAR S32000; - UCHAR S44100; - UCHAR S48000; -} AUTO_PNS_TAB; - -static const AUTO_PNS_TAB levelTable_mono[]= { - {0, 11999, 0, 1, 1, 1, 1, 1,}, - {12000, 19999, 0, 1, 1, 1, 1, 1,}, - {20000, 28999, 0, 2, 1, 1, 1, 1,}, - {29000, 40999, 0, 4, 4, 4, 2, 2,}, - {41000, 55999, 0, 9, 9, 7, 7, 7,}, - {56000, 61999, 0, 0, 0, 0, 9, 9,}, - {62000, 75999, 0, 0, 0, 0, 0, 0,}, - {76000, 92999, 0, 0, 0, 0, 0, 0,}, - {93000, 999999, 0, 0, 0, 0, 0, 0,}, -}; - -static const AUTO_PNS_TAB levelTable_stereo[]= { - {0, 11999, 0, 1, 1, 1, 1, 1,}, - {12000, 19999, 0, 3, 1, 1, 1, 1,}, - {20000, 28999, 0, 3, 3, 3, 2, 2,}, - {29000, 40999, 0, 7, 6, 6, 5, 5,}, - {41000, 55999, 0, 9, 9, 7, 7, 7,}, - {56000, 79999, 0, 0, 0, 0, 0, 0,}, - {80000, 99999, 0, 0, 0, 0, 0, 0,}, - {100000,999999, 0, 0, 0, 0, 0, 0,}, -}; - - -static const PNS_INFO_TAB pnsInfoTab[] = { -/*0 pns off */ -/*1*/ { 4000, FL2FXCONST_SGL(0.04), FL2FXCONST_SGL(0.06), 1150, 1200, FL2FXCONST_SGL(0.02), 8, - USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | USE_TNS_PNS /*| JUST_LONG_WINDOW*/ }, -/*2*/ { 4000, FL2FXCONST_SGL(0.04), FL2FXCONST_SGL(0.07), 1130, 1300, FL2FXCONST_SGL(0.05), 8, - USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | USE_TNS_PNS /*| JUST_LONG_WINDOW*/ }, -/*3*/ { 4100, FL2FXCONST_SGL(0.04), FL2FXCONST_SGL(0.07), 1100, 1400, FL2FXCONST_SGL(0.10), 8, - USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | USE_TNS_PNS /*| JUST_LONG_WINDOW*/ }, -/*4*/ { 4100, FL2FXCONST_SGL(0.03), FL2FXCONST_SGL(0.10), 1100, 1400, FL2FXCONST_SGL(0.15), 8, - USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | USE_TNS_PNS /*| JUST_LONG_WINDOW*/ }, -/*5*/ { 4300, FL2FXCONST_SGL(0.03), FL2FXCONST_SGL(0.10), 1100, 1400, FL2FXCONST_SGL(0.15), 8, - USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | USE_TNS_PNS | JUST_LONG_WINDOW }, -/*6*/ { 5000, FL2FXCONST_SGL(0.03), FL2FXCONST_SGL(0.10), 1100, 1400, FL2FXCONST_SGL(0.25), 8, - USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | USE_TNS_PNS | JUST_LONG_WINDOW }, -/*7*/ { 5500, FL2FXCONST_SGL(0.03), FL2FXCONST_SGL(0.12), 1100, 1400, FL2FXCONST_SGL(0.35), 8, - USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | USE_TNS_PNS | JUST_LONG_WINDOW }, -/*8*/ { 6000, FL2FXCONST_SGL(0.03), FL2FXCONST_SGL(0.12), 1080, 1400, FL2FXCONST_SGL(0.40), 8, - USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | USE_TNS_PNS | JUST_LONG_WINDOW }, -/*9*/ { 6000, FL2FXCONST_SGL(0.03), FL2FXCONST_SGL(0.14), 1070, 1400, FL2FXCONST_SGL(0.45), 8, - USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | USE_TNS_PNS | JUST_LONG_WINDOW }, -}; - -static const AUTO_PNS_TAB levelTable_lowComplexity[]= { - {0, 27999, 0, 0, 0, 0, 0, 0,}, - {28000, 31999, 0, 2, 2, 2, 2, 2,}, - {32000, 47999, 0, 3, 3, 3, 3, 3,}, - {48000, 48000, 0, 4, 4, 4, 4, 4,}, - {48001, 999999, 0, 0, 0, 0, 0, 0,}, -}; - -/* conversion of old LC tuning tables to new (LD enc) structure (only entries which are actually used were converted) */ -static const PNS_INFO_TAB pnsInfoTab_lowComplexity[] = { -/*0 pns off */ - /* DEFAULT parameter set */ -/*1*/ { 4100, FL2FXCONST_SGL(0.03), FL2FXCONST_SGL(0.16), 1100, 1400, FL2FXCONST_SGL(0.5), 16, - USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | USE_TNS_PNS | JUST_LONG_WINDOW }, -/*2*/ { 4100, FL2FXCONST_SGL(0.05), FL2FXCONST_SGL(0.10), 1410, 1400, FL2FXCONST_SGL(0.5), 16, - USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | USE_TNS_PNS | JUST_LONG_WINDOW }, -/*3*/ { 4100, FL2FXCONST_SGL(0.05), FL2FXCONST_SGL(0.10), 1100, 1400, FL2FXCONST_SGL(0.5), 16, - USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | USE_TNS_PNS | JUST_LONG_WINDOW }, - /* LOWSUBST -> PNS is used less often than with DEFAULT parameter set (for br: 48000 - 79999) */ -/*4*/ { 4100, FL2FXCONST_SGL(0.20), FL2FXCONST_SGL(0.10), 1410, 1400, FL2FXCONST_SGL(0.5), 16, - USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | USE_TNS_PNS | JUST_LONG_WINDOW }, -}; - -/**************************************************************************** - function to look up used pns level -****************************************************************************/ -int FDKaacEnc_lookUpPnsUse (int bitRate, int sampleRate, int numChan, const int isLC) { - - int hUsePns=0, size, i; - const AUTO_PNS_TAB *levelTable; - - if (isLC) { - levelTable = &levelTable_lowComplexity[0]; - size = sizeof(levelTable_lowComplexity); - } else - { /* (E)LD */ - levelTable = (numChan > 1) ? &levelTable_stereo[0] : &levelTable_mono[0]; - size = (numChan > 1) ? sizeof(levelTable_stereo) : sizeof(levelTable_mono); - } - - for(i = 0; i < (int) (size/sizeof(AUTO_PNS_TAB)); i++) { - if(((ULONG)bitRate >= levelTable[i].brFrom) && - ((ULONG)bitRate <= levelTable[i].brTo) ) - break; - } - - /* sanity check */ - if ((int)(sizeof(pnsInfoTab)/sizeof(PNS_INFO_TAB)) < i ) { - return (PNS_TABLE_ERROR); - } - - switch (sampleRate) { - case 16000: hUsePns = levelTable[i].S16000; break; - case 22050: hUsePns = levelTable[i].S22050; break; - case 24000: hUsePns = levelTable[i].S24000; break; - case 32000: hUsePns = levelTable[i].S32000; break; - case 44100: hUsePns = levelTable[i].S44100; break; - case 48000: hUsePns = levelTable[i].S48000; break; - default: - if (isLC) { - hUsePns = levelTable[i].S48000; - } - break; - } - - return (hUsePns); -} - - -/***************************************************************************** - - functionname: FDKaacEnc_GetPnsParam - description: Gets PNS parameters depending on bitrate and bandwidth - returns: error status - input: Noiseparams struct, bitrate, sampling rate, - number of sfb's, pointer to sfb offset - output: PNS parameters - -*****************************************************************************/ -AAC_ENCODER_ERROR FDKaacEnc_GetPnsParam(NOISEPARAMS *np, - INT bitRate, - INT sampleRate, - INT sfbCnt, - const INT *sfbOffset, - INT *usePns, - INT numChan, - const int isLC) - -{ - int i, hUsePns; - const PNS_INFO_TAB *pnsInfo; - - if (isLC) { - np->detectionAlgorithmFlags = IS_LOW_COMLEXITY; - pnsInfo = pnsInfoTab_lowComplexity; - } - else - { - np->detectionAlgorithmFlags = 0; - pnsInfo = pnsInfoTab; - } - - if (*usePns<=0) - return AAC_ENC_OK; - - /* new pns params */ - hUsePns = FDKaacEnc_lookUpPnsUse (bitRate, sampleRate, numChan, isLC); - if (hUsePns == 0) { - *usePns = 0; - return AAC_ENC_OK; - } - if (hUsePns == PNS_TABLE_ERROR) - return AAC_ENC_PNS_TABLE_ERROR; - - /* select correct row of tuning table */ - pnsInfo += hUsePns-1; - - np->startSfb = FDKaacEnc_FreqToBandWithRounding( pnsInfo->startFreq, - sampleRate, - sfbCnt, - sfbOffset ); - - np->detectionAlgorithmFlags |= pnsInfo->detectionAlgorithmFlags; - - np->refPower = FX_SGL2FX_DBL(pnsInfo->refPower); - np->refTonality = FX_SGL2FX_DBL(pnsInfo->refTonality); - np->tnsGainThreshold = pnsInfo->tnsGainThreshold; - np->tnsPNSGainThreshold = pnsInfo->tnsPNSGainThreshold; - np->minSfbWidth = pnsInfo->minSfbWidth; - - np->gapFillThr = (FIXP_SGL)pnsInfo->gapFillThr; - - /* assuming a constant dB/Hz slope in the signal's PSD curve, - the detection threshold needs to be corrected for the width of the band */ - for ( i = 0; i < (sfbCnt-1); i++) - { - INT qtmp, sfbWidth; - FIXP_DBL tmp; - - sfbWidth = sfbOffset[i+1]-sfbOffset[i]; - - tmp = fPow(np->refPower, 0, sfbWidth, DFRACT_BITS-1-5, &qtmp); - np->powDistPSDcurve[i] = (FIXP_SGL)((LONG)(scaleValue(tmp, qtmp) >> 16)); - } - np->powDistPSDcurve[sfbCnt] = np->powDistPSDcurve[sfbCnt-1]; - - return AAC_ENC_OK; -} diff --git a/libAACenc/src/pnsparam.h b/libAACenc/src/pnsparam.h deleted file mode 100644 index 08bb83e..0000000 --- a/libAACenc/src/pnsparam.h +++ /dev/null @@ -1,141 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M. Lohwasser - contents/description: PNS parameters depending on bitrate and bandwidth - -******************************************************************************/ - -#ifndef __PNSPARAM_H -#define __PNSPARAM_H - -#include "aacenc.h" -#include "common_fix.h" -#include "psy_const.h" - -#define NUM_PNSINFOTAB 4 -#define PNS_TABLE_ERROR -1 - -/* detection algorithm flags */ -#define USE_POWER_DISTRIBUTION (1<<0) -#define USE_PSYCH_TONALITY (1<<1) -#define USE_TNS_GAIN_THR (1<<2) -#define USE_TNS_PNS (1<<3) -#define JUST_LONG_WINDOW (1<<4) -/* additional algorithm flags */ -#define IS_LOW_COMLEXITY (1<<5) - -typedef struct -{ - /* PNS start band */ - short startSfb; - - /* detection algorithm flags */ - USHORT detectionAlgorithmFlags; - - /* Parameters for detection */ - FIXP_DBL refPower; - FIXP_DBL refTonality; - INT tnsGainThreshold; - INT tnsPNSGainThreshold; - INT minSfbWidth; - FIXP_SGL powDistPSDcurve[MAX_GROUPED_SFB]; - FIXP_SGL gapFillThr; -} NOISEPARAMS; - -int FDKaacEnc_lookUpPnsUse (int bitRate, int sampleRate, int numChan, const int isLC); - -/****** Definition of prototypes ******/ - -AAC_ENCODER_ERROR FDKaacEnc_GetPnsParam(NOISEPARAMS *np, - INT bitRate, - INT sampleRate, - INT sfbCnt, - const INT *sfbOffset, - INT *usePns, - INT numChan, - const INT isLC); - -#endif diff --git a/libAACenc/src/pre_echo_control.cpp b/libAACenc/src/pre_echo_control.cpp deleted file mode 100644 index 3dfd8ed..0000000 --- a/libAACenc/src/pre_echo_control.cpp +++ /dev/null @@ -1,170 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M.Werner - contents/description: Pre echo control - -******************************************************************************/ - -#include "pre_echo_control.h" -#include "psy_configuration.h" - -void FDKaacEnc_InitPreEchoControl(FIXP_DBL *RESTRICT pbThresholdNm1, - INT *calcPreEcho, - INT numPb, - FIXP_DBL *RESTRICT sfbPcmQuantThreshold, - INT *mdctScalenm1) -{ - *mdctScalenm1 = PCM_QUANT_THR_SCALE>>1; - - FDKmemcpy(pbThresholdNm1, sfbPcmQuantThreshold, numPb*sizeof(FIXP_DBL)); - - *calcPreEcho = 1; -} - - -void FDKaacEnc_PreEchoControl(FIXP_DBL *RESTRICT pbThresholdNm1, - INT calcPreEcho, - INT numPb, - INT maxAllowedIncreaseFactor, - FIXP_SGL minRemainingThresholdFactor, - FIXP_DBL *RESTRICT pbThreshold, - INT mdctScale, - INT *mdctScalenm1) -{ - int i; - FIXP_DBL tmpThreshold1, tmpThreshold2; - int scaling; - - /* If lastWindowSequence in previous frame was start- or stop-window, - skip preechocontrol calculation */ - if (calcPreEcho==0) { - /* copy thresholds to internal memory */ - FDKmemcpy(pbThresholdNm1, pbThreshold, numPb*sizeof(FIXP_DBL)); - *mdctScalenm1 = mdctScale; - return; - } - - if ( mdctScale > *mdctScalenm1 ) { - /* if current thresholds are downscaled more than the ones from the last block */ - scaling = 2*(mdctScale-*mdctScalenm1); - for(i = 0; i < numPb; i++) { - - /* multiplication with return data type fract ist equivalent to int multiplication */ - FDK_ASSERT(scaling>=0); - tmpThreshold1 = maxAllowedIncreaseFactor * (pbThresholdNm1[i]>>scaling); - tmpThreshold2 = fMult(minRemainingThresholdFactor, pbThreshold[i]); - - FIXP_DBL tmp = pbThreshold[i]; - - /* copy thresholds to internal memory */ - pbThresholdNm1[i] = tmp; - - tmp = fixMin(tmp, tmpThreshold1); - pbThreshold[i] = fixMax(tmp, tmpThreshold2); - } - } - else { - /* if thresholds of last block are more downscaled than the current ones */ - scaling = 2*(*mdctScalenm1-mdctScale); - for(i = 0; i < numPb; i++) { - - /* multiplication with return data type fract ist equivalent to int multiplication */ - tmpThreshold1 = (maxAllowedIncreaseFactor>>1) * pbThresholdNm1[i]; - tmpThreshold2 = fMult(minRemainingThresholdFactor, pbThreshold[i]); - - /* copy thresholds to internal memory */ - pbThresholdNm1[i] = pbThreshold[i]; - - FDK_ASSERT(scaling>=0); - if((pbThreshold[i]>>(scaling+1)) > tmpThreshold1) { - pbThreshold[i] = tmpThreshold1<<(scaling+1); - } - pbThreshold[i] = fixMax(pbThreshold[i], tmpThreshold2); - } - } - - *mdctScalenm1 = mdctScale; -} diff --git a/libAACenc/src/pre_echo_control.h b/libAACenc/src/pre_echo_control.h deleted file mode 100644 index 9224db0..0000000 --- a/libAACenc/src/pre_echo_control.h +++ /dev/null @@ -1,114 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M.Werner - contents/description: Pre echo control - -******************************************************************************/ - -#ifndef __PRE_ECHO_CONTROL_H -#define __PRE_ECHO_CONTROL_H - -#include "common_fix.h" - - -void FDKaacEnc_InitPreEchoControl(FIXP_DBL *pbThresholdnm1, - INT *calcPreEcho, - INT numPb, - FIXP_DBL *sfbPcmQuantThreshold, - INT *mdctScalenm1); - - -void FDKaacEnc_PreEchoControl(FIXP_DBL *pbThresholdNm1, - INT calcPreEcho, - INT numPb, - INT maxAllowedIncreaseFactor, - FIXP_SGL minRemainingThresholdFactor, - FIXP_DBL *pbThreshold, - INT mdctScale, - INT *mdctScalenm1); - -#endif - diff --git a/libAACenc/src/psy_configuration.cpp b/libAACenc/src/psy_configuration.cpp deleted file mode 100644 index 9a72c68..0000000 --- a/libAACenc/src/psy_configuration.cpp +++ /dev/null @@ -1,657 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M.Werner - contents/description: Psychoaccoustic configuration - -******************************************************************************/ - -#include "psy_configuration.h" -#include "adj_thr.h" -#include "aacEnc_rom.h" - -#include "genericStds.h" - -#include "FDK_trigFcts.h" - -typedef struct{ - LONG sampleRate; - const SFB_PARAM_LONG *paramLong; - const SFB_PARAM_SHORT *paramShort; -}SFB_INFO_TAB; - - -static const SFB_INFO_TAB sfbInfoTab[] = { - {8000, &p_FDKaacEnc_8000_long_1024, &p_FDKaacEnc_8000_short_128}, - {11025, &p_FDKaacEnc_11025_long_1024, &p_FDKaacEnc_11025_short_128}, - {12000, &p_FDKaacEnc_12000_long_1024, &p_FDKaacEnc_12000_short_128}, - {16000, &p_FDKaacEnc_16000_long_1024, &p_FDKaacEnc_16000_short_128}, - {22050, &p_FDKaacEnc_22050_long_1024, &p_FDKaacEnc_22050_short_128}, - {24000, &p_FDKaacEnc_24000_long_1024, &p_FDKaacEnc_24000_short_128}, - {32000, &p_FDKaacEnc_32000_long_1024, &p_FDKaacEnc_32000_short_128}, - {44100, &p_FDKaacEnc_44100_long_1024, &p_FDKaacEnc_44100_short_128}, - {48000, &p_FDKaacEnc_48000_long_1024, &p_FDKaacEnc_48000_short_128}, - {64000, &p_FDKaacEnc_64000_long_1024, &p_FDKaacEnc_64000_short_128}, - {88200, &p_FDKaacEnc_88200_long_1024, &p_FDKaacEnc_88200_short_128}, - {96000, &p_FDKaacEnc_96000_long_1024, &p_FDKaacEnc_96000_short_128} - -}; - -/* 22050 and 24000 Hz */ -static const SFB_PARAM_LONG p_22050_long_512 = { - 31, - { 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, - 4, 8, 8, 8, 12, 12, 12, 16, 20, 24, - 28, 32, 32, 32, 32, 32, 32, 32, 32, 32, - 32} -}; - -/* 32000 Hz */ -static const SFB_PARAM_LONG p_32000_long_512 = { - 37, - { 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, - 4, 4, 4, 4, 8, 8, 8, 8, 8, 12, - 12, 12, 12, 16, 16, 16, 20, 24, 24, 28, - 32, 32, 32, 32, 32, 32, 32} -}; - -/* 44100 Hz */ -static const SFB_PARAM_LONG p_44100_long_512 = { - 36, - {4, 4, 4, 4, 4, 4, 4, 4, 4, 4, - 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, - 12, 12, 12, 12, 16, 20, 24, 28, 32, 32, - 32, 32, 32, 32, 32, 52} -}; - -static const SFB_INFO_TAB sfbInfoTabLD512[] = { - { 8000, &p_22050_long_512, NULL}, - {11025, &p_22050_long_512, NULL}, - {12000, &p_22050_long_512, NULL}, - {16000, &p_22050_long_512, NULL}, - {22050, &p_22050_long_512, NULL}, - {24000, &p_22050_long_512, NULL}, - {32000, &p_32000_long_512, NULL}, - {44100, &p_44100_long_512, NULL}, - {48000, &p_44100_long_512, NULL}, - {64000, &p_44100_long_512, NULL}, - {88200, &p_44100_long_512, NULL}, - {96000, &p_44100_long_512, NULL}, - -}; - - -/* 22050 and 24000 Hz */ -static const SFB_PARAM_LONG p_22050_long_480 = { - 30, - { 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, - 4, 8, 8, 8, 12, 12, 12, 16, 20, 24, - 28, 32, 32, 32, 32, 32, 32, 32, 32, 32} -}; - -/* 32000 Hz */ -static const SFB_PARAM_LONG p_32000_long_480 = { - 37, - { 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, - 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, - 8, 8, 12, 12, 12, 16, 16, 20, 24, 32, - 32, 32, 32, 32, 32, 32, 32} -}; - -/* 44100 Hz */ -static const SFB_PARAM_LONG p_44100_long_480 = { - 35, - { 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, - 4, 4, 4, 4, 8, 8, 8, 8, 8, 12, - 12, 12, 12, 12, 16, 16, 24, 28, 32, 32, - 32, 32, 32, 32, 48} -}; - -static const SFB_INFO_TAB sfbInfoTabLD480[] = { - { 8000, &p_22050_long_480, NULL}, - {11025, &p_22050_long_480, NULL}, - {12000, &p_22050_long_480, NULL}, - {16000, &p_22050_long_480, NULL}, - {22050, &p_22050_long_480, NULL}, - {24000, &p_22050_long_480, NULL}, - {32000, &p_32000_long_480, NULL}, - {44100, &p_44100_long_480, NULL}, - {48000, &p_44100_long_480, NULL}, - {64000, &p_44100_long_480, NULL}, - {88200, &p_44100_long_480, NULL}, - {96000, &p_44100_long_480, NULL}, - -}; - -/* Fixed point precision definitions */ -#define Q_BARCVAL (25) - -static AAC_ENCODER_ERROR FDKaacEnc_initSfbTable(LONG sampleRate, INT blockType, INT granuleLength, INT *sfbOffset, INT *sfbCnt) -{ - INT i, specStartOffset = 0; - const UCHAR* sfbWidth = NULL; - const SFB_INFO_TAB *sfbInfo = NULL; - int size; - - /* - select table - */ - switch(granuleLength) { - case 1024: - case 960: - sfbInfo = sfbInfoTab; - size = (INT)(sizeof(sfbInfoTab)/sizeof(SFB_INFO_TAB)); - break; - case 512: - sfbInfo = sfbInfoTabLD512; - size = sizeof(sfbInfoTabLD512); - break; - case 480: - sfbInfo = sfbInfoTabLD480; - size = sizeof(sfbInfoTabLD480); - break; - default: - return AAC_ENC_INVALID_FRAME_LENGTH; - } - - for(i = 0; i < size; i++){ - if(sfbInfo[i].sampleRate == sampleRate){ - switch(blockType){ - case LONG_WINDOW: - case START_WINDOW: - case STOP_WINDOW: - sfbWidth = sfbInfo[i].paramLong->sfbWidth; - *sfbCnt = sfbInfo[i].paramLong->sfbCnt; - break; - case SHORT_WINDOW: - sfbWidth = sfbInfo[i].paramShort->sfbWidth; - *sfbCnt = sfbInfo[i].paramShort->sfbCnt; - granuleLength /= TRANS_FAC; - break; - } - break; - } - } - if (i == size) { - return AAC_ENC_UNSUPPORTED_SAMPLINGRATE; - } - - /* - calc sfb offsets - */ - for(i = 0; i < *sfbCnt; i++){ - sfbOffset[i] = specStartOffset; - specStartOffset += sfbWidth[i]; - if (specStartOffset >= granuleLength) { - i++; - break; - } - } - *sfbCnt = fixMin(i,*sfbCnt); - sfbOffset[*sfbCnt] = fixMin(specStartOffset,granuleLength); - - return AAC_ENC_OK; -} - - -/***************************************************************************** - - functionname: FDKaacEnc_BarcLineValue - description: Calculates barc value for one frequency line - returns: barc value of line - input: number of lines in transform, index of line to check, Fs - output: - -*****************************************************************************/ -static FIXP_DBL FDKaacEnc_BarcLineValue(INT noOfLines, INT fftLine, LONG samplingFreq) -{ - - FIXP_DBL FOURBY3EM4 = (FIXP_DBL)0x45e7b273; /* 4.0/3 * 0.0001 in q43 */ - FIXP_DBL PZZZ76 = (FIXP_DBL)0x639d5e4a; /* 0.00076 in q41 */ - FIXP_DBL ONE3P3 = (FIXP_DBL)0x35333333; /* 13.3 in q26 */ - FIXP_DBL THREEP5 = (FIXP_DBL)0x1c000000; /* 3.5 in q27 */ - FIXP_DBL INV480 = (FIXP_DBL)0x44444444; // 1/480 in q39 - - FIXP_DBL center_freq, x1, x2; - FIXP_DBL bvalFFTLine, atan1, atan2; - - /* Theoritical maximum of center_freq (samp_freq*0.5) is 96khz * 0.5 = 48000 */ - /* Theoritical maximum of x1 is 1.3333333e-4f * center_freq = 6.4, can keep in q28 */ - /* Theoritical maximum of x2 is 0.00076f * center_freq = 36.48, can keep in q25 */ - - center_freq = fftLine * samplingFreq; /* q11 or q8 */ - - switch (noOfLines) { - case 1024: - center_freq = center_freq << 2; /* q13 */ - break; - case 128: - center_freq = center_freq << 5; /* q13 */ - break; - case 512: - center_freq = (fftLine * samplingFreq) << 3; // q13 - break; - case 480: - center_freq = fMult(center_freq, INV480) << 4; // q13 - break; - default: - center_freq = (FIXP_DBL)0; - } - - x1 = fMult(center_freq, FOURBY3EM4); /* q13 * q43 - (DFRACT_BITS-1) = q25 */ - x2 = fMult(center_freq, PZZZ76) << 2; /* q13 * q41 - (DFRACT_BITS-1) + 2 = q25 */ - - atan1 = fixp_atan(x1); - atan2 = fixp_atan(x2); - - /* q25 (q26 * q30 - (DFRACT_BITS-1)) + q25 (q27 * q30 * q30) */ - bvalFFTLine = fMult(ONE3P3, atan2) + fMult(THREEP5, fMult(atan1, atan1)); - return(bvalFFTLine); - -} - -/* - do not consider energies below a certain input signal level, - i.e. of -96dB or 1 bit at 16 bit PCM resolution, - might need to be configurable to e.g. 24 bit PCM Input or a lower - resolution for low bit rates -*/ -static void FDKaacEnc_InitMinPCMResolution(int numPb, - int *pbOffset, - FIXP_DBL *sfbPCMquantThreshold) -{ - /* PCM_QUANT_NOISE = FDKpow(10.0f, - 20.f / 10.0f) * ABS_LOW * NORM_PCM_ENERGY * FDKpow(2,PCM_QUANT_THR_SCALE) */ - #define PCM_QUANT_NOISE ((FIXP_DBL)0x00547062) - - for( int i = 0; i < numPb; i++ ) { - sfbPCMquantThreshold[i] = (pbOffset[i+1] - pbOffset[i]) * PCM_QUANT_NOISE; - } -} - -static FIXP_DBL getMaskFactor( - const FIXP_DBL dbVal_fix, - const INT dbVal_e, - const FIXP_DBL ten_fix, - const INT ten_e - ) -{ - INT q_msk; - FIXP_DBL mask_factor; - - mask_factor = fPow(ten_fix, DFRACT_BITS-1-ten_e, -dbVal_fix, DFRACT_BITS-1-dbVal_e, &q_msk); - q_msk = fixMin(DFRACT_BITS-1,fixMax(-(DFRACT_BITS-1),q_msk)); - - if ( (q_msk>0) && (mask_factor>(FIXP_DBL)MAXVAL_DBL>>q_msk) ) { - mask_factor = (FIXP_DBL)MAXVAL_DBL; - } - else { - mask_factor = scaleValue(mask_factor, q_msk); - } - - return (mask_factor); -} - -static void FDKaacEnc_initSpreading(INT numPb, - FIXP_DBL *pbBarcValue, - FIXP_DBL *pbMaskLoFactor, - FIXP_DBL *pbMaskHiFactor, - FIXP_DBL *pbMaskLoFactorSprEn, - FIXP_DBL *pbMaskHiFactorSprEn, - const LONG bitrate, - const INT blockType) - -{ - INT i; - FIXP_DBL MASKLOWSPREN, MASKHIGHSPREN; - - FIXP_DBL MASKHIGH = (FIXP_DBL)0x30000000; /* 1.5 in q29 */ - FIXP_DBL MASKLOW = (FIXP_DBL)0x60000000; /* 3.0 in q29 */ - FIXP_DBL MASKLOWSPRENLONG = (FIXP_DBL)0x60000000; /* 3.0 in q29 */ - FIXP_DBL MASKHIGHSPRENLONG = (FIXP_DBL)0x40000000; /* 2.0 in q29 */ - FIXP_DBL MASKHIGHSPRENLONGLOWBR = (FIXP_DBL)0x30000000; /* 1.5 in q29 */ - FIXP_DBL MASKLOWSPRENSHORT = (FIXP_DBL)0x40000000; /* 2.0 in q29 */ - FIXP_DBL MASKHIGHSPRENSHORT = (FIXP_DBL)0x30000000; /* 1.5 in q29 */ - FIXP_DBL TEN = (FIXP_DBL)0x50000000; /* 10.0 in q27 */ - - if (blockType != SHORT_WINDOW) - { - MASKLOWSPREN = MASKLOWSPRENLONG; - MASKHIGHSPREN = (bitrate>20000)?MASKHIGHSPRENLONG:MASKHIGHSPRENLONGLOWBR; - } - else - { - MASKLOWSPREN = MASKLOWSPRENSHORT; - MASKHIGHSPREN = MASKHIGHSPRENSHORT; - } - - for(i=0; i 0) - { - pbMaskHiFactor[i] = getMaskFactor( - fMult(MASKHIGH, (pbBarcValue[i] - pbBarcValue[i-1])), 23, - TEN, 27); - - pbMaskLoFactor[i-1] = getMaskFactor( - fMult(MASKLOW, (pbBarcValue[i] - pbBarcValue[i-1])), 23, - TEN, 27); - - pbMaskHiFactorSprEn[i] = getMaskFactor( - fMult(MASKHIGHSPREN, (pbBarcValue[i] - pbBarcValue[i-1])), 23, - TEN, 27); - - pbMaskLoFactorSprEn[i-1] = getMaskFactor( - fMult(MASKLOWSPREN, (pbBarcValue[i] - pbBarcValue[i-1])), 23, - TEN, 27); - } - else - { - pbMaskHiFactor[i] = (FIXP_DBL)0; - pbMaskLoFactor[numPb-1] = (FIXP_DBL)0; - pbMaskHiFactorSprEn[i] = (FIXP_DBL)0; - pbMaskLoFactorSprEn[numPb-1] = (FIXP_DBL)0; - } - } -} - -static void FDKaacEnc_initBarcValues(INT numPb, - INT *pbOffset, - INT numLines, - INT samplingFrequency, - FIXP_DBL *pbBval) -{ - INT i; - FIXP_DBL MAX_BARC = (FIXP_DBL)0x30000000; /* 24.0 in q25 */ - - for(i=0; i> 1) + (v2 >> 1); - pbBval[i] = fixMin(cur_bark, MAX_BARC); - } -} - -static void FDKaacEnc_initMinSnr(const LONG bitrate, - const LONG samplerate, - const INT numLines, - const INT *sfbOffset, - const INT sfbActive, - const INT blockType, - FIXP_DBL *sfbMinSnrLdData) -{ - INT sfb; - - /* Fix conversion variables */ - INT qbfac, qperwin, qdiv, qpeprt_const, qpeprt; - INT qtmp, qsnr, sfbWidth; - - FIXP_DBL MAX_BARC = (FIXP_DBL)0x30000000; /* 24.0 in q25 */ - FIXP_DBL MAX_BARCP1 = (FIXP_DBL)0x32000000; /* 25.0 in q25 */ - FIXP_DBL BITS2PEFAC = (FIXP_DBL)0x4b851eb8; /* 1.18 in q30 */ - FIXP_DBL PERS2P4 = (FIXP_DBL)0x624dd2f2; /* 0.024 in q36 */ - FIXP_DBL ONEP5 = (FIXP_DBL)0x60000000; /* 1.5 in q30 */ - FIXP_DBL MAX_SNR = (FIXP_DBL)0x33333333; /* 0.8 in q30 */ - FIXP_DBL MIN_SNR = (FIXP_DBL)0x003126e9; /* 0.003 in q30 */ - - FIXP_DBL barcFactor, pePerWindow, pePart, barcWidth; - FIXP_DBL pePart_const, tmp, snr, one_qsnr, one_point5; - - /* relative number of active barks */ - barcFactor = fDivNorm(fixMin(FDKaacEnc_BarcLineValue(numLines, sfbOffset[sfbActive], samplerate), MAX_BARC), - MAX_BARCP1, &qbfac); - - qbfac = DFRACT_BITS-1-qbfac; - - pePerWindow = fDivNorm(bitrate, samplerate, &qperwin); - qperwin = DFRACT_BITS-1-qperwin; - pePerWindow = fMult(pePerWindow, BITS2PEFAC); qperwin = qperwin + 30 - (DFRACT_BITS-1); - pePerWindow = fMult(pePerWindow, PERS2P4); qperwin = qperwin + 36 - (DFRACT_BITS-1); - - switch (numLines) { - case 1024: - qperwin = qperwin - 10; - break; - case 128: - qperwin = qperwin - 7; - break; - case 512: - qperwin = qperwin - 9; - break; - case 480: - qperwin = qperwin - 9; - pePerWindow = fMult(pePerWindow, FL2FXCONST_DBL(480.f/512.f)); - break; - } - - /* for short blocks it is assumed that more bits are available */ - if (blockType == SHORT_WINDOW) - { - pePerWindow = fMult(pePerWindow, ONEP5); - qperwin = qperwin + 30 - (DFRACT_BITS-1); - } - pePart_const = fDivNorm(pePerWindow, barcFactor, &qdiv); qpeprt_const = qperwin - qbfac + DFRACT_BITS-1-qdiv; - - for (sfb = 0; sfb < sfbActive; sfb++) - { - barcWidth = FDKaacEnc_BarcLineValue(numLines, sfbOffset[sfb+1], samplerate) - - FDKaacEnc_BarcLineValue(numLines, sfbOffset[sfb], samplerate); - - /* adapt to sfb bands */ - pePart = fMult(pePart_const, barcWidth); qpeprt = qpeprt_const + 25 - (DFRACT_BITS-1); - - /* pe -> snr calculation */ - sfbWidth = (sfbOffset[sfb+1] - sfbOffset[sfb]); - pePart = fDivNorm(pePart, sfbWidth, &qdiv); qpeprt += DFRACT_BITS-1-qdiv; - - tmp = f2Pow(pePart, DFRACT_BITS-1-qpeprt, &qtmp); - qtmp = DFRACT_BITS-1-qtmp; - - /* Subtract 1.5 */ - qsnr = fixMin(qtmp, 30); - tmp = tmp >> (qtmp - qsnr); - - if((30+1-qsnr) > (DFRACT_BITS-1)) - one_point5 = (FIXP_DBL)0; - else - one_point5 = (FIXP_DBL)(ONEP5 >> (30+1-qsnr)); - - snr = (tmp>>1) - (one_point5); qsnr -= 1; - - /* max(snr, 1.0) */ - if(qsnr > 0) - one_qsnr = (FIXP_DBL)(1 << qsnr); - else - one_qsnr = (FIXP_DBL)0; - - snr = fixMax(one_qsnr, snr); - - /* 1/snr */ - snr = fDivNorm(one_qsnr, snr, &qsnr); - qsnr = DFRACT_BITS-1-qsnr; - snr = (qsnr > 30)? (snr>>(qsnr-30)):snr; - - /* upper limit is -1 dB */ - snr = (snr > MAX_SNR) ? MAX_SNR : snr; - - /* lower limit is -25 dB */ - snr = (snr < MIN_SNR) ? MIN_SNR : snr; - snr = snr << 1; - - sfbMinSnrLdData[sfb] = CalcLdData(snr); - } -} - -AAC_ENCODER_ERROR FDKaacEnc_InitPsyConfiguration(INT bitrate, - INT samplerate, - INT bandwidth, - INT blocktype, - INT granuleLength, - INT useIS, - PSY_CONFIGURATION *psyConf, - FB_TYPE filterbank) -{ - AAC_ENCODER_ERROR ErrorStatus; - INT sfb; - FIXP_DBL sfbBarcVal[MAX_SFB]; - const INT frameLengthLong = granuleLength; - const INT frameLengthShort = granuleLength/TRANS_FAC; - - FDKmemclear(psyConf, sizeof(PSY_CONFIGURATION)); - psyConf->granuleLength = granuleLength; - psyConf->filterbank = filterbank; - - psyConf->allowIS = (useIS) && ( (bitrate/bandwidth) < 5 ); - - /* init sfb table */ - ErrorStatus = FDKaacEnc_initSfbTable(samplerate,blocktype,granuleLength,psyConf->sfbOffset,&psyConf->sfbCnt); - if (ErrorStatus != AAC_ENC_OK) - return ErrorStatus; - - /* calculate barc values for each pb */ - FDKaacEnc_initBarcValues(psyConf->sfbCnt, - psyConf->sfbOffset, - psyConf->sfbOffset[psyConf->sfbCnt], - samplerate, - sfbBarcVal); - - FDKaacEnc_InitMinPCMResolution(psyConf->sfbCnt, - psyConf->sfbOffset, - psyConf->sfbPcmQuantThreshold); - - /* calculate spreading function */ - FDKaacEnc_initSpreading(psyConf->sfbCnt, - sfbBarcVal, - psyConf->sfbMaskLowFactor, - psyConf->sfbMaskHighFactor, - psyConf->sfbMaskLowFactorSprEn, - psyConf->sfbMaskHighFactorSprEn, - bitrate, - blocktype); - - /* init ratio */ - - psyConf->maxAllowedIncreaseFactor = 2; /* integer */ - psyConf->minRemainingThresholdFactor = (FIXP_SGL)0x0148; /* FL2FXCONST_SGL(0.01f); */ /* fract */ - - psyConf->clipEnergy = (FIXP_DBL)0x773593ff; /* FL2FXCONST_DBL(1.0e9*NORM_PCM_ENERGY); */ - - if (blocktype!=SHORT_WINDOW) { - psyConf->lowpassLine = (INT)((2*bandwidth*frameLengthLong)/samplerate); - psyConf->lowpassLineLFE = LFE_LOWPASS_LINE; - } - else { - psyConf->lowpassLine = (INT)((2*bandwidth*frameLengthShort)/samplerate); - psyConf->lowpassLineLFE = 0; /* LFE only in lonf blocks */ - /* psyConf->clipEnergy /= (TRANS_FAC * TRANS_FAC); */ - psyConf->clipEnergy >>= 6; - } - - for (sfb = 0; sfb < psyConf->sfbCnt; sfb++){ - if (psyConf->sfbOffset[sfb] >= psyConf->lowpassLine) - break; - } - psyConf->sfbActive = FDKmax(sfb, 1); - - for (sfb = 0; sfb < psyConf->sfbCnt; sfb++){ - if (psyConf->sfbOffset[sfb] >= psyConf->lowpassLineLFE) - break; - } - psyConf->sfbActiveLFE = sfb; - psyConf->sfbActive = FDKmax(psyConf->sfbActive, psyConf->sfbActiveLFE); - - /* calculate minSnr */ - FDKaacEnc_initMinSnr(bitrate, - samplerate, - psyConf->sfbOffset[psyConf->sfbCnt], - psyConf->sfbOffset, - psyConf->sfbActive, - blocktype, - psyConf->sfbMinSnrLdData); - - return AAC_ENC_OK; -} - diff --git a/libAACenc/src/psy_configuration.h b/libAACenc/src/psy_configuration.h deleted file mode 100644 index 3629246..0000000 --- a/libAACenc/src/psy_configuration.h +++ /dev/null @@ -1,165 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M.Werner - contents/description: Psychoaccoustic configuration - -******************************************************************************/ - -#ifndef _PSY_CONFIGURATION_H -#define _PSY_CONFIGURATION_H - - -#include "aacenc.h" -#include "common_fix.h" - -#include "psy_const.h" -#include "aacenc_tns.h" -#include "aacenc_pns.h" - -#define THR_SHIFTBITS 4 -#define PCM_QUANT_THR_SCALE 16 - -#define C_RATIO (FIXP_DBL)0x02940a10 /* FL2FXCONST_DBL(0.001258925f) << THR_SHIFTBITS; */ /* pow(10.0f, -(29.0f/10.0f)) */ - -typedef struct{ - - INT sfbCnt; /* number of existing sf bands */ - INT sfbActive; /* number of sf bands containing energy after lowpass */ - INT sfbActiveLFE; - INT sfbOffset[MAX_SFB+1]; - - INT filterbank; /* LC, LD or ELD */ - - FIXP_DBL sfbPcmQuantThreshold[MAX_SFB]; - - INT maxAllowedIncreaseFactor; /* preecho control */ - FIXP_SGL minRemainingThresholdFactor; - - INT lowpassLine; - INT lowpassLineLFE; - FIXP_DBL clipEnergy; /* for level dependend tmn */ - - FIXP_DBL sfbMaskLowFactor[MAX_SFB]; - FIXP_DBL sfbMaskHighFactor[MAX_SFB]; - - FIXP_DBL sfbMaskLowFactorSprEn[MAX_SFB]; - FIXP_DBL sfbMaskHighFactorSprEn[MAX_SFB]; - - FIXP_DBL sfbMinSnrLdData[MAX_SFB]; /* minimum snr (formerly known as bmax) */ - - TNS_CONFIG tnsConf; - PNS_CONFIG pnsConf; - - INT granuleLength; - INT allowIS; - -}PSY_CONFIGURATION; - - -typedef struct{ - UCHAR sfbCnt; /* Number of scalefactor bands */ - UCHAR sfbWidth[MAX_SFB_LONG]; /* Width of scalefactor bands for long blocks */ -}SFB_PARAM_LONG; - -typedef struct{ - UCHAR sfbCnt; /* Number of scalefactor bands */ - UCHAR sfbWidth[MAX_SFB_SHORT]; /* Width of scalefactor bands for short blocks */ -}SFB_PARAM_SHORT; - - -AAC_ENCODER_ERROR FDKaacEnc_InitPsyConfiguration(INT bitrate, - INT samplerate, - INT bandwidth, - INT blocktype, - INT granuleLength, - INT useIS, - PSY_CONFIGURATION *psyConf, - FB_TYPE filterbank); - -#endif /* _PSY_CONFIGURATION_H */ - - - diff --git a/libAACenc/src/psy_const.h b/libAACenc/src/psy_const.h deleted file mode 100644 index d9c9f43..0000000 --- a/libAACenc/src/psy_const.h +++ /dev/null @@ -1,160 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M.Werner - contents/description: Global psychoaccoustic constants - -******************************************************************************/ -#ifndef _PSYCONST_H -#define _PSYCONST_H - - -#define TRUE 1 -#define FALSE 0 - - #define TRANS_FAC 8 /* encoder short long ratio */ - -#define FRAME_MAXLEN_SHORT ((1024)/TRANS_FAC) -#define FRAME_LEN_SHORT_128 ((1024)/TRANS_FAC) - -/* Filterbank type*/ -enum FB_TYPE { - FB_LC = 0, - FB_LD = 1, - FB_ELD = 2 -}; - -/* Block types */ -#define N_BLOCKTYPES 6 -enum -{ - LONG_WINDOW = 0, - START_WINDOW, - SHORT_WINDOW, - STOP_WINDOW, - _LOWOV_WINDOW, /* Do not use this block type out side of block_switch.cpp */ - WRONG_WINDOW -}; - -/* Window shapes */ -enum -{ - SINE_WINDOW = 0, - KBD_WINDOW = 1, - LOL_WINDOW = 2 /* Low OverLap window shape for AAC-LD */ -}; - -/* - MS stuff -*/ -enum -{ - SI_MS_MASK_NONE = 0, - SI_MS_MASK_SOME = 1, - SI_MS_MASK_ALL = 2 -}; - - - #define MAX_NO_OF_GROUPS 4 - #define MAX_SFB_LONG 51 /* 51 for a memory optimized implementation, maybe 64 for convenient debugging */ - #define MAX_SFB_SHORT 15 /* 15 for a memory optimized implementation, maybe 16 for convenient debugging */ - -#define MAX_SFB (MAX_SFB_SHORT > MAX_SFB_LONG ? MAX_SFB_SHORT : MAX_SFB_LONG) /* = 51 */ -#define MAX_GROUPED_SFB (MAX_NO_OF_GROUPS*MAX_SFB_SHORT > MAX_SFB_LONG ? \ - MAX_NO_OF_GROUPS*MAX_SFB_SHORT : MAX_SFB_LONG) /* = 60 */ - -#define MAX_INPUT_BUFFER_SIZE (2*(1024)) /* 2048 */ - - -#define PCM_LEVEL 1.0f -#define NORM_PCM (PCM_LEVEL/32768.0f) -#define NORM_PCM_ENERGY (NORM_PCM*NORM_PCM) -#define LOG_NORM_PCM -15 - -#define TNS_PREDGAIN_SCALE (1000) - -#define LFE_LOWPASS_LINE 12 - -#endif /* _PSYCONST_H */ diff --git a/libAACenc/src/psy_data.h b/libAACenc/src/psy_data.h deleted file mode 100644 index 7183955..0000000 --- a/libAACenc/src/psy_data.h +++ /dev/null @@ -1,152 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M.Werner - contents/description: Psychoaccoustic data - -******************************************************************************/ - -#ifndef _PSY_DATA_H -#define _PSY_DATA_H - - -#include "block_switch.h" - -/* Be careful with MAX_SFB_LONG as length of the .Long arrays. - * sfbEnergy.Long and sfbEnergyMS.Long and sfbThreshold.Long are used as a temporary storage for the regrouped - * short energies and thresholds between FDKaacEnc_groupShortData() and BuildInterface() in FDKaacEnc_psyMain(). - * The space required for this is MAX_GROUPED_SFB ( = MAX_NO_OF_GROUPS*MAX_SFB_SHORT ). - * However, this is not important if unions are used (which is not possible with pfloat). */ - -typedef shouldBeUnion{ - FIXP_DBL Long[MAX_GROUPED_SFB]; - FIXP_DBL Short[TRANS_FAC][MAX_SFB_SHORT]; -}SFB_THRESHOLD; - -typedef shouldBeUnion{ - FIXP_DBL Long[MAX_GROUPED_SFB]; - FIXP_DBL Short[TRANS_FAC][MAX_SFB_SHORT]; -}SFB_ENERGY; - -typedef shouldBeUnion{ - FIXP_DBL Long[MAX_GROUPED_SFB]; - FIXP_DBL Short[TRANS_FAC][MAX_SFB_SHORT]; -}SFB_LD_ENERGY; - -typedef shouldBeUnion{ - INT Long[MAX_GROUPED_SFB]; - INT Short[TRANS_FAC][MAX_SFB_SHORT]; -}SFB_MAX_SCALE; - - -typedef struct{ - INT_PCM* psyInputBuffer; - FIXP_DBL overlapAddBuffer[1024]; - - BLOCK_SWITCHING_CONTROL blockSwitchingControl; /* block switching */ - FIXP_DBL sfbThresholdnm1[MAX_SFB]; /* FDKaacEnc_PreEchoControl */ - INT mdctScalenm1; /* scale of last block's mdct (FDKaacEnc_PreEchoControl) */ - INT calcPreEcho; - INT isLFE; -}PSY_STATIC; - - -typedef struct{ - FIXP_DBL *mdctSpectrum; - SFB_THRESHOLD sfbThreshold; /* adapt */ - SFB_ENERGY sfbEnergy; /* sfb energies */ - SFB_LD_ENERGY sfbEnergyLdData; /* sfb energies in ldData format */ - SFB_MAX_SCALE sfbMaxScaleSpec; - SFB_ENERGY sfbEnergyMS; /* mid/side sfb energies */ - FIXP_DBL sfbEnergyMSLdData[MAX_GROUPED_SFB]; /* mid/side sfb energies in ldData format */ - SFB_ENERGY sfbSpreadEnergy; - INT mdctScale; /* exponent of data in mdctSpectrum */ - INT groupedSfbOffset[MAX_GROUPED_SFB+1]; - INT sfbActive; - INT lowpassLine; -}PSY_DATA; - - -#endif /* _PSY_DATA_H */ diff --git a/libAACenc/src/psy_main.cpp b/libAACenc/src/psy_main.cpp deleted file mode 100644 index 446c894..0000000 --- a/libAACenc/src/psy_main.cpp +++ /dev/null @@ -1,1387 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M.Werner - contents/description: Psychoaccoustic major function block - -******************************************************************************/ - -#include "psy_const.h" - -#include "block_switch.h" -#include "transform.h" -#include "spreading.h" -#include "pre_echo_control.h" -#include "band_nrg.h" -#include "psy_configuration.h" -#include "psy_data.h" -#include "ms_stereo.h" -#include "interface.h" -#include "psy_main.h" -#include "grp_data.h" -#include "tns_func.h" -#include "pns_func.h" -#include "tonality.h" -#include "aacEnc_ram.h" -#include "intensity.h" - - - -/* blending to reduce gibbs artifacts */ -#define FADE_OUT_LEN 6 -static const FIXP_DBL fadeOutFactor[FADE_OUT_LEN] = {1840644096, 1533870080, 1227096064, 920322048, 613548032, 306774016}; - -/* forward definitions */ - - -/***************************************************************************** - - functionname: FDKaacEnc_PsyNew - description: allocates memory for psychoacoustic - returns: an error code - input: pointer to a psych handle - -*****************************************************************************/ -AAC_ENCODER_ERROR FDKaacEnc_PsyNew(PSY_INTERNAL **phpsy, - const INT nElements, - const INT nChannels - ,UCHAR *dynamic_RAM - ) -{ - AAC_ENCODER_ERROR ErrorStatus; - PSY_INTERNAL *hPsy; - INT i; - - hPsy = GetRam_aacEnc_PsyInternal(); - *phpsy = hPsy; - if (hPsy == NULL) { - ErrorStatus = AAC_ENC_NO_MEMORY; - goto bail; - } - - for (i=0; ipsyElement[i] = GetRam_aacEnc_PsyElement(i); - if (hPsy->psyElement[i] == NULL) { - ErrorStatus = AAC_ENC_NO_MEMORY; - goto bail; - } - } - - for (i=0; ipStaticChannels[i] = GetRam_aacEnc_PsyStatic(i); - if (hPsy->pStaticChannels[i]==NULL) { - ErrorStatus = AAC_ENC_NO_MEMORY; - goto bail; - } - /* AUDIO INPUT BUFFER */ - hPsy->pStaticChannels[i]->psyInputBuffer = GetRam_aacEnc_PsyInputBuffer(i); - if (hPsy->pStaticChannels[i]->psyInputBuffer==NULL) { - ErrorStatus = AAC_ENC_NO_MEMORY; - goto bail; - } - } - - /* reusable psych memory */ - hPsy->psyDynamic = GetRam_aacEnc_PsyDynamic(0, dynamic_RAM); - - return AAC_ENC_OK; - -bail: - FDKaacEnc_PsyClose(phpsy, NULL); - - return ErrorStatus; -} - -/***************************************************************************** - - functionname: FDKaacEnc_PsyOutNew - description: allocates memory for psyOut struc - returns: an error code - input: pointer to a psych handle - -*****************************************************************************/ -AAC_ENCODER_ERROR FDKaacEnc_PsyOutNew(PSY_OUT **phpsyOut, - const INT nElements, - const INT nChannels, - const INT nSubFrames - ,UCHAR *dynamic_RAM - ) -{ - AAC_ENCODER_ERROR ErrorStatus; - int n, i; - int elInc = 0, chInc = 0; - - for (n=0; npPsyOutChannels[i] = GetRam_aacEnc_PsyOutChannel(chInc++); - } - - for (i=0; ipsyOutElement[i] = GetRam_aacEnc_PsyOutElements(elInc++); - if (phpsyOut[n]->psyOutElement[i] == NULL) { - ErrorStatus = AAC_ENC_NO_MEMORY; - goto bail; - } - } - } /* nSubFrames */ - - return AAC_ENC_OK; - -bail: - FDKaacEnc_PsyClose(NULL, phpsyOut); - return ErrorStatus; -} - - -AAC_ENCODER_ERROR FDKaacEnc_psyInitStates(PSY_INTERNAL *hPsy, - PSY_STATIC* psyStatic, - AUDIO_OBJECT_TYPE audioObjectType) -{ - /* init input buffer */ - FDKmemclear(psyStatic->psyInputBuffer, MAX_INPUT_BUFFER_SIZE*sizeof(INT_PCM)); - - FDKaacEnc_InitBlockSwitching(&psyStatic->blockSwitchingControl, - isLowDelay(audioObjectType) - ); - - return AAC_ENC_OK; -} - - -AAC_ENCODER_ERROR FDKaacEnc_psyInit(PSY_INTERNAL *hPsy, - PSY_OUT **phpsyOut, - const INT nSubFrames, - const INT nMaxChannels, - const AUDIO_OBJECT_TYPE audioObjectType, - CHANNEL_MAPPING *cm) -{ - AAC_ENCODER_ERROR ErrorStatus = AAC_ENC_OK; - int i, ch, n, chInc = 0, resetChannels = 3; - - if ( (nMaxChannels>2) && (cm->nChannels==2) ) { - chInc = 1; - FDKaacEnc_psyInitStates(hPsy, hPsy->pStaticChannels[0], audioObjectType); - } - - if ( (nMaxChannels==2) ) { - resetChannels = 0; - } - - for (i=0; inElements; i++) { - for (ch=0; chelInfo[i].nChannelsInEl; ch++) { - if (cm->elInfo[i].elType!=ID_LFE) { - hPsy->psyElement[i]->psyStatic[ch] = hPsy->pStaticChannels[chInc]; - if (chInc>=resetChannels) { - FDKaacEnc_psyInitStates(hPsy, hPsy->psyElement[i]->psyStatic[ch], audioObjectType); - } - hPsy->psyElement[i]->psyStatic[ch]->isLFE = 0; - } - else { - hPsy->psyElement[i]->psyStatic[ch] = hPsy->pStaticChannels[nMaxChannels-1]; - hPsy->psyElement[i]->psyStatic[ch]->isLFE = 1; - } - chInc++; - } - } - - for (n=0; nnElements; i++) { - for (ch=0; chelInfo[i].nChannelsInEl; ch++) { - phpsyOut[n]->psyOutElement[i]->psyOutChannel[ch] = phpsyOut[n]->pPsyOutChannels[chInc++]; - } - } - } - - return ErrorStatus; -} - - -/***************************************************************************** - - functionname: FDKaacEnc_psyMainInit - description: initializes psychoacoustic - returns: an error code - -*****************************************************************************/ - -AAC_ENCODER_ERROR FDKaacEnc_psyMainInit(PSY_INTERNAL *hPsy, - AUDIO_OBJECT_TYPE audioObjectType, - CHANNEL_MAPPING *cm, - INT sampleRate, - INT granuleLength, - INT bitRate, - INT tnsMask, - INT bandwidth, - INT usePns, - INT useIS, - UINT syntaxFlags, - ULONG initFlags) -{ - AAC_ENCODER_ERROR ErrorStatus; - int i, ch; - int channelsEff = cm->nChannelsEff; - int tnsChannels = 0; - FB_TYPE filterBank; - - - switch(FDKaacEnc_GetMonoStereoMode(cm->encMode)) { - /* ... and map to tnsChannels */ - case EL_MODE_MONO: tnsChannels = 1; break; - case EL_MODE_STEREO: tnsChannels = 2; break; - default: tnsChannels = 0; - } - - switch (audioObjectType) - { - default: filterBank = FB_LC; break; - case AOT_ER_AAC_LD: filterBank = FB_LD; break; - case AOT_ER_AAC_ELD: filterBank = FB_ELD; break; - } - - hPsy->granuleLength = granuleLength; - - ErrorStatus = FDKaacEnc_InitPsyConfiguration(bitRate/channelsEff, sampleRate, bandwidth, LONG_WINDOW, hPsy->granuleLength, useIS, &(hPsy->psyConf[0]), filterBank); - if (ErrorStatus != AAC_ENC_OK) - return ErrorStatus; - - ErrorStatus = FDKaacEnc_InitTnsConfiguration( - (bitRate*tnsChannels)/channelsEff, - sampleRate, - tnsChannels, - LONG_WINDOW, - hPsy->granuleLength, - isLowDelay(audioObjectType), - (syntaxFlags&AC_SBR_PRESENT)?1:0, - &(hPsy->psyConf[0].tnsConf), - &hPsy->psyConf[0], - (INT)(tnsMask&2), - (INT)(tnsMask&8) ); - - if (ErrorStatus != AAC_ENC_OK) - return ErrorStatus; - - if (granuleLength > 512) { - ErrorStatus = FDKaacEnc_InitPsyConfiguration(bitRate/channelsEff, sampleRate, bandwidth, SHORT_WINDOW, hPsy->granuleLength, useIS, &hPsy->psyConf[1], filterBank); - if (ErrorStatus != AAC_ENC_OK) - return ErrorStatus; - - ErrorStatus = FDKaacEnc_InitTnsConfiguration( - (bitRate*tnsChannels)/channelsEff, - sampleRate, - tnsChannels, - SHORT_WINDOW, - hPsy->granuleLength, - isLowDelay(audioObjectType), - (syntaxFlags&AC_SBR_PRESENT)?1:0, - &hPsy->psyConf[1].tnsConf, - &hPsy->psyConf[1], - (INT)(tnsMask&1), - (INT)(tnsMask&4) ); - - if (ErrorStatus != AAC_ENC_OK) - return ErrorStatus; - - } - - - for (i=0; inElements; i++) { - for (ch=0; chelInfo[i].nChannelsInEl; ch++) { - if (initFlags) { - /* reset states */ - FDKaacEnc_psyInitStates(hPsy, hPsy->psyElement[i]->psyStatic[ch], audioObjectType); - } - - FDKaacEnc_InitPreEchoControl(hPsy->psyElement[i]->psyStatic[ch]->sfbThresholdnm1, - &hPsy->psyElement[i]->psyStatic[ch]->calcPreEcho, - hPsy->psyConf[0].sfbCnt, - hPsy->psyConf[0].sfbPcmQuantThreshold, - &hPsy->psyElement[i]->psyStatic[ch]->mdctScalenm1); - } - } - - ErrorStatus = FDKaacEnc_InitPnsConfiguration(&hPsy->psyConf[0].pnsConf, - bitRate/channelsEff, - sampleRate, - usePns, - hPsy->psyConf[0].sfbCnt, - hPsy->psyConf[0].sfbOffset, - cm->elInfo[0].nChannelsInEl, - (hPsy->psyConf[0].filterbank == FB_LC)); - if (ErrorStatus != AAC_ENC_OK) - return ErrorStatus; - - ErrorStatus = FDKaacEnc_InitPnsConfiguration(&hPsy->psyConf[1].pnsConf, - bitRate/channelsEff, - sampleRate, - usePns, - hPsy->psyConf[1].sfbCnt, - hPsy->psyConf[1].sfbOffset, - cm->elInfo[1].nChannelsInEl, - (hPsy->psyConf[1].filterbank == FB_LC)); - return ErrorStatus; -} - - -static -void FDKaacEnc_deinterleaveInputBuffer(INT_PCM *pOutputSamples, - INT_PCM *pInputSamples, - INT nSamples, - INT nChannels) -{ - INT k; - /* deinterlave input samples and write to output buffer */ - for (k=0; kpsyOutChannel; - FIXP_SGL sfbTonality[(2)][MAX_SFB_LONG]; - - PSY_STATIC **RESTRICT psyStatic = psyElement->psyStatic; - - PSY_DATA *RESTRICT psyData[(2)]; - TNS_DATA *RESTRICT tnsData[(2)]; - PNS_DATA *RESTRICT pnsData[(2)]; - - INT zeroSpec = TRUE; /* means all spectral lines are zero */ - - INT blockSwitchingOffset; - - PSY_CONFIGURATION *RESTRICT hThisPsyConf[(2)]; - INT windowLength[(2)]; - INT nWindows[(2)]; - INT wOffset; - - INT maxSfb[(2)]; - INT *pSfbMaxScaleSpec[(2)]; - FIXP_DBL *pSfbEnergy[(2)]; - FIXP_DBL *pSfbSpreadEnergy[(2)]; - FIXP_DBL *pSfbEnergyLdData[(2)]; - FIXP_DBL *pSfbEnergyMS[(2)]; - FIXP_DBL *pSfbThreshold[(2)]; - - INT isShortWindow[(2)]; - - - if (hPsyConfLong->filterbank == FB_LC) { - blockSwitchingOffset = psyConf->granuleLength + (9*psyConf->granuleLength/(2*TRANS_FAC)); - } else { - blockSwitchingOffset = psyConf->granuleLength; - } - - for(ch = 0; ch < channels; ch++) - { - psyData[ch] = &psyDynamic->psyData[ch]; - tnsData[ch] = &psyDynamic->tnsData[ch]; - pnsData[ch] = &psyDynamic->pnsData[ch]; - - psyData[ch]->mdctSpectrum = psyOutChannel[ch]->mdctSpectrum; - } - - /* block switching */ - if (hPsyConfLong->filterbank != FB_ELD) - { - int err; - - for(ch = 0; ch < channels; ch++) - { - C_ALLOC_SCRATCH_START(pTimeSignal, INT_PCM, (1024)) - - /* deinterleave input data and use for block switching */ - FDKaacEnc_deinterleaveInputBuffer( pTimeSignal, - &pInput[chIdx[ch]], - psyConf->granuleLength, - totalChannels); - - - FDKaacEnc_BlockSwitching (&psyStatic[ch]->blockSwitchingControl, - psyConf->granuleLength, - psyStatic[ch]->isLFE, - pTimeSignal - ); - - - /* fill up internal input buffer, to 2xframelength samples */ - FDKmemcpy(psyStatic[ch]->psyInputBuffer+blockSwitchingOffset, - pTimeSignal, - (2*psyConf->granuleLength-blockSwitchingOffset)*sizeof(INT_PCM)); - - C_ALLOC_SCRATCH_END(pTimeSignal, INT_PCM, (1024)) - } - - /* synch left and right block type */ - err = FDKaacEnc_SyncBlockSwitching(&psyStatic[0]->blockSwitchingControl, - &psyStatic[1]->blockSwitchingControl, - channels, - commonWindow); - - if (err) { - return AAC_ENC_UNSUPPORTED_AOT; /* mixed up LC and LD */ - } - - } - else { - for(ch = 0; ch < channels; ch++) - { - /* deinterleave input data and use for block switching */ - FDKaacEnc_deinterleaveInputBuffer( psyStatic[ch]->psyInputBuffer + blockSwitchingOffset, - &pInput[chIdx[ch]], - psyConf->granuleLength, - totalChannels); - } - } - - for(ch = 0; ch < channels; ch++) - isShortWindow[ch]=(psyStatic[ch]->blockSwitchingControl.lastWindowSequence == SHORT_WINDOW); - - /* set parameters according to window length */ - for(ch = 0; ch < channels; ch++) - { - if(isShortWindow[ch]) { - hThisPsyConf[ch] = hPsyConfShort; - windowLength[ch] = psyConf->granuleLength/TRANS_FAC; - nWindows[ch] = TRANS_FAC; - maxSfb[ch] = MAX_SFB_SHORT; - - pSfbMaxScaleSpec[ch] = psyData[ch]->sfbMaxScaleSpec.Short[0]; - pSfbEnergy[ch] = psyData[ch]->sfbEnergy.Short[0]; - pSfbSpreadEnergy[ch] = psyData[ch]->sfbSpreadEnergy.Short[0]; - pSfbEnergyLdData[ch] = psyData[ch]->sfbEnergyLdData.Short[0]; - pSfbEnergyMS[ch] = psyData[ch]->sfbEnergyMS.Short[0]; - pSfbThreshold[ch] = psyData[ch]->sfbThreshold.Short[0]; - - } else - { - hThisPsyConf[ch] = hPsyConfLong; - windowLength[ch] = psyConf->granuleLength; - nWindows[ch] = 1; - maxSfb[ch] = MAX_GROUPED_SFB; - - pSfbMaxScaleSpec[ch] = psyData[ch]->sfbMaxScaleSpec.Long; - pSfbEnergy[ch] = psyData[ch]->sfbEnergy.Long; - pSfbSpreadEnergy[ch] = psyData[ch]->sfbSpreadEnergy.Long; - pSfbEnergyLdData[ch] = psyData[ch]->sfbEnergyLdData.Long; - pSfbEnergyMS[ch] = psyData[ch]->sfbEnergyMS.Long; - pSfbThreshold[ch] = psyData[ch]->sfbThreshold.Long; - } - } - - /* Transform and get mdctScaling for all channels and windows. */ - for(ch = 0; ch < channels; ch++) - { - /* update number of active bands */ - if (psyStatic[ch]->isLFE) { - psyData[ch]->sfbActive = hThisPsyConf[ch]->sfbActiveLFE; - psyData[ch]->lowpassLine = hThisPsyConf[ch]->lowpassLineLFE; - } else - { - psyData[ch]->sfbActive = hThisPsyConf[ch]->sfbActive; - psyData[ch]->lowpassLine = hThisPsyConf[ch]->lowpassLine; - } - - for(w = 0; w < nWindows[ch]; w++) { - - wOffset = w*windowLength[ch]; - - FDKaacEnc_Transform_Real( psyStatic[ch]->psyInputBuffer + wOffset, - psyData[ch]->mdctSpectrum+wOffset, - psyStatic[ch]->blockSwitchingControl.lastWindowSequence, - psyStatic[ch]->blockSwitchingControl.windowShape, - &psyStatic[ch]->blockSwitchingControl.lastWindowShape, - psyConf->granuleLength, - &mdctSpectrum_e, - hThisPsyConf[ch]->filterbank - ,psyStatic[ch]->overlapAddBuffer - ); - - /* Low pass / highest sfb */ - FDKmemclear(&psyData[ch]->mdctSpectrum[psyData[ch]->lowpassLine+wOffset], - (windowLength[ch]-psyData[ch]->lowpassLine)*sizeof(FIXP_DBL)); - - if ( (hPsyConfLong->filterbank != FB_LC) && (psyData[ch]->lowpassLine >= FADE_OUT_LEN) ) { - /* Do blending to reduce gibbs artifacts */ - for (int i=0; imdctSpectrum[psyData[ch]->lowpassLine+wOffset - FADE_OUT_LEN + i] = fMult(psyData[ch]->mdctSpectrum[psyData[ch]->lowpassLine+wOffset - FADE_OUT_LEN + i], fadeOutFactor[i]); - } - } - - - /* Check for zero spectrum. These loops will usually terminate very, very early. */ - for(line=0; (linelowpassLine) && (zeroSpec==TRUE); line++) { - if (psyData[ch]->mdctSpectrum[line+wOffset] != (FIXP_DBL)0) { - zeroSpec = FALSE; - break; - } - } - - } /* w loop */ - - psyData[ch]->mdctScale = mdctSpectrum_e; - - /* rotate internal time samples */ - FDKmemmove(psyStatic[ch]->psyInputBuffer, - psyStatic[ch]->psyInputBuffer+psyConf->granuleLength, - psyConf->granuleLength*sizeof(INT_PCM)); - - - /* ... and get remaining samples from input buffer */ - FDKaacEnc_deinterleaveInputBuffer( psyStatic[ch]->psyInputBuffer+psyConf->granuleLength, - &pInput[ (2*psyConf->granuleLength-blockSwitchingOffset)*totalChannels + chIdx[ch] ], - blockSwitchingOffset-psyConf->granuleLength, - totalChannels); - - } /* ch */ - - /* Do some rescaling to get maximum possible accuracy for energies */ - if ( zeroSpec == FALSE) { - - /* Calc possible spectrum leftshift for each sfb (1 means: 1 bit left shift is possible without overflow) */ - INT minSpecShift = MAX_SHIFT_DBL; - INT nrgShift = MAX_SHIFT_DBL; - INT finalShift = MAX_SHIFT_DBL; - FIXP_DBL currNrg = 0; - FIXP_DBL maxNrg = 0; - - for(ch = 0; ch < channels; ch++) { - for(w = 0; w < nWindows[ch]; w++) { - wOffset = w*windowLength[ch]; - FDKaacEnc_CalcSfbMaxScaleSpec(psyData[ch]->mdctSpectrum+wOffset, - hThisPsyConf[ch]->sfbOffset, - pSfbMaxScaleSpec[ch]+w*maxSfb[ch], - psyData[ch]->sfbActive); - - for (sfb = 0; sfbsfbActive; sfb++) - minSpecShift = fixMin(minSpecShift, (pSfbMaxScaleSpec[ch]+w*maxSfb[ch])[sfb]); - } - - } - - /* Calc possible energy leftshift for each sfb (1 means: 1 bit left shift is possible without overflow) */ - for(ch = 0; ch < channels; ch++) { - for(w = 0; w < nWindows[ch]; w++) { - wOffset = w*windowLength[ch]; - currNrg = FDKaacEnc_CheckBandEnergyOptim(psyData[ch]->mdctSpectrum+wOffset, - pSfbMaxScaleSpec[ch]+w*maxSfb[ch], - hThisPsyConf[ch]->sfbOffset, - psyData[ch]->sfbActive, - pSfbEnergy[ch]+w*maxSfb[ch], - pSfbEnergyLdData[ch]+w*maxSfb[ch], - minSpecShift-4); - - maxNrg = fixMax(maxNrg, currNrg); - } - } - - if ( maxNrg != (FIXP_DBL)0 ) { - nrgShift = (CountLeadingBits(maxNrg)>>1) + (minSpecShift-4); - } - - /* 2check: Hasn't this decision to be made for both channels? */ - /* For short windows 1 additional bit headroom is necessary to prevent overflows when summing up energies in FDKaacEnc_groupShortData() */ - if(isShortWindow[0]) nrgShift--; - - /* both spectrum and energies mustn't overflow */ - finalShift = fixMin(minSpecShift, nrgShift); - - /* do not shift more than 3 bits more to the left than signal without blockfloating point - * would be to avoid overflow of scaled PCM quantization thresholds */ - if (finalShift > psyData[0]->mdctScale + 3 ) - finalShift = psyData[0]->mdctScale + 3; - - FDK_ASSERT(finalShift >= 0); /* right shift is not allowed */ - - /* correct sfbEnergy and sfbEnergyLdData with new finalShift */ - FIXP_DBL ldShift = finalShift * FL2FXCONST_DBL(2.0/64); - for(ch = 0; ch < channels; ch++) { - for(w = 0; w < nWindows[ch]; w++) { - for(sfb=0; sfbsfbActive; sfb++) { - INT scale = fixMax(0, (pSfbMaxScaleSpec[ch]+w*maxSfb[ch])[sfb]-4); - scale = fixMin((scale-finalShift)<<1, DFRACT_BITS-1); - if (scale >= 0) (pSfbEnergy[ch]+w*maxSfb[ch])[sfb] >>= (scale); - else (pSfbEnergy[ch]+w*maxSfb[ch])[sfb] <<= (-scale); - (pSfbThreshold[ch]+w*maxSfb[ch])[sfb] = fMult((pSfbEnergy[ch]+w*maxSfb[ch])[sfb], C_RATIO); - (pSfbEnergyLdData[ch]+w*maxSfb[ch])[sfb] += ldShift; - } - } - } - - if ( finalShift != 0 ) { - for (ch = 0; ch < channels; ch++) { - for(w = 0; w < nWindows[ch]; w++) { - wOffset = w*windowLength[ch]; - for(line=0; linelowpassLine; line++) { - psyData[ch]->mdctSpectrum[line+wOffset] <<= finalShift; - } - /* update sfbMaxScaleSpec */ - for (sfb = 0; sfbsfbActive; sfb++) - (pSfbMaxScaleSpec[ch]+w*maxSfb[ch])[sfb] -= finalShift; - } - /* update mdctScale */ - psyData[ch]->mdctScale -= finalShift; - } - } - - } else { - /* all spectral lines are zero */ - for (ch = 0; ch < channels; ch++) { - psyData[ch]->mdctScale = 0; /* otherwise mdctScale would be for example 7 and PCM quantization thresholds would be shifted - * 14 bits to the right causing some of them to become 0 (which causes problems later) */ - /* clear sfbMaxScaleSpec */ - for(w = 0; w < nWindows[ch]; w++) { - for (sfb = 0; sfbsfbActive; sfb++) { - (pSfbMaxScaleSpec[ch]+w*maxSfb[ch])[sfb] = 0; - (pSfbEnergy[ch]+w*maxSfb[ch])[sfb] = (FIXP_DBL)0; - (pSfbEnergyLdData[ch]+w*maxSfb[ch])[sfb] = FL2FXCONST_DBL(-1.0f); - (pSfbThreshold[ch]+w*maxSfb[ch])[sfb] = (FIXP_DBL)0; - } - } - } - } - - /* Advance psychoacoustics: Tonality and TNS */ - if (psyStatic[0]->isLFE) { - tnsData[0]->dataRaw.Long.subBlockInfo.tnsActive[HIFILT] = 0; - tnsData[0]->dataRaw.Long.subBlockInfo.tnsActive[LOFILT] = 0; - } - else - { - - for(ch = 0; ch < channels; ch++) { - if (!isShortWindow[ch]) { - /* tonality */ - FDKaacEnc_CalculateFullTonality( psyData[ch]->mdctSpectrum, - pSfbMaxScaleSpec[ch], - pSfbEnergyLdData[ch], - sfbTonality[ch], - psyData[ch]->sfbActive, - hThisPsyConf[ch]->sfbOffset, - hThisPsyConf[ch]->pnsConf.usePns); - } - } - - if (hPsyConfLong->tnsConf.tnsActive || hPsyConfShort->tnsConf.tnsActive) { - INT tnsActive[TRANS_FAC]; - INT nrgScaling[2] = {0,0}; - INT tnsSpecShift = 0; - - for(ch = 0; ch < channels; ch++) { - for(w = 0; w < nWindows[ch]; w++) { - - wOffset = w*windowLength[ch]; - /* TNS */ - FDKaacEnc_TnsDetect( - tnsData[ch], - &hThisPsyConf[ch]->tnsConf, - &psyOutChannel[ch]->tnsInfo, - hThisPsyConf[ch]->sfbCnt, - psyData[ch]->mdctSpectrum+wOffset, - w, - psyStatic[ch]->blockSwitchingControl.lastWindowSequence - ); - } - } - - if (channels == 2) { - FDKaacEnc_TnsSync( - tnsData[1], - tnsData[0], - &psyOutChannel[1]->tnsInfo, - &psyOutChannel[0]->tnsInfo, - - psyStatic[1]->blockSwitchingControl.lastWindowSequence, - psyStatic[0]->blockSwitchingControl.lastWindowSequence, - &hThisPsyConf[1]->tnsConf); - } - - FDK_ASSERT(1==commonWindow); /* all checks for TNS do only work for common windows (which is always set)*/ - for(w = 0; w < nWindows[0]; w++) - { - if (isShortWindow[0]) - tnsActive[w] = tnsData[0]->dataRaw.Short.subBlockInfo[w].tnsActive[HIFILT] || - tnsData[0]->dataRaw.Short.subBlockInfo[w].tnsActive[LOFILT] || - tnsData[channels-1]->dataRaw.Short.subBlockInfo[w].tnsActive[HIFILT] || - tnsData[channels-1]->dataRaw.Short.subBlockInfo[w].tnsActive[LOFILT]; - else - tnsActive[w] = tnsData[0]->dataRaw.Long.subBlockInfo.tnsActive[HIFILT] || - tnsData[0]->dataRaw.Long.subBlockInfo.tnsActive[LOFILT] || - tnsData[channels-1]->dataRaw.Long.subBlockInfo.tnsActive[HIFILT] || - tnsData[channels-1]->dataRaw.Long.subBlockInfo.tnsActive[LOFILT]; - } - - for(ch = 0; ch < channels; ch++) { - if (tnsActive[0] && !isShortWindow[ch]) { - /* Scale down spectrum if tns is active in one of the two channels with same lastWindowSequence */ - /* first part of threshold calculation; it's not necessary to update sfbMaxScaleSpec */ - INT shift = 1; - for(sfb=0; sfblowpassLine; sfb++) { - psyData[ch]->mdctSpectrum[sfb] = psyData[ch]->mdctSpectrum[sfb] >> shift; - } - - /* update thresholds */ - for (sfb=0; sfbsfbActive; sfb++) { - pSfbThreshold[ch][sfb] >>= (2*shift); - } - - psyData[ch]->mdctScale += shift; /* update mdctScale */ - - /* calc sfbEnergies after tnsEncode again ! */ - - } - } - - for(ch = 0; ch < channels; ch++) { - for(w = 0; w < nWindows[ch]; w++) - { - wOffset = w*windowLength[ch]; - FDKaacEnc_TnsEncode( - &psyOutChannel[ch]->tnsInfo, - tnsData[ch], - hThisPsyConf[ch]->sfbCnt, - &hThisPsyConf[ch]->tnsConf, - hThisPsyConf[ch]->sfbOffset[psyData[ch]->sfbActive],/*hThisPsyConf[ch]->lowpassLine*/ /* filter stops before that line ! */ - psyData[ch]->mdctSpectrum+wOffset, - w, - psyStatic[ch]->blockSwitchingControl.lastWindowSequence); - - if(tnsActive[w]) { - /* Calc sfb-bandwise mdct-energies for left and right channel again, */ - /* if tns active in current channel or in one channel with same lastWindowSequence left and right */ - FDKaacEnc_CalcSfbMaxScaleSpec(psyData[ch]->mdctSpectrum+wOffset, - hThisPsyConf[ch]->sfbOffset, - pSfbMaxScaleSpec[ch]+w*maxSfb[ch], - psyData[ch]->sfbActive); - } - } - } - - for(ch = 0; ch < channels; ch++) { - for(w = 0; w < nWindows[ch]; w++) { - - if (tnsActive[w]) { - - if (isShortWindow[ch]) { - FDKaacEnc_CalcBandEnergyOptimShort(psyData[ch]->mdctSpectrum+w*windowLength[ch], - pSfbMaxScaleSpec[ch]+w*maxSfb[ch], - hThisPsyConf[ch]->sfbOffset, - psyData[ch]->sfbActive, - pSfbEnergy[ch]+w*maxSfb[ch]); - } - else { - nrgScaling[ch] = /* with tns, energy calculation can overflow; -> scaling */ - FDKaacEnc_CalcBandEnergyOptimLong(psyData[ch]->mdctSpectrum, - pSfbMaxScaleSpec[ch], - hThisPsyConf[ch]->sfbOffset, - psyData[ch]->sfbActive, - pSfbEnergy[ch], - pSfbEnergyLdData[ch]); - tnsSpecShift = fixMax(tnsSpecShift, nrgScaling[ch]); /* nrgScaling is set only if nrg would have an overflow */ - } - } /* if tnsActive */ - } - } /* end channel loop */ - - /* adapt scaling to prevent nrg overflow, only for long blocks */ - for(ch = 0; ch < channels; ch++) { - if ( (tnsSpecShift!=0) && !isShortWindow[ch] ) { - /* scale down spectrum, nrg's and thresholds, if there was an overflow in sfbNrg calculation after tns */ - for(line=0; linelowpassLine; line++) { - psyData[ch]->mdctSpectrum[line] >>= tnsSpecShift; - } - INT scale = (tnsSpecShift-nrgScaling[ch])<<1; - for(sfb=0; sfbsfbActive; sfb++) { - pSfbEnergyLdData[ch][sfb] -= scale*FL2FXCONST_DBL(1.0/LD_DATA_SCALING); - pSfbEnergy[ch][sfb] >>= scale; - pSfbThreshold[ch][sfb] >>= (tnsSpecShift<<1); - } - psyData[ch]->mdctScale += tnsSpecShift; /* update mdctScale; not necessary to update sfbMaxScaleSpec */ - - } - } /* end channel loop */ - - } /* TNS active */ - } /* !isLFE */ - - - - - - - /* Advance thresholds */ - for(ch = 0; ch < channels; ch++) { - INT headroom; - - FIXP_DBL clipEnergy; - INT energyShift = psyData[ch]->mdctScale*2 ; - INT clipNrgShift = energyShift - THR_SHIFTBITS ; - - if(isShortWindow[ch]) - headroom = 6; - else - headroom = 0; - - if (clipNrgShift >= 0) - clipEnergy = hThisPsyConf[ch]->clipEnergy >> clipNrgShift ; - else if (clipNrgShift>=-headroom) - clipEnergy = hThisPsyConf[ch]->clipEnergy << -clipNrgShift ; - else - clipEnergy = (FIXP_DBL)MAXVAL_DBL ; - - for(w = 0; w < nWindows[ch]; w++) - { - INT i; - /* limit threshold to avoid clipping */ - for (i=0; isfbActive; i++) { - *(pSfbThreshold[ch]+w*maxSfb[ch]+i) = fixMin(*(pSfbThreshold[ch]+w*maxSfb[ch]+i), clipEnergy); - } - - /* spreading */ - FDKaacEnc_SpreadingMax(psyData[ch]->sfbActive, - hThisPsyConf[ch]->sfbMaskLowFactor, - hThisPsyConf[ch]->sfbMaskHighFactor, - pSfbThreshold[ch]+w*maxSfb[ch]); - - - /* PCM quantization threshold */ - energyShift += PCM_QUANT_THR_SCALE; - if (energyShift>=0) { - energyShift = fixMin(DFRACT_BITS-1,energyShift); - for (i=0; isfbActive;i++) { - *(pSfbThreshold[ch]+w*maxSfb[ch]+i) = fixMax(*(pSfbThreshold[ch]+w*maxSfb[ch]+i) >> THR_SHIFTBITS, - (hThisPsyConf[ch]->sfbPcmQuantThreshold[i] >> energyShift)); - } - } else { - energyShift = fixMin(DFRACT_BITS-1,-energyShift); - for (i=0; isfbActive;i++) { - *(pSfbThreshold[ch]+w*maxSfb[ch]+i) = fixMax(*(pSfbThreshold[ch]+w*maxSfb[ch]+i) >> THR_SHIFTBITS, - (hThisPsyConf[ch]->sfbPcmQuantThreshold[i] << energyShift)); - } - } - - if (!psyStatic[ch]->isLFE) - { - /* preecho control */ - if(psyStatic[ch]->blockSwitchingControl.lastWindowSequence == STOP_WINDOW) { - /* prevent FDKaacEnc_PreEchoControl from comparing stop - thresholds with short thresholds */ - for (i=0; isfbActive;i++) { - psyStatic[ch]->sfbThresholdnm1[i] = (FIXP_DBL)MAXVAL_DBL; - } - - psyStatic[ch]->mdctScalenm1 = 0; - psyStatic[ch]->calcPreEcho = 0; - } - - FDKaacEnc_PreEchoControl( psyStatic[ch]->sfbThresholdnm1, - psyStatic[ch]->calcPreEcho, - psyData[ch]->sfbActive, - hThisPsyConf[ch]->maxAllowedIncreaseFactor, - hThisPsyConf[ch]->minRemainingThresholdFactor, - pSfbThreshold[ch]+w*maxSfb[ch], - psyData[ch]->mdctScale, - &psyStatic[ch]->mdctScalenm1); - - psyStatic[ch]->calcPreEcho = 1; - - if(psyStatic[ch]->blockSwitchingControl.lastWindowSequence == START_WINDOW) - { - /* prevent FDKaacEnc_PreEchoControl in next frame to compare start - thresholds with short thresholds */ - for (i=0; isfbActive;i++) { - psyStatic[ch]->sfbThresholdnm1[i] = (FIXP_DBL)MAXVAL_DBL; - } - - psyStatic[ch]->mdctScalenm1 = 0; - psyStatic[ch]->calcPreEcho = 0; - } - - } - - /* spread energy to avoid hole detection */ - FDKmemcpy(pSfbSpreadEnergy[ch]+w*maxSfb[ch], pSfbEnergy[ch]+w*maxSfb[ch], psyData[ch]->sfbActive*sizeof(FIXP_DBL)); - - FDKaacEnc_SpreadingMax(psyData[ch]->sfbActive, - hThisPsyConf[ch]->sfbMaskLowFactorSprEn, - hThisPsyConf[ch]->sfbMaskHighFactorSprEn, - pSfbSpreadEnergy[ch]+w*maxSfb[ch]); - } - } - - /* Calc bandwise energies for mid and side channel. Do it only if 2 channels exist */ - if (channels==2) { - for(w = 0; w < nWindows[1]; w++) { - wOffset = w*windowLength[1]; - FDKaacEnc_CalcBandNrgMSOpt(psyData[0]->mdctSpectrum+wOffset, - psyData[1]->mdctSpectrum+wOffset, - pSfbMaxScaleSpec[0]+w*maxSfb[0], - pSfbMaxScaleSpec[1]+w*maxSfb[1], - hThisPsyConf[1]->sfbOffset, - psyData[0]->sfbActive, - pSfbEnergyMS[0]+w*maxSfb[0], - pSfbEnergyMS[1]+w*maxSfb[1], - (psyStatic[1]->blockSwitchingControl.lastWindowSequence != SHORT_WINDOW), - psyData[0]->sfbEnergyMSLdData, - psyData[1]->sfbEnergyMSLdData); - } - } - - /* group short data (maxSfb[ch] for short blocks is determined here) */ - for(ch=0;chblockSwitchingControl.noOfGroups * hPsyConfShort->sfbCnt; - /* At this point, energies and thresholds are copied/regrouped from the ".Short" to the ".Long" arrays */ - FDKaacEnc_groupShortData( psyData[ch]->mdctSpectrum, - &psyData[ch]->sfbThreshold, - &psyData[ch]->sfbEnergy, - &psyData[ch]->sfbEnergyMS, - &psyData[ch]->sfbSpreadEnergy, - hPsyConfShort->sfbCnt, - psyData[ch]->sfbActive, - hPsyConfShort->sfbOffset, - hPsyConfShort->sfbMinSnrLdData, - psyData[ch]->groupedSfbOffset, - &maxSfbPerGroup[ch], - psyOutChannel[ch]->sfbMinSnrLdData, - psyStatic[ch]->blockSwitchingControl.noOfGroups, - psyStatic[ch]->blockSwitchingControl.groupLen, - psyConf[1].granuleLength); - - - /* calculate ldData arrays (short values are in .Long-arrays after FDKaacEnc_groupShortData) */ - for (sfbGrp = 0; sfbGrp < noSfb; sfbGrp += hPsyConfShort->sfbCnt) { - LdDataVector(&psyData[ch]->sfbEnergy.Long[sfbGrp], &psyOutChannel[ch]->sfbEnergyLdData[sfbGrp], psyData[ch]->sfbActive); - } - - /* calc sfbThrld and set Values smaller 2^-31 to 2^-33*/ - for (sfbGrp = 0; sfbGrp < noSfb; sfbGrp += hPsyConfShort->sfbCnt) { - LdDataVector(&psyData[ch]->sfbThreshold.Long[sfbGrp], &psyOutChannel[ch]->sfbThresholdLdData[sfbGrp], psyData[ch]->sfbActive); - for (sfb=0;sfbsfbActive;sfb++) { - psyOutChannel[ch]->sfbThresholdLdData[sfbGrp+sfb] = - fixMax(psyOutChannel[ch]->sfbThresholdLdData[sfbGrp+sfb], FL2FXCONST_DBL(-0.515625f)); - } - } - - if ( channels==2 ) { - for (sfbGrp = 0; sfbGrp < noSfb; sfbGrp += hPsyConfShort->sfbCnt) { - LdDataVector(&psyData[ch]->sfbEnergyMS.Long[sfbGrp], &psyData[ch]->sfbEnergyMSLdData[sfbGrp], psyData[ch]->sfbActive); - } - } - - FDKmemcpy(psyOutChannel[ch]->sfbOffsets, psyData[ch]->groupedSfbOffset, (MAX_GROUPED_SFB+1)*sizeof(INT)); - - } else { - /* maxSfb[ch] for long blocks */ - for (sfb = psyData[ch]->sfbActive-1; sfb >= 0; sfb--) { - for (line = hPsyConfLong->sfbOffset[sfb+1]-1; line >= hPsyConfLong->sfbOffset[sfb]; line--) { - if (psyData[ch]->mdctSpectrum[line] != FL2FXCONST_SGL(0.0f)) break; - } - if (line > hPsyConfLong->sfbOffset[sfb]) break; - } - maxSfbPerGroup[ch] = sfb + 1; - /* ensure at least one section in ICS; workaround for existing decoder crc implementation */ - maxSfbPerGroup[ch] = fixMax(fixMin(5,psyData[ch]->sfbActive),maxSfbPerGroup[ch]); - - /* sfbNrgLdData is calculated in FDKaacEnc_advancePsychLong, copy in psyOut structure */ - FDKmemcpy(psyOutChannel[ch]->sfbEnergyLdData, psyData[ch]->sfbEnergyLdData.Long, psyData[ch]->sfbActive*sizeof(FIXP_DBL)); - - FDKmemcpy(psyOutChannel[ch]->sfbOffsets, hPsyConfLong->sfbOffset, (MAX_GROUPED_SFB+1)*sizeof(INT)); - - /* sfbMinSnrLdData modified in adjust threshold, copy necessary */ - FDKmemcpy(psyOutChannel[ch]->sfbMinSnrLdData, hPsyConfLong->sfbMinSnrLdData, psyData[ch]->sfbActive*sizeof(FIXP_DBL)); - - /* sfbEnergyMSLdData ist already calculated in FDKaacEnc_CalcBandNrgMSOpt; only in long case */ - - /* calc sfbThrld and set Values smaller 2^-31 to 2^-33*/ - LdDataVector(psyData[ch]->sfbThreshold.Long, psyOutChannel[ch]->sfbThresholdLdData, psyData[ch]->sfbActive); - for (i=0;isfbActive;i++) { - psyOutChannel[ch]->sfbThresholdLdData[i] = - fixMax(psyOutChannel[ch]->sfbThresholdLdData[i], FL2FXCONST_DBL(-0.515625f)); - } - - - } - - - } - - - /* - Intensity parameter intialization. - */ - for(ch=0;chisBook, MAX_GROUPED_SFB*sizeof(INT)); - FDKmemclear(psyOutChannel[ch]->isScale, MAX_GROUPED_SFB*sizeof(INT)); - } - - for(ch=0;chisLFE) - { - /* PNS Decision */ - FDKaacEnc_PnsDetect( &(psyConf[0].pnsConf), - pnsData[ch], - psyStatic[ch]->blockSwitchingControl.lastWindowSequence, - psyData[ch]->sfbActive, - maxSfbPerGroup[ch], /* count of Sfb which are not zero. */ - psyOutChannel[ch]->sfbThresholdLdData, - psyConf[win].sfbOffset, - psyData[ch]->mdctSpectrum, - psyData[ch]->sfbMaxScaleSpec.Long, - sfbTonality[ch], - psyOutChannel[ch]->tnsInfo.order[0][0], - tnsData[ch]->dataRaw.Long.subBlockInfo.predictionGain[HIFILT], - tnsData[ch]->dataRaw.Long.subBlockInfo.tnsActive[HIFILT], - psyOutChannel[ch]->sfbEnergyLdData, - psyOutChannel[ch]->noiseNrg ); - } /* !isLFE */ - } - - /* - stereo Processing - */ - if(channels == 2) - { - psyOutElement->toolsInfo.msDigest = MS_NONE; - psyOutElement->commonWindow = commonWindow; - if (psyOutElement->commonWindow) - maxSfbPerGroup[0] = maxSfbPerGroup[1] = - fixMax(maxSfbPerGroup[0], maxSfbPerGroup[1]); - - if(psyStatic[0]->blockSwitchingControl.lastWindowSequence != SHORT_WINDOW) - { - /* PNS preprocessing depending on ms processing: PNS not in Short Window! */ - FDKaacEnc_PreProcessPnsChannelPair( - psyData[0]->sfbActive, - (&psyData[0]->sfbEnergy)->Long, - (&psyData[1]->sfbEnergy)->Long, - psyOutChannel[0]->sfbEnergyLdData, - psyOutChannel[1]->sfbEnergyLdData, - psyData[0]->sfbEnergyMS.Long, - &(psyConf[0].pnsConf), - pnsData[0], - pnsData[1]); - - FDKaacEnc_IntensityStereoProcessing( - psyData[0]->sfbEnergy.Long, - psyData[1]->sfbEnergy.Long, - psyData[0]->mdctSpectrum, - psyData[1]->mdctSpectrum, - psyData[0]->sfbThreshold.Long, - psyData[1]->sfbThreshold.Long, - psyOutChannel[1]->sfbThresholdLdData, - psyData[0]->sfbSpreadEnergy.Long, - psyData[1]->sfbSpreadEnergy.Long, - psyOutChannel[0]->sfbEnergyLdData, - psyOutChannel[1]->sfbEnergyLdData, - &psyOutElement->toolsInfo.msDigest, - psyOutElement->toolsInfo.msMask, - psyConf[0].sfbCnt, - psyConf[0].sfbCnt, - maxSfbPerGroup[0], - psyConf[0].sfbOffset, - psyConf[0].allowIS && commonWindow, - psyOutChannel[1]->isBook, - psyOutChannel[1]->isScale, - pnsData); - - FDKaacEnc_MsStereoProcessing( - psyData, - psyOutChannel, - psyOutChannel[1]->isBook, - &psyOutElement->toolsInfo.msDigest, - psyOutElement->toolsInfo.msMask, - psyData[0]->sfbActive, - psyData[0]->sfbActive, - maxSfbPerGroup[0], - psyOutChannel[0]->sfbOffsets); - - /* PNS postprocessing */ - FDKaacEnc_PostProcessPnsChannelPair(psyData[0]->sfbActive, - &(psyConf[0].pnsConf), - pnsData[0], - pnsData[1], - psyOutElement->toolsInfo.msMask, - &psyOutElement->toolsInfo.msDigest); - - } else { - FDKaacEnc_IntensityStereoProcessing( - psyData[0]->sfbEnergy.Long, - psyData[1]->sfbEnergy.Long, - psyData[0]->mdctSpectrum, - psyData[1]->mdctSpectrum, - psyData[0]->sfbThreshold.Long, - psyData[1]->sfbThreshold.Long, - psyOutChannel[1]->sfbThresholdLdData, - psyData[0]->sfbSpreadEnergy.Long, - psyData[1]->sfbSpreadEnergy.Long, - psyOutChannel[0]->sfbEnergyLdData, - psyOutChannel[1]->sfbEnergyLdData, - &psyOutElement->toolsInfo.msDigest, - psyOutElement->toolsInfo.msMask, - psyStatic[0]->blockSwitchingControl.noOfGroups*hPsyConfShort->sfbCnt, - psyConf[1].sfbCnt, - maxSfbPerGroup[0], - psyData[0]->groupedSfbOffset, - psyConf[0].allowIS && commonWindow, - psyOutChannel[1]->isBook, - psyOutChannel[1]->isScale, - pnsData); - - /* it's OK to pass the ".Long" arrays here. They contain grouped short data since FDKaacEnc_groupShortData() */ - FDKaacEnc_MsStereoProcessing( psyData, - psyOutChannel, - psyOutChannel[1]->isBook, - &psyOutElement->toolsInfo.msDigest, - psyOutElement->toolsInfo.msMask, - psyStatic[0]->blockSwitchingControl.noOfGroups*hPsyConfShort->sfbCnt, - hPsyConfShort->sfbCnt, - maxSfbPerGroup[0], - psyOutChannel[0]->sfbOffsets); - } - } - - /* - PNS Coding - */ - for(ch=0;chisLFE) { - /* no PNS coding */ - for(sfb = 0; sfb < psyData[ch]->sfbActive; sfb++) { - psyOutChannel[ch]->noiseNrg[sfb] = NO_NOISE_PNS; - } - } else - { - FDKaacEnc_CodePnsChannel(psyData[ch]->sfbActive, - &(psyConf[ch].pnsConf), - pnsData[ch]->pnsFlag, - psyData[ch]->sfbEnergyLdData.Long, - psyOutChannel[ch]->noiseNrg, /* this is the energy that will be written to the bitstream */ - psyOutChannel[ch]->sfbThresholdLdData); - } - } - - /* - build output - */ - for(ch=0;chmaxSfbPerGroup = maxSfbPerGroup[ch]; - psyOutChannel[ch]->mdctScale = psyData[ch]->mdctScale; - - if(isShortWindow[ch]==0) { - - psyOutChannel[ch]->sfbCnt = hPsyConfLong->sfbActive; - psyOutChannel[ch]->sfbPerGroup = hPsyConfLong->sfbActive; - psyOutChannel[ch]->lastWindowSequence = psyStatic[ch]->blockSwitchingControl.lastWindowSequence; - psyOutChannel[ch]->windowShape = psyStatic[ch]->blockSwitchingControl.windowShape; - } - else { - INT sfbCnt = psyStatic[ch]->blockSwitchingControl.noOfGroups*hPsyConfShort->sfbCnt; - - psyOutChannel[ch]->sfbCnt = sfbCnt; - psyOutChannel[ch]->sfbPerGroup = hPsyConfShort->sfbCnt; - psyOutChannel[ch]->lastWindowSequence = SHORT_WINDOW; - psyOutChannel[ch]->windowShape = SINE_WINDOW; - } - - /* generate grouping mask */ - mask = 0; - for (grp = 0; grp < psyStatic[ch]->blockSwitchingControl.noOfGroups; grp++) - { - mask <<= 1; - for (j=1; jblockSwitchingControl.groupLen[grp]; j++) { - mask = (mask<<1) | 1 ; - } - } - psyOutChannel[ch]->groupingMask = mask; - - /* build interface */ - FDKmemcpy(psyOutChannel[ch]->groupLen,psyStatic[ch]->blockSwitchingControl.groupLen,MAX_NO_OF_GROUPS*sizeof(INT)); - FDKmemcpy(psyOutChannel[ch]->sfbEnergy,(&psyData[ch]->sfbEnergy)->Long, MAX_GROUPED_SFB*sizeof(FIXP_DBL)); - FDKmemcpy(psyOutChannel[ch]->sfbSpreadEnergy,(&psyData[ch]->sfbSpreadEnergy)->Long, MAX_GROUPED_SFB*sizeof(FIXP_DBL)); -// FDKmemcpy(psyOutChannel[ch]->mdctSpectrum, psyData[ch]->mdctSpectrum, (1024)*sizeof(FIXP_DBL)); - } - - return AAC_ENC_OK; -} - - -void FDKaacEnc_PsyClose(PSY_INTERNAL **phPsyInternal, - PSY_OUT **phPsyOut) -{ - int n, i; - - - if(phPsyInternal!=NULL) { - PSY_INTERNAL *hPsyInternal = *phPsyInternal; - - if (hPsyInternal) - { - for (i=0; i<(8); i++) { - if (hPsyInternal->pStaticChannels[i]) { - if (hPsyInternal->pStaticChannels[i]->psyInputBuffer) - FreeRam_aacEnc_PsyInputBuffer(&hPsyInternal->pStaticChannels[i]->psyInputBuffer); /* AUDIO INPUT BUFFER */ - - FreeRam_aacEnc_PsyStatic(&hPsyInternal->pStaticChannels[i]); /* PSY_STATIC */ - } - } - - for (i=0; i<(8); i++) { - if (hPsyInternal->psyElement[i]) - FreeRam_aacEnc_PsyElement(&hPsyInternal->psyElement[i]); /* PSY_ELEMENT */ - } - - - FreeRam_aacEnc_PsyInternal(phPsyInternal); - } - } - - if (phPsyOut!=NULL) { - for (n=0; n<(1); n++) { - if (phPsyOut[n]) - { - for (i=0; i<(8); i++) { - if (phPsyOut[n]->pPsyOutChannels[i]) - FreeRam_aacEnc_PsyOutChannel(&phPsyOut[n]->pPsyOutChannels[i]); /* PSY_OUT_CHANNEL */ - } - - for (i=0; i<(8); i++) { - if (phPsyOut[n]->psyOutElement[i]) - FreeRam_aacEnc_PsyOutElements(&phPsyOut[n]->psyOutElement[i]); /* PSY_OUT_ELEMENTS */ - } - - FreeRam_aacEnc_PsyOut(&phPsyOut[n]); - } - } - } -} diff --git a/libAACenc/src/psy_main.h b/libAACenc/src/psy_main.h deleted file mode 100644 index 7bdcc38..0000000 --- a/libAACenc/src/psy_main.h +++ /dev/null @@ -1,174 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M.Werner - contents/description: Psychoaccoustic major function block - -******************************************************************************/ - -#ifndef _PSYMAIN_H -#define _PSYMAIN_H - - -#include "psy_configuration.h" -#include "qc_data.h" -#include "aacenc_pns.h" - -/* - psych internal -*/ -typedef struct { - - PSY_STATIC* psyStatic[(2)]; - -}PSY_ELEMENT; - -typedef struct { - - PSY_DATA psyData[(2)]; - TNS_DATA tnsData[(2)]; - PNS_DATA pnsData[(2)]; - -}PSY_DYNAMIC; - - -typedef struct { - - PSY_CONFIGURATION psyConf[2]; /* LONG / SHORT */ - PSY_ELEMENT* psyElement[(8)]; - PSY_STATIC* pStaticChannels[(8)]; - PSY_DYNAMIC* psyDynamic; - INT granuleLength; - -}PSY_INTERNAL; - - -AAC_ENCODER_ERROR FDKaacEnc_PsyNew(PSY_INTERNAL **phpsy, - const INT nElements, - const INT nChannels - ,UCHAR *dynamic_RAM - ); - -AAC_ENCODER_ERROR FDKaacEnc_PsyOutNew(PSY_OUT **phpsyOut, - const INT nElements, - const INT nChannels, - const INT nSubFrames - ,UCHAR *dynamic_RAM - ); - -AAC_ENCODER_ERROR FDKaacEnc_psyInit(PSY_INTERNAL *hPsy, - PSY_OUT **phpsyOut, - const INT nSubFrames, - const INT nMaxChannels, - const AUDIO_OBJECT_TYPE audioObjectType, - CHANNEL_MAPPING *cm); - -AAC_ENCODER_ERROR FDKaacEnc_psyMainInit(PSY_INTERNAL *hPsy, - AUDIO_OBJECT_TYPE audioObjectType, - CHANNEL_MAPPING *cm, - INT sampleRate, - INT granuleLength, - INT bitRate, - INT tnsMask, - INT bandwidth, - INT usePns, - INT useIS, - UINT syntaxFlags, - ULONG initFlags); - -AAC_ENCODER_ERROR FDKaacEnc_psyMain(INT channels, - PSY_ELEMENT *psyElement, - PSY_DYNAMIC *psyDynamic, - PSY_CONFIGURATION *psyConf, - PSY_OUT_ELEMENT *psyOutElement, - INT_PCM *pInput, - INT *chIdx, - INT totalChannels - ); - -void FDKaacEnc_PsyClose(PSY_INTERNAL **phPsyInternal, - PSY_OUT **phPsyOut); - -#endif /* _PSYMAIN_H */ diff --git a/libAACenc/src/qc_data.h b/libAACenc/src/qc_data.h deleted file mode 100644 index 00d6090..0000000 --- a/libAACenc/src/qc_data.h +++ /dev/null @@ -1,280 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M. Werner - contents/description: Quantizing & coding data - -******************************************************************************/ - -#ifndef _QC_DATA_H -#define _QC_DATA_H - - -#include "psy_const.h" -#include "dyn_bits.h" -#include "adj_thr_data.h" -#include "line_pe.h" -#include "FDK_audio.h" -#include "interface.h" - - -typedef enum { - QCDATA_BR_MODE_INVALID = -1, - QCDATA_BR_MODE_CBR = 0, - QCDATA_BR_MODE_VBR_1 = 1, /* 32 kbps/channel */ - QCDATA_BR_MODE_VBR_2 = 2, /* 40 kbps/channel */ - QCDATA_BR_MODE_VBR_3 = 3, /* 48 kbps/channel */ - QCDATA_BR_MODE_VBR_4 = 4, /* 64 kbps/channel */ - QCDATA_BR_MODE_VBR_5 = 5, /* 96 kbps/channel */ - QCDATA_BR_MODE_FF = 6, /* Fixed frame mode. */ - QCDATA_BR_MODE_SFR = 7 /* Superframe mode. */ - - -} QCDATA_BR_MODE; - -typedef struct { - MP4_ELEMENT_ID elType; - INT instanceTag; - INT nChannelsInEl; - INT ChannelIndex[2]; - FIXP_DBL relativeBits; -} ELEMENT_INFO; - -typedef struct { - CHANNEL_MODE encMode; - INT nChannels; - INT nChannelsEff; - INT nElements; - ELEMENT_INFO elInfo[(8)]; -} CHANNEL_MAPPING; - -typedef struct { - INT paddingRest; -} PADDING; - - -/* Quantizing & coding stage */ - -struct QC_INIT{ - CHANNEL_MAPPING* channelMapping; - INT sceCpe; /* not used yet */ - INT maxBits; /* maximum number of bits in reservoir */ - INT averageBits; /* average number of bits we should use */ - INT bitRes; - INT sampleRate; /* output sample rate */ - INT advancedBitsToPe; /* if set, calc bits2PE factor depending on samplerate */ - INT staticBits; /* Bits per frame consumed by transport layers. */ - QCDATA_BR_MODE bitrateMode; - INT meanPe; - INT chBitrate; - INT invQuant; - INT maxIterations; /* Maximum number of allowed iterations before FDKaacEnc_crashRecovery() is applied. */ - FIXP_DBL maxBitFac; - INT bitrate; - INT nSubFrames; /* helper variable */ - INT minBits; /* minimal number of bits in one frame*/ - - PADDING padding; -}; - -typedef struct -{ - FIXP_DBL mdctSpectrum[(1024)]; - - SHORT quantSpec[(1024)]; - - UINT maxValueInSfb[MAX_GROUPED_SFB]; - INT scf[MAX_GROUPED_SFB]; - INT globalGain; - SECTION_DATA sectionData; - - FIXP_DBL sfbFormFactorLdData[MAX_GROUPED_SFB]; - - FIXP_DBL sfbThresholdLdData[MAX_GROUPED_SFB]; - FIXP_DBL sfbMinSnrLdData[MAX_GROUPED_SFB]; - FIXP_DBL sfbEnergyLdData[MAX_GROUPED_SFB]; - FIXP_DBL sfbEnergy[MAX_GROUPED_SFB]; - FIXP_DBL sfbWeightedEnergyLdData[MAX_GROUPED_SFB]; - - FIXP_DBL sfbEnFacLd[MAX_GROUPED_SFB]; - - FIXP_DBL sfbSpreadEnergy[MAX_GROUPED_SFB]; - -} QC_OUT_CHANNEL; - - -typedef struct -{ - EXT_PAYLOAD_TYPE type; /* type of the extension payload */ - INT nPayloadBits; /* size of the payload */ - UCHAR *pPayload; /* pointer to payload */ - -} QC_OUT_EXTENSION; - - -typedef struct -{ - INT staticBitsUsed; /* for verification purposes */ - INT dynBitsUsed; /* for verification purposes */ - - INT extBitsUsed; /* bit consumption of extended fill elements */ - INT nExtensions; /* number of extension payloads for this element */ - QC_OUT_EXTENSION extension[(1)]; /* reffering extension payload */ - - INT grantedDynBits; - - INT grantedPe; - INT grantedPeCorr; - - PE_DATA peData; - - QC_OUT_CHANNEL *qcOutChannel[(2)]; - - -} QC_OUT_ELEMENT; - -typedef struct -{ - QC_OUT_ELEMENT *qcElement[(8)]; - QC_OUT_CHANNEL *pQcOutChannels[(8)]; - QC_OUT_EXTENSION extension[(2+2)]; /* global extension payload */ - INT nExtensions; /* number of extension payloads for this AU */ - INT maxDynBits; /* maximal allowed dynamic bits in frame */ - INT grantedDynBits; /* granted dynamic bits in frame */ - INT totFillBits; /* fill bits */ - INT elementExtBits; /* element associated extension payload bits, e.g. sbr, drc ... */ - INT globalExtBits; /* frame/au associated extension payload bits (anc data ...) */ - INT staticBits; /* aac side info bits */ - - INT totalNoRedPe; - INT totalGrantedPeCorr; - - INT usedDynBits; /* number of dynamic bits in use */ - INT alignBits; /* AU alignment bits */ - INT totalBits; /* sum of static, dyn, sbr, fill, align and dse bits */ - -} QC_OUT; - -typedef struct { - INT chBitrateEl; /* channel bitrate in element (totalbitrate*el_relativeBits/el_channels) */ - INT maxBitsEl; /* used in crash recovery */ - INT bitResLevelEl; /* update bitreservoir level in each call of FDKaacEnc_QCMain */ - INT maxBitResBitsEl; /* nEffChannels*6144 - averageBitsInFrame */ - FIXP_DBL relativeBitsEl; /* Bits relative to total Bits*/ -} ELEMENT_BITS; - -typedef struct -{ - /* this is basically struct QC_INIT */ - - INT globHdrBits; - INT maxBitsPerFrame; /* maximal allowed bits per frame, 6144*nChannelsEff */ - INT minBitsPerFrame; /* minimal allowd bits per fram, superframing - DRM */ - INT nElements; - QCDATA_BR_MODE bitrateMode; - INT bitDistributionMode; /* 0: full bitreservoir, 1: reduced bitreservoir, 2: disabled bitreservoir */ - INT bitResTot; - INT bitResTotMax; - INT maxIterations; /* Maximum number of allowed iterations before FDKaacEnc_crashRecovery() is applied. */ - INT invQuant; - - FIXP_DBL vbrQualFactor; - FIXP_DBL maxBitFac; - - PADDING padding; - - ELEMENT_BITS *elementBits[(8)]; - BITCNTR_STATE *hBitCounter; - ADJ_THR_STATE *hAdjThr; - - INT dZoneQuantEnable; /* enable dead zone quantizer */ - -} QC_STATE; - -#endif /* _QC_DATA_H */ - - - - diff --git a/libAACenc/src/qc_main.cpp b/libAACenc/src/qc_main.cpp deleted file mode 100644 index 9cd73f6..0000000 --- a/libAACenc/src/qc_main.cpp +++ /dev/null @@ -1,1641 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M. Werner - contents/description: Quantizing & coding - -******************************************************************************/ - -#include "qc_main.h" -#include "quantize.h" -#include "interface.h" -#include "adj_thr.h" -#include "sf_estim.h" -#include "bit_cnt.h" -#include "dyn_bits.h" -#include "channel_map.h" -#include "aacEnc_ram.h" - -#include "genericStds.h" - - -typedef struct { - QCDATA_BR_MODE bitrateMode; - LONG vbrQualFactor; -} TAB_VBR_QUAL_FACTOR; - -static const TAB_VBR_QUAL_FACTOR tableVbrQualFactor[] = { - {QCDATA_BR_MODE_VBR_1, FL2FXCONST_DBL(0.160f)}, /* 32 kbps mono AAC-LC + SBR + PS */ - {QCDATA_BR_MODE_VBR_2, FL2FXCONST_DBL(0.148f)}, /* 64 kbps stereo AAC-LC + SBR */ - {QCDATA_BR_MODE_VBR_3, FL2FXCONST_DBL(0.135f)}, /* 80 - 96 kbps stereo AAC-LC */ - {QCDATA_BR_MODE_VBR_4, FL2FXCONST_DBL(0.111f)}, /* 128 kbps stereo AAC-LC */ - {QCDATA_BR_MODE_VBR_5, FL2FXCONST_DBL(0.070f)} /* 192 kbps stereo AAC-LC */ -}; - -static INT isConstantBitrateMode( - const QCDATA_BR_MODE bitrateMode - ) -{ - return ( ((bitrateMode==QCDATA_BR_MODE_CBR) || (bitrateMode==QCDATA_BR_MODE_SFR) || (bitrateMode==QCDATA_BR_MODE_FF)) ? 1 : 0 ); -} - - - -typedef enum{ - FRAME_LEN_BYTES_MODULO = 1, - FRAME_LEN_BYTES_INT = 2 -}FRAME_LEN_RESULT_MODE; - -/* forward declarations */ - -static INT FDKaacEnc_calcMaxValueInSfb(INT sfbCnt, - INT maxSfbPerGroup, - INT sfbPerGroup, - INT *RESTRICT sfbOffset, - SHORT *RESTRICT quantSpectrum, - UINT *RESTRICT maxValue); - -static void FDKaacEnc_crashRecovery(INT nChannels, - PSY_OUT_ELEMENT* psyOutElement, - QC_OUT* qcOut, - QC_OUT_ELEMENT *qcElement, - INT bitsToSave, - AUDIO_OBJECT_TYPE aot, - UINT syntaxFlags, - SCHAR epConfig); - -static -AAC_ENCODER_ERROR FDKaacEnc_reduceBitConsumption(int* iterations, - const int maxIterations, - int gainAdjustment, - int* chConstraintsFulfilled, - int* calculateQuant, - int nChannels, - PSY_OUT_ELEMENT* psyOutElement, - QC_OUT* qcOut, - QC_OUT_ELEMENT* qcOutElement, - ELEMENT_BITS* elBits, - AUDIO_OBJECT_TYPE aot, - UINT syntaxFlags, - SCHAR epConfig); - - -void FDKaacEnc_QCClose (QC_STATE **phQCstate, QC_OUT **phQC); - -/***************************************************************************** - - functionname: FDKaacEnc_calcFrameLen - description: - returns: - input: - output: - -*****************************************************************************/ -static INT FDKaacEnc_calcFrameLen(INT bitRate, - INT sampleRate, - INT granuleLength, - FRAME_LEN_RESULT_MODE mode) -{ - - INT result; - - result = ((granuleLength)>>3)*(bitRate); - - switch(mode) { - case FRAME_LEN_BYTES_MODULO: - result %= sampleRate; - break; - case FRAME_LEN_BYTES_INT: - result /= sampleRate; - break; - } - return(result); -} - -/***************************************************************************** - - functionname:FDKaacEnc_framePadding - description: Calculates if padding is needed for actual frame - returns: - input: - output: - -*****************************************************************************/ -static INT FDKaacEnc_framePadding(INT bitRate, - INT sampleRate, - INT granuleLength, - INT *paddingRest) -{ - INT paddingOn; - INT difference; - - paddingOn = 0; - - difference = FDKaacEnc_calcFrameLen( bitRate, - sampleRate, - granuleLength, - FRAME_LEN_BYTES_MODULO ); - *paddingRest-=difference; - - if (*paddingRest <= 0 ) { - paddingOn = 1; - *paddingRest += sampleRate; - } - - return( paddingOn ); -} - - -/********************************************************************************* - - functionname: FDKaacEnc_QCOutNew - description: - return: - -**********************************************************************************/ -AAC_ENCODER_ERROR FDKaacEnc_QCOutNew(QC_OUT **phQC, - const INT nElements, - const INT nChannels, - const INT nSubFrames - ,UCHAR *dynamic_RAM - ) -{ - AAC_ENCODER_ERROR ErrorStatus; - int n, i; - int elInc = 0, chInc = 0; - - for (n=0; npQcOutChannels[i] = GetRam_aacEnc_QCchannel(chInc, dynamic_RAM); - if ( phQC[n]->pQcOutChannels[i] == NULL - ) - { - ErrorStatus = AAC_ENC_NO_MEMORY; - goto QCOutNew_bail; - } - chInc++; - } /* nChannels */ - - for (i=0; iqcElement[i] = GetRam_aacEnc_QCelement(elInc); - if (phQC[n]->qcElement[i] == NULL) - { - ErrorStatus = AAC_ENC_NO_MEMORY; - goto QCOutNew_bail; - } - elInc++; - } /* nElements */ - - } /* nSubFrames */ - - - return AAC_ENC_OK; - -QCOutNew_bail: - return ErrorStatus; -} - -/********************************************************************************* - - functionname: FDKaacEnc_QCOutInit - description: - return: - -**********************************************************************************/ -AAC_ENCODER_ERROR FDKaacEnc_QCOutInit(QC_OUT *phQC[(1)], - const INT nSubFrames, - const CHANNEL_MAPPING *cm) -{ - INT n,i,ch; - - for (n=0; nnElements; i++) { - for (ch=0; chelInfo[i].nChannelsInEl; ch++) { - phQC[n]->qcElement[i]->qcOutChannel[ch] = phQC[n]->pQcOutChannels[chInc]; - chInc++; - } /* chInEl */ - } /* nElements */ - } /* nSubFrames */ - - return AAC_ENC_OK; -} - -/********************************************************************************* - - functionname: FDKaacEnc_QCNew - description: - return: - -**********************************************************************************/ -AAC_ENCODER_ERROR FDKaacEnc_QCNew(QC_STATE **phQC, - INT nElements - ,UCHAR* dynamic_RAM - ) -{ - AAC_ENCODER_ERROR ErrorStatus; - int i; - - QC_STATE* hQC = GetRam_aacEnc_QCstate(); - *phQC = hQC; - if (hQC == NULL) { - ErrorStatus = AAC_ENC_NO_MEMORY; - goto QCNew_bail; - } - - if (FDKaacEnc_AdjThrNew(&hQC->hAdjThr, nElements)) { - ErrorStatus = AAC_ENC_NO_MEMORY; - goto QCNew_bail; - } - - if (FDKaacEnc_BCNew(&(hQC->hBitCounter), dynamic_RAM)) { - ErrorStatus = AAC_ENC_NO_MEMORY; - goto QCNew_bail; - } - - for (i=0; ielementBits[i] = GetRam_aacEnc_ElementBits(i); - if (hQC->elementBits[i] == NULL) { - ErrorStatus = AAC_ENC_NO_MEMORY; - goto QCNew_bail; - } - } - - return AAC_ENC_OK; - -QCNew_bail: - FDKaacEnc_QCClose(phQC, NULL); - return ErrorStatus; -} - -/********************************************************************************* - - functionname: FDKaacEnc_QCInit - description: - return: - -**********************************************************************************/ -AAC_ENCODER_ERROR FDKaacEnc_QCInit(QC_STATE *hQC, - struct QC_INIT *init) -{ - int i; - hQC->maxBitsPerFrame = init->maxBits; - hQC->minBitsPerFrame = init->minBits; - hQC->nElements = init->channelMapping->nElements; - hQC->bitResTotMax = init->bitRes; - hQC->bitResTot = init->bitRes; - hQC->maxBitFac = init->maxBitFac; - hQC->bitrateMode = init->bitrateMode; - hQC->invQuant = init->invQuant; - hQC->maxIterations = init->maxIterations; - - if ( isConstantBitrateMode(hQC->bitrateMode) ) { - INT bitresPerChannel = (hQC->bitResTotMax / init->channelMapping->nChannelsEff); - /* 0: full bitreservoir, 1: reduced bitreservoir, 2: disabled bitreservoir */ - hQC->bitDistributionMode = (bitresPerChannel>BITRES_MIN_LD) ? 0 : (bitresPerChannel>0) ? 1 : 2; - } - else { - hQC->bitDistributionMode = 0; /* full bitreservoir */ - } - - - hQC->padding.paddingRest = init->padding.paddingRest; - - hQC->globHdrBits = init->staticBits; /* Bit overhead due to transport */ - - FDKaacEnc_InitElementBits(hQC, - init->channelMapping, - init->bitrate, - (init->averageBits/init->nSubFrames) - hQC->globHdrBits, - hQC->maxBitsPerFrame/init->channelMapping->nChannelsEff); - - hQC->vbrQualFactor = FL2FXCONST_DBL(0.f); - for (i=0; i<(int)(sizeof(tableVbrQualFactor)/sizeof(TAB_VBR_QUAL_FACTOR)); i++) { - if (hQC->bitrateMode==tableVbrQualFactor[i].bitrateMode) { - hQC->vbrQualFactor = (FIXP_DBL)tableVbrQualFactor[i].vbrQualFactor; - break; - } - } - - if (init->channelMapping->nChannelsEff == 1 && - (init->bitrate / init->channelMapping->nChannelsEff) < 32000 && - init->advancedBitsToPe != 0 - ) - { - hQC->dZoneQuantEnable = 1; - } else { - hQC->dZoneQuantEnable = 0; - } - - FDKaacEnc_AdjThrInit( - hQC->hAdjThr, - init->meanPe, - hQC->elementBits, /* or channelBitrates, was: channelBitrate */ - hQC->invQuant, - init->channelMapping->nElements, - init->channelMapping->nChannelsEff, - init->sampleRate, /* output sample rate */ - init->advancedBitsToPe, /* if set, calc bits2PE factor depending on samplerate */ - hQC->vbrQualFactor, - hQC->dZoneQuantEnable - ); - - return AAC_ENC_OK; -} - - - -/********************************************************************************* - - functionname: FDKaacEnc_QCMainPrepare - description: - return: - -**********************************************************************************/ -AAC_ENCODER_ERROR FDKaacEnc_QCMainPrepare(ELEMENT_INFO *elInfo, - ATS_ELEMENT* RESTRICT adjThrStateElement, - PSY_OUT_ELEMENT* RESTRICT psyOutElement, - QC_OUT_ELEMENT* RESTRICT qcOutElement, - AUDIO_OBJECT_TYPE aot, - UINT syntaxFlags, - SCHAR epConfig - ) -{ - AAC_ENCODER_ERROR ErrorStatus = AAC_ENC_OK; - INT nChannels = elInfo->nChannelsInEl; - - PSY_OUT_CHANNEL** RESTRICT psyOutChannel = psyOutElement->psyOutChannel; /* may be modified in-place */ - - FDKaacEnc_CalcFormFactor(qcOutElement->qcOutChannel, psyOutChannel, nChannels); - - /* prepare and calculate PE without reduction */ - FDKaacEnc_peCalculation(&qcOutElement->peData, psyOutChannel, qcOutElement->qcOutChannel, &psyOutElement->toolsInfo, adjThrStateElement, nChannels); - - ErrorStatus = FDKaacEnc_ChannelElementWrite( NULL, elInfo, NULL, - psyOutElement, - psyOutElement->psyOutChannel, - syntaxFlags, - aot, - epConfig, - &qcOutElement->staticBitsUsed, - 0 ); - - return ErrorStatus; -} - -/********************************************************************************* - - functionname: FDKaacEnc_AdjustBitrate - description: adjusts framelength via padding on a frame to frame basis, - to achieve a bitrate that demands a non byte aligned - framelength - return: errorcode - -**********************************************************************************/ -AAC_ENCODER_ERROR FDKaacEnc_AdjustBitrate(QC_STATE *RESTRICT hQC, - CHANNEL_MAPPING *RESTRICT cm, - INT *avgTotalBits, - INT bitRate, /* total bitrate */ - INT sampleRate, /* output sampling rate */ - INT granuleLength) /* frame length */ -{ - INT paddingOn; - INT frameLen; - - /* Do we need an extra padding byte? */ - paddingOn = FDKaacEnc_framePadding(bitRate, - sampleRate, - granuleLength, - &hQC->padding.paddingRest); - - frameLen = paddingOn + FDKaacEnc_calcFrameLen(bitRate, - sampleRate, - granuleLength, - FRAME_LEN_BYTES_INT); - - *avgTotalBits = frameLen<<3; - - return AAC_ENC_OK; -} - -static AAC_ENCODER_ERROR FDKaacEnc_distributeElementDynBits(QC_STATE* hQC, - QC_OUT_ELEMENT* qcElement[(8)], - CHANNEL_MAPPING* cm, - INT codeBits) -{ - - INT i, firstEl = cm->nElements-1; - INT totalBits = 0; - - for (i=(cm->nElements-1); i>=0; i--) { - if ((cm->elInfo[i].elType == ID_SCE) || (cm->elInfo[i].elType == ID_CPE) || - (cm->elInfo[i].elType == ID_LFE)) - { - qcElement[i]->grantedDynBits = (INT)fMult(hQC->elementBits[i]->relativeBitsEl, (FIXP_DBL)codeBits); - totalBits += qcElement[i]->grantedDynBits; - firstEl = i; - } - } - qcElement[firstEl]->grantedDynBits += codeBits - totalBits; - - return AAC_ENC_OK; -} - -/** - * \brief Verify whether minBitsPerFrame criterion can be satisfied. - * - * This function evaluates the bit consumption only if minBitsPerFrame parameter is not 0. - * In hyperframing mode the difference between grantedDynBits and usedDynBits of all sub frames - * results the number of fillbits to be written. - * This bits can be distrubitued in superframe to reach minBitsPerFrame bit consumption in single AU's. - * The return value denotes if enough desired fill bits are available to achieve minBitsPerFrame in all frames. - * This check can only be used within superframes. - * - * \param qcOut Pointer to coding data struct. - * \param minBitsPerFrame Minimal number of bits to be consumed in each frame. - * \param nSubFrames Number of frames in superframe - * - * \return - * - 1: all fine - * - 0: criterion not fulfilled - */ -static int checkMinFrameBitsDemand( - QC_OUT** qcOut, - const INT minBitsPerFrame, - const INT nSubFrames - ) -{ - int result = 1; /* all fine*/ - return result; -} - -//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// - -//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// -/********************************************************************************* - - functionname: FDKaacEnc_getMinimalStaticBitdemand - description: calculate minmal size of static bits by reduction , - to zero spectrum and deactivating tns and MS - return: number of static bits - -**********************************************************************************/ -static int FDKaacEnc_getMinimalStaticBitdemand(CHANNEL_MAPPING* cm, - PSY_OUT** psyOut) -{ - AUDIO_OBJECT_TYPE aot = AOT_AAC_LC; - UINT syntaxFlags = 0; - SCHAR epConfig = -1; - int i, bitcount = 0; - - for (i=0; inElements; i++) { - ELEMENT_INFO elInfo = cm->elInfo[i]; - - if ( (elInfo.elType == ID_SCE) - || (elInfo.elType == ID_CPE) - || (elInfo.elType == ID_LFE) ) - { - INT minElBits = 0; - - FDKaacEnc_ChannelElementWrite( NULL, &elInfo, NULL, - psyOut[0]->psyOutElement[i], - psyOut[0]->psyOutElement[i]->psyOutChannel, - syntaxFlags, - aot, - epConfig, - &minElBits, - 1 ); - bitcount += minElBits; - } - } - - return bitcount; -} - -//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// - -static AAC_ENCODER_ERROR FDKaacEnc_prepareBitDistribution(QC_STATE* hQC, - PSY_OUT** psyOut, - QC_OUT** qcOut, - CHANNEL_MAPPING* cm, - QC_OUT_ELEMENT* qcElement[(1)][(8)], - INT avgTotalBits, - INT *totalAvailableBits, - INT *avgTotalDynBits) -{ - int i; - /* get maximal allowed dynamic bits */ - qcOut[0]->grantedDynBits = (fixMin(hQC->maxBitsPerFrame, avgTotalBits) - hQC->globHdrBits)&~7; - qcOut[0]->grantedDynBits -= (qcOut[0]->globalExtBits + qcOut[0]->staticBits + qcOut[0]->elementExtBits); - qcOut[0]->maxDynBits = ((hQC->maxBitsPerFrame)&~7) - (qcOut[0]->globalExtBits + qcOut[0]->staticBits + qcOut[0]->elementExtBits); - /* assure that enough bits are available */ - if ((qcOut[0]->grantedDynBits+hQC->bitResTot) < 0) { - /* crash recovery allows to reduce static bits to a minimum */ - if ( (qcOut[0]->grantedDynBits+hQC->bitResTot) < (FDKaacEnc_getMinimalStaticBitdemand(cm, psyOut)-qcOut[0]->staticBits) ) - return AAC_ENC_BITRES_TOO_LOW; - } - - /* distribute dynamic bits to each element */ - FDKaacEnc_distributeElementDynBits(hQC, - qcElement[0], - cm, - qcOut[0]->grantedDynBits); - - *avgTotalDynBits = 0; /*frameDynBits;*/ - - *totalAvailableBits = avgTotalBits; - - /* sum up corrected granted PE */ - qcOut[0]->totalGrantedPeCorr = 0; - - for (i=0; inElements; i++) - { - ELEMENT_INFO elInfo = cm->elInfo[i]; - int nChannels = elInfo.nChannelsInEl; - - if ((elInfo.elType == ID_SCE) || (elInfo.elType == ID_CPE) || - (elInfo.elType == ID_LFE)) - { - /* for ( all sub frames ) ... */ - FDKaacEnc_DistributeBits(hQC->hAdjThr, - hQC->hAdjThr->adjThrStateElem[i], - psyOut[0]->psyOutElement[i]->psyOutChannel, - &qcElement[0][i]->peData, - &qcElement[0][i]->grantedPe, - &qcElement[0][i]->grantedPeCorr, - nChannels, - psyOut[0]->psyOutElement[i]->commonWindow, - qcElement[0][i]->grantedDynBits, - hQC->elementBits[i]->bitResLevelEl, - hQC->elementBits[i]->maxBitResBitsEl, - hQC->maxBitFac, - hQC->bitDistributionMode); - - *totalAvailableBits += hQC->elementBits[i]->bitResLevelEl; - /* get total corrected granted PE */ - qcOut[0]->totalGrantedPeCorr += qcElement[0][i]->grantedPeCorr; - } /* -end- if(ID_SCE || ID_CPE || ID_LFE) */ - - } /* -end- element loop */ - - *totalAvailableBits = FDKmin(hQC->maxBitsPerFrame, (*totalAvailableBits)); - - return AAC_ENC_OK; -} - -//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// -static AAC_ENCODER_ERROR FDKaacEnc_updateUsedDynBits(INT* sumDynBitsConsumed, - QC_OUT_ELEMENT* qcElement[(8)], - CHANNEL_MAPPING* cm) -{ - INT i; - - *sumDynBitsConsumed = 0; - - for (i=0; inElements; i++) - { - ELEMENT_INFO elInfo = cm->elInfo[i]; - - if ((elInfo.elType == ID_SCE) || (elInfo.elType == ID_CPE) || - (elInfo.elType == ID_LFE)) - { - /* sum up bits consumed */ - *sumDynBitsConsumed += qcElement[i]->dynBitsUsed; - } /* -end- if(ID_SCE || ID_CPE || ID_LFE) */ - - } /* -end- element loop */ - - return AAC_ENC_OK; -} - - -static INT FDKaacEnc_getTotalConsumedDynBits(QC_OUT** qcOut, - INT nSubFrames) -{ - INT c, totalBits=0; - - /* sum up bit consumption for all sub frames */ - for (c=0; cusedDynBits==-1) return -1; - totalBits += qcOut[c]->usedDynBits; - } - - return totalBits; - -} - -static INT FDKaacEnc_getTotalConsumedBits(QC_OUT** qcOut, - QC_OUT_ELEMENT* qcElement[(1)][(8)], - CHANNEL_MAPPING* cm, - INT globHdrBits, - INT nSubFrames) -{ - int c, i; - int totalUsedBits = 0; - - for (c = 0 ; c < nSubFrames ; c++ ) - { - int dataBits = 0; - for (i=0; inElements; i++) - { - if ((cm->elInfo[i].elType == ID_SCE) || (cm->elInfo[i].elType == ID_CPE) || - (cm->elInfo[i].elType == ID_LFE)) - { - dataBits += qcElement[c][i]->dynBitsUsed + qcElement[c][i]->staticBitsUsed + qcElement[c][i]->extBitsUsed; - } - } - dataBits += qcOut[c]->globalExtBits; - - totalUsedBits += (8 - (dataBits) % 8) % 8; - totalUsedBits += dataBits + globHdrBits; /* header bits for every frame */ - } - return totalUsedBits; -} - -static AAC_ENCODER_ERROR FDKaacEnc_BitResRedistribution( - QC_STATE *const hQC, - const CHANNEL_MAPPING *const cm, - const INT avgTotalBits - ) -{ - /* check bitreservoir fill level */ - if (hQC->bitResTot < 0) { - return AAC_ENC_BITRES_TOO_LOW; - } - else if (hQC->bitResTot > hQC->bitResTotMax) { - return AAC_ENC_BITRES_TOO_HIGH; - } - else { - INT i, firstEl = cm->nElements-1; - INT totalBits = 0, totalBits_max = 0; - - int totalBitreservoir = FDKmin(hQC->bitResTot, (hQC->maxBitsPerFrame-avgTotalBits)); - int totalBitreservoirMax = FDKmin(hQC->bitResTotMax, (hQC->maxBitsPerFrame-avgTotalBits)); - - int sc_bitResTot = CountLeadingBits(totalBitreservoir); - int sc_bitResTotMax = CountLeadingBits(totalBitreservoirMax); - - for (i=(cm->nElements-1); i>=0; i--) { - if ((cm->elInfo[i].elType == ID_SCE) || (cm->elInfo[i].elType == ID_CPE) || - (cm->elInfo[i].elType == ID_LFE)) - { - hQC->elementBits[i]->bitResLevelEl = (INT)fMult(hQC->elementBits[i]->relativeBitsEl, (FIXP_DBL)(totalBitreservoir<>sc_bitResTot; - totalBits += hQC->elementBits[i]->bitResLevelEl; - - hQC->elementBits[i]->maxBitResBitsEl = (INT)fMult(hQC->elementBits[i]->relativeBitsEl, (FIXP_DBL)(totalBitreservoirMax<>sc_bitResTotMax; - totalBits_max += hQC->elementBits[i]->maxBitResBitsEl; - - firstEl = i; - } - } - hQC->elementBits[firstEl]->bitResLevelEl += totalBitreservoir - totalBits; - hQC->elementBits[firstEl]->maxBitResBitsEl += totalBitreservoirMax - totalBits_max; - } - - return AAC_ENC_OK; -} - - -AAC_ENCODER_ERROR FDKaacEnc_QCMain(QC_STATE* RESTRICT hQC, - PSY_OUT** psyOut, - QC_OUT** qcOut, - INT avgTotalBits, - CHANNEL_MAPPING* cm - ,AUDIO_OBJECT_TYPE aot, - UINT syntaxFlags, - SCHAR epConfig - ) -{ - int i, c; - AAC_ENCODER_ERROR ErrorStatus = AAC_ENC_OK; - INT avgTotalDynBits = 0; /* maximal allowed dynamic bits for all frames */ - INT totalAvailableBits = 0; - INT nSubFrames = 1; - - /*-------------------------------------------- */ - /* redistribute total bitreservoir to elements */ - ErrorStatus = FDKaacEnc_BitResRedistribution(hQC, cm, avgTotalBits); - if (ErrorStatus != AAC_ENC_OK) { - return ErrorStatus; - } - - /*-------------------------------------------- */ - /* fastenc needs one time threshold simulation, - in case of multiple frames, one more guess has to be calculated */ - - /*-------------------------------------------- */ - /* helper pointer */ - QC_OUT_ELEMENT* qcElement[(1)][(8)]; - - /* work on a copy of qcChannel and qcElement */ - for (i=0; inElements; i++) - { - ELEMENT_INFO elInfo = cm->elInfo[i]; - - if ((elInfo.elType == ID_SCE) || (elInfo.elType == ID_CPE) || - (elInfo.elType == ID_LFE)) - { - /* for ( all sub frames ) ... */ - for (c = 0 ; c < nSubFrames ; c++ ) - { - { - qcElement[c][i] = qcOut[c]->qcElement[i]; - } - } - } - } - - /*-------------------------------------------- */ - /*-------------------------------------------- */ - if ( isConstantBitrateMode(hQC->bitrateMode) ) - { - /* calc granted dynamic bits for sub frame and - distribute it to each element */ - ErrorStatus = FDKaacEnc_prepareBitDistribution( - hQC, - psyOut, - qcOut, - cm, - qcElement, - avgTotalBits, - &totalAvailableBits, - &avgTotalDynBits); - - if (ErrorStatus != AAC_ENC_OK) { - return ErrorStatus; - } - } - else { - qcOut[0]->grantedDynBits = ((hQC->maxBitsPerFrame - (hQC->globHdrBits))&~7) - - (qcOut[0]->globalExtBits + qcOut[0]->staticBits + qcOut[0]->elementExtBits); - qcOut[0]->maxDynBits = qcOut[0]->grantedDynBits; - - totalAvailableBits = hQC->maxBitsPerFrame; - avgTotalDynBits = 0; - } - -#ifdef PNS_PRECOUNT_ENABLE - /* Calculate estimated pns bits and substract them from grantedDynBits to get a more accurate number of available bits. */ - if (syntaxFlags & (AC_LD|AC_ELD)) - { - int estimatedPnsBits = 0, ch; - - for (ch=0; chnChannels; ch++) { - qcOut[0]->pQcOutChannels[ch]->sectionData.noiseNrgBits = noisePreCount(psyOut[0]->pPsyOutChannels[ch]->noiseNrg, psyOut[0]->pPsyOutChannels[ch]->maxSfbPerGroup); - estimatedPnsBits += qcOut[0]->pQcOutChannels[ch]->sectionData.noiseNrgBits; - } - qcOut[0]->grantedDynBits -= estimatedPnsBits; - } -#endif - - /* for ( all sub frames ) ... */ - for (c = 0 ; c < nSubFrames ; c++ ) - { - /* for CBR and VBR mode */ - FDKaacEnc_AdjustThresholds(hQC->hAdjThr->adjThrStateElem, - qcElement[c], - qcOut[c], - psyOut[c]->psyOutElement, - isConstantBitrateMode(hQC->bitrateMode), - hQC->hAdjThr->maxIter2ndGuess, - cm); - - } /* -end- sub frame counter */ - - /*-------------------------------------------- */ - INT iterations[(1)][(8)]; - INT chConstraintsFulfilled[(1)][(8)][(2)]; - INT calculateQuant[(1)][(8)][(2)]; - INT constraintsFulfilled[(1)][(8)]; - /*-------------------------------------------- */ - - - /* for ( all sub frames ) ... */ - for (c = 0 ; c < nSubFrames ; c++ ) - { - for (i=0; inElements; i++) - { - ELEMENT_INFO elInfo = cm->elInfo[i]; - INT ch, nChannels = elInfo.nChannelsInEl; - - if ((elInfo.elType == ID_SCE) || (elInfo.elType == ID_CPE) || - (elInfo.elType == ID_LFE)) - { - /* Turn thresholds into scalefactors, optimize bit consumption and verify conformance */ - FDKaacEnc_EstimateScaleFactors(psyOut[c]->psyOutElement[i]->psyOutChannel, - qcElement[c][i]->qcOutChannel, - hQC->invQuant, - hQC->dZoneQuantEnable, - cm->elInfo[i].nChannelsInEl); - - - /*-------------------------------------------- */ - constraintsFulfilled[c][i] = 1; - iterations[c][i] = 0 ; - - for (ch = 0; ch < nChannels; ch++) - { - chConstraintsFulfilled[c][i][ch] = 1; - calculateQuant[c][i][ch] = 1; - } - - /*-------------------------------------------- */ - - } /* -end- if(ID_SCE || ID_CPE || ID_LFE) */ - - } /* -end- element loop */ - - qcOut[c]->usedDynBits = -1; - - } /* -end- sub frame counter */ - - - - INT quantizationDone = 0; - INT sumDynBitsConsumedTotal = 0; - INT decreaseBitConsumption = -1; /* no direction yet! */ - - /*-------------------------------------------- */ - /* -start- Quantization loop ... */ - /*-------------------------------------------- */ - do /* until max allowed bits per frame and maxDynBits!=-1*/ - { - quantizationDone = 0; - - c = 0; /* get frame to process */ - - for (i=0; inElements; i++) - { - ELEMENT_INFO elInfo = cm->elInfo[i]; - INT ch, nChannels = elInfo.nChannelsInEl; - - if ((elInfo.elType == ID_SCE) || (elInfo.elType == ID_CPE) || - (elInfo.elType == ID_LFE)) - { - do /* until spectral values < MAX_QUANT */ - { - /*-------------------------------------------- */ - if (!constraintsFulfilled[c][i]) - { - FDKaacEnc_reduceBitConsumption(&iterations[c][i], - hQC->maxIterations, - (decreaseBitConsumption) ? 1 : -1, - chConstraintsFulfilled[c][i], - calculateQuant[c][i], - nChannels, - psyOut[c]->psyOutElement[i], - qcOut[c], - qcElement[c][i], - hQC->elementBits[i], - aot, - syntaxFlags, - epConfig); - } - - /*-------------------------------------------- */ - /*-------------------------------------------- */ - constraintsFulfilled[c][i] = 1 ; - - /*-------------------------------------------- */ - /* quantize spectrum (per each channel) */ - for (ch = 0; ch < nChannels; ch++) - { - /*-------------------------------------------- */ - chConstraintsFulfilled[c][i][ch] = 1; - - /*-------------------------------------------- */ - - if (calculateQuant[c][i][ch]) - { - QC_OUT_CHANNEL* qcOutCh = qcElement[c][i]->qcOutChannel[ch]; - PSY_OUT_CHANNEL* psyOutCh = psyOut[c]->psyOutElement[i]->psyOutChannel[ch]; - - calculateQuant[c][i][ch] = 0; /* calculate quantization only if necessary */ - - /*-------------------------------------------- */ - FDKaacEnc_QuantizeSpectrum(psyOutCh->sfbCnt, - psyOutCh->maxSfbPerGroup, - psyOutCh->sfbPerGroup, - psyOutCh->sfbOffsets, - qcOutCh->mdctSpectrum, - qcOutCh->globalGain, - qcOutCh->scf, - qcOutCh->quantSpec, - hQC->dZoneQuantEnable); - - /*-------------------------------------------- */ - if (FDKaacEnc_calcMaxValueInSfb(psyOutCh->sfbCnt, - psyOutCh->maxSfbPerGroup, - psyOutCh->sfbPerGroup, - psyOutCh->sfbOffsets, - qcOutCh->quantSpec, - qcOutCh->maxValueInSfb) > MAX_QUANT) - { - chConstraintsFulfilled[c][i][ch] = 0; - constraintsFulfilled[c][i] = 0 ; - /* if quanizted value out of range; increase global gain! */ - decreaseBitConsumption = 1; - } - - /*-------------------------------------------- */ - - } /* if calculateQuant[c][i][ch] */ - - } /* channel loop */ - - /*-------------------------------------------- */ - /* quantize spectrum (per each channel) */ - - /*-------------------------------------------- */ - - } while (!constraintsFulfilled[c][i]) ; /* does not regard bit consumption */ - - - /*-------------------------------------------- */ - /*-------------------------------------------- */ - qcElement[c][i]->dynBitsUsed = 0 ; /* reset dynamic bits */ - - /* quantization valid in current channel! */ - for (ch = 0; ch < nChannels; ch++) - { - QC_OUT_CHANNEL* qcOutCh = qcElement[c][i]->qcOutChannel[ch]; - PSY_OUT_CHANNEL *psyOutCh = psyOut[c]->psyOutElement[i]->psyOutChannel[ch]; - - /* count dynamic bits */ - INT chDynBits = FDKaacEnc_dynBitCount(hQC->hBitCounter, - qcOutCh->quantSpec, - qcOutCh->maxValueInSfb, - qcOutCh->scf, - psyOutCh->lastWindowSequence, - psyOutCh->sfbCnt, - psyOutCh->maxSfbPerGroup, - psyOutCh->sfbPerGroup, - psyOutCh->sfbOffsets, - &qcOutCh->sectionData, - psyOutCh->noiseNrg, - psyOutCh->isBook, - psyOutCh->isScale, - syntaxFlags) ; - - /* sum up dynamic channel bits */ - qcElement[c][i]->dynBitsUsed += chDynBits; - } - - /* save dynBitsUsed for correction of bits2pe relation */ - if(hQC->hAdjThr->adjThrStateElem[i]->dynBitsLast==-1) { - hQC->hAdjThr->adjThrStateElem[i]->dynBitsLast = qcElement[c][i]->dynBitsUsed; - } - } /* -end- if(ID_SCE || ID_CPE || ID_LFE) */ - - } /* -end- element loop */ - - /* update dynBits of current subFrame */ - FDKaacEnc_updateUsedDynBits(&qcOut[c]->usedDynBits, - qcElement[c], - cm); - - /* get total consumed bits, dyn bits in all sub frames have to be valid */ - sumDynBitsConsumedTotal = FDKaacEnc_getTotalConsumedDynBits(qcOut, nSubFrames); - - if (sumDynBitsConsumedTotal==-1) - { - quantizationDone = 0; /* bit consumption not valid in all sub frames */ - } - else - { - int sumBitsConsumedTotal = FDKaacEnc_getTotalConsumedBits(qcOut, qcElement, cm, hQC->globHdrBits, nSubFrames); - - /* in all frames are valid dynamic bits */ - if ( ((sumBitsConsumedTotal < totalAvailableBits) || qcOut[c]->usedDynBits==0) && (decreaseBitConsumption==1) && checkMinFrameBitsDemand(qcOut,hQC->minBitsPerFrame,nSubFrames) - /*()*/ ) - { - quantizationDone = 1; /* exit bit adjustment */ - } - if (sumBitsConsumedTotal > totalAvailableBits && (decreaseBitConsumption==0) ) -// /*()*/ ) - { - quantizationDone = 0; /* reset! */ - break; - } - } - - - /*-------------------------------------------- */ - - int emergencyIterations = 1; - int dynBitsOvershoot = 0; - - for (c = 0 ; c < nSubFrames ; c++ ) - { - for (i=0; inElements; i++) - { - ELEMENT_INFO elInfo = cm->elInfo[i]; - - if ((elInfo.elType == ID_SCE) || (elInfo.elType == ID_CPE) || - (elInfo.elType == ID_LFE)) - { - /* iteration limitation */ - emergencyIterations &= ((iterations[c][i] < hQC->maxIterations) ? 0 : 1); - } - } - /* detection if used dyn bits exceeds the maximal allowed criterion */ - dynBitsOvershoot |= ((qcOut[c]->usedDynBits > qcOut[c]->maxDynBits) ? 1 : 0); - } - - if (quantizationDone==0 || dynBitsOvershoot) - { - - int sumBitsConsumedTotal = FDKaacEnc_getTotalConsumedBits(qcOut, qcElement, cm, hQC->globHdrBits, nSubFrames); - - if ( (sumDynBitsConsumedTotal >= avgTotalDynBits) || (sumDynBitsConsumedTotal==0) ) { - quantizationDone = 1; - } - if (emergencyIterations && (sumBitsConsumedTotal < totalAvailableBits)) { - quantizationDone = 1; - } - if ((sumBitsConsumedTotal > totalAvailableBits) || !checkMinFrameBitsDemand(qcOut,hQC->minBitsPerFrame,nSubFrames)) { - quantizationDone = 0; - } - if ((sumBitsConsumedTotal < totalAvailableBits) && checkMinFrameBitsDemand(qcOut,hQC->minBitsPerFrame,nSubFrames)) { - decreaseBitConsumption = 0; - } - else { - decreaseBitConsumption = 1; - } - - if (dynBitsOvershoot) { - quantizationDone = 0; - decreaseBitConsumption = 1; - } - - /* reset constraints fullfilled flags */ - FDKmemclear(constraintsFulfilled, sizeof(constraintsFulfilled)); - FDKmemclear(chConstraintsFulfilled, sizeof(chConstraintsFulfilled)); - - - }/* quantizationDone */ - - } while (!quantizationDone) ; - - /*-------------------------------------------- */ - /* ... -end- Quantization loop */ - /*-------------------------------------------- */ - - /*-------------------------------------------- */ - /*-------------------------------------------- */ - - return AAC_ENC_OK; -} - - -static AAC_ENCODER_ERROR FDKaacEnc_reduceBitConsumption(int* iterations, - const int maxIterations, - int gainAdjustment, - int* chConstraintsFulfilled, - int* calculateQuant, - int nChannels, - PSY_OUT_ELEMENT* psyOutElement, - QC_OUT* qcOut, - QC_OUT_ELEMENT* qcOutElement, - ELEMENT_BITS* elBits, - AUDIO_OBJECT_TYPE aot, - UINT syntaxFlags, - SCHAR epConfig) -{ - int ch; - - /** SOLVING PROBLEM **/ - if ((*iterations)++ >= maxIterations) - { - if (qcOutElement->dynBitsUsed==0) { - } - /* crash recovery */ - else { - INT bitsToSave = 0; - if ( (bitsToSave = fixMax((qcOutElement->dynBitsUsed + 8) - (elBits->bitResLevelEl + qcOutElement->grantedDynBits), - (qcOutElement->dynBitsUsed + qcOutElement->staticBitsUsed + 8) - (elBits->maxBitsEl))) > 0 ) - { - FDKaacEnc_crashRecovery(nChannels, - psyOutElement, - qcOut, - qcOutElement, - bitsToSave, - aot, - syntaxFlags, - epConfig) ; - } - else - { - for (ch = 0; ch < nChannels; ch++) - { - qcOutElement->qcOutChannel[ch]->globalGain += 1; - } - } - for (ch = 0; ch < nChannels; ch++) - { - calculateQuant[ch] = 1; - } - } - } - else /* iterations >= maxIterations */ - { - /* increase gain (+ next iteration) */ - for (ch = 0; ch < nChannels; ch++) - { - if(!chConstraintsFulfilled[ch]) - { - qcOutElement->qcOutChannel[ch]->globalGain += gainAdjustment ; - calculateQuant[ch] = 1; /* global gain has changed, recalculate quantization in next iteration! */ - } - } - } - - return AAC_ENC_OK; -} - -AAC_ENCODER_ERROR FDKaacEnc_updateFillBits(CHANNEL_MAPPING* cm, - QC_STATE* qcKernel, - ELEMENT_BITS* RESTRICT elBits[(8)], - QC_OUT** qcOut) -{ - switch (qcKernel->bitrateMode) { - case QCDATA_BR_MODE_SFR: - break; - - case QCDATA_BR_MODE_FF: - break; - - case QCDATA_BR_MODE_VBR_1: - case QCDATA_BR_MODE_VBR_2: - case QCDATA_BR_MODE_VBR_3: - case QCDATA_BR_MODE_VBR_4: - case QCDATA_BR_MODE_VBR_5: - qcOut[0]->totFillBits = (qcOut[0]->grantedDynBits - qcOut[0]->usedDynBits)&7; /* precalculate alignment bits */ - qcOut[0]->totalBits = qcOut[0]->staticBits + qcOut[0]->usedDynBits + qcOut[0]->totFillBits + qcOut[0]->elementExtBits + qcOut[0]->globalExtBits; - qcOut[0]->totFillBits += ( fixMax(0, qcKernel->minBitsPerFrame - qcOut[0]->totalBits) + 7) & ~7; - break; - - case QCDATA_BR_MODE_CBR: - case QCDATA_BR_MODE_INVALID: - default: - INT bitResSpace = qcKernel->bitResTotMax - qcKernel->bitResTot ; - /* processing fill-bits */ - INT deltaBitRes = qcOut[0]->grantedDynBits - qcOut[0]->usedDynBits ; - qcOut[0]->totFillBits = fixMax((deltaBitRes&7), (deltaBitRes - (fixMax(0,bitResSpace-7)&~7))); - qcOut[0]->totalBits = qcOut[0]->staticBits + qcOut[0]->usedDynBits + qcOut[0]->totFillBits + qcOut[0]->elementExtBits + qcOut[0]->globalExtBits; - qcOut[0]->totFillBits += ( fixMax(0, qcKernel->minBitsPerFrame - qcOut[0]->totalBits) + 7) & ~7; - break; - } /* switch (qcKernel->bitrateMode) */ - - return AAC_ENC_OK; -} - - - - -/********************************************************************************* - - functionname: FDKaacEnc_calcMaxValueInSfb - description: - return: - -**********************************************************************************/ - -static INT FDKaacEnc_calcMaxValueInSfb(INT sfbCnt, - INT maxSfbPerGroup, - INT sfbPerGroup, - INT *RESTRICT sfbOffset, - SHORT *RESTRICT quantSpectrum, - UINT *RESTRICT maxValue) -{ - INT sfbOffs,sfb; - INT maxValueAll = 0; - - for (sfbOffs=0;sfbOffsbitrateMode) { - case QCDATA_BR_MODE_FF: - case QCDATA_BR_MODE_VBR_1: - case QCDATA_BR_MODE_VBR_2: - case QCDATA_BR_MODE_VBR_3: - case QCDATA_BR_MODE_VBR_4: - case QCDATA_BR_MODE_VBR_5: - /* variable bitrate */ - qcKernel->bitResTot = FDKmin(qcKernel->maxBitsPerFrame, qcKernel->bitResTotMax); - break; - - case QCDATA_BR_MODE_CBR: - case QCDATA_BR_MODE_SFR: - case QCDATA_BR_MODE_INVALID: - default: - int c = 0; - /* constant bitrate */ - { - qcKernel->bitResTot += qcOut[c]->grantedDynBits - (qcOut[c]->usedDynBits + qcOut[c]->totFillBits + qcOut[c]->alignBits); - } - break; - } -} - -/********************************************************************************* - - functionname: FDKaacEnc_FinalizeBitConsumption - description: - return: - -**********************************************************************************/ -AAC_ENCODER_ERROR FDKaacEnc_FinalizeBitConsumption(CHANNEL_MAPPING *cm, - QC_STATE *qcKernel, - QC_OUT *qcOut, - QC_OUT_ELEMENT** qcElement, - HANDLE_TRANSPORTENC hTpEnc, - AUDIO_OBJECT_TYPE aot, - UINT syntaxFlags, - SCHAR epConfig) -{ - QC_OUT_EXTENSION fillExtPayload; - INT totFillBits, alignBits; - - /* Get total consumed bits in AU */ - qcOut->totalBits = qcOut->staticBits + qcOut->usedDynBits + qcOut->totFillBits + - qcOut->elementExtBits + qcOut->globalExtBits; - - if (qcKernel->bitrateMode==QCDATA_BR_MODE_CBR) { - - /* Now we can get the exact transport bit amount, and hopefully it is equal to the estimated value */ - INT exactTpBits = transportEnc_GetStaticBits(hTpEnc, qcOut->totalBits); - - if (exactTpBits != qcKernel->globHdrBits) { - INT diffFillBits = 0; - - /* How many bits can be taken by bitreservoir */ - const INT bitresSpace = qcKernel->bitResTotMax - (qcKernel->bitResTot + (qcOut->grantedDynBits - (qcOut->usedDynBits + qcOut->totFillBits) ) ); - - /* Number of bits which can be moved to bitreservoir. */ - const INT bitsToBitres = qcKernel->globHdrBits - exactTpBits; - FDK_ASSERT(bitsToBitres>=0); /* is always positive */ - - /* If bitreservoir can not take all bits, move ramaining bits to fillbits */ - diffFillBits = FDKmax(0, bitsToBitres - bitresSpace); - - /* Assure previous alignment */ - diffFillBits = (diffFillBits+7)&~7; - - /* Move as many bits as possible to bitreservoir */ - qcKernel->bitResTot += (bitsToBitres-diffFillBits); - - /* Write remaing bits as fill bits */ - qcOut->totFillBits += diffFillBits; - qcOut->totalBits += diffFillBits; - qcOut->grantedDynBits += diffFillBits; - - /* Get new header bits */ - qcKernel->globHdrBits = transportEnc_GetStaticBits(hTpEnc, qcOut->totalBits); - - if (qcKernel->globHdrBits != exactTpBits) { - /* In previous step, fill bits and corresponding total bits were changed when bitreservoir was completely filled. - Now we can take the too much taken bits caused by header overhead from bitreservoir. - */ - qcKernel->bitResTot -= (qcKernel->globHdrBits - exactTpBits); - } - } - - } /* MODE_CBR */ - - /* Update exact number of consumed header bits. */ - qcKernel->globHdrBits = transportEnc_GetStaticBits(hTpEnc, qcOut->totalBits); - - /* Save total fill bits and distribut to alignment and fill bits */ - totFillBits = qcOut->totFillBits; - - /* fake a fill extension payload */ - FDKmemclear(&fillExtPayload, sizeof(QC_OUT_EXTENSION)); - - fillExtPayload.type = EXT_FILL_DATA; - fillExtPayload.nPayloadBits = totFillBits; - - /* ask bitstream encoder how many of that bits can be written in a fill extension data entity */ - qcOut->totFillBits = FDKaacEnc_writeExtensionData( NULL, - &fillExtPayload, - 0, 0, - syntaxFlags, - aot, - epConfig ); - - /* now distribute extra fillbits and alignbits */ - alignBits = 7 - (qcOut->staticBits + qcOut->usedDynBits + qcOut->elementExtBits - + qcOut->totFillBits + qcOut->globalExtBits -1)%8; - - /* Maybe we could remove this */ - if( ((alignBits + qcOut->totFillBits - totFillBits)==8) && (qcOut->totFillBits>8) ) - qcOut->totFillBits -= 8; - - qcOut->totalBits = qcOut->staticBits + qcOut->usedDynBits + qcOut->totFillBits + - alignBits + qcOut->elementExtBits + qcOut->globalExtBits; - - if ( (qcOut->totalBits>qcKernel->maxBitsPerFrame) || (qcOut->totalBitsminBitsPerFrame) ) { - return AAC_ENC_QUANT_ERROR; - } - - qcOut->alignBits = alignBits; - - return AAC_ENC_OK; -} - - - -/********************************************************************************* - - functionname: FDKaacEnc_crashRecovery - description: fulfills constraints by means of brute force... - => bits are saved by cancelling out spectral lines!! - (beginning at the highest frequencies) - return: errorcode - -**********************************************************************************/ - -static void FDKaacEnc_crashRecovery(INT nChannels, - PSY_OUT_ELEMENT* psyOutElement, - QC_OUT* qcOut, - QC_OUT_ELEMENT *qcElement, - INT bitsToSave, - AUDIO_OBJECT_TYPE aot, - UINT syntaxFlags, - SCHAR epConfig) -{ - INT ch ; - INT savedBits = 0 ; - INT sfb, sfbGrp ; - INT bitsPerScf[(2)][MAX_GROUPED_SFB] ; - INT sectionToScf[(2)][MAX_GROUPED_SFB] ; - INT *sfbOffset ; - INT sect, statBitsNew ; - QC_OUT_CHANNEL **qcChannel = qcElement->qcOutChannel; - PSY_OUT_CHANNEL **psyChannel = psyOutElement->psyOutChannel; - - /* create a table which converts frq-bins to bit-demand... [bitsPerScf] */ - /* ...and another one which holds the corresponding sections [sectionToScf] */ - for (ch = 0; ch < nChannels; ch++) - { - sfbOffset = psyChannel[ch]->sfbOffsets ; - - for (sect = 0; sect < qcChannel[ch]->sectionData.noOfSections; sect++) - { - INT sfb ; - INT codeBook = qcChannel[ch]->sectionData.huffsection[sect].codeBook ; - - for (sfb = qcChannel[ch]->sectionData.huffsection[sect].sfbStart; - sfb < qcChannel[ch]->sectionData.huffsection[sect].sfbStart + - qcChannel[ch]->sectionData.huffsection[sect].sfbCnt; - sfb++) - { - bitsPerScf[ch][sfb] = 0; - if ( (codeBook != CODE_BOOK_PNS_NO) /*&& - (sfb < (qcChannel[ch]->sectionData.noOfGroups*qcChannel[ch]->sectionData.maxSfbPerGroup))*/ ) - { - INT sfbStartLine = sfbOffset[sfb] ; - INT noOfLines = sfbOffset[sfb+1] - sfbStartLine ; - bitsPerScf[ch][sfb] = FDKaacEnc_countValues(&(qcChannel[ch]->quantSpec[sfbStartLine]), noOfLines, codeBook) ; - } - sectionToScf[ch][sfb] = sect ; - } - - } - } - - /* LOWER [maxSfb] IN BOTH CHANNELS!! */ - /* Attention: in case of stereo: maxSfbL == maxSfbR, GroupingL == GroupingR ; */ - - for (sfb = qcChannel[0]->sectionData.maxSfbPerGroup-1; sfb >= 0; sfb--) - { - for (sfbGrp = 0; sfbGrp < psyChannel[0]->sfbCnt; sfbGrp += psyChannel[0]->sfbPerGroup) - { - for (ch = 0; ch < nChannels; ch++) - { - int sect = sectionToScf[ch][sfbGrp+sfb]; - qcChannel[ch]->sectionData.huffsection[sect].sfbCnt-- ; - savedBits += bitsPerScf[ch][sfbGrp+sfb] ; - - if (qcChannel[ch]->sectionData.huffsection[sect].sfbCnt == 0) { - savedBits += (psyChannel[ch]->lastWindowSequence!=SHORT_WINDOW) ? FDKaacEnc_sideInfoTabLong[0] - : FDKaacEnc_sideInfoTabShort[0]; - } - } - } - - /* ...have enough bits been saved? */ - if (savedBits >= bitsToSave) - break ; - - } /* sfb loop */ - - /* if not enough bits saved, - clean whole spectrum and remove side info overhead */ - if (sfb == -1) { - sfb = 0 ; - } - - for (ch = 0; ch < nChannels; ch++) - { - qcChannel[ch]->sectionData.maxSfbPerGroup = sfb ; - psyChannel[ch]->maxSfbPerGroup = sfb ; - /* when no spectrum is coded save tools info in bitstream */ - if(sfb==0) { - FDKmemclear(&psyChannel[ch]->tnsInfo, sizeof(TNS_INFO)); - FDKmemclear(&psyOutElement->toolsInfo, sizeof(TOOLSINFO)); - } - } - /* dynamic bits will be updated in iteration loop */ - - { /* if stop sfb has changed save bits in side info, e.g. MS or TNS coding */ - ELEMENT_INFO elInfo; - - FDKmemclear(&elInfo, sizeof(ELEMENT_INFO)); - elInfo.nChannelsInEl = nChannels; - elInfo.elType = (nChannels == 2) ? ID_CPE : ID_SCE; - - FDKaacEnc_ChannelElementWrite( NULL, &elInfo, NULL, - psyOutElement, - psyChannel, - syntaxFlags, - aot, - epConfig, - &statBitsNew, - 0 ); - } - - savedBits = qcElement->staticBitsUsed - statBitsNew; - - /* update static and dynamic bits */ - qcElement->staticBitsUsed -= savedBits; - qcElement->grantedDynBits += savedBits; - - qcOut->staticBits -= savedBits; - qcOut->grantedDynBits += savedBits; - qcOut->maxDynBits += savedBits; - - -} - - - -void FDKaacEnc_QCClose (QC_STATE **phQCstate, QC_OUT **phQC) -{ - int n, i; - - if (phQC!=NULL) { - - for (n=0;n<(1);n++) { - if (phQC[n] != NULL) { - QC_OUT *hQC = phQC[n]; - for (i=0; i<(8); i++) { - } - - for (i=0; i<(8); i++) { - if (hQC->qcElement[i]) - FreeRam_aacEnc_QCelement(&hQC->qcElement[i]); - } - - FreeRam_aacEnc_QCout(&phQC[n]); - } - } - } - - if (phQCstate!=NULL) { - if (*phQCstate != NULL) { - QC_STATE *hQCstate = *phQCstate; - - if (hQCstate->hAdjThr != NULL) - FDKaacEnc_AdjThrClose(&hQCstate->hAdjThr); - - if (hQCstate->hBitCounter != NULL) - FDKaacEnc_BCClose(&hQCstate->hBitCounter); - - for (i=0; i<(8); i++) { - if (hQCstate->elementBits[i]!=NULL) { - FreeRam_aacEnc_ElementBits(&hQCstate->elementBits[i]); - } - } - FreeRam_aacEnc_QCstate(phQCstate); - } - } -} - diff --git a/libAACenc/src/qc_main.h b/libAACenc/src/qc_main.h deleted file mode 100644 index 4e8c042..0000000 --- a/libAACenc/src/qc_main.h +++ /dev/null @@ -1,170 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M. Werner - contents/description: Quantizing & coding - -******************************************************************************/ -#ifndef _QC_MAIN_H -#define _QC_MAIN_H - - -#include "aacenc.h" -#include "qc_data.h" -#include "interface.h" -#include "psy_main.h" -#include "tpenc_lib.h" - -/* Quantizing & coding stage */ - -AAC_ENCODER_ERROR FDKaacEnc_QCOutNew(QC_OUT **phQC, - const INT nElements, - const INT nChannels, - const INT nSubFrames - ,UCHAR *dynamic_RAM - ); - -AAC_ENCODER_ERROR FDKaacEnc_QCOutInit(QC_OUT *phQC[(1)], - const INT nSubFrames, - const CHANNEL_MAPPING *cm); - -AAC_ENCODER_ERROR FDKaacEnc_QCNew(QC_STATE **phQC, - INT nElements - ,UCHAR* dynamic_RAM - ); - -AAC_ENCODER_ERROR FDKaacEnc_QCInit(QC_STATE *hQC, struct QC_INIT *init); - -AAC_ENCODER_ERROR FDKaacEnc_QCMainPrepare( - ELEMENT_INFO *elInfo, - ATS_ELEMENT* RESTRICT adjThrStateElement, - PSY_OUT_ELEMENT* RESTRICT psyOutElement, - QC_OUT_ELEMENT* RESTRICT qcOutElement, /* returns error code */ - AUDIO_OBJECT_TYPE aot, - UINT syntaxFlags, - SCHAR epConfig - ); - - -AAC_ENCODER_ERROR FDKaacEnc_QCMain(QC_STATE* RESTRICT hQC, - PSY_OUT** psyOut, - QC_OUT** qcOut, - INT avgTotalBits, - CHANNEL_MAPPING* cm - ,AUDIO_OBJECT_TYPE aot, - UINT syntaxFlags, - SCHAR epConfig - ); - -AAC_ENCODER_ERROR FDKaacEnc_updateFillBits(CHANNEL_MAPPING* cm, - QC_STATE* qcKernel, - ELEMENT_BITS* RESTRICT elBits[(8)], - QC_OUT** qcOut); - - -void FDKaacEnc_updateBitres( CHANNEL_MAPPING *cm, - QC_STATE *qcKernel, - QC_OUT **qcOut); - -AAC_ENCODER_ERROR FDKaacEnc_FinalizeBitConsumption( CHANNEL_MAPPING *cm, - QC_STATE *hQC, - QC_OUT *qcOut, - QC_OUT_ELEMENT** qcElement, - HANDLE_TRANSPORTENC hTpEnc, - AUDIO_OBJECT_TYPE aot, - UINT syntaxFlags, - SCHAR epConfig - ); - -AAC_ENCODER_ERROR FDKaacEnc_AdjustBitrate(QC_STATE *RESTRICT hQC, - CHANNEL_MAPPING *RESTRICT cm, - INT *avgTotalBits, - INT bitRate, - INT sampleRate, - INT granuleLength); - -void FDKaacEnc_QCClose (QC_STATE **phQCstate, QC_OUT **phQC); - -#endif /* _QC_MAIN_H */ diff --git a/libAACenc/src/quantize.cpp b/libAACenc/src/quantize.cpp deleted file mode 100644 index a74da0e..0000000 --- a/libAACenc/src/quantize.cpp +++ /dev/null @@ -1,405 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M.Werner - contents/description: Quantization - -******************************************************************************/ - -#include "quantize.h" - -#include "aacEnc_rom.h" - -/***************************************************************************** - - functionname: FDKaacEnc_quantizeLines - description: quantizes spectrum lines - returns: - input: global gain, number of lines to process, spectral data - output: quantized spectrum - -*****************************************************************************/ -static void FDKaacEnc_quantizeLines(INT gain, - INT noOfLines, - FIXP_DBL *mdctSpectrum, - SHORT *quaSpectrum, - INT dZoneQuantEnable) -{ - int line; - FIXP_DBL k = FL2FXCONST_DBL(0.0f); - FIXP_QTD quantizer = FDKaacEnc_quantTableQ[(-gain)&3]; - INT quantizershift = ((-gain)>>2)+1; - const INT kShift=16; - - if (dZoneQuantEnable) - k = FL2FXCONST_DBL(0.23f)>>kShift; - else - k = FL2FXCONST_DBL(-0.0946f + 0.5f)>>kShift; - - for (line = 0; line < noOfLines; line++) - { - FIXP_DBL accu = fMultDiv2(mdctSpectrum[line],quantizer); - - if (accu < FL2FXCONST_DBL(0.0f)) - { - accu=-accu; - /* normalize */ - INT accuShift = CntLeadingZeros(accu) - 1; /* CountLeadingBits() is not necessary here since test value is always > 0 */ - accu <<= accuShift; - INT tabIndex = (INT)(accu>>(DFRACT_BITS-2-MANT_DIGITS))&(~MANT_SIZE); - INT totalShift = quantizershift-accuShift+1; - accu = fMultDiv2(FDKaacEnc_mTab_3_4[tabIndex],FDKaacEnc_quantTableE[totalShift&3]); - totalShift = (16-4)-(3*(totalShift>>2)); - FDK_ASSERT(totalShift >=0); /* MAX_QUANT_VIOLATION */ - accu >>= fixMin(totalShift,DFRACT_BITS-1); - quaSpectrum[line] = (SHORT)(-((LONG)(k + accu) >> (DFRACT_BITS-1-16))); - } - else if(accu > FL2FXCONST_DBL(0.0f)) - { - /* normalize */ - INT accuShift = CntLeadingZeros(accu) - 1; /* CountLeadingBits() is not necessary here since test value is always > 0 */ - accu <<= accuShift; - INT tabIndex = (INT)(accu>>(DFRACT_BITS-2-MANT_DIGITS))&(~MANT_SIZE); - INT totalShift = quantizershift-accuShift+1; - accu = fMultDiv2(FDKaacEnc_mTab_3_4[tabIndex],FDKaacEnc_quantTableE[totalShift&3]); - totalShift = (16-4)-(3*(totalShift>>2)); - FDK_ASSERT(totalShift >=0); /* MAX_QUANT_VIOLATION */ - accu >>= fixMin(totalShift,DFRACT_BITS-1); - quaSpectrum[line] = (SHORT)((LONG)(k + accu) >> (DFRACT_BITS-1-16)); - } - else - quaSpectrum[line]=0; - } -} - - -/***************************************************************************** - - functionname:iFDKaacEnc_quantizeLines - description: iquantizes spectrum lines - mdctSpectrum = iquaSpectrum^4/3 *2^(0.25*gain) - input: global gain, number of lines to process,quantized spectrum - output: spectral data - -*****************************************************************************/ -static void FDKaacEnc_invQuantizeLines(INT gain, - INT noOfLines, - SHORT *quantSpectrum, - FIXP_DBL *mdctSpectrum) - -{ - INT iquantizermod; - INT iquantizershift; - INT line; - - iquantizermod = gain&3; - iquantizershift = gain>>2; - - for (line = 0; line < noOfLines; line++) { - - if(quantSpectrum[line] < 0) { - FIXP_DBL accu; - INT ex,specExp,tabIndex; - FIXP_DBL s,t; - - accu = (FIXP_DBL) -quantSpectrum[line]; - - ex = CountLeadingBits(accu); - accu <<= ex; - specExp = (DFRACT_BITS-1) - ex; - - FDK_ASSERT(specExp < 14); /* this fails if abs(value) > 8191 */ - - tabIndex = (INT)(accu>>(DFRACT_BITS-2-MANT_DIGITS))&(~MANT_SIZE); - - /* calculate "mantissa" ^4/3 */ - s = FDKaacEnc_mTab_4_3Elc[tabIndex]; - - /* get approperiate exponent multiplier for specExp^3/4 combined with scfMod */ - t = FDKaacEnc_specExpMantTableCombElc[iquantizermod][specExp]; - - /* multiply "mantissa" ^4/3 with exponent multiplier */ - accu = fMult(s,t); - - /* get approperiate exponent shifter */ - specExp = FDKaacEnc_specExpTableComb[iquantizermod][specExp]-1; /* -1 to avoid overflows in accu */ - - if ((-iquantizershift-specExp) < 0) - accu <<= -(-iquantizershift-specExp); - else - accu >>= -iquantizershift-specExp; - - mdctSpectrum[line] = -accu; - } - else if (quantSpectrum[line] > 0) { - FIXP_DBL accu; - INT ex,specExp,tabIndex; - FIXP_DBL s,t; - - accu = (FIXP_DBL)(INT)quantSpectrum[line]; - - ex = CountLeadingBits(accu); - accu <<= ex; - specExp = (DFRACT_BITS-1) - ex; - - FDK_ASSERT(specExp < 14); /* this fails if abs(value) > 8191 */ - - tabIndex = (INT)(accu>>(DFRACT_BITS-2-MANT_DIGITS))&(~MANT_SIZE); - - /* calculate "mantissa" ^4/3 */ - s = FDKaacEnc_mTab_4_3Elc[tabIndex]; - - /* get approperiate exponent multiplier for specExp^3/4 combined with scfMod */ - t = FDKaacEnc_specExpMantTableCombElc[iquantizermod][specExp]; - - /* multiply "mantissa" ^4/3 with exponent multiplier */ - accu = fMult(s,t); - - /* get approperiate exponent shifter */ - specExp = FDKaacEnc_specExpTableComb[iquantizermod][specExp]-1; /* -1 to avoid overflows in accu */ - - if (( -iquantizershift-specExp) < 0) - accu <<= -(-iquantizershift-specExp); - else - accu >>= -iquantizershift-specExp; - - mdctSpectrum[line] = accu; - } - else { - mdctSpectrum[line] = FL2FXCONST_DBL(0.0f); - } - } -} - -/***************************************************************************** - - functionname: FDKaacEnc_QuantizeSpectrum - description: quantizes the entire spectrum - returns: - input: number of scalefactor bands to be quantized, ... - output: quantized spectrum - -*****************************************************************************/ -void FDKaacEnc_QuantizeSpectrum(INT sfbCnt, - INT maxSfbPerGroup, - INT sfbPerGroup, - INT *sfbOffset, - FIXP_DBL *mdctSpectrum, - INT globalGain, - INT *scalefactors, - SHORT *quantizedSpectrum, - INT dZoneQuantEnable) -{ - INT sfbOffs,sfb; - - /* in FDKaacEnc_quantizeLines quaSpectrum is calculated with: - spec^(3/4) * 2^(-3/16*QSS) * 2^(3/4*scale) + k - simplify scaling calculation and reduce QSS before: - spec^(3/4) * 2^(-3/16*(QSS - 4*scale)) */ - - for(sfbOffs=0;sfbOffsMAX_QUANT) { - return FL2FXCONST_DBL(0.0f); - } - /* inverse quantization */ - FDKaacEnc_invQuantizeLines(gain,1,&quantSpectrum[i],&invQuantSpec); - - /* dist */ - diff = fixp_abs(fixp_abs(invQuantSpec) - fixp_abs(mdctSpectrum[i]>>1)); - - scale = CountLeadingBits(diff); - diff = scaleValue(diff, scale); - diff = fPow2(diff); - scale = fixMin(2*(scale-1), DFRACT_BITS-1); - - diff = scaleValue(diff, -scale); - - xfsf = xfsf + diff; - } - - xfsf = CalcLdData(xfsf); - - return xfsf; -} - -/***************************************************************************** - - functionname: FDKaacEnc_calcSfbQuantEnergyAndDist - description: calculates energy and distortion of quantized values - returns: - input: gain, number of lines to process, quantized spectral data, - spectral data - output: energy, distortion - -*****************************************************************************/ -void FDKaacEnc_calcSfbQuantEnergyAndDist(FIXP_DBL *mdctSpectrum, - SHORT *quantSpectrum, - INT noOfLines, - INT gain, - FIXP_DBL *en, - FIXP_DBL *dist) -{ - INT i,scale; - FIXP_DBL invQuantSpec; - FIXP_DBL diff; - - FIXP_DBL energy = FL2FXCONST_DBL(0.0f); - FIXP_DBL distortion = FL2FXCONST_DBL(0.0f); - - for (i=0; iMAX_QUANT) { - *en = FL2FXCONST_DBL(0.0f); - *dist = FL2FXCONST_DBL(0.0f); - return; - } - - /* inverse quantization */ - FDKaacEnc_invQuantizeLines(gain,1,&quantSpectrum[i],&invQuantSpec); - - /* energy */ - energy += fPow2(invQuantSpec); - - /* dist */ - diff = fixp_abs(fixp_abs(invQuantSpec) - fixp_abs(mdctSpectrum[i]>>1)); - - scale = CountLeadingBits(diff); - diff = scaleValue(diff, scale); - diff = fPow2(diff); - - scale = fixMin(2*(scale-1), DFRACT_BITS-1); - - diff = scaleValue(diff, -scale); - - distortion += diff; - } - - *en = CalcLdData(energy)+FL2FXCONST_DBL(0.03125f); - *dist = CalcLdData(distortion); -} - diff --git a/libAACenc/src/quantize.h b/libAACenc/src/quantize.h deleted file mode 100644 index 16d3d4e..0000000 --- a/libAACenc/src/quantize.h +++ /dev/null @@ -1,121 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M.Werner - contents/description: Quantization - -******************************************************************************/ - -#ifndef _QUANTIZE_H_ -#define _QUANTIZE_H_ - -#include "common_fix.h" - -/* quantizing */ - -#define MAX_QUANT 8191 - -void FDKaacEnc_QuantizeSpectrum(INT sfbCnt, - INT maxSfbPerGroup, - INT sfbPerGroup, - INT *sfbOffset, FIXP_DBL *mdctSpectrum, - INT globalGain, INT *scalefactors, - SHORT *quantizedSpectrum, - INT dZoneQuantEnable); - -FIXP_DBL FDKaacEnc_calcSfbDist(FIXP_DBL *mdctSpectrum, - SHORT *quantSpectrum, - INT noOfLines, - INT gain, - INT dZoneQuantEnable); - -void FDKaacEnc_calcSfbQuantEnergyAndDist(FIXP_DBL *mdctSpectrum, - SHORT *quantSpectrum, - INT noOfLines, - INT gain, - FIXP_DBL *en, - FIXP_DBL *dist); - -#endif /* _QUANTIZE_H_ */ diff --git a/libAACenc/src/sf_estim.cpp b/libAACenc/src/sf_estim.cpp deleted file mode 100644 index 1cb243b..0000000 --- a/libAACenc/src/sf_estim.cpp +++ /dev/null @@ -1,1330 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M. Werner - contents/description: Scale factor estimation - -******************************************************************************/ - -#include "sf_estim.h" -#include "aacEnc_rom.h" -#include "quantize.h" -#include "bit_cnt.h" - - - - -#define AS_PE_FAC_SHIFT 7 -#define DIST_FAC_SHIFT 3 -#define AS_PE_FAC_FLOAT (float)(1 << AS_PE_FAC_SHIFT) -static const INT MAX_SCF_DELTA = 60; - - -static const FIXP_DBL PE_C1 = FL2FXCONST_DBL(3.0f/AS_PE_FAC_FLOAT); /* (log(8.0)/log(2)) >> AS_PE_FAC_SHIFT */ -static const FIXP_DBL PE_C2 = FL2FXCONST_DBL(1.3219281f/AS_PE_FAC_FLOAT); /* (log(2.5)/log(2)) >> AS_PE_FAC_SHIFT */ -static const FIXP_DBL PE_C3 = FL2FXCONST_DBL(0.5593573f); /* 1-C2/C1 */ - - -/* - Function; FDKaacEnc_FDKaacEnc_CalcFormFactorChannel - - Description: Calculates the formfactor - - sf: scale factor of the mdct spectrum - sfbFormFactorLdData is scaled with the factor 1/(((2^sf)^0.5) * (2^FORM_FAC_SHIFT)) -*/ -static void -FDKaacEnc_FDKaacEnc_CalcFormFactorChannel(FIXP_DBL *RESTRICT sfbFormFactorLdData, - PSY_OUT_CHANNEL *RESTRICT psyOutChan) -{ - INT j, sfb, sfbGrp; - FIXP_DBL formFactor; - - int tmp0 = psyOutChan->sfbCnt; - int tmp1 = psyOutChan->maxSfbPerGroup; - int step = psyOutChan->sfbPerGroup; - for(sfbGrp = 0; sfbGrp < tmp0; sfbGrp += step) { - for (sfb = 0; sfb < tmp1; sfb++) { - formFactor = FL2FXCONST_DBL(0.0f); - /* calc sum of sqrt(spec) */ - for(j=psyOutChan->sfbOffsets[sfbGrp+sfb]; jsfbOffsets[sfbGrp+sfb+1]; j++ ) { - formFactor += sqrtFixp(fixp_abs(psyOutChan->mdctSpectrum[j]))>>FORM_FAC_SHIFT; - } - sfbFormFactorLdData[sfbGrp+sfb] = CalcLdData(formFactor); - } - /* set sfbFormFactor for sfbs with zero spec to zero. Just for debugging. */ - for ( ; sfb < psyOutChan->sfbPerGroup; sfb++) { - sfbFormFactorLdData[sfbGrp+sfb] = FL2FXCONST_DBL(-1.0f); - } - } -} - -/* - Function: FDKaacEnc_CalcFormFactor - - Description: Calls FDKaacEnc_FDKaacEnc_CalcFormFactorChannel() for each channel -*/ - -void -FDKaacEnc_CalcFormFactor(QC_OUT_CHANNEL *qcOutChannel[(2)], - PSY_OUT_CHANNEL *psyOutChannel[(2)], - const INT nChannels) -{ - INT j; - for (j=0; jsfbFormFactorLdData, psyOutChannel[j]); - } -} - -/* - Function: FDKaacEnc_calcSfbRelevantLines - - Description: Calculates sfbNRelevantLines - - sfbNRelevantLines is scaled with the factor 1/((2^FORM_FAC_SHIFT) * 2.0) -*/ -static void -FDKaacEnc_calcSfbRelevantLines( const FIXP_DBL *const sfbFormFactorLdData, - const FIXP_DBL *const sfbEnergyLdData, - const FIXP_DBL *const sfbThresholdLdData, - const INT *const sfbOffsets, - const INT sfbCnt, - const INT sfbPerGroup, - const INT maxSfbPerGroup, - FIXP_DBL *sfbNRelevantLines) -{ - INT sfbOffs, sfb; - FIXP_DBL sfbWidthLdData; - FIXP_DBL asPeFacLdData = FL2FXCONST_DBL(0.109375); /* AS_PE_FAC_SHIFT*ld64(2) */ - FIXP_DBL accu; - - /* sfbNRelevantLines[i] = 2^( (sfbFormFactorLdData[i] - 0.25 * (sfbEnergyLdData[i] - ld64(sfbWidth[i]/(2^7)) - AS_PE_FAC_SHIFT*ld64(2)) * 64); */ - - FDKmemclear(sfbNRelevantLines, sfbCnt * sizeof(FIXP_DBL)); - - for (sfbOffs=0; sfbOffs (FIXP_DBL)sfbThresholdLdData[sfbOffs+sfb]) { - INT sfbWidth = sfbOffsets[sfbOffs+sfb+1] - sfbOffsets[sfbOffs+sfb]; - - /* avgFormFactorLdData = sqrtFixp(sqrtFixp(sfbEnergyLdData[sfbOffs+sfb]/sfbWidth)); */ - /* sfbNRelevantLines[sfbOffs+sfb] = sfbFormFactor[sfbOffs+sfb] / avgFormFactorLdData; */ - sfbWidthLdData = (FIXP_DBL)(sfbWidth << (DFRACT_BITS-1-AS_PE_FAC_SHIFT)); - sfbWidthLdData = CalcLdData(sfbWidthLdData); - - accu = sfbEnergyLdData[sfbOffs+sfb] - sfbWidthLdData - asPeFacLdData; - accu = sfbFormFactorLdData[sfbOffs+sfb] - (accu >> 2); - - sfbNRelevantLines[sfbOffs+sfb] = CalcInvLdData(accu) >> 1; - } - } - } -} - -/* - Function: FDKaacEnc_countSingleScfBits - - Description: - - scfBitsFract is scaled by 1/(2^(2*AS_PE_FAC_SHIFT)) -*/ -static FIXP_DBL FDKaacEnc_countSingleScfBits(INT scf, INT scfLeft, INT scfRight) -{ - FIXP_DBL scfBitsFract; - - scfBitsFract = (FIXP_DBL) ( FDKaacEnc_bitCountScalefactorDelta(scfLeft-scf) - + FDKaacEnc_bitCountScalefactorDelta(scf-scfRight) ); - - scfBitsFract = scfBitsFract << (DFRACT_BITS-1-(2*AS_PE_FAC_SHIFT)); - - return scfBitsFract; /* output scaled by 1/(2^(2*AS_PE_FAC)) */ -} - -/* - Function: FDKaacEnc_calcSingleSpecPe - - specPe is scaled by 1/(2^(2*AS_PE_FAC_SHIFT)) -*/ -static FIXP_DBL FDKaacEnc_calcSingleSpecPe(INT scf, FIXP_DBL sfbConstPePart, FIXP_DBL nLines) -{ - FIXP_DBL specPe = FL2FXCONST_DBL(0.0f); - FIXP_DBL ldRatio; - FIXP_DBL scfFract; - - scfFract = (FIXP_DBL)(scf << (DFRACT_BITS-1-AS_PE_FAC_SHIFT)); - - ldRatio = sfbConstPePart - fMult(FL2FXCONST_DBL(0.375f),scfFract); - - if (ldRatio >= PE_C1) { - specPe = fMult(FL2FXCONST_DBL(0.7f),fMult(nLines,ldRatio)); - } - else { - specPe = fMult(FL2FXCONST_DBL(0.7f),fMult(nLines,(PE_C2 + fMult(PE_C3,ldRatio)))); - } - - return specPe; /* output scaled by 1/(2^(2*AS_PE_FAC)) */ -} - -/* - Function: FDKaacEnc_countScfBitsDiff - - scfBitsDiff is scaled by 1/(2^(2*AS_PE_FAC_SHIFT)) -*/ -static FIXP_DBL FDKaacEnc_countScfBitsDiff(INT *scfOld, - INT *scfNew, - INT sfbCnt, - INT startSfb, - INT stopSfb) -{ - FIXP_DBL scfBitsFract; - INT scfBitsDiff = 0; - INT sfb = 0, sfbLast; - INT sfbPrev, sfbNext; - - /* search for first relevant sfb */ - sfbLast = startSfb; - while ((sfbLast=0) && (scfOld[sfbPrev]==FDK_INT_MIN)) - sfbPrev--; - if (sfbPrev>=0) - scfBitsDiff += FDKaacEnc_bitCountScalefactorDelta(scfNew[sfbPrev]-scfNew[sfbLast]) - - FDKaacEnc_bitCountScalefactorDelta(scfOld[sfbPrev]-scfOld[sfbLast]); - /* now loop through all sfbs and count diffs of relevant sfbs */ - for (sfb=sfbLast+1; sfbsfbEnergy[sfb] * 6.75f / sfbFormFactor[sfb]) * LOG2_1; */ - /* 0.02152255861f = log(6.75)/log(2)/AS_PE_FAC_FLOAT; LOG2_1 is 1.0 for log2 */ - /* 0.09375f = log(64.0)/log(2.0)/64.0 = scale of sfbFormFactorLdData */ - if (sfbConstPePart[sfb] == (FIXP_DBL)FDK_INT_MIN) - sfbConstPePart[sfb] = ((psyOutChan->sfbEnergyLdData[sfb] - sfbFormFactorLdData[sfb] - FL2FXCONST_DBL(0.09375f)) >> 1) + FL2FXCONST_DBL(0.02152255861f); - - scfFract = (FIXP_DBL) (scfOld[sfb] << (DFRACT_BITS-1-AS_PE_FAC_SHIFT)); - ldRatioOld = sfbConstPePart[sfb] - fMult(FL2FXCONST_DBL(0.375f),scfFract); - - scfFract = (FIXP_DBL) (scfNew[sfb] << (DFRACT_BITS-1-AS_PE_FAC_SHIFT)); - ldRatioNew = sfbConstPePart[sfb] - fMult(FL2FXCONST_DBL(0.375f),scfFract); - - if (ldRatioOld >= PE_C1) - pOld = ldRatioOld; - else - pOld = PE_C2 + fMult(PE_C3,ldRatioOld); - - if (ldRatioNew >= PE_C1) - pNew = ldRatioNew; - else - pNew = PE_C2 + fMult(PE_C3,ldRatioNew); - - specPeDiff += fMult(FL2FXCONST_DBL(0.7f),fMult(sfbNRelevantLines[sfb],(pNew - pOld))); - } - } - - return specPeDiff; -} - -/* - Function: FDKaacEnc_improveScf - - Description: Calculate the distortion by quantization and inverse quantization of the spectrum with - various scalefactors. The scalefactor which provides the best results will be used. -*/ -static INT FDKaacEnc_improveScf(FIXP_DBL *spec, - SHORT *quantSpec, - SHORT *quantSpecTmp, - INT sfbWidth, - FIXP_DBL threshLdData, - INT scf, - INT minScf, - FIXP_DBL *distLdData, - INT *minScfCalculated, - INT dZoneQuantEnable - ) -{ - FIXP_DBL sfbDistLdData; - INT scfBest = scf; - INT k; - FIXP_DBL distFactorLdData = FL2FXCONST_DBL(-0.0050301265); /* ld64(1/1.25) */ - - /* calc real distortion */ - sfbDistLdData = FDKaacEnc_calcSfbDist(spec, - quantSpec, - sfbWidth, - scf, - dZoneQuantEnable); - *minScfCalculated = scf; - /* nmr > 1.25 -> try to improve nmr */ - if (sfbDistLdData > (threshLdData-distFactorLdData)) { - INT scfEstimated = scf; - FIXP_DBL sfbDistBestLdData = sfbDistLdData; - INT cnt; - /* improve by bigger scf ? */ - cnt = 0; - - while ((sfbDistLdData > (threshLdData-distFactorLdData)) && (cnt++ < 3)) { - scf++; - sfbDistLdData = FDKaacEnc_calcSfbDist(spec, - quantSpecTmp, - sfbWidth, - scf, - dZoneQuantEnable); - - if (sfbDistLdData < sfbDistBestLdData) { - scfBest = scf; - sfbDistBestLdData = sfbDistLdData; - for (k=0; k (threshLdData-distFactorLdData)) && (cnt++ < 1) && (scf > minScf)) { - scf--; - sfbDistLdData = FDKaacEnc_calcSfbDist(spec, - quantSpecTmp, - sfbWidth, - scf, - dZoneQuantEnable); - - if (sfbDistLdData < sfbDistBestLdData) { - scfBest = scf; - sfbDistBestLdData = sfbDistLdData; - for (k=0; k try to find bigger scf to use less bits */ - FIXP_DBL sfbDistBestLdData = sfbDistLdData; - FIXP_DBL sfbDistAllowedLdData = fixMin(sfbDistLdData-distFactorLdData,threshLdData); - int cnt; - for (cnt=0; cnt<3; cnt++) { - scf++; - sfbDistLdData = FDKaacEnc_calcSfbDist(spec, - quantSpecTmp, - sfbWidth, - scf, - dZoneQuantEnable); - - if (sfbDistLdData < sfbDistAllowedLdData) { - *minScfCalculated = scfBest+1; - scfBest = scf; - sfbDistBestLdData = sfbDistLdData; - for (k=0; ksfbCnt; i++) { - prevScfLast[i] = FDK_INT_MAX; - prevScfNext[i] = FDK_INT_MAX; - deltaPeLast[i] = (FIXP_DBL)FDK_INT_MAX; - } - - sfbLast = -1; - sfbAct = -1; - sfbNext = -1; - scfLast = 0; - scfNext = 0; - scfMin = FDK_INT_MAX; - scfMax = FDK_INT_MAX; - do { - /* search for new relevant sfb */ - sfbNext++; - while ((sfbNext < psyOutChan->sfbCnt) && (scf[sfbNext] == FDK_INT_MIN)) - sfbNext++; - if ((sfbLast>=0) && (sfbAct>=0) && (sfbNextsfbCnt)) { - /* relevant scfs to the left and to the right */ - scfAct = scf[sfbAct]; - scfLast = scf + sfbLast; - scfNext = scf + sfbNext; - scfMin = fixMin(*scfLast, *scfNext); - scfMax = fixMax(*scfLast, *scfNext); - } - else if ((sfbLast==-1) && (sfbAct>=0) && (sfbNextsfbCnt)) { - /* first relevant scf */ - scfAct = scf[sfbAct]; - scfLast = &scfAct; - scfNext = scf + sfbNext; - scfMin = *scfNext; - scfMax = *scfNext; - } - else if ((sfbLast>=0) && (sfbAct>=0) && (sfbNext==psyOutChan->sfbCnt)) { - /* last relevant scf */ - scfAct = scf[sfbAct]; - scfLast = scf + sfbLast; - scfNext = &scfAct; - scfMin = *scfLast; - scfMax = *scfLast; - } - if (sfbAct>=0) - scfMin = fixMax(scfMin, minScf[sfbAct]); - - if ((sfbAct >= 0) && - (sfbLast>=0 || sfbNextsfbCnt) && - (scfAct > scfMin) && - (scfAct <= scfMin+MAX_SCF_DELTA) && - (scfAct >= scfMax-MAX_SCF_DELTA) && - (*scfLast != prevScfLast[sfbAct] || - *scfNext != prevScfNext[sfbAct] || - deltaPe < deltaPeLast[sfbAct])) { - /* bigger than neighbouring scf found, try to use smaller scf */ - success = 0; - - sfbWidth = psyOutChan->sfbOffsets[sfbAct+1] - psyOutChan->sfbOffsets[sfbAct]; - sfbOffs = psyOutChan->sfbOffsets[sfbAct]; - - /* estimate required bits for actual scf */ - enLdData = qcOutChannel->sfbEnergyLdData[sfbAct]; - - /* sfbConstPePart[sfbAct] = (float)log(6.75f*en/sfbFormFactor[sfbAct]) * LOG2_1; */ - /* 0.02152255861f = log(6.75)/log(2)/AS_PE_FAC_FLOAT; LOG2_1 is 1.0 for log2 */ - /* 0.09375f = log(64.0)/log(2.0)/64.0 = scale of sfbFormFactorLdData */ - if (sfbConstPePart[sfbAct] == (FIXP_DBL)FDK_INT_MIN) { - sfbConstPePart[sfbAct] = ((enLdData - sfbFormFactorLdData[sfbAct] - FL2FXCONST_DBL(0.09375f)) >> 1) + FL2FXCONST_DBL(0.02152255861f); - } - - sfbPeOld = FDKaacEnc_calcSingleSpecPe(scfAct,sfbConstPePart[sfbAct],sfbNRelevantLines[sfbAct]) - +FDKaacEnc_countSingleScfBits(scfAct, *scfLast, *scfNext); - - deltaPeNew = deltaPe; - updateMinScfCalculated = 1; - - do { - /* estimate required bits for smaller scf */ - scfAct--; - /* check only if the same check was not done before */ - if (scfAct < minScfCalculated[sfbAct] && scfAct>=scfMax-MAX_SCF_DELTA){ - /* estimate required bits for new scf */ - sfbPeNew = FDKaacEnc_calcSingleSpecPe(scfAct,sfbConstPePart[sfbAct],sfbNRelevantLines[sfbAct]) - +FDKaacEnc_countSingleScfBits(scfAct,*scfLast, *scfNext); - - /* use new scf if no increase in pe and - quantization error is smaller */ - deltaPeTmp = deltaPe + sfbPeNew - sfbPeOld; - /* 0.0006103515625f = 10.0f/(2^(2*AS_PE_FAC_SHIFT)) */ - if (deltaPeTmp < FL2FXCONST_DBL(0.0006103515625f)) { - /* distortion of new scf */ - sfbDistNew = FDKaacEnc_calcSfbDist(qcOutChannel->mdctSpectrum+sfbOffs, - quantSpecTmp+sfbOffs, - sfbWidth, - scfAct, - dZoneQuantEnable); - - if (sfbDistNew < sfbDist[sfbAct]) { - /* success, replace scf by new one */ - scf[sfbAct] = scfAct; - sfbDist[sfbAct] = sfbDistNew; - - for (k=0; k scfMin); - - deltaPe = deltaPeNew; - - /* save parameters to avoid multiple computations of the same sfb */ - prevScfLast[sfbAct] = *scfLast; - prevScfNext[sfbAct] = *scfNext; - deltaPeLast[sfbAct] = deltaPe; - } - - if (success && restartOnSuccess) { - /* start again at first sfb */ - sfbLast = -1; - sfbAct = -1; - sfbNext = -1; - scfLast = 0; - scfNext = 0; - scfMin = FDK_INT_MAX; - scfMax = FDK_INT_MAX; - success = 0; - } - else { - /* shift sfbs for next band */ - sfbLast = sfbAct; - sfbAct = sfbNext; - } - } while (sfbNext < psyOutChan->sfbCnt); -} - -/* - Function: FDKaacEnc_assimilateMultipleScf - -*/ -static void FDKaacEnc_assimilateMultipleScf(PSY_OUT_CHANNEL *psyOutChan, - QC_OUT_CHANNEL *qcOutChannel, - SHORT *quantSpec, - SHORT *quantSpecTmp, - INT dZoneQuantEnable, - INT *scf, - INT *minScf, - FIXP_DBL *sfbDist, - FIXP_DBL *sfbConstPePart, - FIXP_DBL *sfbFormFactorLdData, - FIXP_DBL *sfbNRelevantLines) -{ - INT sfb, startSfb, stopSfb; - INT scfTmp[MAX_GROUPED_SFB], scfMin, scfMax, scfAct; - INT possibleRegionFound; - INT sfbWidth, sfbOffs, i, k; - FIXP_DBL sfbDistNew[MAX_GROUPED_SFB], distOldSum, distNewSum; - INT deltaScfBits; - FIXP_DBL deltaSpecPe; - FIXP_DBL deltaPe = FL2FXCONST_DBL(0.0f); - FIXP_DBL deltaPeNew; - INT sfbCnt = psyOutChan->sfbCnt; - - /* calc min and max scalfactors */ - scfMin = FDK_INT_MAX; - scfMax = FDK_INT_MIN; - for (sfb=0; sfb scfAct)) - sfb++; - stopSfb = sfb; - - /* check if in all sfb of a valid region scfAct >= minScf[sfb] */ - possibleRegionFound = 0; - if (startSfb < sfbCnt) { - possibleRegionFound = 1; - for (sfb=startSfb; sfb> DIST_FAC_SHIFT; - - sfbWidth = psyOutChan->sfbOffsets[sfb+1] - psyOutChan->sfbOffsets[sfb]; - sfbOffs = psyOutChan->sfbOffsets[sfb]; - - sfbDistNew[sfb] = FDKaacEnc_calcSfbDist(qcOutChannel->mdctSpectrum+sfbOffs, - quantSpecTmp+sfbOffs, - sfbWidth, - scfAct, - dZoneQuantEnable); - - if (sfbDistNew[sfb] >qcOutChannel->sfbThresholdLdData[sfb]) { - /* no improvement, skip further dist. calculations */ - distNewSum = distOldSum << 1; - break; - } - distNewSum += CalcInvLdData(sfbDistNew[sfb]) >> DIST_FAC_SHIFT; - } - } - /* distortion smaller ? -> use new scalefactors */ - if (distNewSum < distOldSum) { - deltaPe = deltaPeNew; - for (sfb=startSfb; sfbsfbOffsets[sfb+1] - - psyOutChan->sfbOffsets[sfb]; - sfbOffs = psyOutChan->sfbOffsets[sfb]; - scf[sfb] = scfAct; - sfbDist[sfb] = sfbDistNew[sfb]; - - for (k=0; k scfMin); - } -} - -/* - Function: FDKaacEnc_FDKaacEnc_assimilateMultipleScf2 - -*/ -static void FDKaacEnc_FDKaacEnc_assimilateMultipleScf2(PSY_OUT_CHANNEL *psyOutChan, - QC_OUT_CHANNEL *qcOutChannel, - SHORT *quantSpec, - SHORT *quantSpecTmp, - INT dZoneQuantEnable, - INT *scf, - INT *minScf, - FIXP_DBL *sfbDist, - FIXP_DBL *sfbConstPePart, - FIXP_DBL *sfbFormFactorLdData, - FIXP_DBL *sfbNRelevantLines) -{ - INT sfb, startSfb, stopSfb; - INT scfTmp[MAX_GROUPED_SFB], scfAct, scfNew; - INT scfPrev, scfNext, scfPrevNextMin, scfPrevNextMax, scfLo, scfHi; - INT scfMin, scfMax; - INT *sfbOffs = psyOutChan->sfbOffsets; - FIXP_DBL sfbDistNew[MAX_GROUPED_SFB], sfbDistMax[MAX_GROUPED_SFB]; - FIXP_DBL distOldSum, distNewSum; - INT deltaScfBits; - FIXP_DBL deltaSpecPe; - FIXP_DBL deltaPe = FL2FXCONST_DBL(0.0f); - FIXP_DBL deltaPeNew = FL2FXCONST_DBL(0.0f); - INT sfbCnt = psyOutChan->sfbCnt; - INT bSuccess, bCheckScf; - INT i,k; - - /* calc min and max scalfactors */ - scfMin = FDK_INT_MAX; - scfMax = FDK_INT_MIN; - for (sfb=0; sfb= scfAct) - scfLo = fixMin(scfAct, scfPrevNextMin); - else - scfLo = scfPrevNextMax; - - if (startSfb < sfbCnt && scfHi-scfLo <= MAX_SCF_DELTA) { /* region found */ - /* 1. try to save bits by coarser quantization */ - if (scfHi > scf[startSfb]) { - /* calculate the allowed distortion */ - for (sfb=startSfb; sfbsfbThreshold[sfb]*sfbDist[sfb]*sfbDist[sfb],1.0f/3.0f); */ - /* sfbDistMax[sfb] = fixMax(sfbDistMax[sfb],qcOutChannel->sfbEnergy[sfb]*FL2FXCONST_DBL(1.e-3f)); */ - /* -0.15571537944 = ld64(1.e-3f)*/ - sfbDistMax[sfb] = fMult(FL2FXCONST_DBL(1.0f/3.0f),qcOutChannel->sfbThresholdLdData[sfb])+fMult(FL2FXCONST_DBL(1.0f/3.0f),sfbDist[sfb])+fMult(FL2FXCONST_DBL(1.0f/3.0f),sfbDist[sfb]); - sfbDistMax[sfb] = fixMax(sfbDistMax[sfb],qcOutChannel->sfbEnergyLdData[sfb]-FL2FXCONST_DBL(0.15571537944)); - sfbDistMax[sfb] = fixMin(sfbDistMax[sfb],qcOutChannel->sfbThresholdLdData[sfb]); - } - } - - /* loop over all possible scf values for this region */ - bCheckScf = 1; - for (scfNew=scf[startSfb]+1; scfNew<=scfHi; scfNew++) { - for (k=0; kmdctSpectrum+sfbOffs[sfb], - quantSpecTmp+sfbOffs[sfb], - sfbOffs[sfb+1]-sfbOffs[sfb], - scfNew, - dZoneQuantEnable); - - if (sfbDistNew[sfb] > sfbDistMax[sfb]) { - /* no improvement, skip further dist. calculations */ - bSuccess = 0; - if (sfbDistNew[sfb] == qcOutChannel->sfbEnergyLdData[sfb]) { - /* if whole sfb is already quantized to 0, further - checks with even coarser quant. are useless*/ - bCheckScf = 0; - } - break; - } - } - } - if (bCheckScf==0) /* further calculations useless ? */ - break; - /* distortion small enough ? -> use new scalefactors */ - if (bSuccess) { - deltaPe = deltaPeNew; - for (sfb=startSfb; sfb= minScf[sfb] */ - for (sfb=startSfb; sfb> DIST_FAC_SHIFT; - - sfbDistNew[sfb] = FDKaacEnc_calcSfbDist(qcOutChannel->mdctSpectrum+sfbOffs[sfb], - quantSpecTmp+sfbOffs[sfb], - sfbOffs[sfb+1]-sfbOffs[sfb], - scfNew, - dZoneQuantEnable); - - if (sfbDistNew[sfb] > qcOutChannel->sfbThresholdLdData[sfb]) { - /* no improvement, skip further dist. calculations */ - distNewSum = distOldSum << 1; - break; - } - distNewSum += CalcInvLdData(sfbDistNew[sfb]) >> DIST_FAC_SHIFT; - } - } - /* distortion smaller ? -> use new scalefactors */ - if (distNewSum < fMult(FL2FXCONST_DBL(0.8f),distOldSum)) { - deltaPe = deltaPeNew; - for (sfb=startSfb; sfbmdctSpectrum+sfbOffs[sfb], - quantSpec+sfbOffs[sfb], - sfbOffs[sfb+1]-sfbOffs[sfb], scfNew, - &sfbEnQ, &sfbDistNew[sfb]); - - distOldSum += CalcInvLdData(sfbDist[sfb]) >> DIST_FAC_SHIFT; - distNewSum += CalcInvLdData(sfbDistNew[sfb]) >> DIST_FAC_SHIFT; - - /* 0.00259488556167 = ld64(1.122f) */ - /* -0.00778722686652 = ld64(0.7079f) */ - if ((sfbDistNew[sfb] > (sfbDist[sfb]+FL2FXCONST_DBL(0.00259488556167f))) || (sfbEnQ < (qcOutChannel->sfbEnergyLdData[sfb] - FL2FXCONST_DBL(0.00778722686652f)))){ - bSuccess = 0; - break; - } - } - } - /* distortion smaller ? -> use new scalefactors */ - if (distNewSum < distOldSum && bSuccess) { - deltaPe = deltaPeNew; - for (sfb=startSfb; sfb C1/2^8 */ - - - - if (invQuant>0) { - FDKmemclear(quantSpec, (1024)*sizeof(SHORT)); - } - - /* scfs without energy or with thresh>energy are marked with FDK_INT_MIN */ - for(i=0; isfbCnt; i++) { - scf[i] = FDK_INT_MIN; - } - - for (i=0; isfbCnt; sfbOffs+=psyOutChannel->sfbPerGroup) { - for(sfb=0; sfbmaxSfbPerGroup; sfb++) { - - threshLdData = qcOutChannel->sfbThresholdLdData[sfbOffs+sfb]; - energyLdData = qcOutChannel->sfbEnergyLdData[sfbOffs+sfb]; - - sfbDistLdData[sfbOffs+sfb] = energyLdData; - - - if (energyLdData > threshLdData) { - FIXP_DBL tmp; - - /* energyPart = (float)log10(sfbFormFactor[sfbOffs+sfb]); */ - /* 0.09375f = log(64.0)/log(2.0)/64.0 = scale of sfbFormFactorLdData */ - energyPartLdData = sfbFormFactorLdData[sfbOffs+sfb] + FL2FXCONST_DBL(0.09375f); - - /* influence of allowed distortion */ - /* thresholdPart = (float)log10(6.75*thresh+FLT_MIN); */ - thresholdPartLdData = threshConstLdData + threshLdData; - - /* scf calc */ - /* scfFloat = 8.8585f * (thresholdPart - energyPart); */ - scfFract = thresholdPartLdData - energyPartLdData; - /* conversion from log2 to log10 */ - scfFract = fMult(convConst,scfFract); - /* (8.8585f * scfFract)/8 = 8/8 * scfFract + 0.8585 * scfFract/8 */ - scfFract = scfFract + fMult(FL2FXCONST_DBL(0.8585f),scfFract >> 3); - - /* integer scalefactor */ - /* scfInt = (int)floor(scfFloat); */ - scfInt = (INT)(scfFract>>((DFRACT_BITS-1)-3-LD_DATA_SHIFT)); /* 3 bits => scfFract/8.0; 6 bits => ld64 */ - - /* maximum of spectrum */ - maxSpec = FL2FXCONST_DBL(0.0f); - - for(j=psyOutChannel->sfbOffsets[sfbOffs+sfb]; jsfbOffsets[sfbOffs+sfb+1]; j++ ){ - absSpec = fixp_abs(qcOutChannel->mdctSpectrum[j]); - maxSpec = (absSpec > maxSpec) ? absSpec : maxSpec; - } - - /* lower scf limit to avoid quantized values bigger than MAX_QUANT */ - /* C1 = -69.33295f, C2 = 5.77078f = 4/log(2) */ - /* minSfMaxQuant[sfbOffs+sfb] = (int)ceil(C1 + C2*log(maxSpec)); */ - /* C1/2^8 + 4/log(2.0)*log(maxSpec)/2^8 => C1/2^8 + log(maxSpec)/log(2.0)*4/2^8 => C1/2^8 + log(maxSpec)/log(2.0)/64.0 */ - - //minSfMaxQuant[sfbOffs+sfb] = ((INT) ((c1Const + CalcLdData(maxSpec)) >> ((DFRACT_BITS-1)-8))) + 1; - tmp = CalcLdData(maxSpec); - if (c1Const>FL2FXCONST_DBL(-1.f)-tmp) { - minSfMaxQuant[sfbOffs+sfb] = ((INT) ((c1Const + tmp) >> ((DFRACT_BITS-1)-8))) + 1; - } - else { - minSfMaxQuant[sfbOffs+sfb] = ((INT) (FL2FXCONST_DBL(-1.f) >> ((DFRACT_BITS-1)-8))) + 1; - } - - scfInt = fixMax(scfInt, minSfMaxQuant[sfbOffs+sfb]); - - - /* find better scalefactor with analysis by synthesis */ - if (invQuant>0) { - scfInt = FDKaacEnc_improveScf(qcOutChannel->mdctSpectrum+psyOutChannel->sfbOffsets[sfbOffs+sfb], - quantSpec+psyOutChannel->sfbOffsets[sfbOffs+sfb], - quantSpecTmp+psyOutChannel->sfbOffsets[sfbOffs+sfb], - psyOutChannel->sfbOffsets[sfbOffs+sfb+1]-psyOutChannel->sfbOffsets[sfbOffs+sfb], - threshLdData, scfInt, minSfMaxQuant[sfbOffs+sfb], - &sfbDistLdData[sfbOffs+sfb], &minScfCalculated[sfbOffs+sfb], - dZoneQuantEnable - ); - } - scf[sfbOffs+sfb] = scfInt; - } - } - } - - - if (invQuant>1) { - /* try to decrease scf differences */ - FIXP_DBL sfbConstPePart[MAX_GROUPED_SFB]; - FIXP_DBL sfbNRelevantLines[MAX_GROUPED_SFB]; - - for (i=0; isfbCnt; i++) - sfbConstPePart[i] = (FIXP_DBL)FDK_INT_MIN; - - FDKaacEnc_calcSfbRelevantLines( sfbFormFactorLdData, - qcOutChannel->sfbEnergyLdData, - qcOutChannel->sfbThresholdLdData, - psyOutChannel->sfbOffsets, - psyOutChannel->sfbCnt, - psyOutChannel->sfbPerGroup, - psyOutChannel->maxSfbPerGroup, - sfbNRelevantLines); - - - FDKaacEnc_assimilateSingleScf(psyOutChannel, qcOutChannel, quantSpec, quantSpecTmp, - dZoneQuantEnable, - scf, - minSfMaxQuant, sfbDistLdData, sfbConstPePart, - sfbFormFactorLdData, sfbNRelevantLines, minScfCalculated, 1); - - if(invQuant > 1) { - FDKaacEnc_assimilateMultipleScf(psyOutChannel, qcOutChannel, quantSpec, quantSpecTmp, - dZoneQuantEnable, - scf, - minSfMaxQuant, sfbDistLdData, sfbConstPePart, - sfbFormFactorLdData, sfbNRelevantLines); - - FDKaacEnc_assimilateMultipleScf(psyOutChannel, qcOutChannel, quantSpec, quantSpecTmp, - dZoneQuantEnable, - scf, - minSfMaxQuant, sfbDistLdData, sfbConstPePart, - sfbFormFactorLdData, sfbNRelevantLines); - - - FDKaacEnc_FDKaacEnc_assimilateMultipleScf2(psyOutChannel, qcOutChannel, quantSpec, quantSpecTmp, - dZoneQuantEnable, - scf, - minSfMaxQuant, sfbDistLdData, sfbConstPePart, - sfbFormFactorLdData, sfbNRelevantLines); - } - } - - - /* get min scalefac */ - minSf = FDK_INT_MAX; - for (sfbOffs=0; sfbOffssfbCnt; sfbOffs+=psyOutChannel->sfbPerGroup) { - for (sfb = 0; sfb < psyOutChannel->maxSfbPerGroup; sfb++) { - if (scf[sfbOffs+sfb]!=FDK_INT_MIN) - minSf = fixMin(minSf,scf[sfbOffs+sfb]); - } - } - - /* limit scf delta */ - for (sfbOffs=0; sfbOffssfbCnt; sfbOffs+=psyOutChannel->sfbPerGroup) { - for (sfb = 0; sfb < psyOutChannel->maxSfbPerGroup; sfb++) { - if ((scf[sfbOffs+sfb] != FDK_INT_MIN) && (minSf+MAX_SCF_DELTA) < scf[sfbOffs+sfb]) { - scf[sfbOffs+sfb] = minSf + MAX_SCF_DELTA; - if (invQuant > 0) { /* changed bands need to be quantized again */ - sfbDistLdData[sfbOffs+sfb] = - FDKaacEnc_calcSfbDist(qcOutChannel->mdctSpectrum+psyOutChannel->sfbOffsets[sfbOffs+sfb], - quantSpec+psyOutChannel->sfbOffsets[sfbOffs+sfb], - psyOutChannel->sfbOffsets[sfbOffs+sfb+1]-psyOutChannel->sfbOffsets[sfbOffs+sfb], - scf[sfbOffs+sfb], - dZoneQuantEnable - ); - } - } - } - } - - - /* get max scalefac for global gain */ - maxSf = FDK_INT_MIN; - for (sfbOffs=0; sfbOffssfbCnt; sfbOffs+=psyOutChannel->sfbPerGroup) { - for (sfb = 0; sfb < psyOutChannel->maxSfbPerGroup; sfb++) { - maxSf = fixMax(maxSf,scf[sfbOffs+sfb]); - } - } - - /* calc loop scalefactors, if spec is not all zero (i.e. maxSf == -99) */ - if( maxSf > FDK_INT_MIN ) { - *globalGain = maxSf; - for (sfbOffs=0; sfbOffssfbCnt; sfbOffs+=psyOutChannel->sfbPerGroup) { - for (sfb = 0; sfb < psyOutChannel->maxSfbPerGroup; sfb++) { - if( scf[sfbOffs+sfb] == FDK_INT_MIN ) { - scf[sfbOffs+sfb] = 0; - /* set band explicitely to zero */ - for(j=psyOutChannel->sfbOffsets[sfbOffs+sfb]; jsfbOffsets[sfbOffs+sfb+1]; j++ ) { - qcOutChannel->mdctSpectrum[j] = FL2FXCONST_DBL(0.0f); - } - } - else { - scf[sfbOffs+sfb] = maxSf - scf[sfbOffs+sfb]; - } - } - } - } - else{ - *globalGain = 0; - /* set spectrum explicitely to zero */ - for (sfbOffs=0; sfbOffssfbCnt; sfbOffs+=psyOutChannel->sfbPerGroup) { - for (sfb = 0; sfb < psyOutChannel->maxSfbPerGroup; sfb++) { - scf[sfbOffs+sfb] = 0; - /* set band explicitely to zero */ - for(j=psyOutChannel->sfbOffsets[sfbOffs+sfb]; jsfbOffsets[sfbOffs+sfb+1]; j++ ) { - qcOutChannel->mdctSpectrum[j] = FL2FXCONST_DBL(0.0f); - } - } - } - } - - /* free quantSpecTmp from scratch */ - C_ALLOC_SCRATCH_END(quantSpecTmp, SHORT, (1024)); - - -} - -void -FDKaacEnc_EstimateScaleFactors(PSY_OUT_CHANNEL *psyOutChannel[], - QC_OUT_CHANNEL* qcOutChannel[], - const int invQuant, - const INT dZoneQuantEnable, - const int nChannels) -{ - int ch; - - for (ch = 0; ch < nChannels; ch++) - { - FDKaacEnc_FDKaacEnc_EstimateScaleFactorsChannel(qcOutChannel[ch], - psyOutChannel[ch], - qcOutChannel[ch]->scf, - &qcOutChannel[ch]->globalGain, - qcOutChannel[ch]->sfbFormFactorLdData - ,invQuant, - qcOutChannel[ch]->quantSpec, - dZoneQuantEnable - ); - } - -} - diff --git a/libAACenc/src/sf_estim.h b/libAACenc/src/sf_estim.h deleted file mode 100644 index ef8d366..0000000 --- a/libAACenc/src/sf_estim.h +++ /dev/null @@ -1,118 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M. Werner - contents/description: Scale factor estimation - -******************************************************************************/ - -#ifndef _SF_ESTIM_H -#define _SF_ESTIM_H - -#include "common_fix.h" - - -#include "psy_const.h" -#include "qc_data.h" -#include "interface.h" - -#define FORM_FAC_SHIFT 6 - - -void -FDKaacEnc_CalcFormFactor(QC_OUT_CHANNEL *qcOutChannel[(2)], - PSY_OUT_CHANNEL *psyOutChannel[(2)], - const INT nChannels); - -void -FDKaacEnc_EstimateScaleFactors(PSY_OUT_CHANNEL *psyOutChannel[], - QC_OUT_CHANNEL* qcOutChannel[], - const int invQuant, - const INT dZoneQuantEnable, - const int nChannels); - - - -#endif diff --git a/libAACenc/src/spreading.cpp b/libAACenc/src/spreading.cpp deleted file mode 100644 index 852da1e..0000000 --- a/libAACenc/src/spreading.cpp +++ /dev/null @@ -1,114 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M.Werner - contents/description: Spreading of energy - -******************************************************************************/ - -#include "spreading.h" - -void FDKaacEnc_SpreadingMax(const INT pbCnt, - const FIXP_DBL *RESTRICT maskLowFactor, - const FIXP_DBL *RESTRICT maskHighFactor, - FIXP_DBL *RESTRICT pbSpreadEnergy) -{ - int i; - FIXP_DBL delay; - - /* slope to higher frequencies */ - delay = pbSpreadEnergy[0]; - for (i=1; i=0; i--) { - delay = fixMax(pbSpreadEnergy[i], fMult(maskLowFactor[i],delay)); - pbSpreadEnergy[i] = delay; - } -} diff --git a/libAACenc/src/spreading.h b/libAACenc/src/spreading.h deleted file mode 100644 index e1b506c..0000000 --- a/libAACenc/src/spreading.h +++ /dev/null @@ -1,102 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M.Werner - contents/description: Spreading of energy and weighted tonality - -******************************************************************************/ - -#ifndef _SPREADING_H -#define _SPREADING_H - -#include "common_fix.h" - - -void FDKaacEnc_SpreadingMax(const INT pbCnt, - const FIXP_DBL *RESTRICT maskLowFactor, - const FIXP_DBL *RESTRICT maskHighFactor, - FIXP_DBL *RESTRICT pbSpreadEnergy); - -#endif /* #ifndef _SPREADING_H */ diff --git a/libAACenc/src/tns_func.h b/libAACenc/src/tns_func.h deleted file mode 100644 index 5e5265d..0000000 --- a/libAACenc/src/tns_func.h +++ /dev/null @@ -1,145 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: Alex Goeschel - contents/description: Temporal noise shaping - -******************************************************************************/ - -#ifndef _TNS_FUNC_H -#define _TNS_FUNC_H - -#include "common_fix.h" - -#include "psy_configuration.h" - -AAC_ENCODER_ERROR FDKaacEnc_InitTnsConfiguration(INT bitrate, - INT samplerate, - INT channels, - INT blocktype, - INT granuleLength, - INT isLowDelay, - INT ldSbrPresent, - TNS_CONFIG *tnsConfig, - PSY_CONFIGURATION *psyConfig, - INT active, - INT useTnsPeak ); - -INT FDKaacEnc_TnsDetect( - TNS_DATA *tnsData, - const TNS_CONFIG *tC, - TNS_INFO* tnsInfo, - INT sfbCnt, - FIXP_DBL *spectrum, - INT subBlockNumber, - INT blockType - ); - - - -void FDKaacEnc_TnsSync( - TNS_DATA *tnsDataDest, - const TNS_DATA *tnsDataSrc, - TNS_INFO *tnsInfoDest, - TNS_INFO *tnsInfoSrc, - const INT blockTypeDest, - const INT blockTypeSrc, - const TNS_CONFIG *tC - ); - -INT FDKaacEnc_TnsEncode( - TNS_INFO* tnsInfo, - TNS_DATA* tnsData, - const INT numOfSfb, - const TNS_CONFIG *tC, - const INT lowPassLine, - FIXP_DBL* spectrum, - const INT subBlockNumber, - const INT blockType - ); - - - -#endif /* _TNS_FUNC_H */ diff --git a/libAACenc/src/tonality.cpp b/libAACenc/src/tonality.cpp deleted file mode 100644 index 7246a34..0000000 --- a/libAACenc/src/tonality.cpp +++ /dev/null @@ -1,204 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - author: M. Werner - contents/description: Convert chaos measure to the tonality index - -******************************************************************************/ - -#include "tonality.h" -#include "chaosmeasure.h" - -static const FIXP_DBL normlog = (FIXP_DBL)0xd977d949; /*FL2FXCONST_DBL(-0.4342944819f * FDKlog(2.0)/FDKlog(2.7182818)); */ - -static void FDKaacEnc_CalcSfbTonality(FIXP_DBL *RESTRICT spectrum, - INT *RESTRICT sfbMaxScaleSpec, - FIXP_DBL *RESTRICT chaosMeasure, - FIXP_SGL *RESTRICT sfbTonality, - INT sfbCnt, - const INT *RESTRICT sfbOffset, - FIXP_DBL *RESTRICT sfbEnergyLD64 ); - - -void FDKaacEnc_CalculateFullTonality(FIXP_DBL *RESTRICT spectrum, - INT *RESTRICT sfbMaxScaleSpec, - FIXP_DBL *RESTRICT sfbEnergyLD64, - FIXP_SGL *RESTRICT sfbTonality, - INT sfbCnt, - const INT *sfbOffset, - INT usePns) -{ - INT j; -#if defined(ARCH_PREFER_MULT_32x16) - FIXP_SGL alpha_0 = FL2FXCONST_SGL(0.25f); /* used in smooth ChaosMeasure */ - FIXP_SGL alpha_1 = FL2FXCONST_SGL(1.0f-0.25f); /* used in smooth ChaosMeasure */ -#else - FIXP_DBL alpha_0 = FL2FXCONST_DBL(0.25f); /* used in smooth ChaosMeasure */ - FIXP_DBL alpha_1 = FL2FXCONST_DBL(1.0f-0.25f); /* used in smooth ChaosMeasure */ -#endif - INT numberOfLines = sfbOffset[sfbCnt]; - - if (!usePns) - return; - - C_ALLOC_SCRATCH_START(chaosMeasurePerLine, FIXP_DBL, (1024)); - /* calculate chaos measure */ - FDKaacEnc_CalculateChaosMeasure(spectrum, - numberOfLines, - chaosMeasurePerLine); - - /* smooth ChaosMeasure */ - for (j=1;j 7/2 = 4 (spc*spc) */ - - FIXP_DBL chaosMeasureSfb = FL2FXCONST_DBL(0.0); - - /* calc chaosMeasurePerSfb */ - for (j=(sfbOffset[i+1]-sfbOffset[i])-1; j>=0; j--) { - FIXP_DBL tmp = (*spectrum++)< FL2FXCONST_DBL(-0.0519051) ) /* > ld(0.05)+ld(2) */ - { - if (chaosMeasureSfbLD64 <= FL2FXCONST_DBL(0.0) ) - sfbTonality[i] = FX_DBL2FX_SGL(fMultDiv2( chaosMeasureSfbLD64 , normlog ) << 7); - else - sfbTonality[i] = FL2FXCONST_SGL(0.0); - } - else - sfbTonality[i] = (FIXP_SGL)MAXVAL_SGL; - } - else - sfbTonality[i] = (FIXP_SGL)MAXVAL_SGL; - } -} diff --git a/libAACenc/src/tonality.h b/libAACenc/src/tonality.h deleted file mode 100644 index fbe78ee..0000000 --- a/libAACenc/src/tonality.h +++ /dev/null @@ -1,108 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - author: M. Lohwasser - contents/description: Calculate tonality index - -******************************************************************************/ - -#ifndef __TONALITY_H -#define __TONALITY_H - -#include "common_fix.h" - - -#include "chaosmeasure.h" - - -void FDKaacEnc_CalculateFullTonality( FIXP_DBL *RESTRICT spectrum, - INT *RESTRICT sfbMaxScaleSpec, - FIXP_DBL *RESTRICT sfbEnergyLD64, - FIXP_SGL *RESTRICT sfbTonality, - INT sfbCnt, - const INT *sfbOffset, - INT usePns); - -#endif diff --git a/libAACenc/src/transform.cpp b/libAACenc/src/transform.cpp deleted file mode 100644 index 690b82e..0000000 --- a/libAACenc/src/transform.cpp +++ /dev/null @@ -1,264 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/***************************************************************************** - - Description: FDKaacLdEnc_MdctTransform480: - The module FDKaacLdEnc_MdctTransform will perform the MDCT. - The MDCT supports the sine window and - the zero padded window. The algorithm of the MDCT - can be divided in Windowing, PreModulation, Fft and - PostModulation. - -******************************************************************************/ - -#include "transform.h" - -#include "dct.h" -#include "psy_const.h" -#include "aacEnc_rom.h" -#include "FDK_tools_rom.h" - -INT FDKaacEnc_Transform_Real (const INT_PCM * pTimeData, - FIXP_DBL *RESTRICT mdctData, - const INT blockType, - const INT windowShape, - INT *prevWindowShape, - const INT frameLength, - INT *mdctData_e, - INT filterType - ,FIXP_DBL * RESTRICT overlapAddBuffer - ) -{ - const INT_PCM * RESTRICT timeData; - - INT i; - /* tl: transform length - fl: left window slope length - nl: left window slope offset - fr: right window slope length - nr: right window slope offset - See FDK_tools/doc/intern/mdct.tex for more detail. */ - int tl, fl, nl, fr, nr; - - const FIXP_WTP * RESTRICT pLeftWindowPart; - const FIXP_WTP * RESTRICT pRightWindowPart; - - /* - * MDCT scale: - * + 1: fMultDiv2() in windowing. - * + 1: Because of factor 1/2 in Princen-Bradley compliant windowed TDAC. - */ - *mdctData_e = 1+1; - - tl = frameLength; - timeData = pTimeData; - - switch( blockType ) { - case LONG_WINDOW: - { - int offset = (windowShape == LOL_WINDOW) ? ((frameLength * 3)>>2) : 0; - fl = frameLength - offset; - fr = frameLength - offset; - } - break; - case STOP_WINDOW: - fl = frameLength >> 3; - fr = frameLength; - break; - case START_WINDOW: /* or StopStartSequence */ - fl = frameLength; - fr = frameLength >> 3; - break; - case SHORT_WINDOW: - fl = fr = frameLength >> 3; - tl >>= 3; - timeData = pTimeData + 3*fl + (fl/2); - break; - default: - FDK_ASSERT(0); - return -1; - break; - } - - /* Taken from FDK_tools/src/mdct.cpp Derive NR and NL */ - nr = (tl - fr)>>1; - nl = (tl - fl)>>1; - - pLeftWindowPart = FDKgetWindowSlope(fl, *prevWindowShape); - pRightWindowPart = FDKgetWindowSlope(fr, windowShape); - - /* windowing */ - if (filterType != FB_ELD) - { - /* Left window slope offset */ - for (i=0; i> ( 1 ); -#else - mdctData[(tl/2)+i] = - (FIXP_DBL) timeData[tl-i-1] << (DFRACT_BITS - SAMPLE_BITS - 1); -#endif - } - /* Left window slope */ - for (i=0; i> (1); -#else - mdctData[(tl/2)-1-i] = - (FIXP_DBL) timeData[tl+i] << (DFRACT_BITS - SAMPLE_BITS - 1); -#endif - } - /* Right window slope */ - for (i=0; i> (-WTS1)); - outval += (fMultDiv2((FIXP_PCM)timeData[L+N*3/4+i], pWindowELD[N+N/2+i]) >> (-WTS1) ); - outval += (fMultDiv2(overlapAddBuffer[N/2+i], pWindowELD[2*N+i])>> (-WTS2-1)); - - overlapAddBuffer[N/2+i] = overlapAddBuffer[i]; - - overlapAddBuffer[i] = z0; - mdctData[i] = overlapAddBuffer[N/2+i] + (fMultDiv2(overlapAddBuffer[N+N/2-1-i], pWindowELD[2*N+N/2+i]) >> (-WTS2-1)); - - mdctData[N-1-i] = outval; - overlapAddBuffer[N+N/2-1-i] = outval; - } - - for(i=N/4;i> (-WTS1)) ; - outval += (fMultDiv2(overlapAddBuffer[N/2+i], pWindowELD[2*N+i]) >> (-WTS2-1)); - - overlapAddBuffer[N/2+i] = overlapAddBuffer[i] + (fMult((FIXP_PCM)timeData[L-N/4+i], pWindowELD[N/2+i])<< (WTS0-1) ); - - overlapAddBuffer[i] = z0; - mdctData[i] = overlapAddBuffer[N/2+i] + (fMultDiv2(overlapAddBuffer[N+N/2-1-i], pWindowELD[2*N+N/2+i]) >> (-WTS2-1)); - - mdctData[N-1-i] = outval; - overlapAddBuffer[N+N/2-1-i] = outval; - } - } - - dct_IV(mdctData, tl, mdctData_e); - - *prevWindowShape = windowShape; - - return 0; -} - diff --git a/libAACenc/src/transform.h b/libAACenc/src/transform.h deleted file mode 100644 index 5053174..0000000 --- a/libAACenc/src/transform.h +++ /dev/null @@ -1,123 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: M. Werner - contents/description: MDCT Transform - -******************************************************************************/ - -#ifndef _TRANSFORM_H -#define _TRANSFORM_H - -#include "common_fix.h" - -#define WTS0 1 -#define WTS1 0 -#define WTS2 -2 - -/** - * \brief: Performe MDCT transform of time domain data. - * \param timeData pointer to time domain input signal. - * \param mdctData pointer to store frequency domain output data. - * \param blockType index indicating the type of block. Either - * LONG_WINDOW, START_WINDOW, SHORT_WINDOW or STOP_WINDOW. - * \param windowShape index indicating the window slope type to be used. - * Values allowed are either SINE_WINDOW or KBD_WINDOW. - * \param frameLength length of the block. - * \param mdctData_e pointer to an INT where the exponent of the frequency - * domain output data is stored into. - * \return 0 in case of success, non-zero in case of error (inconsistent parameters). - */ -INT FDKaacEnc_Transform_Real (const INT_PCM *timeData, - FIXP_DBL *RESTRICT mdctData, - const INT blockType, - const INT windowShape, - INT *prevWindowShape, - const INT frameLength, - INT *mdctData_e, - INT filterType - ,FIXP_DBL * RESTRICT overlapAddBuffer - ); -#endif diff --git a/libMpegTPDec/include/tp_data.h b/libMpegTPDec/include/tp_data.h index c6e89b5..6653392 100644 --- a/libMpegTPDec/include/tp_data.h +++ b/libMpegTPDec/include/tp_data.h @@ -102,7 +102,7 @@ amm-info@iis.fraunhofer.de /* #define TP_CELP_ENABLE */ /* #define TP_HVXC_ENABLE */ /* #define TP_SLS_ENABLE */ -#define TP_ELD_ENABLE +/* #define TP_ELD_ENABLE */ /* #define TP_USAC_ENABLE */ /* #define TP_RSVD50_ENABLE */ @@ -207,26 +207,6 @@ typedef struct { -#ifdef TP_ELD_ENABLE - -typedef enum { - ELDEXT_TERM = 0x0, /* Termination tag */ - ELDEXT_SAOC = 0x1, /* SAOC config */ - ELDEXT_LDSAC = 0x2 /* LD MPEG Surround config */ - /* reserved */ -} ASC_ELD_EXT_TYPE; - -typedef struct { - UCHAR m_frameLengthFlag; - - UCHAR m_sbrPresentFlag; - UCHAR m_useLdQmfTimeAlign; /* Use LD-MPS QMF in SBR to achive time alignment */ - UCHAR m_sbrSamplingRate; - UCHAR m_sbrCrcFlag; - -} CSEldSpecificConfig; -#endif /* TP_ELD_ENABLE */ - @@ -240,9 +220,6 @@ typedef struct { #ifdef TP_GA_ENABLE CSGaSpecificConfig m_gaSpecificConfig; /**< General audio specific configuration. */ #endif /* TP_GA_ENABLE */ -#ifdef TP_ELD_ENABLE - CSEldSpecificConfig m_eldSpecificConfig; /**< ELD specific configuration. */ -#endif /* TP_ELD_ENABLE */ } m_sc; /* Common ASC parameters */ @@ -281,24 +258,12 @@ typedef INT (*cbSsc_t)( const INT muxMode, const INT configBytes ); -typedef INT (*cbSbr_t)( - void * self, - HANDLE_FDK_BITSTREAM hBs, - const INT sampleRateIn, - const INT sampleRateOut, - const INT samplesPerFrame, - const AUDIO_OBJECT_TYPE coreCodec, - const MP4_ELEMENT_ID elementID, - const INT elementIndex - ); typedef struct { cbUpdateConfig_t cbUpdateConfig; /*!< Function pointer for Config change notify callback. */ void *cbUpdateConfigData; /*!< User data pointer for Config change notify callback. */ cbSsc_t cbSsc; /*!< Function pointer for SSC parser callback. */ void *cbSscData; /*!< User data pointer for SSC parser callback. */ - cbSbr_t cbSbr; /*!< Function pointer for SBR header parser callback. */ - void *cbSbrData; /*!< User data pointer for SBR header parser callback. */ } CSTpCallBacks; static const UINT SamplingRateTable[] = diff --git a/libMpegTPDec/include/tpdec_lib.h b/libMpegTPDec/include/tpdec_lib.h index 2ad397d..2958acb 100644 --- a/libMpegTPDec/include/tpdec_lib.h +++ b/libMpegTPDec/include/tpdec_lib.h @@ -352,15 +352,6 @@ int transportDec_RegisterSscCallback ( const cbSsc_t cbSscParse, void* user_data ); -/** - * \brief Register SBR header parser callback. - * \param hTp Handle of transport decoder. - * \param cbUpdateConfig Pointer to a callback function to handle SBR header parsing. - * \param user_data void pointer for user data passed to the callback as first parameter. - * \return 0 on success. - */ -int transportDec_RegisterSbrCallback( HANDLE_TRANSPORTDEC hTpDec, const cbSbr_t cbSbr, void* user_data); - /** * \brief Fill internal input buffer with bitstream data from the external input buffer. * The function only copies such data as long as the decoder-internal input buffer is not full. diff --git a/libMpegTPDec/src/tpdec_asc.cpp b/libMpegTPDec/src/tpdec_asc.cpp index a292bcb..18805c4 100644 --- a/libMpegTPDec/src/tpdec_asc.cpp +++ b/libMpegTPDec/src/tpdec_asc.cpp @@ -1061,118 +1061,6 @@ TRANSPORTDEC_ERROR GaSpecificConfig_Parse( CSGaSpecificConfig *self, - -#ifdef TP_ELD_ENABLE - -static INT ld_sbr_header( const CSAudioSpecificConfig *asc, - HANDLE_FDK_BITSTREAM hBs, - CSTpCallBacks *cb ) -{ - const int channelConfiguration = asc->m_channelConfiguration; - int i = 0; - INT error = 0; - - if (channelConfiguration == 2) { - error = cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, AOT_ER_AAC_ELD, ID_CPE, i++); - } else { - error = cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, AOT_ER_AAC_ELD, ID_SCE, i++); - } - - switch ( channelConfiguration ) { - case 14: - case 12: - case 7: - error |= cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, AOT_ER_AAC_ELD, ID_CPE, i++); - case 6: - case 5: - error |= cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, AOT_ER_AAC_ELD, ID_CPE, i++); - case 3: - error |= cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, AOT_ER_AAC_ELD, ID_CPE, i++); - break; - - case 11: - error |= cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, AOT_ER_AAC_ELD, ID_CPE, i++); - case 4: - error |= cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, AOT_ER_AAC_ELD, ID_CPE, i++); - error |= cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, AOT_ER_AAC_ELD, ID_SCE, i++); - break; - } - - return error; -} - -static -TRANSPORTDEC_ERROR EldSpecificConfig_Parse( - CSAudioSpecificConfig *asc, - HANDLE_FDK_BITSTREAM hBs, - CSTpCallBacks *cb - ) -{ - TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK; - CSEldSpecificConfig *esc = &asc->m_sc.m_eldSpecificConfig; - ASC_ELD_EXT_TYPE eldExtType; - int eldExtLen, len, cnt; - - FDKmemclear(esc, sizeof(CSEldSpecificConfig)); - - esc->m_frameLengthFlag = FDKreadBits(hBs, 1 ); - if (esc->m_frameLengthFlag) { - asc->m_samplesPerFrame = 480; - } else { - asc->m_samplesPerFrame = 512; - } - - asc->m_vcb11Flag = FDKreadBits(hBs, 1 ); - asc->m_rvlcFlag = FDKreadBits(hBs, 1 ); - asc->m_hcrFlag = FDKreadBits(hBs, 1 ); - - esc->m_sbrPresentFlag = FDKreadBits(hBs, 1 ); - - if (esc->m_sbrPresentFlag == 1) { - esc->m_sbrSamplingRate = FDKreadBits(hBs, 1 ); /* 0: single rate, 1: dual rate */ - esc->m_sbrCrcFlag = FDKreadBits(hBs, 1 ); - - asc->m_extensionSamplingFrequency = asc->m_samplingFrequency << esc->m_sbrSamplingRate; - - if (cb->cbSbr != NULL){ - if ( 0 != ld_sbr_header(asc, hBs, cb) ) { - return TRANSPORTDEC_PARSE_ERROR; - } - } else { - return TRANSPORTDEC_UNSUPPORTED_FORMAT; - } - } - esc->m_useLdQmfTimeAlign = 0; - - /* new ELD syntax */ - /* parse ExtTypeConfigData */ - while ((eldExtType = (ASC_ELD_EXT_TYPE)FDKreadBits(hBs, 4 )) != ELDEXT_TERM) { - eldExtLen = len = FDKreadBits(hBs, 4 ); - if ( len == 0xf ) { - len = FDKreadBits(hBs, 8 ); - eldExtLen += len; - - if ( len == 0xff ) { - len = FDKreadBits(hBs, 16 ); - eldExtLen += len; - } - } - - switch (eldExtType) { - default: - for(cnt=0; cntm_sc.m_eldSpecificConfig.m_frameLengthFlag; - self->m_sbrPresentFlag = self->m_sc.m_eldSpecificConfig.m_sbrPresentFlag; - self->m_extensionSamplingFrequency = (self->m_sc.m_eldSpecificConfig.m_sbrSamplingRate+1) * self->m_samplingFrequency; - break; -#endif /* TP_ELD_ENABLE */ default: return TRANSPORTDEC_UNSUPPORTED_FORMAT; diff --git a/libMpegTPDec/src/tpdec_lib.cpp b/libMpegTPDec/src/tpdec_lib.cpp index 5760752..3465fdc 100644 --- a/libMpegTPDec/src/tpdec_lib.cpp +++ b/libMpegTPDec/src/tpdec_lib.cpp @@ -299,16 +299,6 @@ int transportDec_RegisterSscCallback( HANDLE_TRANSPORTDEC hTpDec, const cbSsc_t return 0; } -int transportDec_RegisterSbrCallback( HANDLE_TRANSPORTDEC hTpDec, const cbSbr_t cbSbr, void* user_data) -{ - if (hTpDec == NULL) { - return -1; - } - hTpDec->callbacks.cbSbr = cbSbr; - hTpDec->callbacks.cbSbrData = user_data; - return 0; -} - TRANSPORTDEC_ERROR transportDec_FillData( const HANDLE_TRANSPORTDEC hTp, UCHAR *pBuffer, diff --git a/libMpegTPEnc/include/mpegFileWrite.h b/libMpegTPEnc/include/mpegFileWrite.h deleted file mode 100644 index f886a0b..0000000 --- a/libMpegTPEnc/include/mpegFileWrite.h +++ /dev/null @@ -1,140 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/***************************** MPEG-4 AAC Decoder ************************** - - Author(s): Manuel Jander - Description: Bitstream data provider for MP4 decoders - -******************************************************************************/ - -#include "machine_type.h" -#include "FDK_audio.h" - -/*!< If MPFWRITE_MP4FF_ENABLE is set, include support for MPEG ISO fileformat. - If not set, no .mp4, .m4a and .3gp files can be used for input. */ -/* #define MPFWRITE_MP4FF_ENABLE */ - -typedef struct STRUCT_FILEWRITE *HANDLE_FILEWRITE; - -#ifdef __cplusplus -extern "C" { -#endif - -/** - * \brief Open an MPEG audio file. - * \param mpegFileWrite_Filename String of the filename to be opened. - * \param fileFmt Transport format to use. - * \param conf - * \param confSize - * \return MPEG file write handle. - */ -HANDLE_FILEWRITE mpegFileWrite_Open( char *mpegFileWrite_Filename, - FILE_FORMAT fileFmt, - TRANSPORT_TYPE transportType, - UCHAR *conf, - UINT confSize - ); - -/** - * \brief Write to an MPEG audio file. - * \param inBuffer Buffer to write. - * \param bufferSize Size of buffer to write in bytes. - * \return 0 on sucess, -1 on unsupported file format or write error. - */ -int mpegFileWrite_Write( HANDLE_FILEWRITE hFileWrite, - UCHAR *inBuffer, - int bufferSize - ); - -/** - * \brief Deallocate memory and close file. - * \param hFileWrite MPEG file write handle. - * \return 0 on sucess. - */ -int mpegFileWrite_Close( HANDLE_FILEWRITE *hFileWrite ); - - -#ifdef __cplusplus -} -#endif diff --git a/libMpegTPEnc/include/tp_data.h b/libMpegTPEnc/include/tp_data.h deleted file mode 100644 index c6e89b5..0000000 --- a/libMpegTPEnc/include/tp_data.h +++ /dev/null @@ -1,350 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/***************************** MPEG-4 AAC Decoder ************************** - - Author(s): Manuel Jander - Description: MPEG Transport data tables - -******************************************************************************/ - -#ifndef __TP_DATA_H__ -#define __TP_DATA_H__ - -#include "machine_type.h" -#include "FDK_audio.h" -#include "FDK_bitstream.h" - -/* - * Configuration - */ -#define TP_GA_ENABLE -/* #define TP_CELP_ENABLE */ -/* #define TP_HVXC_ENABLE */ -/* #define TP_SLS_ENABLE */ -#define TP_ELD_ENABLE -/* #define TP_USAC_ENABLE */ -/* #define TP_RSVD50_ENABLE */ - -#if defined(TP_GA_ENABLE) || defined(TP_SLS_ENABLE) -#define TP_PCE_ENABLE /**< Enable full PCE support */ -#endif - -/** - * ProgramConfig struct. - */ -/* ISO/IEC 14496-3 4.4.1.1 Table 4.2 Program config element */ -#define PC_FSB_CHANNELS_MAX 16 /* Front/Side/Back channels */ -#define PC_LFE_CHANNELS_MAX 4 -#define PC_ASSOCDATA_MAX 8 -#define PC_CCEL_MAX 16 /* CC elements */ -#define PC_COMMENTLENGTH 256 - -typedef struct -{ -#ifdef TP_PCE_ENABLE - /* PCE bitstream elements: */ - UCHAR ElementInstanceTag; - UCHAR Profile; - UCHAR SamplingFrequencyIndex; - UCHAR NumFrontChannelElements; - UCHAR NumSideChannelElements; - UCHAR NumBackChannelElements; - UCHAR NumLfeChannelElements; - UCHAR NumAssocDataElements; - UCHAR NumValidCcElements; - - UCHAR MonoMixdownPresent; - UCHAR MonoMixdownElementNumber; - - UCHAR StereoMixdownPresent; - UCHAR StereoMixdownElementNumber; - - UCHAR MatrixMixdownIndexPresent; - UCHAR MatrixMixdownIndex; - UCHAR PseudoSurroundEnable; - - UCHAR FrontElementIsCpe[PC_FSB_CHANNELS_MAX]; - UCHAR FrontElementTagSelect[PC_FSB_CHANNELS_MAX]; - UCHAR FrontElementHeightInfo[PC_FSB_CHANNELS_MAX]; - - UCHAR SideElementIsCpe[PC_FSB_CHANNELS_MAX]; - UCHAR SideElementTagSelect[PC_FSB_CHANNELS_MAX]; - UCHAR SideElementHeightInfo[PC_FSB_CHANNELS_MAX]; - - UCHAR BackElementIsCpe[PC_FSB_CHANNELS_MAX]; - UCHAR BackElementTagSelect[PC_FSB_CHANNELS_MAX]; - UCHAR BackElementHeightInfo[PC_FSB_CHANNELS_MAX]; - - UCHAR LfeElementTagSelect[PC_LFE_CHANNELS_MAX]; - - UCHAR AssocDataElementTagSelect[PC_ASSOCDATA_MAX]; - - UCHAR CcElementIsIndSw[PC_CCEL_MAX]; - UCHAR ValidCcElementTagSelect[PC_CCEL_MAX]; - - UCHAR CommentFieldBytes; - UCHAR Comment[PC_COMMENTLENGTH]; -#endif /* TP_PCE_ENABLE */ - - /* Helper variables for administration: */ - UCHAR isValid; /*!< Flag showing if PCE has been read successfully. */ - UCHAR NumChannels; /*!< Amount of audio channels summing all channel elements including LFEs */ - UCHAR NumEffectiveChannels; /*!< Amount of audio channels summing only SCEs and CPEs */ - UCHAR elCounter; - -} CProgramConfig; - -typedef enum { - ASCEXT_UNKOWN = -1, - ASCEXT_SBR = 0x2b7, - ASCEXT_PS = 0x548, - ASCEXT_MPS = 0x76a, - ASCEXT_SAOC = 0x7cb, - ASCEXT_LDMPS = 0x7cc - -} TP_ASC_EXTENSION_ID; - -#ifdef TP_GA_ENABLE -/** - * GaSpecificConfig struct - */ -typedef struct { - UINT m_frameLengthFlag ; - UINT m_dependsOnCoreCoder ; - UINT m_coreCoderDelay ; - - UINT m_extensionFlag ; - UINT m_extensionFlag3 ; - - UINT m_layer; - UINT m_numOfSubFrame; - UINT m_layerLength; - -} CSGaSpecificConfig; -#endif /* TP_GA_ENABLE */ - - - - -#ifdef TP_ELD_ENABLE - -typedef enum { - ELDEXT_TERM = 0x0, /* Termination tag */ - ELDEXT_SAOC = 0x1, /* SAOC config */ - ELDEXT_LDSAC = 0x2 /* LD MPEG Surround config */ - /* reserved */ -} ASC_ELD_EXT_TYPE; - -typedef struct { - UCHAR m_frameLengthFlag; - - UCHAR m_sbrPresentFlag; - UCHAR m_useLdQmfTimeAlign; /* Use LD-MPS QMF in SBR to achive time alignment */ - UCHAR m_sbrSamplingRate; - UCHAR m_sbrCrcFlag; - -} CSEldSpecificConfig; -#endif /* TP_ELD_ENABLE */ - - - - -/** - * Audio configuration struct, suitable for encoder and decoder configuration. - */ -typedef struct { - - /* XYZ Specific Data */ - union { -#ifdef TP_GA_ENABLE - CSGaSpecificConfig m_gaSpecificConfig; /**< General audio specific configuration. */ -#endif /* TP_GA_ENABLE */ -#ifdef TP_ELD_ENABLE - CSEldSpecificConfig m_eldSpecificConfig; /**< ELD specific configuration. */ -#endif /* TP_ELD_ENABLE */ - } m_sc; - - /* Common ASC parameters */ -#ifdef TP_PCE_ENABLE - CProgramConfig m_progrConfigElement; /**< Program configuration. */ -#endif /* TP_PCE_ENABLE */ - - AUDIO_OBJECT_TYPE m_aot; /**< Audio Object Type. */ - UINT m_samplingFrequency; /**< Samplerate. */ - UINT m_samplesPerFrame; /**< Amount of samples per frame. */ - UINT m_directMapping; /**< Document this please !! */ - - AUDIO_OBJECT_TYPE m_extensionAudioObjectType; /**< Audio object type */ - UINT m_extensionSamplingFrequency; /**< Samplerate */ - - SCHAR m_channelConfiguration; /**< Channel configuration index */ - - SCHAR m_epConfig; /**< Error protection index */ - SCHAR m_vcb11Flag; /**< aacSectionDataResilienceFlag */ - SCHAR m_rvlcFlag; /**< aacScalefactorDataResilienceFlag */ - SCHAR m_hcrFlag; /**< aacSpectralDataResilienceFlag */ - - SCHAR m_sbrPresentFlag; /**< Flag indicating the presence of SBR data in the bitstream */ - SCHAR m_psPresentFlag; /**< Flag indicating the presence of parametric stereo data in the bitstream */ - UCHAR m_samplingFrequencyIndex; /**< Samplerate index */ - UCHAR m_extensionSamplingFrequencyIndex; /**< Samplerate index */ - SCHAR m_extensionChannelConfiguration; /**< Channel configuration index */ - -} CSAudioSpecificConfig; - -typedef INT (*cbUpdateConfig_t)(void*, const CSAudioSpecificConfig*); -typedef INT (*cbSsc_t)( - void*, HANDLE_FDK_BITSTREAM, - const AUDIO_OBJECT_TYPE coreCodec, - const INT samplingFrequency, - const INT muxMode, - const INT configBytes - ); -typedef INT (*cbSbr_t)( - void * self, - HANDLE_FDK_BITSTREAM hBs, - const INT sampleRateIn, - const INT sampleRateOut, - const INT samplesPerFrame, - const AUDIO_OBJECT_TYPE coreCodec, - const MP4_ELEMENT_ID elementID, - const INT elementIndex - ); - -typedef struct { - cbUpdateConfig_t cbUpdateConfig; /*!< Function pointer for Config change notify callback. */ - void *cbUpdateConfigData; /*!< User data pointer for Config change notify callback. */ - cbSsc_t cbSsc; /*!< Function pointer for SSC parser callback. */ - void *cbSscData; /*!< User data pointer for SSC parser callback. */ - cbSbr_t cbSbr; /*!< Function pointer for SBR header parser callback. */ - void *cbSbrData; /*!< User data pointer for SBR header parser callback. */ -} CSTpCallBacks; - -static const UINT SamplingRateTable[] = -{ 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000, 7350, 0, 0, - 0 -}; - -static inline -int getSamplingRateIndex( UINT samplingRate ) -{ - UINT sf_index, tableSize=sizeof(SamplingRateTable)/sizeof(UINT); - - for (sf_index=0; sf_indextableSize-1) { - return tableSize-1; - } - - return sf_index; -} - -/* - * Get Channel count from channel configuration - */ -static inline int getNumberOfTotalChannels(int channelConfig) -{ - switch (channelConfig) { - case 1: case 2: case 3: - case 4: case 5: case 6: - return channelConfig; - case 7: case 12: case 14: - return 8; - case 11: - return 7; - default: - return 0; - } -} - -static inline -int getNumberOfEffectiveChannels(const int channelConfig) -{ /* index: 0,1,2,3,4,5,6,7,8,9,10,11,12,13,14,15 */ - const int n[] = {0,1,2,3,4,5,5,7,0,0, 0, 6, 7, 0, 7, 0}; - return n[channelConfig]; -} - -#endif /* __TP_DATA_H__ */ diff --git a/libMpegTPEnc/include/tpenc_lib.h b/libMpegTPEnc/include/tpenc_lib.h deleted file mode 100644 index 2833e82..0000000 --- a/libMpegTPEnc/include/tpenc_lib.h +++ /dev/null @@ -1,296 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/************************** MPEG-4 Transport Encoder ************************ - - Author(s): Manuel Jander - Description: MPEG Transport encode - -******************************************************************************/ - -#ifndef __TPENC_LIB_H__ -#define __TPENC_LIB_H__ - -#include "tp_data.h" -#include "FDK_bitstream.h" - -#define TRANSPORTENC_INBUF_SIZE 8192 - -typedef enum { - TRANSPORTENC_OK = 0, /*!< All fine. */ - TRANSPORTENC_NO_MEM, /*!< Out of memory. */ - TRANSPORTENC_UNKOWN_ERROR = 1, /*!< Unknown error (embarrasing). */ - TRANSPORTENC_INVALID_PARAMETER, /*!< An invalid parameter was passed to a function . */ - TRANSPORTENC_PARSE_ERROR, /*!< Bitstream data contained inconsistencies (wrong syntax). */ - TRANSPORTENC_UNSUPPORTED_FORMAT, /*!< Unsupported transport format. */ - TRANSPORTENC_NOT_ENOUGH_BITS, /*!< Out of bits. Provide more bits and try again. */ - - TRANSPORTENC_INVALID_CONFIG, /*!< Error in configuration. */ - TRANSPORTENC_LATM_INVALID_NR_OF_SUBFRAMES, /*!< LATM: number of subframes out of range. */ - TRANSPORTENC_LOAS_NOT_AVAILABLE, /*!< LOAS format not supported. */ - TRANSPORTENC_INVALID_LATM_ALIGNMENT, /*!< AudioMuxElement length not aligned to 1 byte. */ - - TRANSPORTENC_INVALID_TRANSMISSION_FRAME_LENGTH, /*!< Invalid transmission frame length (< 0). */ - TRANSPORTENC_INVALID_CELP_FRAME_LENGTH, /*!< Invalid CELP frame length found (>= 62). */ - TRANSPORTENC_INVALID_FRAME_BITS, /*!< Frame bits is not 40 and not 80. */ - TRANSPORTENC_INVALID_AOT, /*!< Unknown AOT found. */ - TRANSPORTENC_INVALID_AU_LENGTH /*!< Invalid Access Unit length (not byte-aligned). */ - -} TRANSPORTENC_ERROR; - -typedef struct TRANSPORTENC *HANDLE_TRANSPORTENC; - -/** - * \brief Determine a reasonable channel configuration on the basis of channel_mode. - * \param noChannels Number of audio channels. - * \return CHANNEL_MODE value that matches the given amount of audio channels. - */ -CHANNEL_MODE transportEnc_GetChannelMode( int noChannels ); - -/** - * \brief Register SBR heaqder writer callback. - * \param hTp Handle of transport decoder. - * \param cbUpdateConfig Pointer to a callback function to handle SBR header writing. - * \param user_data void pointer for user data passed to the callback as first parameter. - * \return 0 on success. - */ -int transportEnc_RegisterSbrCallback ( - HANDLE_TRANSPORTENC hTpEnc, - const cbSbr_t cbSbr, - void* user_data - ); - -/** - * \brief Register SSC writer callback. - * \param hTp Handle of transport decoder. - * \param cbUpdateConfig Pointer to a callback function to handle SSC writing. - * \param user_data void pointer for user data passed to the callback as first parameter. - * \return 0 on success. - */ -int transportEnc_RegisterSscCallback ( - HANDLE_TRANSPORTENC hTpEnc, - const cbSsc_t cbSsc, - void* user_data - ); - -/** - * \brief Write ASC from given parameters. - * \param asc A HANDLE_FDK_BITSTREAM where the ASC is written to. - * \param config Structure containing the codec configuration settings. - * \param cb callback information structure. - * \return 0 on success. - */ -int transportEnc_writeASC ( - HANDLE_FDK_BITSTREAM asc, - CODER_CONFIG *config, - CSTpCallBacks *cb - ); - - -/* Defintion of flags that can be passed to transportEnc_Open() */ -#define TP_FLAG_MPEG4 1 /** MPEG4 (instead of MPEG2) */ -#define TP_FLAG_LATM_AMV 2 /** LATM AudioMuxVersion */ -#define TP_FLAG_LATM_AMVA 4 /** LATM AudioMuxVersionA */ - -/** - * \brief Allocate transport encoder. - * \param phTpEnc Pointer to transport encoder handle. - * \return Error code. - */ -TRANSPORTENC_ERROR transportEnc_Open( HANDLE_TRANSPORTENC *phTpEnc ); - -/** - * \brief Init transport encoder. - * \param bsBuffer Pointer to transport encoder. - * \param bsBuffer Pointer to bitstream buffer. - * \param bsBufferSize Size in bytes of bsBuffer. - * \param transportFmt Format of the transport to be written. - * \param config Pointer to a valid CODER_CONFIG struct. - * \param flags Transport encoder flags. - * \return Error code. - */ -TRANSPORTENC_ERROR transportEnc_Init( - HANDLE_TRANSPORTENC hTpEnc, - UCHAR *bsBuffer, - INT bsBufferSize, - TRANSPORT_TYPE transportFmt, - CODER_CONFIG *config, - UINT flags - ); - -/** - * \brief Get transport encoder bitstream. - * \param hTp Pointer to a transport encoder handle. - * \return The handle to the requested FDK bitstream. - */ -HANDLE_FDK_BITSTREAM transportEnc_GetBitstream( HANDLE_TRANSPORTENC hTp ); - -/** - * \brief Get amount of bits required by the transport headers. - * \param hTp Handle of transport encoder. - * \param auBits Amount of payload bits required for the current subframe. - * \return Error code. - */ -INT transportEnc_GetStaticBits( HANDLE_TRANSPORTENC hTp, int auBits ); - -/** - * \brief Close transport encoder. This function assures that all allocated memory is freed. - * \param phTp Pointer to a previously allocated transport encoder handle. - */ -void transportEnc_Close( HANDLE_TRANSPORTENC *phTp ); - -/** - * \brief Write one access unit. - * \param hTp Handle of transport encoder. - * \param total_bits Amount of total access unit bits. - * \param bufferFullness Value of current buffer fullness in bits. - * \param noConsideredChannels Number of bitrate wise considered channels (all minus LFE channels). - * \return Error code. - */ -TRANSPORTENC_ERROR transportEnc_WriteAccessUnit( HANDLE_TRANSPORTENC hTp, - INT total_bits, - int bufferFullness, - int noConsideredChannels ); - -/** - * \brief Inform the transportEnc layer that writing of access unit has finished. This function - * is required to be called when the encoder has finished writing one Access - * one Access Unit for bitstream housekeeping. - * \param hTp Transport handle. - * \param pBits Pointer to an int, where the current amount of frame bits is passed - * and where the current amount of subframe bits is returned. - * - * OR: This integer is modified by the amount of extra bit alignment that may occurr. - * - * \return Error code. - */ -TRANSPORTENC_ERROR transportEnc_EndAccessUnit( HANDLE_TRANSPORTENC hTp, int *pBits); - -/* - * \brief Get a payload frame. - * \param hTpEnc Transport encoder handle. - * \param nBytes Pointer to an int to hold the frame size in bytes. Returns zero - * if currently there is no complete frame for output (number of sub frames > 1). - * \return Error code. - */ -TRANSPORTENC_ERROR transportEnc_GetFrame(HANDLE_TRANSPORTENC hTpEnc, int *nbytes); - -/* ADTS CRC support */ - -/** - * \brief Set current bitstream position as start of a new data region. - * \param hTpEnc Transport encoder handle. - * \param mBits Size in bits of the data region. Set to 0 if it should not be of a fixed size. - * \return Data region ID, which should be used when calling transportEnc_CrcEndReg(). - */ -int transportEnc_CrcStartReg(HANDLE_TRANSPORTENC hTpEnc, int mBits); - -/** - * \brief Set end of data region. - * \param hTpEnc Transport encoder handle. - * \param reg Data region ID, opbtained from transportEnc_CrcStartReg(). - * \return void - */ -void transportEnc_CrcEndReg(HANDLE_TRANSPORTENC hTpEnc, int reg); - -/** - * \brief Get AudioSpecificConfig or StreamMuxConfig from transport encoder handle and write it to dataBuffer. - * \param hTpEnc Transport encoder handle. - * \param cc Pointer to the current and valid configuration contained in a CODER_CONFIG struct. - * \param dataBuffer Bitbuffer holding binary configuration. - * \param confType Pointer to an UINT where the configuration type is returned (0:ASC, 1:SMC). - * \return Error code. - */ -TRANSPORTENC_ERROR transportEnc_GetConf( HANDLE_TRANSPORTENC hTpEnc, - CODER_CONFIG *cc, - FDK_BITSTREAM *dataBuffer, - UINT *confType ); - -/** - * \brief Get information (version among other things) of the transport encoder library. - * \param info Pointer to an allocated LIB_INFO struct. - * \return Error code. - */ -TRANSPORTENC_ERROR transportEnc_GetLibInfo( LIB_INFO *info ); - -#endif /* #ifndef __TPENC_LIB_H__ */ diff --git a/libMpegTPEnc/src/tpenc_adif.cpp b/libMpegTPEnc/src/tpenc_adif.cpp deleted file mode 100644 index b48a32e..0000000 --- a/libMpegTPEnc/src/tpenc_adif.cpp +++ /dev/null @@ -1,182 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - contents/description: ADIF Transport Headers writing - -******************************************************************************/ - -#include "tpenc_adif.h" - -#include "tpenc_lib.h" -#include "tpenc_asc.h" - - - -int adifWrite_EncodeHeader(ADIF_INFO *adif, - HANDLE_FDK_BITSTREAM hBs, - INT adif_buffer_fullness) -{ - /* ADIF/PCE/ADTS definitions */ - const char adifId[5]="ADIF"; - const int copyRightIdPresent=0; - const int originalCopy=0; - const int home=0; - - int i; - - INT sampleRate = adif->samplingRate; - INT totalBitRate = adif->bitRate; - - if (adif->headerWritten) - return 0; - - /* Align inside PCE with respect to the first bit of the header */ - UINT alignAnchor = FDKgetValidBits(hBs); - - /* Signal variable bitrate if buffer fullnes exceeds 20 bit */ - adif->bVariableRate = ( adif_buffer_fullness >= (INT)(0x1<<20) ) ? 1 : 0; - - FDKwriteBits(hBs, adifId[0],8); - FDKwriteBits(hBs, adifId[1],8); - FDKwriteBits(hBs, adifId[2],8); - FDKwriteBits(hBs, adifId[3],8); - - - FDKwriteBits(hBs, copyRightIdPresent ? 1:0,1); - - if(copyRightIdPresent) { - for(i=0;i<72;i++) { - FDKwriteBits(hBs,0,1); - } - } - FDKwriteBits(hBs, originalCopy ? 1:0,1); - FDKwriteBits(hBs, home ? 1:0,1); - FDKwriteBits(hBs, adif->bVariableRate?1:0, 1); - FDKwriteBits(hBs, totalBitRate,23); - - /* we write only one PCE at the moment */ - FDKwriteBits(hBs, 0, 4); - - if(!adif->bVariableRate) { - FDKwriteBits(hBs, adif_buffer_fullness, 20); - } - - /* Write PCE */ - transportEnc_writePCE(hBs, adif->cm, sampleRate, adif->instanceTag, adif->profile, 0, 0, alignAnchor); - - return 0; -} - -int adifWrite_GetHeaderBits(ADIF_INFO *adif) -{ - /* ADIF definitions */ - const int copyRightIdPresent=0; - - if (adif->headerWritten) - return 0; - - int bits = 0; - - bits += 8*4; /* ADIF ID */ - - bits += 1; /* Copyright present */ - - if (copyRightIdPresent) - bits += 72; /* Copyright ID */ - - bits += 26; - - bits += 4; /* Number of PCE's */ - - if(!adif->bVariableRate) { - bits += 20; - } - - /* write PCE */ - bits = transportEnc_GetPCEBits(adif->cm, 0, bits); - - return bits; -} - diff --git a/libMpegTPEnc/src/tpenc_adif.h b/libMpegTPEnc/src/tpenc_adif.h deleted file mode 100644 index d590354..0000000 --- a/libMpegTPEnc/src/tpenc_adif.h +++ /dev/null @@ -1,135 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: Alex Goeschel - contents/description: Transport Headers support - -******************************************************************************/ - -#ifndef TPENC_ADIF_H -#define TPENC_ADIF_H - -#include "machine_type.h" -#include "FDK_bitstream.h" - -#include "tp_data.h" - -typedef struct { - CHANNEL_MODE cm; - INT samplingRate; - INT bitRate; - int profile; - int bVariableRate; - int instanceTag; - int headerWritten; -} ADIF_INFO; - -/** - * \brief encodes ADIF Header - * - * \param adif pointer to ADIF_INFO structure - * \param hBitStream handle of bitstream, where the ADIF header is written into - * \param adif_buffer_fullness buffer fullness value for the ADIF header - * - * \return 0 on success - */ -int adifWrite_EncodeHeader( - ADIF_INFO *adif, - HANDLE_FDK_BITSTREAM hBitStream, - INT adif_buffer_fullness - ); - -/** - * \brief Get bit demand of a ADIF header - * - * \param adif pointer to ADIF_INFO structure - * - * \return amount of bits required to write the ADIF header according to the data - * contained in the adif parameter - */ -int adifWrite_GetHeaderBits( ADIF_INFO *adif ); - -#endif /* TPENC_ADIF_H */ - diff --git a/libMpegTPEnc/src/tpenc_adts.cpp b/libMpegTPEnc/src/tpenc_adts.cpp deleted file mode 100644 index f4f3178..0000000 --- a/libMpegTPEnc/src/tpenc_adts.cpp +++ /dev/null @@ -1,315 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: Alex Groeschel - contents/description: ADTS Transport Headers support - -******************************************************************************/ - -#include "tpenc_adts.h" - - -#include "tpenc_lib.h" -#include "tpenc_asc.h" - - -int adtsWrite_CrcStartReg( - HANDLE_ADTS pAdts, /*!< pointer to adts stucture */ - HANDLE_FDK_BITSTREAM hBs, /*!< handle to current bit buffer structure */ - int mBits /*!< number of bits in crc region */ - ) -{ - if (pAdts->protection_absent) { - return 0; - } - return ( FDKcrcStartReg(&pAdts->crcInfo, hBs, mBits) ); -} - -void adtsWrite_CrcEndReg( - HANDLE_ADTS pAdts, /*!< pointer to adts crc info stucture */ - HANDLE_FDK_BITSTREAM hBs, /*!< handle to current bit buffer structure */ - int reg /*!< crc region */ - ) -{ - if (pAdts->protection_absent == 0) - { - FDKcrcEndReg(&pAdts->crcInfo, hBs, reg); - } -} - -int adtsWrite_GetHeaderBits( HANDLE_ADTS hAdts ) -{ - int bits = 0; - - if (hAdts->currentBlock == 0) { - /* Static and variable header bits */ - bits = 56; - if (!hAdts->protection_absent) { - /* Add header/ single raw data block CRC bits */ - bits += 16; - if (hAdts->num_raw_blocks>0) { - /* Add bits of raw data block position markers */ - bits += (hAdts->num_raw_blocks)*16; - } - } - } - if (!hAdts->protection_absent && hAdts->num_raw_blocks>0) { - /* Add raw data block CRC bits. Not really part of the header, put they cause bit overhead to be accounted. */ - bits += 16; - } - - hAdts->headerBits = bits; - - return bits; -} - -INT adtsWrite_Init(HANDLE_ADTS hAdts, CODER_CONFIG *config) -{ - /* Sanity checks */ - if ( config->nSubFrames < 1 - || config->nSubFrames > 4 - || (int)config->aot > 4 - || (int)config->aot < 1 ) { - return -1; - } - - /* fixed header */ - if (config->flags & CC_MPEG_ID) { - hAdts->mpeg_id = 0; /* MPEG 4 */ - } else { - hAdts->mpeg_id = 1; /* MPEG 2 */ - } - hAdts->layer=0; - hAdts->protection_absent = ! (config->flags & CC_PROTECTION); - hAdts->profile = ((int)config->aot) - 1; - hAdts->sample_freq_index = getSamplingRateIndex(config->samplingRate); - hAdts->sample_freq = config->samplingRate; - hAdts->private_bit=0; - hAdts->channel_mode = config->channelMode; - hAdts->original=0; - hAdts->home=0; - /* variable header */ - hAdts->copyright_id=0; - hAdts->copyright_start=0; - - hAdts->num_raw_blocks=config->nSubFrames-1; /* 0 means 1 raw data block */ - - FDKcrcInit(&hAdts->crcInfo, 0x8005, 0xFFFF, 16); - - hAdts->currentBlock = 0; - - - return 0; -} - -int adtsWrite_EncodeHeader(HANDLE_ADTS hAdts, - HANDLE_FDK_BITSTREAM hBitStream, - int buffer_fullness, - int frame_length) -{ - INT crcIndex = 0; - - - hAdts->headerBits = adtsWrite_GetHeaderBits(hAdts); - - FDK_ASSERT(((frame_length+hAdts->headerBits)/8)<0x2000); /*13 bit*/ - FDK_ASSERT(buffer_fullness<0x800); /* 11 bit */ - - if (!hAdts->protection_absent) { - FDKcrcReset(&hAdts->crcInfo); - } - - if (hAdts->currentBlock == 0) { - FDKresetBitbuffer(hBitStream, BS_WRITER); - } - - hAdts->subFrameStartBit = FDKgetValidBits(hBitStream); - - /* Skip new header if this is raw data block 1..n */ - if (hAdts->currentBlock == 0) - { - FDKresetBitbuffer(hBitStream, BS_WRITER); - - if (hAdts->num_raw_blocks == 0) { - crcIndex = adtsWrite_CrcStartReg(hAdts, hBitStream, 0); - } - - /* fixed header */ - FDKwriteBits(hBitStream, 0xFFF, 12); - FDKwriteBits(hBitStream, hAdts->mpeg_id, 1); - FDKwriteBits(hBitStream, hAdts->layer, 2); - FDKwriteBits(hBitStream, hAdts->protection_absent, 1); - FDKwriteBits(hBitStream, hAdts->profile, 2); - FDKwriteBits(hBitStream, hAdts->sample_freq_index, 4); - FDKwriteBits(hBitStream, hAdts->private_bit, 1); - FDKwriteBits(hBitStream, getChannelConfig(hAdts->channel_mode), 3); - FDKwriteBits(hBitStream, hAdts->original, 1); - FDKwriteBits(hBitStream, hAdts->home, 1); - /* variable header */ - FDKwriteBits(hBitStream, hAdts->copyright_id, 1); - FDKwriteBits(hBitStream, hAdts->copyright_start, 1); - FDKwriteBits(hBitStream, (frame_length + hAdts->headerBits)>>3, 13); - FDKwriteBits(hBitStream, buffer_fullness, 11); - FDKwriteBits(hBitStream, hAdts->num_raw_blocks, 2); - - if (!hAdts->protection_absent) { - int i; - - /* End header CRC portion for single raw data block and write dummy zero values for unknown fields. */ - if (hAdts->num_raw_blocks == 0) { - adtsWrite_CrcEndReg(hAdts, hBitStream, crcIndex); - } else { - for (i=0; inum_raw_blocks; i++) { - FDKwriteBits(hBitStream, 0, 16); - } - } - FDKwriteBits(hBitStream, 0, 16); - } - } /* End of ADTS header */ - - return 0; -} - -void adtsWrite_EndRawDataBlock(HANDLE_ADTS hAdts, - HANDLE_FDK_BITSTREAM hBs, - int *pBits) -{ - if (!hAdts->protection_absent) { - FDK_BITSTREAM bsWriter; - - FDKinitBitStream(&bsWriter, hBs->hBitBuf.Buffer, hBs->hBitBuf.bufSize, 0, BS_WRITER); - FDKpushFor(&bsWriter, 56); - - if (hAdts->num_raw_blocks == 0) { - FDKwriteBits(&bsWriter, FDKcrcGetCRC(&hAdts->crcInfo), 16); - } else { - int distance; - - /* Write CRC of current raw data block */ - FDKwriteBits(hBs, FDKcrcGetCRC(&hAdts->crcInfo), 16); - - /* Write distance to current data block */ - if (hAdts->currentBlock < hAdts->num_raw_blocks) { - FDKpushFor(&bsWriter, hAdts->currentBlock*16); - distance = FDKgetValidBits(hBs) - (56 + (hAdts->num_raw_blocks)*16 + 16); - FDKwriteBits(&bsWriter, distance>>3, 16); - } - } - FDKsyncCache(&bsWriter); - } - - /* Write total frame lenth for multiple raw data blocks and header CRC */ - if (hAdts->num_raw_blocks > 0 && hAdts->currentBlock == hAdts->num_raw_blocks) { - FDK_BITSTREAM bsWriter; - int crcIndex = 0; - - FDKinitBitStream(&bsWriter, hBs->hBitBuf.Buffer, hBs->hBitBuf.bufSize, 0, BS_WRITER); - - if (!hAdts->protection_absent) { - FDKcrcReset(&hAdts->crcInfo); - crcIndex = FDKcrcStartReg(&hAdts->crcInfo, &bsWriter, 0); - } - /* Write total frame length */ - FDKpushFor(&bsWriter, 56-28+2); - FDKwriteBits(&bsWriter, FDKgetValidBits(hBs)>>3, 13); - - /* Write header CRC */ - if (!hAdts->protection_absent) { - FDKpushFor(&bsWriter, 11+2 + (hAdts->num_raw_blocks)*16); - FDKcrcEndReg(&hAdts->crcInfo, &bsWriter, crcIndex); - FDKwriteBits(&bsWriter, FDKcrcGetCRC(&hAdts->crcInfo), 16); - } - FDKsyncCache(&bsWriter); - } - - /* Correct *pBits to reflect the amount of bits of the current subframe */ - *pBits -= hAdts->subFrameStartBit; - if (!hAdts->protection_absent && hAdts->num_raw_blocks > 0) { - /* Fixup CRC bits, since they come after each raw data block */ - *pBits += 16; - } - hAdts->currentBlock++; -} - diff --git a/libMpegTPEnc/src/tpenc_adts.h b/libMpegTPEnc/src/tpenc_adts.h deleted file mode 100644 index c12c7c7..0000000 --- a/libMpegTPEnc/src/tpenc_adts.h +++ /dev/null @@ -1,218 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** MPEG Audio Encoder ************************** - - Initial author: Alex Groeschel - contents/description: ADTS Transport writer - -******************************************************************************/ - -#ifndef TPENC_ADTS_H -#define TPENC_ADTS_H - - - -#include "tp_data.h" - -#include "FDK_crc.h" - -typedef struct { - INT sample_freq; - CHANNEL_MODE channel_mode; - UCHAR decoderCanDoMpeg4; - UCHAR mpeg_id; - UCHAR layer; - UCHAR protection_absent; - UCHAR profile; - UCHAR sample_freq_index; - UCHAR private_bit; - UCHAR original; - UCHAR home; - UCHAR copyright_id; - UCHAR copyright_start; - USHORT frame_length; - UCHAR num_raw_blocks; - UCHAR BufferFullnesStartFlag; - int headerBits; /*!< Header bit demand for the current raw data block */ - int currentBlock; /*!< Index of current raw data block */ - int subFrameStartBit; /*!< Bit position where the current raw data block begins */ - FDK_CRCINFO crcInfo; -} STRUCT_ADTS; - -typedef STRUCT_ADTS *HANDLE_ADTS; - -/** - * \brief Initialize ADTS data structure - * - * \param hAdts ADTS data handle - * \param config a valid CODER_CONFIG struct from where the required - * information for the ADTS header is extrated from - * - * \return 0 in case of success. - */ -INT adtsWrite_Init( - HANDLE_ADTS hAdts, - CODER_CONFIG *config - ); - -/** - * \brief Get the total bit overhead caused by ADTS - * - * \hAdts handle to ADTS data - * - * \return Amount of additional bits required for the current raw data block - */ -int adtsWrite_GetHeaderBits( HANDLE_ADTS hAdts ); - -/** - * \brief Write an ADTS header into the given bitstream. May not write a header - * in case of multiple raw data blocks. - * - * \param hAdts ADTS data handle - * \param hBitStream bitstream handle into which the ADTS may be written into - * \param buffer_fullness the buffer fullness value for the ADTS header - * \param the current raw data block length - * - * \return 0 in case of success. - */ -INT adtsWrite_EncodeHeader( - HANDLE_ADTS hAdts, - HANDLE_FDK_BITSTREAM hBitStream, - int bufferFullness, - int frame_length - ); -/** - * \brief Finish a ADTS raw data block - * - * \param hAdts ADTS data handle - * \param hBs bitstream handle into which the ADTS may be written into - * \param pBits a pointer to a integer holding the current bitstream buffer bit count, - * which is corrected to the current raw data block boundary. - * - */ -void adtsWrite_EndRawDataBlock( - HANDLE_ADTS hAdts, - HANDLE_FDK_BITSTREAM hBs, - int *bits - ); - - -/** - * \brief Start CRC region with a maximum number of bits - * If mBits is positive zero padding will be used for CRC calculation, if there - * are less than mBits bits available. - * If mBits is negative no zero padding is done. - * If mBits is zero the memory for the buffer is allocated dynamically, the - * number of bits is not limited. - * - * \param pAdts ADTS data handle - * \param hBs bitstream handle of which the CRC region ends - * \param mBits limit of number of bits to be considered for the requested CRC region - * - * \return ID for the created region, -1 in case of an error - */ -int adtsWrite_CrcStartReg( - HANDLE_ADTS pAdts, - HANDLE_FDK_BITSTREAM hBs, - int mBits - ); - -/** - * \brief Ends CRC region identified by reg - * - * \param pAdts ADTS data handle - * \param hBs bitstream handle of which the CRC region ends - * \param reg a CRC region ID returned previously by adtsWrite_CrcStartReg() - */ -void adtsWrite_CrcEndReg( - HANDLE_ADTS pAdts, - HANDLE_FDK_BITSTREAM hBs, - int reg - ); - - - - -#endif /* TPENC_ADTS_H */ - diff --git a/libMpegTPEnc/src/tpenc_asc.cpp b/libMpegTPEnc/src/tpenc_asc.cpp deleted file mode 100644 index bc4302e..0000000 --- a/libMpegTPEnc/src/tpenc_asc.cpp +++ /dev/null @@ -1,576 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/***************************** MPEG-4 AAC Encoder ************************** - - Author(s): - Description: - -******************************************************************************/ - -#include "tp_data.h" - -#include "tpenc_lib.h" -#include "tpenc_asc.h" -#include "FDK_bitstream.h" -#include "genericStds.h" - -#define PCE_MAX_ELEMENTS 8 - -/** - * Describe a PCE based on placed channel elements and element type sequence. - */ -typedef struct { - - UCHAR num_front_channel_elements; /*!< Number of front channel elements. */ - UCHAR num_side_channel_elements; /*!< Number of side channel elements. */ - UCHAR num_back_channel_elements; /*!< Number of back channel elements. */ - UCHAR num_lfe_channel_elements; /*!< Number of lfe channel elements. */ - MP4_ELEMENT_ID el_list[PCE_MAX_ELEMENTS];/*!< List contains sequence describing the elements - in present channel mode. (MPEG order) */ -} PCE_CONFIGURATION; - - -/** - * Map an incoming channel mode to a existing PCE configuration entry. - */ -typedef struct { - - CHANNEL_MODE channel_mode; /*!< Present channel mode. */ - PCE_CONFIGURATION pce_configuration; /*!< Program config element description. */ - -} CHANNEL_CONFIGURATION; - - -/** - * \brief Table contains all supported channel modes and according PCE configuration description. - * - * The number of channel element parameter describes the kind of consecutively elements. - * E.g. MODE_1_2_2_2_1 means: - * - First 3 elements (SCE,CPE,CPE) are front channel elements. - * - Next element (CPE) is a back channel element. - * - Last element (LFE) is a lfe channel element. - */ -static const CHANNEL_CONFIGURATION pceConfigTab[] = -{ - { MODE_1, { 1, 0, 0, 0, { ID_SCE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } }, - { MODE_2, { 1, 0, 0, 0, { ID_CPE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } }, - { MODE_1_2, { 2, 0, 0, 0, { ID_SCE, ID_CPE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } }, - { MODE_1_2_1, { 2, 0, 1, 0, { ID_SCE, ID_CPE, ID_SCE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } }, - { MODE_1_2_2, { 2, 0, 1, 0, { ID_SCE, ID_CPE, ID_CPE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } }, - { MODE_1_2_2_1, { 2, 0, 1, 1, { ID_SCE, ID_CPE, ID_CPE, ID_LFE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } }, - { MODE_1_2_2_2_1, { 3, 0, 1, 1, { ID_SCE, ID_CPE, ID_CPE, ID_CPE, ID_LFE, ID_NONE, ID_NONE, ID_NONE } } }, - - - { MODE_1_1, { 2, 0, 0, 0, { ID_SCE, ID_SCE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } }, - { MODE_1_1_1_1, { 2, 2, 0, 0, { ID_SCE, ID_SCE, ID_SCE, ID_SCE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } }, - { MODE_1_1_1_1_1_1, { 2, 2, 2, 0, { ID_SCE, ID_SCE, ID_SCE, ID_SCE, ID_SCE, ID_SCE, ID_NONE, ID_NONE } } }, - { MODE_1_1_1_1_1_1_1_1, { 3, 2, 3, 0, { ID_SCE, ID_SCE, ID_SCE, ID_SCE, ID_SCE, ID_SCE, ID_SCE, ID_SCE } } }, - - { MODE_2_2, { 1, 0, 1, 0, { ID_CPE, ID_CPE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } }, - { MODE_2_2_2, { 1, 1, 1, 0, { ID_CPE, ID_CPE, ID_CPE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } }, - { MODE_2_2_2_2, { 4, 0, 0, 0, { ID_CPE, ID_CPE, ID_CPE, ID_CPE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } }, - - { MODE_2_1, { 1, 0, 1, 0, { ID_CPE, ID_SCE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } }, - - { MODE_7_1_REAR_SURROUND, { 2, 0, 2, 1, { ID_SCE, ID_CPE, ID_CPE, ID_CPE, ID_LFE, ID_NONE, ID_NONE, ID_NONE } } }, - { MODE_7_1_FRONT_CENTER, { 3, 0, 1, 1, { ID_SCE, ID_CPE, ID_CPE, ID_CPE, ID_LFE, ID_NONE, ID_NONE, ID_NONE } } }, - -}; - - -/** - * \brief Get program config element description for existing channel mode. - * - * \param channel_mode Current channel mode. - * - * \return - * - Pointer to PCE_CONFIGURATION entry, on success. - * - NULL, on failure. - */ -static const PCE_CONFIGURATION* getPceEntry( - const CHANNEL_MODE channel_mode - ) -{ - UINT i; - const PCE_CONFIGURATION *pce_config = NULL; - - for (i=0; i < (sizeof(pceConfigTab)/sizeof(CHANNEL_CONFIGURATION)); i++) { - if (pceConfigTab[i].channel_mode == channel_mode) { - pce_config = &pceConfigTab[i].pce_configuration; - } - } - - return pce_config; -} - -int getChannelConfig( CHANNEL_MODE channel_mode ) -{ - INT chan_config = 0; - - switch(channel_mode) { - case MODE_1: chan_config = 1; break; - case MODE_2: chan_config = 2; break; - case MODE_1_2: chan_config = 3; break; - case MODE_1_2_1: chan_config = 4; break; - case MODE_1_2_2: chan_config = 5; break; - case MODE_1_2_2_1: chan_config = 6; break; - case MODE_1_2_2_2_1: chan_config = 7; break; - - default: chan_config = 0; - } - - return chan_config; -} - -CHANNEL_MODE transportEnc_GetChannelMode( int noChannels ) -{ - CHANNEL_MODE chMode; - - if (noChannels <= 8 && noChannels > 0) - chMode = (CHANNEL_MODE)((noChannels == 8) ? 7 : noChannels); /* see : iso/mpeg4 v1 audio subpart1*/ - else - chMode = MODE_UNKNOWN; - - return chMode; -} - -#ifdef TP_PCE_ENABLE -int transportEnc_writePCE(HANDLE_FDK_BITSTREAM hBs, - CHANNEL_MODE channelMode, - INT sampleRate, - int instanceTagPCE, - int profile, - int matrixMixdownA, - int pseudoSurroundEnable, - UINT alignAnchor) -{ - int sampleRateIndex, i; - const PCE_CONFIGURATION* config = NULL; - const MP4_ELEMENT_ID* pEl_list = NULL; - UCHAR cpeCnt=0, sceCnt=0, lfeCnt=0; - - sampleRateIndex = getSamplingRateIndex(sampleRate); - if (sampleRateIndex == 15) { - return -1; - } - - if ((config=getPceEntry(channelMode))==NULL) { - return -1; - } - - /* Pointer to first element in element list. */ - pEl_list = &config->el_list[0]; - - FDKwriteBits(hBs, instanceTagPCE, 4); /* Element instance tag */ - FDKwriteBits(hBs, profile, 2); /* Object type */ - FDKwriteBits(hBs, sampleRateIndex, 4); /* Sample rate index*/ - - FDKwriteBits(hBs, config->num_front_channel_elements, 4); /* Front channel Elements */ - FDKwriteBits(hBs, config->num_side_channel_elements , 4); /* No Side Channel Elements */ - FDKwriteBits(hBs, config->num_back_channel_elements , 4); /* No Back channel Elements */ - FDKwriteBits(hBs, config->num_lfe_channel_elements , 2); /* No Lfe channel elements */ - - FDKwriteBits(hBs, 0, 3); /* No assoc data elements */ - FDKwriteBits(hBs, 0, 4); /* No valid cc elements */ - FDKwriteBits(hBs, 0, 1); /* Mono mixdown present */ - FDKwriteBits(hBs, 0, 1); /* Stereo mixdown present */ - - if ( matrixMixdownA!=0 && ((channelMode==MODE_1_2_2)||(channelMode==MODE_1_2_2_1)) ) { - FDKwriteBits(hBs, 1, 1); /* Matrix mixdown present */ - FDKwriteBits(hBs, (matrixMixdownA-1)&0x3, 2); /* matrix_mixdown_idx */ - FDKwriteBits(hBs, (pseudoSurroundEnable)?1:0, 1); /* pseudo_surround_enable */ - } - else { - FDKwriteBits(hBs, 0, 1); /* Matrix mixdown not present */ - } - - for(i=0; inum_front_channel_elements; i++) { - UCHAR isCpe = (*pEl_list++==ID_CPE) ? 1 : 0; - UCHAR tag = (isCpe) ? cpeCnt++ : sceCnt++; - FDKwriteBits(hBs, isCpe, 1); /* Front channel Elements is CPE? */ - FDKwriteBits(hBs, tag, 4); /* Front channel Instance Tag.*/ - } - for(i=0; inum_side_channel_elements; i++) { - UCHAR isCpe = (*pEl_list++==ID_CPE) ? 1 : 0; - UCHAR tag = (isCpe) ? cpeCnt++ : sceCnt++; - FDKwriteBits(hBs, isCpe, 1); /* Front channel Elements is CPE? */ - FDKwriteBits(hBs, tag, 4); /* Front channel Instance Tag.*/ - } - for(i=0; inum_back_channel_elements; i++) { - UCHAR isCpe = (*pEl_list++==ID_CPE) ? 1 : 0; - UCHAR tag = (isCpe) ? cpeCnt++ : sceCnt++; - FDKwriteBits(hBs, isCpe, 1); /* Front channel Elements is CPE? */ - FDKwriteBits(hBs, tag, 4); /* Front channel Instance Tag.*/ - } - for(i=0; inum_lfe_channel_elements; i++) { - FDKwriteBits(hBs, lfeCnt++, 4); /* LFE channel Instance Tag. */ - } - - /* - num_valid_cc_elements always 0. - - num_assoc_data_elements always 0. */ - - /* Byte alignment: relative to alignAnchor - ADTS: align with respect to the first bit of the raw_data_block() - ADIF: align with respect to the first bit of the header - LATM: align with respect to the first bit of the ASC */ - FDKbyteAlign(hBs, alignAnchor); /* Alignment */ - - FDKwriteBits(hBs, 0 ,8); /* Do no write any comment. */ - - /* - comment_field_bytes always 0. */ - - return 0; -} - -int transportEnc_GetPCEBits(CHANNEL_MODE channelMode, - int matrixMixdownA, - int bits) -{ - const PCE_CONFIGURATION* config = NULL; - - if ((config=getPceEntry(channelMode))==NULL) { - return -1; /* unsupported channelmapping */ - } - - bits += 4 + 2 + 4; /* Element instance tag + Object type + Sample rate index */ - bits += 4 + 4 + 4 + 2; /* No (front + side + back + lfe channel) elements */ - bits += 3 + 4; /* No (assoc data + valid cc) elements */ - bits += 1 + 1 + 1 ; /* Mono + Stereo + Matrix mixdown present */ - - if ( matrixMixdownA!=0 && ((channelMode==MODE_1_2_2)||(channelMode==MODE_1_2_2_1)) ) { - bits +=3; /* matrix_mixdown_idx + pseudo_surround_enable */ - } - - bits += (1+4) * (INT)config->num_front_channel_elements; - bits += (1+4) * (INT)config->num_side_channel_elements; - bits += (1+4) * (INT)config->num_back_channel_elements; - bits += (4) * (INT)config->num_lfe_channel_elements; - - /* - num_valid_cc_elements always 0. - - num_assoc_data_elements always 0. */ - - if ((bits%8) != 0) { - bits += (8 - (bits%8)); /* Alignment */ - } - - bits += 8; /* Comment field bytes */ - - /* - comment_field_bytes alwys 0. */ - - return bits; -} -#endif /* TP_PCE_ENABLE */ - -static void writeAot(HANDLE_FDK_BITSTREAM hBitstreamBuffer, AUDIO_OBJECT_TYPE aot) -{ - int tmp = (int) aot; - - if (tmp > 31) { - FDKwriteBits( hBitstreamBuffer, AOT_ESCAPE, 5 ); - FDKwriteBits( hBitstreamBuffer, tmp-32, 6 ); /* AudioObjectType */ - } else { - FDKwriteBits( hBitstreamBuffer, tmp, 5 ); - } -} - -static void writeSampleRate(HANDLE_FDK_BITSTREAM hBitstreamBuffer, int sampleRate) -{ - int sampleRateIndex = getSamplingRateIndex(sampleRate); - - FDKwriteBits( hBitstreamBuffer, sampleRateIndex, 4 ); - if( sampleRateIndex == 15 ) { - FDKwriteBits( hBitstreamBuffer, sampleRate, 24 ); - } -} - -#ifdef TP_GA_ENABLE -static -int transportEnc_writeGASpecificConfig( - HANDLE_FDK_BITSTREAM asc, - CODER_CONFIG *config, - int extFlg, - UINT alignAnchor - ) -{ - int aot = config->aot; - int samplesPerFrame = config->samplesPerFrame; - - /* start of GASpecificConfig according to ISO/IEC 14496-3 Subpart 4, 4.4.1 */ - FDKwriteBits( asc, ((samplesPerFrame==960 || samplesPerFrame==480)?1:0), 1); /* frameLengthFlag: 1 for a 960/480 (I)MDCT, 0 for a 1024/512 (I)MDCT*/ - FDKwriteBits( asc, 0, 1); /* dependsOnCoreCoder: Sampling Rate Coder Specific, see in ISO/IEC 14496-3 Subpart 4, 4.4.1 */ - FDKwriteBits( asc, extFlg, 1 ); /* Extension Flag: Shall be 1 for aot = 17,19,20,21,22,23 */ - - /* Write PCE if channel config is not 1-7 */ - if (getChannelConfig(config->channelMode) == 0) { - transportEnc_writePCE(asc, config->channelMode, config->samplingRate, 0, 1, config->matrixMixdownA, (config->flags&CC_PSEUDO_SURROUND)?1:0, alignAnchor); - } - if (extFlg) { - if (aot == AOT_ER_BSAC) { - FDKwriteBits( asc, config->BSACnumOfSubFrame, 5 ); /* numOfSubFrame */ - FDKwriteBits( asc, config->BSAClayerLength, 11 ); /* layer_length */ - } - if ((aot == AOT_ER_AAC_LC) || (aot == AOT_ER_AAC_LTP) || - (aot == AOT_ER_AAC_SCAL) || (aot == AOT_ER_AAC_LD)) - { - FDKwriteBits( asc, (config->flags & CC_VCB11) ? 1 : 0, 1 ); /* aacSectionDataResillienceFlag */ - FDKwriteBits( asc, (config->flags & CC_RVLC) ? 1 : 0, 1 ); /* aacScaleFactorDataResillienceFlag */ - FDKwriteBits( asc, (config->flags & CC_HCR) ? 1 : 0, 1 ); /* aacSpectralDataResillienceFlag */ - } - FDKwriteBits( asc, 0, 1 ); /* extensionFlag3: reserved. Shall be '0' */ - } - return 0; -} -#endif /* TP_GA_ENABLE */ - -#ifdef TP_ELD_ENABLE - -static -int transportEnc_writeELDSpecificConfig( - HANDLE_FDK_BITSTREAM hBs, - CODER_CONFIG *config, - int epConfig, - CSTpCallBacks *cb - ) -{ - /* ELD specific config */ - if (config->channelMode == MODE_1_1) { - return -1; - } - FDKwriteBits(hBs, (config->samplesPerFrame == 480) ? 1 : 0, 1); - - FDKwriteBits(hBs, (config->flags & CC_VCB11 ) ? 1:0, 1); - FDKwriteBits(hBs, (config->flags & CC_RVLC ) ? 1:0, 1); - FDKwriteBits(hBs, (config->flags & CC_HCR ) ? 1:0, 1); - - FDKwriteBits(hBs, (config->flags & CC_SBR) ? 1:0, 1); /* SBR header flag */ - if ( (config->flags & CC_SBR) ) { - FDKwriteBits(hBs, (config->samplingRate == config->extSamplingRate) ? 0:1, 1); /* Samplerate Flag */ - FDKwriteBits(hBs, (config->flags & CC_SBRCRC) ? 1:0, 1); /* SBR CRC flag*/ - - if (cb->cbSbr != NULL) { - const PCE_CONFIGURATION *pPce; - int e; - - pPce = getPceEntry(config->channelMode); - - for (e=0; eel_list[e] != ID_NONE; e++ ) { - if ( (pPce->el_list[e] == ID_SCE) || (pPce->el_list[e] == ID_CPE) ) { - cb->cbSbr(cb->cbSbrData, hBs, 0, 0, 0, config->aot, pPce->el_list[e], e); - } - } - } - } - - FDKwriteBits(hBs, 0, 4); /* ELDEXT_TERM */ - - return 0; -} -#endif /* TP_ELD_ENABLE */ - - -int transportEnc_writeASC ( - HANDLE_FDK_BITSTREAM asc, - CODER_CONFIG *config, - CSTpCallBacks *cb - ) -{ - UINT extFlag = 0; - int err; - int epConfig = 0; - - /* Required for the PCE. */ - UINT alignAnchor = FDKgetValidBits(asc); - - /* Extension Flag: Shall be 1 for aot = 17,19,20,21,22,23,39 */ - switch (config->aot) { - case AOT_ER_AAC_LC: - case AOT_ER_AAC_LTP: - case AOT_ER_AAC_SCAL: - case AOT_ER_TWIN_VQ: - case AOT_ER_BSAC: - case AOT_ER_AAC_LD: - case AOT_ER_AAC_ELD: - case AOT_USAC: - extFlag = 1; - break; - default: - break; - } - - if (config->sbrSignaling==SIG_EXPLICIT_HIERARCHICAL && config->sbrPresent) - writeAot(asc, config->extAOT); - else - writeAot(asc, config->aot); - - { - writeSampleRate(asc, config->samplingRate); - } - - /* Try to guess a reasonable channel mode if not given */ - if (config->channelMode == MODE_INVALID) { - config->channelMode = transportEnc_GetChannelMode(config->noChannels); - if (config->channelMode == MODE_INVALID) - return -1; - } - - FDKwriteBits( asc, getChannelConfig(config->channelMode), 4 ); - - if (config->sbrSignaling==SIG_EXPLICIT_HIERARCHICAL && config->sbrPresent) { - writeSampleRate(asc, config->extSamplingRate); - writeAot(asc, config->aot); - } - - switch (config->aot) { -#ifdef TP_GA_ENABLE - case AOT_AAC_MAIN: - case AOT_AAC_LC: - case AOT_AAC_SSR: - case AOT_AAC_LTP: - case AOT_AAC_SCAL: - case AOT_TWIN_VQ: - case AOT_ER_AAC_LC: - case AOT_ER_AAC_LTP: - case AOT_ER_AAC_SCAL: - case AOT_ER_TWIN_VQ: - case AOT_ER_BSAC: - case AOT_ER_AAC_LD: - err = transportEnc_writeGASpecificConfig(asc, config, extFlag, alignAnchor); - if (err) - return err; - break; -#endif /* TP_GA_ENABLE */ -#ifdef TP_ELD_ENABLE - case AOT_ER_AAC_ELD: - err = transportEnc_writeELDSpecificConfig(asc, config, epConfig, cb); - if (err) - return err; - break; -#endif /* TP_ELD_ENABLE */ - default: - return -1; - } - - switch (config->aot) { - case AOT_ER_AAC_LC: - case AOT_ER_AAC_LTP: - case AOT_ER_AAC_SCAL: - case AOT_ER_TWIN_VQ: - case AOT_ER_BSAC: - case AOT_ER_AAC_LD: - case AOT_ER_CELP: - case AOT_ER_HVXC: - case AOT_ER_HILN: - case AOT_ER_PARA: - case AOT_ER_AAC_ELD: - FDKwriteBits( asc, 0, 2 ); /* epconfig 0 */ - break; - default: - break; - } - - /* backward compatible explicit signaling of extension AOT */ - if (config->sbrSignaling==SIG_EXPLICIT_BW_COMPATIBLE) - { - TP_ASC_EXTENSION_ID ascExtId = ASCEXT_UNKOWN; - - if (config->sbrPresent) { - ascExtId=ASCEXT_SBR; - FDKwriteBits( asc, ascExtId, 11 ); - writeAot(asc, config->extAOT); - FDKwriteBits( asc, 1, 1 ); /* sbrPresentFlag=1 */ - writeSampleRate(asc, config->extSamplingRate); - if (config->psPresent) { - ascExtId=ASCEXT_PS; - FDKwriteBits( asc, ascExtId, 11 ); - FDKwriteBits( asc, 1, 1 ); /* psPresentFlag=1 */ - } - } - - } - - /* Make sure all bits are sync'ed */ - FDKsyncCache( asc ); - - return 0; -} diff --git a/libMpegTPEnc/src/tpenc_asc.h b/libMpegTPEnc/src/tpenc_asc.h deleted file mode 100644 index 47fe7a1..0000000 --- a/libMpegTPEnc/src/tpenc_asc.h +++ /dev/null @@ -1,142 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/***************************** MPEG-4 AAC Encoder ************************** - - Author(s): Manuel Jander - Description: Audio Specific Config writer - -******************************************************************************/ - -#ifndef TPENC_ASC_H -#define TPENC_ASC_H - -/** - * \brief Get channel config from channel mode. - * - * \param channel_mode channel mode - * - * \return chanel config - */ -int getChannelConfig( CHANNEL_MODE channel_mode ); - -/** - * \brief Write a Program Config Element. - * - * \param hBs bitstream handle into which the PCE is appended - * \param channelMode the channel mode to be used - * \param sampleRate the sample rate - * \param instanceTagPCE the instance tag of the Program Config Element - * \param profile the MPEG Audio profile to be used - * \param matrix mixdown gain - * \param pseudo surround indication - * \param reference bitstream position for alignment - * \return zero on success, non-zero on failure. - */ -int transportEnc_writePCE( - HANDLE_FDK_BITSTREAM hBs, - CHANNEL_MODE channelMode, - INT sampleRate, - int instanceTagPCE, - int profile, - int matrixMixdownA, - int pseudoSurroundEnable, - UINT alignAnchor - ); - -/** - * \brief Get the bit count required by a Program Config Element - * - * \param channelMode the channel mode to be used - * \param matrix mixdown gain - * \param bit offset at which the PCE would start - * \return the amount of bits required for the PCE including the given bit offset. - */ -int transportEnc_GetPCEBits( - CHANNEL_MODE channelMode, - int matrixMixdownA, - int bits - ); - -#endif /* TPENC_ASC_H */ - diff --git a/libMpegTPEnc/src/tpenc_latm.cpp b/libMpegTPEnc/src/tpenc_latm.cpp deleted file mode 100644 index f292019..0000000 --- a/libMpegTPEnc/src/tpenc_latm.cpp +++ /dev/null @@ -1,881 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/***************************** MPEG-4 AAC Encoder ************************** - - Author(s): - Description: - -******************************************************************************/ - -#include "tpenc_latm.h" - - -#include "genericStds.h" - -static const short celpFrameLengthTable[64] = { - 154, 170, 186, 147, 156, 165, 114, 120, - 186, 126, 132, 138, 142, 146, 154, 166, - 174, 182, 190, 198, 206, 210, 214, 110, - 114, 118, 120, 122, 218, 230, 242, 254, - 266, 278, 286, 294, 318, 342, 358, 374, - 390, 406, 422, 136, 142, 148, 154, 160, - 166, 170, 174, 186, 198, 206, 214, 222, - 230, 238, 216, 160, 280, 338, 0, 0 -}; - -/******* - write value to transport stream - first two bits define the size of the value itself - then the value itself, with a size of 0-3 bytes -*******/ -static -UINT transportEnc_LatmWriteValue(HANDLE_FDK_BITSTREAM hBs, int value) -{ - UCHAR valueBytes = 4; - unsigned int bitsWritten = 0; - int i; - - if ( value < (1<<8) ) { - valueBytes = 1; - } else if ( value < (1<<16) ) { - valueBytes = 2; - } else if ( value < (1<<24) ) { - valueBytes = 3; - } else { - valueBytes = 4; - } - - FDKwriteBits(hBs, valueBytes-1, 2 ); /* size of value in Bytes */ - for (i=0; i>((valueBytes-1-i)<<3)), 8); - } - - bitsWritten = (valueBytes<<3)+2; - - return bitsWritten; -} - -static -UINT transportEnc_LatmCountFixBitDemandHeader ( HANDLE_LATM_STREAM hAss ) -{ - int bitDemand = 0; - int insertSetupData = 0 ; - - /* only if start of new latm frame */ - if (hAss->subFrameCnt==0) - { - /* AudioSyncStream */ - - if (hAss->tt == TT_MP4_LOAS) { - bitDemand += 11 ; /* syncword */ - bitDemand += 13 ; /* audioMuxLengthBytes */ - } - - /* AudioMuxElement*/ - - /* AudioMuxElement::Stream Mux Config */ - if (hAss->muxConfigPeriod > 0) { - insertSetupData = (hAss->latmFrameCounter == 0); - } else { - insertSetupData = 0; - } - - if (hAss->tt != TT_MP4_LATM_MCP0) { - /* AudioMuxElement::useSameStreamMux Flag */ - bitDemand+=1; - - if( insertSetupData ) { - bitDemand += hAss->streamMuxConfigBits; - } - } - - /* AudioMuxElement::otherDataBits */ - bitDemand += 8*hAss->otherDataLenBytes; - - /* AudioMuxElement::ByteAlign */ - if ( bitDemand % 8 ) { - hAss->fillBits = 8 - (bitDemand % 8); - bitDemand += hAss->fillBits ; - } else { - hAss->fillBits = 0; - } - } - - return bitDemand ; -} - -static -UINT transportEnc_LatmCountVarBitDemandHeader ( HANDLE_LATM_STREAM hAss , unsigned int streamDataLength ) -{ - int bitDemand = 0; - int prog, layer; - - /* Payload Length Info*/ - if( hAss->allStreamsSameTimeFraming ) { - for( prog=0; prognoProgram; prog++ ) { - for( layer=0; layerm_linfo[prog][layer]); - - if( p_linfo->streamID >= 0 ) { - switch( p_linfo->frameLengthType ) { - case 0: - if ( streamDataLength > 0 ) { - streamDataLength -= bitDemand ; - while( streamDataLength >= (255<<3) ) { - bitDemand+=8; - streamDataLength -= (255<<3); - } - bitDemand += 8; - } - break; - - case 1: - case 4: - case 6: - bitDemand += 2; - break; - - default: - return 0; - } - } - } - } - } else { - /* there are many possibilities to use this mechanism. */ - switch( hAss->varMode ) { - case LATMVAR_SIMPLE_SEQUENCE: { - /* Use the sequence generated by the encoder */ - //int streamCntPosition = transportEnc_SetWritePointer( hAss->hAssemble, 0 ); - //int streamCntPosition = FDKgetValidBits( hAss->hAssemble ); - bitDemand+=4; - - hAss->varStreamCnt = 0; - for( prog=0; prognoProgram; prog++ ) { - for( layer=0; layerm_linfo[prog][layer]); - - if( p_linfo->streamID >= 0 ) { - - bitDemand+=4; /* streamID */ - switch( p_linfo->frameLengthType ) { - case 0: - streamDataLength -= bitDemand ; - while( streamDataLength >= (255<<3) ) { - bitDemand+=8; - streamDataLength -= (255<<3); - } - - bitDemand += 8; - break; - /*bitDemand += 1; endFlag - break;*/ - - case 1: - case 4: - case 6: - - break; - - default: - return 0; - } - hAss->varStreamCnt++; - } - } - } - bitDemand+=4; - //transportEnc_UpdateBitstreamField( hAss->hAssemble, streamCntPosition, hAss->varStreamCnt-1, 4 ); - //UINT pos = streamCntPosition-FDKgetValidBits(hAss->hAssemble); - //FDKpushBack( hAss->hAssemble, pos); - //FDKwriteBits( hAss->hAssemble, hAss->varStreamCnt-1, 4); - //FDKpushFor( hAss->hAssemble, pos-4); - } - break; - - default: - return 0; - } - } - - return bitDemand ; -} - -TRANSPORTENC_ERROR -CreateStreamMuxConfig( - HANDLE_LATM_STREAM hAss, - HANDLE_FDK_BITSTREAM hBs, - int bufferFullness, - CSTpCallBacks *cb - ) -{ - INT streamIDcnt, tmp; - int layer, prog; - - USHORT coreFrameOffset=0; - - hAss->taraBufferFullness = 0xFF; - hAss->audioMuxVersionA = 0; /* for future extensions */ - hAss->streamMuxConfigBits = 0; - - FDKwriteBits( hBs, hAss->audioMuxVersion, 1 ); /* audioMuxVersion */ - hAss->streamMuxConfigBits += 1; - - if ( hAss->audioMuxVersion == 1 ) { - FDKwriteBits( hBs, hAss->audioMuxVersionA, 1 ); /* audioMuxVersionA */ - hAss->streamMuxConfigBits+=1; - } - - if ( hAss->audioMuxVersionA == 0 ) - { - if ( hAss->audioMuxVersion == 1 ) { - hAss->streamMuxConfigBits+= transportEnc_LatmWriteValue( hBs, hAss->taraBufferFullness );/* taraBufferFullness */ - } - FDKwriteBits( hBs, hAss->allStreamsSameTimeFraming ? 1:0, 1 ); /* allStreamsSameTimeFraming */ - FDKwriteBits( hBs, hAss->noSubframes-1, 6 ); /* Number of Subframes */ - FDKwriteBits( hBs, hAss->noProgram-1, 4 ); /* Number of Programs */ - - hAss->streamMuxConfigBits+=11; - - streamIDcnt = 0; - for( prog=0; prognoProgram; prog++ ) { - int transLayer = 0; - - FDKwriteBits( hBs, hAss->noLayer[prog]-1, 3 ); - hAss->streamMuxConfigBits+=3; - - for( layer=0; layerm_linfo[prog][layer]); - CODER_CONFIG *p_lci = hAss->config[prog][layer]; - - p_linfo->streamID = -1; - - if( hAss->config[prog][layer] != NULL ) { - int useSameConfig = 0; - - if( transLayer > 0 ) { - FDKwriteBits( hBs, useSameConfig ? 1 : 0, 1 ); - hAss->streamMuxConfigBits+=1; - } - if( (useSameConfig == 0) || (transLayer==0) ) { - const UINT alignAnchor = FDKgetValidBits(hBs); - - transportEnc_writeASC( - hBs, - hAss->config[prog][layer], - cb - ); - - if ( hAss->audioMuxVersion == 1 ) { - UINT ascLen = transportEnc_LatmWriteValue(hBs, 0); - FDKbyteAlign(hBs, alignAnchor); - ascLen = FDKgetValidBits(hBs) - alignAnchor - ascLen; - FDKpushBack(hBs, FDKgetValidBits(hBs) - alignAnchor); - - transportEnc_LatmWriteValue(hBs, ascLen); - - transportEnc_writeASC( - hBs, - hAss->config[prog][layer], - cb - ); - - FDKbyteAlign(hBs, alignAnchor); /* asc length fillbits */ - } - - hAss->streamMuxConfigBits += FDKgetValidBits(hBs) - alignAnchor; /* add asc length to smc summary */ - } - transLayer++; - - if( !hAss->allStreamsSameTimeFraming ) { - if( streamIDcnt >= LATM_MAX_STREAM_ID ) - return TRANSPORTENC_INVALID_CONFIG; - } - p_linfo->streamID = streamIDcnt++; - - switch( p_lci->aot ) { - case AOT_AAC_MAIN : - case AOT_AAC_LC : - case AOT_AAC_SSR : - case AOT_AAC_LTP : - case AOT_AAC_SCAL : - case AOT_ER_AAC_LD : - case AOT_ER_AAC_ELD : - case AOT_USAC: - p_linfo->frameLengthType = 0; - - FDKwriteBits( hBs, p_linfo->frameLengthType, 3 ); /* frameLengthType */ - FDKwriteBits( hBs, bufferFullness, 8 ); /* bufferFullness */ - hAss->streamMuxConfigBits+=11; - - if ( !hAss->allStreamsSameTimeFraming ) { - CODER_CONFIG *p_lci_prev = hAss->config[prog][layer-1]; - if ( ((p_lci->aot == AOT_AAC_SCAL) || (p_lci->aot == AOT_ER_AAC_SCAL)) && - ((p_lci_prev->aot == AOT_CELP) || (p_lci_prev->aot == AOT_ER_CELP)) ) { - FDKwriteBits( hBs, coreFrameOffset, 6 ); /* coreFrameOffset */ - hAss->streamMuxConfigBits+=6; - } - } - break; - - case AOT_TWIN_VQ: - p_linfo->frameLengthType = 1; - tmp = ( (p_lci->bitsFrame+7) >> 3 ) - 20; /* transmission frame length in bytes */ - if( (tmp < 0) ) { - return TRANSPORTENC_INVALID_TRANSMISSION_FRAME_LENGTH; - } - FDKwriteBits( hBs, p_linfo->frameLengthType, 3 ); /* frameLengthType */ - FDKwriteBits( hBs, tmp, 9 ); - hAss->streamMuxConfigBits+=12; - - p_linfo->frameLengthBits = (tmp+20) << 3; - break; - - case AOT_CELP: - p_linfo->frameLengthType = 4; - FDKwriteBits( hBs, p_linfo->frameLengthType, 3 ); /* frameLengthType */ - hAss->streamMuxConfigBits+=3; - { - int i; - for( i=0; i<62; i++ ) { - if( celpFrameLengthTable[i] == p_lci->bitsFrame ) - break; - } - if( i>=62 ) { - return TRANSPORTENC_INVALID_CELP_FRAME_LENGTH; - } - - FDKwriteBits( hBs, i, 6 ); /* CELPframeLengthTabelIndex */ - hAss->streamMuxConfigBits+=6; - } - p_linfo->frameLengthBits = p_lci->bitsFrame; - break; - - case AOT_HVXC: - p_linfo->frameLengthType = 6; - FDKwriteBits( hBs, p_linfo->frameLengthType, 3 ); /* frameLengthType */ - hAss->streamMuxConfigBits+=3; - { - int i; - - if( p_lci->bitsFrame == 40 ) { - i = 0; - } else if( p_lci->bitsFrame == 80 ) { - i = 1; - } else { - return TRANSPORTENC_INVALID_FRAME_BITS; - } - FDKwriteBits( hBs, i, 1 ); /* HVXCframeLengthTableIndex */ - hAss->streamMuxConfigBits+=1; - } - p_linfo->frameLengthBits = p_lci->bitsFrame; - break; - - case AOT_NULL_OBJECT: - default: - return TRANSPORTENC_INVALID_AOT; - } - } - } - } - - FDKwriteBits( hBs, (hAss->otherDataLenBytes>0) ? 1:0, 1 ); /* otherDataPresent */ - hAss->streamMuxConfigBits+=1; - - if( hAss->otherDataLenBytes > 0 ) { - - INT otherDataLenTmp = hAss->otherDataLenBytes; - INT escCnt = 0; - INT otherDataLenEsc = 1; - - while(otherDataLenTmp) { - otherDataLenTmp >>= 8; - escCnt ++; - } - - do { - otherDataLenTmp = (hAss->otherDataLenBytes>>(escCnt*8)) & 0xFF; - escCnt--; - otherDataLenEsc = escCnt>0; - - FDKwriteBits( hBs, otherDataLenEsc, 1 ); - FDKwriteBits( hBs, otherDataLenTmp, 8 ); - hAss->streamMuxConfigBits+=9; - } while(otherDataLenEsc); - } - - { - USHORT crcCheckPresent=0; - USHORT crcCheckSum=0; - - FDKwriteBits( hBs, crcCheckPresent, 1 ); /* crcCheckPresent */ - hAss->streamMuxConfigBits+=1; - if ( crcCheckPresent ){ - FDKwriteBits( hBs, crcCheckSum, 8 ); /* crcCheckSum */ - hAss->streamMuxConfigBits+=8; - } - } - - } else { /* if ( audioMuxVersionA == 0 ) */ - - /* for future extensions */ - - } - - return TRANSPORTENC_OK; -} - - -static TRANSPORTENC_ERROR -WriteAuPayloadLengthInfo( HANDLE_FDK_BITSTREAM hBitStream, int AuLengthBits ) -{ - int restBytes; - - if( AuLengthBits % 8 ) - return TRANSPORTENC_INVALID_AU_LENGTH; - - while( AuLengthBits >= 255*8 ) { - FDKwriteBits( hBitStream, 255, 8 ); /* 255 shows incomplete AU */ - AuLengthBits -= (255*8); - } - - restBytes = (AuLengthBits) >> 3; - FDKwriteBits( hBitStream, restBytes, 8 ); - - return TRANSPORTENC_OK; -} - -static -TRANSPORTENC_ERROR transportEnc_LatmSetNrOfSubframes( HANDLE_LATM_STREAM hAss, - INT noSubframes_next) /* nr of access units / payloads within a latm frame */ -{ - /* sanity chk */ - if (noSubframes_next < 1 || noSubframes_next > MAX_NR_OF_SUBFRAMES) { - return TRANSPORTENC_LATM_INVALID_NR_OF_SUBFRAMES; - } - - hAss->noSubframes_next = noSubframes_next; - - /* if at start then we can take over the value immediately, otherwise we have to wait for the next SMC */ - if ( (hAss->subFrameCnt == 0) && (hAss->latmFrameCounter == 0) ) { - hAss->noSubframes = noSubframes_next; - } - - return TRANSPORTENC_OK; -} - -static -int allStreamsSameTimeFraming( HANDLE_LATM_STREAM hAss, UCHAR noProgram, UCHAR noLayer[] /* return */ ) -{ - int prog, layer; - - signed int lastNoSamples = -1; - signed int minFrameSamples = FDK_INT_MAX; - signed int maxFrameSamples = 0; - - signed int highestSamplingRate = -1; - - for( prog=0; progconfig[prog][layer] != NULL ) - { - INT hsfSamplesFrame; - - noLayer[prog]++; - - if( highestSamplingRate < 0 ) - highestSamplingRate = hAss->config[prog][layer]->samplingRate; - - hsfSamplesFrame = hAss->config[prog][layer]->samplesPerFrame * highestSamplingRate / hAss->config[prog][layer]->samplingRate; - - if( hsfSamplesFrame <= minFrameSamples ) minFrameSamples = hsfSamplesFrame; - if( hsfSamplesFrame >= maxFrameSamples ) maxFrameSamples = hsfSamplesFrame; - - if( lastNoSamples == -1 ) { - lastNoSamples = hsfSamplesFrame; - } else { - if( hsfSamplesFrame != lastNoSamples ) { - return 0; - } - } - } - } - } - - return 1; -} - -/** - * Initialize LATM/LOAS Stream and add layer 0 at program 0. - */ -static -TRANSPORTENC_ERROR transportEnc_InitLatmStream( HANDLE_LATM_STREAM hAss, - int fractDelayPresent, - signed int muxConfigPeriod, /* insert setup data every muxConfigPeriod frames */ - UINT audioMuxVersion, - TRANSPORT_TYPE tt - ) -{ - TRANSPORTENC_ERROR ErrorStatus = TRANSPORTENC_OK; - - if (hAss == NULL) - return TRANSPORTENC_INVALID_PARAMETER; - - hAss->tt = tt; - - hAss->noProgram = 1; - - hAss->audioMuxVersion = audioMuxVersion; - - /* Fill noLayer array using hAss->config */ - hAss->allStreamsSameTimeFraming = allStreamsSameTimeFraming( hAss, hAss->noProgram, hAss->noLayer ); - /* Only allStreamsSameTimeFraming==1 is supported */ - FDK_ASSERT(hAss->allStreamsSameTimeFraming); - - hAss->fractDelayPresent = fractDelayPresent; - hAss->otherDataLenBytes = 0; - - hAss->varMode = LATMVAR_SIMPLE_SEQUENCE; - - /* initialize counters */ - hAss->subFrameCnt = 0; - hAss->noSubframes = DEFAULT_LATM_NR_OF_SUBFRAMES; - hAss->noSubframes_next = DEFAULT_LATM_NR_OF_SUBFRAMES; - - /* sync layer related */ - hAss->audioMuxLengthBytes = 0; - - hAss->latmFrameCounter = 0; - hAss->muxConfigPeriod = muxConfigPeriod; - - return ErrorStatus; -} - - -/** - * - */ -UINT transportEnc_LatmCountTotalBitDemandHeader ( HANDLE_LATM_STREAM hAss , unsigned int streamDataLength ) -{ - UINT bitDemand = 0; - - switch (hAss->tt) { - case TT_MP4_LOAS: - case TT_MP4_LATM_MCP0: - case TT_MP4_LATM_MCP1: - if (hAss->subFrameCnt == 0) { - bitDemand = transportEnc_LatmCountFixBitDemandHeader ( hAss ); - } - bitDemand += transportEnc_LatmCountVarBitDemandHeader ( hAss , streamDataLength /*- bitDemand*/); - break; - default: - break; - } - - return bitDemand; -} - -static TRANSPORTENC_ERROR -AdvanceAudioMuxElement ( - HANDLE_LATM_STREAM hAss, - HANDLE_FDK_BITSTREAM hBs, - int auBits, - int bufferFullness, - CSTpCallBacks *cb - ) -{ - TRANSPORTENC_ERROR ErrorStatus = TRANSPORTENC_OK; - int insertMuxSetup; - - /* Insert setup data to assemble Buffer */ - if (hAss->subFrameCnt == 0) - { - if (hAss->muxConfigPeriod > 0) { - insertMuxSetup = (hAss->latmFrameCounter == 0); - } else { - insertMuxSetup = 0; - } - - if (hAss->tt != TT_MP4_LATM_MCP0) { - if( insertMuxSetup ) { - FDKwriteBits( hBs, 0, 1 ); /* useSameStreamMux useNewStreamMux */ - CreateStreamMuxConfig(hAss, hBs, bufferFullness, cb); - if (ErrorStatus != TRANSPORTENC_OK) - return ErrorStatus; - } else { - FDKwriteBits( hBs, 1, 1 ); /* useSameStreamMux */ - } - } - } - - /* PayloadLengthInfo */ - { - int prog, layer; - - for (prog = 0; prog < hAss->noProgram; prog++) { - for (layer = 0; layer < hAss->noLayer[prog]; layer++) { - ErrorStatus = WriteAuPayloadLengthInfo( hBs, auBits ); - if (ErrorStatus != TRANSPORTENC_OK) - return ErrorStatus; - } - } - } - /* At this point comes the access unit. */ - - return TRANSPORTENC_OK; -} - -TRANSPORTENC_ERROR -transportEnc_LatmWrite ( - HANDLE_LATM_STREAM hAss, - HANDLE_FDK_BITSTREAM hBs, - int auBits, - int bufferFullness, - CSTpCallBacks *cb - ) -{ - TRANSPORTENC_ERROR ErrorStatus; - - if (hAss->subFrameCnt == 0) { - /* Start new frame */ - FDKresetBitbuffer(hBs, BS_WRITER); - } - - hAss->latmSubframeStart = FDKgetValidBits(hBs); - - /* Insert syncword and syncword distance - - only if loas - - we must update the syncword distance (=audiomuxlengthbytes) later - */ - if( hAss->tt == TT_MP4_LOAS && hAss->subFrameCnt == 0) - { - /* Start new LOAS frame */ - FDKwriteBits( hBs, 0x2B7, 11 ); - hAss->audioMuxLengthBytes = 0; - hAss->audioMuxLengthBytesPos = FDKgetValidBits( hBs ); /* store read pointer position */ - FDKwriteBits( hBs, hAss->audioMuxLengthBytes, 13 ); - } - - ErrorStatus = AdvanceAudioMuxElement( - hAss, - hBs, - auBits, - bufferFullness, - cb - ); - - if (ErrorStatus != TRANSPORTENC_OK) - return ErrorStatus; - - return ErrorStatus; -} - -void transportEnc_LatmAdjustSubframeBits(HANDLE_LATM_STREAM hAss, - int *bits) -{ - /* Substract bits from possible previous subframe */ - *bits -= hAss->latmSubframeStart; - /* Add fill bits */ - if (hAss->subFrameCnt == 0) - *bits += hAss->fillBits; -} - - -void transportEnc_LatmGetFrame(HANDLE_LATM_STREAM hAss, - HANDLE_FDK_BITSTREAM hBs, - int *bytes) -{ - - hAss->subFrameCnt++; - if (hAss->subFrameCnt >= hAss->noSubframes) - { - - /* Add LOAS frame length if required. */ - if (hAss->tt == TT_MP4_LOAS) - { - int latmBytes; - - latmBytes = (FDKgetValidBits(hBs)+7) >> 3; - - /* write length info into assembler buffer */ - hAss->audioMuxLengthBytes = latmBytes - 3; /* 3=Syncword + length */ - { - FDK_BITSTREAM tmpBuf; - - FDKinitBitStream( &tmpBuf, hBs->hBitBuf.Buffer, hBs->hBitBuf.bufSize, 0, BS_WRITER ) ; - FDKpushFor( &tmpBuf, hAss->audioMuxLengthBytesPos ); - FDKwriteBits( &tmpBuf, hAss->audioMuxLengthBytes, 13 ); - FDKsyncCache( &tmpBuf ); - } - } - - /* Write AudioMuxElement byte alignment fill bits */ - FDKwriteBits(hBs, 0, hAss->fillBits); - - FDK_ASSERT( (FDKgetValidBits(hBs) % 8) == 0); - - hAss->subFrameCnt = 0; - - FDKsyncCache(hBs); - *bytes = (FDKgetValidBits(hBs) + 7)>>3; - //FDKfetchBuffer(hBs, buffer, (UINT*)bytes); - - if (hAss->muxConfigPeriod > 0) - { - hAss->latmFrameCounter++; - - if (hAss->latmFrameCounter >= hAss->muxConfigPeriod) { - hAss->latmFrameCounter = 0; - hAss->noSubframes = hAss->noSubframes_next; - } - } - } else { - /* No data this time */ - *bytes = 0; - } -} - -/** - * Init LATM/LOAS - */ -TRANSPORTENC_ERROR transportEnc_Latm_Init( - HANDLE_LATM_STREAM hAss, - HANDLE_FDK_BITSTREAM hBs, - CODER_CONFIG *layerConfig, - UINT audioMuxVersion, - TRANSPORT_TYPE tt, - CSTpCallBacks *cb - ) -{ - TRANSPORTENC_ERROR ErrorStatus; - int fractDelayPresent = 0; - int prog, layer; - - int setupDataDistanceFrames = layerConfig->headerPeriod; - - FDK_ASSERT(setupDataDistanceFrames>=0); - - for (prog=0; progconfig[prog][layer] = NULL; - hAss->m_linfo[prog][layer].streamID = -1; - } - } - - hAss->config[0][0] = layerConfig; - hAss->m_linfo[0][0].streamID = 0; - - ErrorStatus = transportEnc_InitLatmStream( hAss, - fractDelayPresent, - setupDataDistanceFrames, - (audioMuxVersion)?1:0, - tt - ); - if (ErrorStatus != TRANSPORTENC_OK) - goto bail; - - ErrorStatus = transportEnc_LatmSetNrOfSubframes( - hAss, - layerConfig->nSubFrames - ); - if (ErrorStatus != TRANSPORTENC_OK) - goto bail; - - /* Get the size of the StreamMuxConfig somehow */ - AdvanceAudioMuxElement(hAss, hBs, 0, 0, cb); - //CreateStreamMuxConfig(hAss, hBs, 0); - -bail: - return ErrorStatus; -} - - - - - - diff --git a/libMpegTPEnc/src/tpenc_latm.h b/libMpegTPEnc/src/tpenc_latm.h deleted file mode 100644 index 34eea58..0000000 --- a/libMpegTPEnc/src/tpenc_latm.h +++ /dev/null @@ -1,264 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/***************************** MPEG-4 AAC Encoder ************************** - - Author(s): - Description: - -******************************************************************************/ - -#ifndef TPENC_LATM_H -#define TPENC_LATM_H - - - -#include "tpenc_lib.h" -#include "FDK_bitstream.h" - - -#define DEFAULT_LATM_NR_OF_SUBFRAMES 1 -#define DEFAULT_LATM_SMC_REPEAT 8 - -#define MAX_AAC_LAYERS 9 - -#define LATM_MAX_PROGRAMS 1 -#define LATM_MAX_STREAM_ID 16 - -#define LATM_MAX_LAYERS 1 /*MAX_AAC_LAYERS*/ - -#define MAX_NR_OF_SUBFRAMES 2 /* set this carefully to avoid buffer overflows */ - -typedef enum { LATMVAR_SIMPLE_SEQUENCE } LATM_VAR_MODE; - -typedef struct { - signed int frameLengthType; - signed int frameLengthBits; - signed int varFrameLengthTable[4]; - signed int streamID; -} LATM_LAYER_INFO; - - -typedef struct { - LATM_LAYER_INFO m_linfo[LATM_MAX_PROGRAMS][LATM_MAX_LAYERS]; - CODER_CONFIG *config[LATM_MAX_PROGRAMS][LATM_MAX_LAYERS]; - - LATM_VAR_MODE varMode; - TRANSPORT_TYPE tt; - - int audioMuxLengthBytes; - - int audioMuxLengthBytesPos; - int taraBufferFullness; /* state of the bit reservoir */ - int varStreamCnt; - unsigned int otherDataLenBytes; - - UCHAR latmFrameCounter; /* Current frame number. Counts modulo muxConfigPeriod */ - UCHAR muxConfigPeriod; /* Distance in frames between MuxConfig */ - - UCHAR audioMuxVersion; /* AMV1 supports transmission of taraBufferFullness and ASC lengths */ - UCHAR audioMuxVersionA; /* for future extensions */ - - UCHAR noProgram; - UCHAR noLayer[LATM_MAX_PROGRAMS]; - UCHAR fractDelayPresent; - - UCHAR allStreamsSameTimeFraming; - UCHAR subFrameCnt; /* Current Subframe frame */ - UCHAR noSubframes; /* Number of subframes */ - UINT latmSubframeStart; /* Position of current subframe start */ - UCHAR noSubframes_next; - - UCHAR fillBits; /* AudioMuxElement fill bits */ - UCHAR streamMuxConfigBits; - -} LATM_STREAM; - -typedef LATM_STREAM *HANDLE_LATM_STREAM; - -/** - * \brief Initialize LATM_STREAM Handle. Creates automatically one program with one layer with - * the given layerConfig. The layerConfig must be persisten because references to this pointer - * are made at any time again. - * Use transportEnc_Latm_AddLayer() to add more programs/layers. - * - * \param hLatmStreamInfo HANDLE_LATM_STREAM handle - * \param hBs Bitstream handle - * \param layerConfig a valid CODER_CONFIG struct containing the current audio configuration parameters - * \param audioMuxVersion the LATM audioMuxVersion to be used - * \param tt the specific TRANSPORT_TYPE to be used, either TT_MP4_LOAS, TT_MP4_LATM_MCP1 or TT_MP4_LATM_MCP0 LOAS - * \param cb callback information structure. - * - * \return an TRANSPORTENC_ERROR error code - */ -TRANSPORTENC_ERROR transportEnc_Latm_Init( - HANDLE_LATM_STREAM hLatmStreamInfo, - HANDLE_FDK_BITSTREAM hBs, - CODER_CONFIG *layerConfig, - UINT audioMuxVersion, - TRANSPORT_TYPE tt, - CSTpCallBacks *cb - ); - -/** - * \brief Get bit demand of next LATM/LOAS header - * - * \param hAss HANDLE_LATM_STREAM handle - * \param streamDataLength the length of the payload - * - * \return the number of bits required by the LATM/LOAS headers - */ -unsigned int transportEnc_LatmCountTotalBitDemandHeader ( - HANDLE_LATM_STREAM hAss, - unsigned int streamDataLength - ); - -/** - * \brief Write LATM/LOAS header into given bitstream handle - * - * \param hLatmStreamInfo HANDLE_LATM_STREAM handle - * \param hBitstream Bitstream handle - * \param auBits amount of current payload bits - * \param bufferFullness LATM buffer fullness value - * \param cb callback information structure. - * - * \return an TRANSPORTENC_ERROR error code - */ -TRANSPORTENC_ERROR -transportEnc_LatmWrite ( - HANDLE_LATM_STREAM hAss, - HANDLE_FDK_BITSTREAM hBitstream, - int auBits, - int bufferFullness, - CSTpCallBacks *cb - ); - -/** - * \brief Adjust bit count relative to current subframe - * - * \param hAss HANDLE_LATM_STREAM handle - * \param pBits pointer to an int, where the current frame bit count is contained, - * and where the subframe relative bit count will be returned into - * - * \return void - */ -void transportEnc_LatmAdjustSubframeBits(HANDLE_LATM_STREAM hAss, - int *pBits); - -/** - * \brief Request an LATM frame, which may, or may not be available - * - * \param hAss HANDLE_LATM_STREAM handle - * \param hBs Bitstream handle - * \param pBytes pointer to an int, where the current frame byte count stored into. - * A return value of zero means that currently no LATM/LOAS frame can be returned. - * The latter is expected in case of multiple subframes being used. - * - * \return void - */ -void transportEnc_LatmGetFrame( - HANDLE_LATM_STREAM hAss, - HANDLE_FDK_BITSTREAM hBs, - int *pBytes - ); - -/** - * \brief Write a StreamMuxConfig into the given bitstream handle - * - * \param hAss HANDLE_LATM_STREAM handle - * \param hBs Bitstream handle - * \param bufferFullness LATM buffer fullness value - * \param cb callback information structure. - * - * \return void - */ -TRANSPORTENC_ERROR -CreateStreamMuxConfig( - HANDLE_LATM_STREAM hAss, - HANDLE_FDK_BITSTREAM hBs, - int bufferFullness, - CSTpCallBacks *cb - ); - - -#endif /* TPENC_LATM_H */ diff --git a/libMpegTPEnc/src/tpenc_lib.cpp b/libMpegTPEnc/src/tpenc_lib.cpp deleted file mode 100644 index 24fb32f..0000000 --- a/libMpegTPEnc/src/tpenc_lib.cpp +++ /dev/null @@ -1,642 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/************************** MPEG-4 Transport Encoder ************************ - - Author(s): Manuel Jander - Description: MPEG Transport encode - -******************************************************************************/ - -#include "tpenc_lib.h" - -/* library info */ -#include "version" - -#define MODULE_NAME "transportEnc" - -#include "tpenc_asc.h" -#include "conv_string.h" - -#include "tpenc_adts.h" - -#include "tpenc_adif.h" - -#include "tpenc_latm.h" - - - -typedef struct { - int curSubFrame; - int nSubFrames; - int prevBits; -} RAWPACKETS_INFO; - -struct TRANSPORTENC -{ - CODER_CONFIG config; - TRANSPORT_TYPE transportFmt; /*!< MPEG4 transport type. */ - - FDK_BITSTREAM bitStream; - UCHAR *bsBuffer; - INT bsBufferSize; - - INT pceFrameCounter; /*!< Indicates frame period when PCE must be written in raw_data_block. - -1 means not to write a PCE in raw_dat_block. */ - union { - STRUCT_ADTS adts; - - ADIF_INFO adif; - - LATM_STREAM latm; - - RAWPACKETS_INFO raw; - - - - } writer; - - CSTpCallBacks callbacks; -}; - -typedef struct _TRANSPORTENC_STRUCT TRANSPORTENC_STRUCT; - - -/* - * MEMORY Declaration - */ - -C_ALLOC_MEM(Ram_TransportEncoder, TRANSPORTENC, 1) - -TRANSPORTENC_ERROR transportEnc_Open( HANDLE_TRANSPORTENC *phTpEnc ) -{ - HANDLE_TRANSPORTENC hTpEnc; - - if ( phTpEnc == NULL ){ - return TRANSPORTENC_INVALID_PARAMETER; - } - - hTpEnc = GetRam_TransportEncoder(0); - - if ( hTpEnc == NULL ) { - return TRANSPORTENC_NO_MEM; - } - - *phTpEnc = hTpEnc; - return TRANSPORTENC_OK; -} - -/** - * \brief Get frame period of PCE in raw_data_block. - * - * - Write PCE only if necessary. PCE can be part of the ASC if chConfig==0 whererfore - * no additonal PCE will be written in raw_data_block. - * - A matrixMixdown coefficient can only be written if chConfig is 5.0 or 5.1. - * - The PCE repetition rate in raw_data_block can be controlled via headerPeriod parameter. - * - * \param channelConfig Channel Configuration derived from Channel Mode - * \param transportFmt Format of the transport to be written. - * \param headerPeriod Chosen PCE frame repetition rate. - * \param matrixMixdownA Indicates if a valid Matrix Mixdown coefficient is available. - * - * \return PCE frame repetition rate. -1 means no PCE present in raw_data_block. - */ -static INT getPceRepetitionRate( - const int channelConfig, - const TRANSPORT_TYPE transportFmt, - const int headerPeriod, - const int matrixMixdownA - ) -{ - INT pceFrameCounter = -1; /* variable to be returned */ - - if (headerPeriod>0) { - switch ( channelConfig ) { - case 0: - switch (transportFmt) { - case TT_MP4_ADTS: - case TT_MP4_LATM_MCP0: - case TT_MP4_RAW: - pceFrameCounter = headerPeriod; - break; - case TT_MP4_ADIF: /* ADIF header comprises PCE */ - case TT_MP4_LOAS: /* PCE in ASC if chChonfig==0 */ - case TT_MP4_LATM_MCP1: /* PCE in ASC if chChonfig==0 */ - case TT_DRM: /* PCE not allowed in DRM */ - default: - pceFrameCounter = -1; /* no PCE in raw_data_block */ - } - break; - case 5: /* MODE_1_2_2 */ - case 6: /* MODE_1_2_2_1 */ - /* matrixMixdownCoefficient can only be written if 5.0 and 5.1 config present. */ - if (matrixMixdownA!=0) { - switch (transportFmt) { - case TT_MP4_ADIF: /* ADIF header comprises PCE */ - case TT_MP4_ADTS: - case TT_MP4_LOAS: /* no PCE in ASC because chConfig!=0 */ - case TT_MP4_LATM_MCP1: /* no PCE in ASC because chConfig!=0 */ - case TT_MP4_LATM_MCP0: - case TT_MP4_RAW: - pceFrameCounter = headerPeriod; - break; - case TT_DRM: /* PCE not allowed in DRM */ - default: - pceFrameCounter = -1; /* no PCE in raw_data_block */ - } /* switch transportFmt */ - } /* if matrixMixdownA!=0 */ - break; - default: - pceFrameCounter = -1; /* no PCE in raw_data_block */ - } /* switch getChannelConfig() */ - } /* if headerPeriod>0 */ - else { - pceFrameCounter = -1; /* no PCE in raw_data_block */ - } - - return pceFrameCounter; -} - -TRANSPORTENC_ERROR transportEnc_Init( - HANDLE_TRANSPORTENC hTpEnc, - UCHAR *bsBuffer, - INT bsBufferSize, - TRANSPORT_TYPE transportFmt, - CODER_CONFIG *cconfig, - UINT flags - ) -{ - /* Copy configuration structure */ - FDKmemcpy(&hTpEnc->config, cconfig, sizeof(CODER_CONFIG)); - - /* Init transportEnc struct. */ - hTpEnc->transportFmt = transportFmt; - - hTpEnc->bsBuffer = bsBuffer; - hTpEnc->bsBufferSize = bsBufferSize; - - FDKinitBitStream(&hTpEnc->bitStream, hTpEnc->bsBuffer, hTpEnc->bsBufferSize, 0, BS_WRITER); - - switch (transportFmt) { - - case TT_MP4_ADIF: - /* Sanity checks */ - if ( (hTpEnc->config.aot != AOT_AAC_LC) - ||(hTpEnc->config.samplesPerFrame != 1024)) - { - return TRANSPORTENC_INVALID_PARAMETER; - } - hTpEnc->writer.adif.headerWritten = 0; - hTpEnc->writer.adif.samplingRate = hTpEnc->config.samplingRate; - hTpEnc->writer.adif.bitRate = hTpEnc->config.bitRate; - hTpEnc->writer.adif.profile = ((int)hTpEnc->config.aot) - 1; - hTpEnc->writer.adif.cm = hTpEnc->config.channelMode; - hTpEnc->writer.adif.bVariableRate = 0; - hTpEnc->writer.adif.instanceTag = 0; - break; - - case TT_MP4_ADTS: - /* Sanity checks */ - if ( ( hTpEnc->config.aot != AOT_AAC_LC) - ||(hTpEnc->config.samplesPerFrame != 1024) ) - { - return TRANSPORTENC_INVALID_PARAMETER; - } - if ( adtsWrite_Init(&hTpEnc->writer.adts, &hTpEnc->config) != 0) { - return TRANSPORTENC_INVALID_PARAMETER; - } - break; - - case TT_MP4_LOAS: - case TT_MP4_LATM_MCP0: - case TT_MP4_LATM_MCP1: - { - TRANSPORTENC_ERROR error; - - error = transportEnc_Latm_Init( - &hTpEnc->writer.latm, - &hTpEnc->bitStream, - &hTpEnc->config, - flags & TP_FLAG_LATM_AMV, - transportFmt, - &hTpEnc->callbacks - ); - if (error != TRANSPORTENC_OK) { - return error; - } - } - break; - - case TT_MP4_RAW: - hTpEnc->writer.raw.curSubFrame = 0; - hTpEnc->writer.raw.nSubFrames = hTpEnc->config.nSubFrames; - break; - - - - default: - return TRANSPORTENC_INVALID_PARAMETER; - } - - /* pceFrameCounter indicates if PCE must be written in raw_data_block. */ - hTpEnc->pceFrameCounter = getPceRepetitionRate( - getChannelConfig(hTpEnc->config.channelMode), - transportFmt, - hTpEnc->config.headerPeriod, - hTpEnc->config.matrixMixdownA); - - return TRANSPORTENC_OK; -} - -HANDLE_FDK_BITSTREAM transportEnc_GetBitstream( HANDLE_TRANSPORTENC hTp ) -{ - return &hTp->bitStream; -} - -int transportEnc_RegisterSbrCallback( HANDLE_TRANSPORTENC hTpEnc, const cbSbr_t cbSbr, void* user_data) -{ - if (hTpEnc == NULL) { - return -1; - } - hTpEnc->callbacks.cbSbr = cbSbr; - hTpEnc->callbacks.cbSbrData = user_data; - return 0; -} - - -TRANSPORTENC_ERROR transportEnc_WriteAccessUnit( - HANDLE_TRANSPORTENC hTp, - INT frameUsedBits, - int bufferFullness, - int ncc - ) -{ - TRANSPORTENC_ERROR err = TRANSPORTENC_OK; - - if (!hTp) { - return TRANSPORTENC_INVALID_PARAMETER; - } - HANDLE_FDK_BITSTREAM hBs = &hTp->bitStream; - - /* In case of writing PCE in raw_data_block frameUsedBits must be adapted. */ - if (hTp->pceFrameCounter>=hTp->config.headerPeriod) { - frameUsedBits += transportEnc_GetPCEBits(hTp->config.channelMode, hTp->config.matrixMixdownA, 3); /* Consider 3 bits ID signalling in alignment */ - } - - switch (hTp->transportFmt) { - case TT_MP4_ADIF: - FDKinitBitStream(&hTp->bitStream, hTp->bsBuffer, hTp->bsBufferSize, 0, BS_WRITER); - adifWrite_EncodeHeader( - &hTp->writer.adif, - hBs, - bufferFullness - ); - break; - case TT_MP4_ADTS: - bufferFullness /= ncc; /* Number of Considered Channels */ - bufferFullness /= 32; - bufferFullness = FDKmin(0x7FF, bufferFullness); /* Signal variable rate */ - adtsWrite_EncodeHeader( - &hTp->writer.adts, - &hTp->bitStream, - bufferFullness, - frameUsedBits - ); - break; - case TT_MP4_LOAS: - case TT_MP4_LATM_MCP0: - case TT_MP4_LATM_MCP1: - bufferFullness /= ncc; /* Number of Considered Channels */ - bufferFullness /= 32; - bufferFullness = FDKmin(0xFF, bufferFullness); /* Signal variable rate */ - transportEnc_LatmWrite( - &hTp->writer.latm, - hBs, - frameUsedBits, - bufferFullness, - &hTp->callbacks - ); - break; - case TT_MP4_RAW: - if (hTp->writer.raw.curSubFrame >= hTp->writer.raw.nSubFrames) { - hTp->writer.raw.curSubFrame = 0; - FDKinitBitStream(&hTp->bitStream, hTp->bsBuffer, hTp->bsBufferSize, 0, BS_WRITER); - } - hTp->writer.raw.prevBits = FDKgetValidBits(hBs); - break; - default: - err = TRANSPORTENC_UNSUPPORTED_FORMAT; - break; - } - - /* Write PCE in raw_data_block if required */ - if (hTp->pceFrameCounter>=hTp->config.headerPeriod) { - INT crcIndex = 0; - /* Align inside PCE with repsect to the first bit of the raw_data_block() */ - UINT alignAnchor = FDKgetValidBits(&hTp->bitStream); - - /* Write PCE element ID bits */ - FDKwriteBits(&hTp->bitStream, ID_PCE, 3); - - if ( (hTp->transportFmt==TT_MP4_ADTS) && !hTp->writer.adts.protection_absent) { - crcIndex = adtsWrite_CrcStartReg(&hTp->writer.adts, &hTp->bitStream, 0); - } - - /* Write PCE as first raw_data_block element */ - transportEnc_writePCE(&hTp->bitStream, hTp->config.channelMode, hTp->config.samplingRate, 0, 1, hTp->config.matrixMixdownA, (hTp->config.flags&CC_PSEUDO_SURROUND)?1:0, alignAnchor); - - if ( (hTp->transportFmt==TT_MP4_ADTS) && !hTp->writer.adts.protection_absent) { - adtsWrite_CrcEndReg(&hTp->writer.adts, &hTp->bitStream, crcIndex); - } - hTp->pceFrameCounter = 0; /* reset pce frame counter */ - } - - if (hTp->pceFrameCounter!=-1) { - hTp->pceFrameCounter++; /* Update pceFrameCounter only if PCE writing is active. */ - } - - return err; -} - - -TRANSPORTENC_ERROR transportEnc_EndAccessUnit(HANDLE_TRANSPORTENC hTp, int *bits) -{ - switch (hTp->transportFmt) { - case TT_MP4_LATM_MCP0: - case TT_MP4_LATM_MCP1: - case TT_MP4_LOAS: - transportEnc_LatmAdjustSubframeBits(&hTp->writer.latm, bits); - break; - case TT_MP4_ADTS: - adtsWrite_EndRawDataBlock(&hTp->writer.adts, &hTp->bitStream, bits); - break; - case TT_MP4_ADIF: - /* Substract ADIF header from AU bits, not to be considered. */ - *bits -= adifWrite_GetHeaderBits(&hTp->writer.adif); - hTp->writer.adif.headerWritten = 1; - break; - case TT_MP4_RAW: - *bits -= hTp->writer.raw.prevBits; - break; - default: - break; - } - - return TRANSPORTENC_OK; -} - -TRANSPORTENC_ERROR transportEnc_GetFrame(HANDLE_TRANSPORTENC hTpEnc, int *nbytes) -{ - HANDLE_FDK_BITSTREAM hBs = &hTpEnc->bitStream; - - switch (hTpEnc->transportFmt) { - case TT_MP4_LATM_MCP0: - case TT_MP4_LATM_MCP1: - case TT_MP4_LOAS: - *nbytes = hTpEnc->bsBufferSize; - transportEnc_LatmGetFrame(&hTpEnc->writer.latm, hBs, nbytes); - break; - case TT_MP4_ADTS: - if (hTpEnc->writer.adts.currentBlock >= hTpEnc->writer.adts.num_raw_blocks+1) { - *nbytes = (FDKgetValidBits(hBs) + 7)>>3; - hTpEnc->writer.adts.currentBlock = 0; - } else { - *nbytes = 0; - } - break; - case TT_MP4_ADIF: - FDK_ASSERT((INT)FDKgetValidBits(hBs) >= 0); - *nbytes = (FDKgetValidBits(hBs) + 7)>>3; - break; - case TT_MP4_RAW: - FDKsyncCache(hBs); - hTpEnc->writer.raw.curSubFrame++; - *nbytes = ((FDKgetValidBits(hBs)-hTpEnc->writer.raw.prevBits) + 7)>>3; - break; - default: - break; - } - - return TRANSPORTENC_OK; -} - -INT transportEnc_GetStaticBits( HANDLE_TRANSPORTENC hTp, int auBits ) -{ - INT nbits = 0, nPceBits = 0; - - /* Write PCE within raw_data_block in transport lib. */ - if (hTp->pceFrameCounter>=hTp->config.headerPeriod) { - nPceBits = transportEnc_GetPCEBits(hTp->config.channelMode, hTp->config.matrixMixdownA, 3); /* Consider 3 bits ID signalling in alignment */ - auBits += nPceBits; /* Adapt required raw_data_block bit consumtpion for AU length information e.g. in LATM/LOAS configuration. */ - } - - switch (hTp->transportFmt) { - case TT_MP4_ADIF: - case TT_MP4_RAW: - nbits = 0; /* Do not consider the ADIF header into the total bitrate */ - break; - case TT_MP4_ADTS: - nbits = adtsWrite_GetHeaderBits(&hTp->writer.adts); - break; - case TT_MP4_LOAS: - case TT_MP4_LATM_MCP0: - case TT_MP4_LATM_MCP1: - nbits = transportEnc_LatmCountTotalBitDemandHeader( &hTp->writer.latm, auBits ); - break; - default: - nbits = 0; - break; - } - - /* PCE is written in the transport library therefore the bit consumption is part of the transport static bits. */ - nbits += nPceBits; - - return nbits; -} - -void transportEnc_Close(HANDLE_TRANSPORTENC *phTp) -{ - if (phTp != NULL) - { - if (*phTp != NULL) { - FreeRam_TransportEncoder(phTp); - } - } -} - -int transportEnc_CrcStartReg(HANDLE_TRANSPORTENC hTpEnc, int mBits) -{ - int crcReg = 0; - - switch (hTpEnc->transportFmt) { - case TT_MP4_ADTS: - crcReg = adtsWrite_CrcStartReg(&hTpEnc->writer.adts, &hTpEnc->bitStream, mBits); - break; - default: - break; - } - - return crcReg; -} - -void transportEnc_CrcEndReg(HANDLE_TRANSPORTENC hTpEnc, int reg) -{ - switch (hTpEnc->transportFmt) { - case TT_MP4_ADTS: - adtsWrite_CrcEndReg(&hTpEnc->writer.adts, &hTpEnc->bitStream, reg); - break; - default: - break; - } -} - - -TRANSPORTENC_ERROR transportEnc_GetConf(HANDLE_TRANSPORTENC hTpEnc, - CODER_CONFIG *cc, - FDK_BITSTREAM *dataBuffer, - UINT *confType) -{ - TRANSPORTENC_ERROR tpErr = TRANSPORTENC_OK; - HANDLE_LATM_STREAM hLatmConfig = &hTpEnc->writer.latm; - - *confType = 0; /* set confType variable to default */ - - /* write StreamMuxConfig or AudioSpecificConfig depending on format used */ - switch (hTpEnc->transportFmt) - { - case TT_MP4_LATM_MCP0: - case TT_MP4_LATM_MCP1: - case TT_MP4_LOAS: - tpErr = CreateStreamMuxConfig(hLatmConfig, dataBuffer, 0, &hTpEnc->callbacks); - *confType = 1; /* config is SMC */ - break; - default: - if (transportEnc_writeASC(dataBuffer, cc, &hTpEnc->callbacks) != 0) { - tpErr = TRANSPORTENC_UNKOWN_ERROR; - } - } - - return tpErr; - -} - -TRANSPORTENC_ERROR transportEnc_GetLibInfo( LIB_INFO *info ) -{ - int i; - - if (info == NULL) { - return TRANSPORTENC_INVALID_PARAMETER; - } - /* search for next free tab */ - for (i = 0; i < FDK_MODULE_LAST; i++) { - if (info[i].module_id == FDK_NONE) break; - } - if (i == FDK_MODULE_LAST) { - return TRANSPORTENC_UNKOWN_ERROR; - } - info += i; - - info->module_id = FDK_TPENC; - info->version = LIB_VERSION(TP_LIB_VL0, TP_LIB_VL1, TP_LIB_VL2); - LIB_VERSION_STRING(info); -#ifdef __ANDROID__ - info->build_date = ""; - info->build_time = ""; -#else - info->build_date = __DATE__; - info->build_time = __TIME__; -#endif - info->title = TP_LIB_TITLE; - - /* Set flags */ - info->flags = 0 - | CAPF_ADIF - | CAPF_ADTS - | CAPF_LATM - | CAPF_LOAS - | CAPF_RAWPACKETS - ; - - return TRANSPORTENC_OK; -} - diff --git a/libMpegTPEnc/src/version b/libMpegTPEnc/src/version deleted file mode 100644 index 8742568..0000000 --- a/libMpegTPEnc/src/version +++ /dev/null @@ -1,13 +0,0 @@ - -/* library info */ -#define TP_LIB_VL0 2 -#define TP_LIB_VL1 3 -#define TP_LIB_VL2 6 -#define TP_LIB_TITLE "MPEG Transport" -#ifdef __ANDROID__ -#define TP_LIB_BUILD_DATE "" -#define TP_LIB_BUILD_TIME "" -#else -#define TP_LIB_BUILD_DATE __DATE__ -#define TP_LIB_BUILD_TIME __TIME__ -#endif diff --git a/libSBRdec/include/sbrdecoder.h b/libSBRdec/include/sbrdecoder.h deleted file mode 100644 index 3bb9ba3..0000000 --- a/libSBRdec/include/sbrdecoder.h +++ /dev/null @@ -1,347 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/************************ Fraunhofer IIS SBR decoder library ****************** - - Author(s): - Description: SBR decoder front-end prototypes and definitions. - -******************************************************************************/ - -#ifndef __SBRDECODER_H -#define __SBRDECODER_H - -#include "common_fix.h" - -#include "FDK_bitstream.h" -#include "FDK_audio.h" - - -#define SBR_DEBUG_EXTHLP "\ ---- SBR ---\n\ - 0x00000010 Ancillary data and SBR-Header\n\ - 0x00000020 SBR-Side info\n\ - 0x00000040 Decoded SBR-bitstream data, e.g. envelope data\n\ - 0x00000080 SBR-Bitstream statistics\n\ - 0x00000100 Miscellaneous SBR-messages\n\ - 0x00000200 SBR-Energies and gains in the adjustor\n\ - 0x00000400 Fatal SBR errors\n\ - 0x00000800 Transposer coefficients for inverse filtering\n\ -" - -/* Capability flags */ -#define CAPF_SBR_LP 0x00000001 /*!< Flag indicating library's capability of Low Power mode. */ -#define CAPF_SBR_HQ 0x00000002 /*!< Flag indicating library's capability of High Quality mode. */ -#define CAPF_SBR_DRM_BS 0x00000004 /*!< Flag indicating library's capability to decode DRM SBR data. */ -#define CAPF_SBR_CONCEALMENT 0x00000008 /*!< Flag indicating library's capability to conceal erroneous frames. */ -#define CAPF_SBR_DRC 0x00000010 /*!< Flag indicating library's capability for Dynamic Range Control. */ -#define CAPF_SBR_PS_MPEG 0x00000020 /*!< Flag indicating library's capability to do MPEG Parametric Stereo. */ -#define CAPF_SBR_PS_DRM 0x00000040 /*!< Flag indicating library's capability to do DRM Parametric Stereo. */ - -typedef enum -{ - SBRDEC_OK = 0, /*!< All fine. */ - /* SBRDEC_CONCEAL, */ - /* SBRDEC_NOSYNCH, */ - /* SBRDEC_ILLEGAL_PROGRAM, */ - /* SBRDEC_ILLEGAL_TAG, */ - /* SBRDEC_ILLEGAL_CHN_CONFIG, */ - /* SBRDEC_ILLEGAL_SECTION, */ - /* SBRDEC_ILLEGAL_SCFACTORS, */ - /* SBRDEC_ILLEGAL_PULSE_DATA, */ - /* SBRDEC_MAIN_PROFILE_NOT_IMPLEMENTED, */ - /* SBRDEC_GC_NOT_IMPLEMENTED, */ - /* SBRDEC_ILLEGAL_PLUS_ELE_ID, */ - SBRDEC_CREATE_ERROR, /*!< */ - SBRDEC_NOT_INITIALIZED, /*!< */ - SBRDEC_MEM_ALLOC_FAILED, /*!< Memory allocation failed. Probably not enough memory available. */ - SBRDEC_PARSE_ERROR, /*!< */ - SBRDEC_UNSUPPORTED_CONFIG, /*!< */ - SBRDEC_SET_PARAM_FAIL /*!< */ -} SBR_ERROR; - -typedef enum -{ - SBR_SYSTEM_BITSTREAM_DELAY, /*!< System: Switch to enable an additional SBR bitstream delay of one frame. */ - SBR_QMF_MODE, /*!< Set QMF mode, either complex or low power. */ - SBR_LD_QMF_TIME_ALIGN, /*!< Set QMF type, either LD-MPS or CLDFB. Relevant for ELD streams only. */ - SBR_FLUSH_DATA, /*!< Set internal state to flush the decoder with the next process call. */ - SBR_CLEAR_HISTORY, /*!< Clear all internal states (delay lines, QMF states, ...). */ - SBR_BS_INTERRUPTION /*!< Signal bit stream interruption. Value is ignored. */ -} SBRDEC_PARAM; - -typedef struct SBR_DECODER_INSTANCE *HANDLE_SBRDECODER; - - -#ifdef __cplusplus -extern "C" -{ -#endif - - -/** - * \brief Allocates and initializes one SBR decoder instance. - * \param pSelf Pointer to where a SBR decoder handle is copied into. - * \return Error code. - */ -SBR_ERROR sbrDecoder_Open ( HANDLE_SBRDECODER *pSelf ); - -/** - * \brief Initialize a SBR decoder runtime instance. Must be called before decoding starts. - * - * \param self Handle to a SBR decoder instance. - * \param sampleRateIn Input samplerate of the SBR decoder instance. - * \param sampleRateOut Output samplerate of the SBR decoder instance. - * \param samplesPerFrame Number of samples per frames. - * \param coreCodec Audio Object Type (AOT) of the core codec. - * \param elementID Table with MPEG-4 element Ids in canonical order. - * \param forceReset Flag that enforces a complete decoder reset. - * - * \return Error code. - */ -SBR_ERROR sbrDecoder_InitElement ( - HANDLE_SBRDECODER self, - const int sampleRateIn, - const int sampleRateOut, - const int samplesPerFrame, - const AUDIO_OBJECT_TYPE coreCodec, - const MP4_ELEMENT_ID elementID, - const int elementIndex - ); - -/** - * \brief pass out of band SBR header to SBR decoder - * - * \param self Handle to a SBR decoder instance. - * \param hBs bit stream handle data source. - * \param elementID SBR element ID. - * \param elementIndex SBR element index. - * - * \return Error code. - */ -INT sbrDecoder_Header ( - HANDLE_SBRDECODER self, - HANDLE_FDK_BITSTREAM hBs, - const INT sampleRateIn, - const INT sampleRateOut, - const INT samplesPerFrame, - const AUDIO_OBJECT_TYPE coreCodec, - const MP4_ELEMENT_ID elementID, - const INT elementIndex - ); - -/** - * \brief Set a parameter of the SBR decoder runtime instance. - * \param self SBR decoder handle. - * \param param Parameter which will be set if successfull. - * \param value New parameter value. - * \return Error code. - */ -SBR_ERROR sbrDecoder_SetParam ( HANDLE_SBRDECODER self, - const SBRDEC_PARAM param, - const INT value ); - -/** - * \brief Feed DRC channel data into a SBR decoder runtime instance. - * - * \param self SBR decoder handle. - * \param ch Channel number to which the DRC data is associated to. - * \param numBands Number of DRC bands. - * \param pNextFact_mag Pointer to a table with the DRC factor magnitudes. - * \param nextFact_exp Exponent for all DRC factors. - * \param drcInterpolationScheme DRC interpolation scheme. - * \param winSequence Window sequence from core coder (eight short or one long window). - * \param pBandTop Pointer to a table with the top borders for all DRC bands. - * - * \return Error code. - */ -SBR_ERROR sbrDecoder_drcFeedChannel ( HANDLE_SBRDECODER self, - INT ch, - UINT numBands, - FIXP_DBL *pNextFact_mag, - INT nextFact_exp, - SHORT drcInterpolationScheme, - UCHAR winSequence, - USHORT *pBandTop ); - -/** - * \brief Disable SBR DRC for a certain channel. - * - * \param hSbrDecoder SBR decoder handle. - * \param ch Number of the channel that has to be disabled. - * - * \return None. - */ -void sbrDecoder_drcDisable ( HANDLE_SBRDECODER self, - INT ch ); - - -/** - * \brief Parse one SBR element data extension data block. The bit stream position will - * be placed at the end of the SBR payload block. The remaining bits will be returned - * into *count if a payload length is given (byPayLen > 0). If no SBR payload length is - * given (bsPayLen < 0) then the bit stream position on return will be random after this - * function call in case of errors, and any further decoding will be completely pointless. - * This function accepts either normal ordered SBR data or reverse ordered DRM SBR data. - * - * \param self SBR decoder handle. - * \param hBs Bit stream handle as data source. - * \param count Pointer to an integer where the amount of parsed SBR payload bits is stored into. - * \param bsPayLen If > 0 this value is the SBR payload length. If < 0, the SBR payload length is unknown. - * \param flags CRC flag (0: EXT_SBR_DATA; 1: EXT_SBR_DATA_CRC) - * \param prev_element Previous MPEG-4 element ID. - * \param element_index Index of the current element. - * - * \return Error code. - */ -SBR_ERROR sbrDecoder_Parse ( - HANDLE_SBRDECODER self, - HANDLE_FDK_BITSTREAM hBs, - int *count, - int bsPayLen, - int crcFlag, - MP4_ELEMENT_ID prev_element, - int element_index, - int fGlobalIndependencyFlag - ); - -/** - * \brief This function decodes the given SBR bitstreams and applies SBR to the given time data. - * - * SBR-processing works InPlace. I.e. the calling function has to provide - * a time domain buffer timeData which can hold the completely decoded - * result. - * - * Left and right channel are read and stored according to the - * interleaving flag, frame length and number of channels. - * - * \param self Handle of an open SBR decoder instance. - * \param hSbrBs SBR Bitstream handle. - * \param timeData Pointer to input and finally upsampled output data. - * \param numChannels Pointer to a buffer holding the number of channels in time data buffer. - * \param sampleRate Output samplerate. - * \param channelMapping Channel mapping indices. - * \param interleaved Flag indicating if time data is stored interleaved (1: Interleaved time data, 0: non-interleaved timedata). - * \param coreDecodedOk Flag indicating if the core decoder did not find any error (0: core decoder found errors, 1: no errors). - * \param psDecoded Pointer to a buffer holding a flag. Input: PS is possible, Output: PS has been rendered. - * - * \return Error code. - */ -SBR_ERROR sbrDecoder_Apply ( HANDLE_SBRDECODER self, - INT_PCM *timeData, - int *numChannels, - int *sampleRate, - const UCHAR channelMapping[(8)], - const int interleaved, - const int coreDecodedOk, - UCHAR *psDecoded ); - - -/** - * \brief Close SBR decoder instance and free memory. - * \param self SBR decoder handle. - * \return Error Code. - */ -SBR_ERROR sbrDecoder_Close ( HANDLE_SBRDECODER *self ); - - -/** - * \brief Get SBR decoder library information. - * \param info Pointer to a LIB_INFO struct, where library information is written to. - * \return 0 on success, -1 if invalid handle or if no free element is available to write information to. - */ -INT sbrDecoder_GetLibInfo( LIB_INFO *info ); - -/** - * \brief Determine the modules output signal delay in samples. - * \param self SBR decoder handle. - * \return The number of samples signal delay added by the module. - */ -UINT sbrDecoder_GetDelay( const HANDLE_SBRDECODER self ); - - -#ifdef __cplusplus -} -#endif - -#endif diff --git a/libSBRdec/src/arm/env_calc_arm.cpp b/libSBRdec/src/arm/env_calc_arm.cpp deleted file mode 100644 index 12b17d8..0000000 --- a/libSBRdec/src/arm/env_calc_arm.cpp +++ /dev/null @@ -1,148 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** Fraunhofer IIS *************************** - - Author(s): Arthur Tritthart - Description: (ARM optimised) SBR domain coding - -******************************************************************************/ -#ifndef INCLUSION_GUARD_CALC_ENV_ARM -#define INCLUSION_GUARD_CALC_ENV_ARM - - -/*! - \brief Compute maximal value of a complex array (re/im) of a given width - Negative values are temporarily logically or'ed with 0xFFFFFFFF - instead of negating the value, if the sign bit is set. - \param maxVal Preset maximal value - \param reTmp real input signal - \param imTmp imaginary input signal - \return new maximal value -*/ - -#ifdef FUNCTION_FDK_get_maxval -__asm FIXP_DBL FDK_get_maxval (FIXP_DBL maxVal, FIXP_DBL *reTmp, FIXP_DBL *imTmp, int width ) -{ - - /* Register map: - r0 maxVal - r1 reTmp - r2 imTmp - r3 width - r4 real - r5 imag - */ - PUSH {r4-r5} - - MOVS r3, r3, ASR #1 - ADC r3, r3, #0 - BCS FDK_get_maxval_loop_2nd_part - BEQ FDK_get_maxval_loop_end - -FDK_get_maxval_loop - LDR r4, [r1], #4 - LDR r5, [r2], #4 - EOR r4, r4, r4, ASR #31 - EOR r5, r5, r5, ASR #31 - ORR r0, r0, r4 - ORR r0, r0, r5 - -FDK_get_maxval_loop_2nd_part - LDR r4, [r1], #4 - LDR r5, [r2], #4 - EOR r4, r4, r4, ASR #31 - EOR r5, r5, r5, ASR #31 - ORR r0, r0, r4 - ORR r0, r0, r5 - - SUBS r3, r3, #1 - BNE FDK_get_maxval_loop - -FDK_get_maxval_loop_end - POP {r4-r5} - BX lr -} -#endif /* FUNCTION_FDK_get_maxval */ - -#endif /* INCLUSION_GUARD_CALC_ENV_ARM */ diff --git a/libSBRdec/src/arm/lpp_tran_arm.cpp b/libSBRdec/src/arm/lpp_tran_arm.cpp deleted file mode 100644 index 028a26f..0000000 --- a/libSBRdec/src/arm/lpp_tran_arm.cpp +++ /dev/null @@ -1,154 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/******************************** Fraunhofer IIS *************************** - - Author(s): Arthur Tritthart - Description: (ARM optimised) LPP transposer subroutines - -******************************************************************************/ - - -#if defined(__arm__) - - -#define FUNCTION_LPPTRANSPOSER_func1 - -#ifdef FUNCTION_LPPTRANSPOSER_func1 - -/* Note: This code requires only 43 cycles per iteration instead of 61 on ARM926EJ-S */ -#ifdef __GNUC__ -__attribute__ ((noinline)) -#endif -static void lppTransposer_func1( - FIXP_DBL *lowBandReal, - FIXP_DBL *lowBandImag, - FIXP_DBL **qmfBufferReal, - FIXP_DBL **qmfBufferImag, - int loops, - int hiBand, - int dynamicScale, - int descale, - FIXP_SGL a0r, - FIXP_SGL a0i, - FIXP_SGL a1r, - FIXP_SGL a1i) -{ - - FIXP_DBL real1, real2, imag1, imag2, accu1, accu2; - - real2 = lowBandReal[-2]; - real1 = lowBandReal[-1]; - imag2 = lowBandImag[-2]; - imag1 = lowBandImag[-1]; - for(int i=0; i < loops; i++) - { - accu1 = fMultDiv2( a0r,real1); - accu2 = fMultDiv2( a0i,imag1); - accu1 = fMultAddDiv2(accu1,a1r,real2); - accu2 = fMultAddDiv2(accu2,a1i,imag2); - real2 = fMultDiv2( a1i,real2); - accu1 = accu1 - accu2; - accu1 = accu1 >> dynamicScale; - - accu2 = fMultAddDiv2(real2,a1r,imag2); - real2 = real1; - imag2 = imag1; - accu2 = fMultAddDiv2(accu2,a0i,real1); - real1 = lowBandReal[i]; - accu2 = fMultAddDiv2(accu2,a0r,imag1); - imag1 = lowBandImag[i]; - accu2 = accu2 >> dynamicScale; - - accu1 <<= 1; - accu2 <<= 1; - - qmfBufferReal[i][hiBand] = accu1 + (real1>>descale); - qmfBufferImag[i][hiBand] = accu2 + (imag1>>descale); - } -} -#endif /* #ifdef FUNCTION_LPPTRANSPOSER_func1 */ -#endif /* __arm__ */ - - - diff --git a/libSBRdec/src/env_calc.cpp b/libSBRdec/src/env_calc.cpp deleted file mode 100644 index 73bd7ba..0000000 --- a/libSBRdec/src/env_calc.cpp +++ /dev/null @@ -1,2317 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief Envelope calculation - - The envelope adjustor compares the energies present in the transposed - highband to the reference energies conveyed with the bitstream. - The highband is amplified (sometimes) or attenuated (mostly) to the - desired level. - - The spectral shape of the reference energies can be changed several times per - frame if necessary. Each set of energy values corresponding to a certain range - in time will be called an envelope here. - The bitstream supports several frequency scales and two resolutions. Normally, - one or more QMF-subbands are grouped to one SBR-band. An envelope contains - reference energies for each SBR-band. - In addition to the energy envelopes, noise envelopes are transmitted that - define the ratio of energy which is generated by adding noise instead of - transposing the lowband. The noise envelopes are given in a coarser time - and frequency resolution. - If a signal contains strong tonal components, synthetic sines can be - generated in individual SBR bands. - - An overlap buffer of 6 QMF-timeslots is used to allow a more - flexible alignment of the envelopes in time that is not restricted to the - core codec's frame borders. - Therefore the envelope adjustor has access to the spectral data of the - current frame as well as the last 6 QMF-timeslots of the previous frame. - However, in average only the data of 1 frame is being processed as - the adjustor is called once per frame. - - Depending on the frequency range set in the bitstream, only QMF-subbands between - lowSubband and highSubband are adjusted. - - Scaling of spectral data to maximize SNR (see #QMF_SCALE_FACTOR) as well as a special Mantissa-Exponent format - ( see calculateSbrEnvelope() ) are being used. The main entry point for this modules is calculateSbrEnvelope(). - - \sa sbr_scale.h, #QMF_SCALE_FACTOR, calculateSbrEnvelope(), \ref documentationOverview -*/ - - -#include "env_calc.h" - -#include "sbrdec_freq_sca.h" -#include "env_extr.h" -#include "transcendent.h" -#include "sbr_ram.h" -#include "sbr_rom.h" - -#include "genericStds.h" /* need FDKpow() for debug outputs */ - -#if defined(__arm__) -#include "arm/env_calc_arm.cpp" -#endif - -typedef struct -{ - FIXP_DBL nrgRef[MAX_FREQ_COEFFS]; - FIXP_DBL nrgEst[MAX_FREQ_COEFFS]; - FIXP_DBL nrgGain[MAX_FREQ_COEFFS]; - FIXP_DBL noiseLevel[MAX_FREQ_COEFFS]; - FIXP_DBL nrgSine[MAX_FREQ_COEFFS]; - - SCHAR nrgRef_e[MAX_FREQ_COEFFS]; - SCHAR nrgEst_e[MAX_FREQ_COEFFS]; - SCHAR nrgGain_e[MAX_FREQ_COEFFS]; - SCHAR noiseLevel_e[MAX_FREQ_COEFFS]; - SCHAR nrgSine_e[MAX_FREQ_COEFFS]; -} -ENV_CALC_NRGS; - -static void equalizeFiltBufferExp(FIXP_DBL *filtBuffer, - SCHAR *filtBuffer_e, - FIXP_DBL *NrgGain, - SCHAR *NrgGain_e, - int subbands); - -static void calcNrgPerSubband(FIXP_DBL **analysBufferReal, - FIXP_DBL **analysBufferImag, - int lowSubband, int highSubband, - int start_pos, int next_pos, - SCHAR frameExp, - FIXP_DBL *nrgEst, - SCHAR *nrgEst_e ); - -static void calcNrgPerSfb(FIXP_DBL **analysBufferReal, - FIXP_DBL **analysBufferImag, - int nSfb, - UCHAR *freqBandTable, - int start_pos, int next_pos, - SCHAR input_e, - FIXP_DBL *nrg_est, - SCHAR *nrg_est_e ); - -static void calcSubbandGain(FIXP_DBL nrgRef, SCHAR nrgRef_e, ENV_CALC_NRGS* nrgs, int c, - FIXP_DBL tmpNoise, SCHAR tmpNoise_e, - UCHAR sinePresentFlag, - UCHAR sineMapped, - int noNoiseFlag); - -static void calcAvgGain(ENV_CALC_NRGS* nrgs, - int lowSubband, - int highSubband, - FIXP_DBL *sumRef_m, - SCHAR *sumRef_e, - FIXP_DBL *ptrAvgGain_m, - SCHAR *ptrAvgGain_e); - -static void adjustTimeSlot_EldGrid(FIXP_DBL *ptrReal, - ENV_CALC_NRGS* nrgs, - UCHAR *ptrHarmIndex, - int lowSubbands, - int noSubbands, - int scale_change, - int noNoiseFlag, - int *ptrPhaseIndex, - int scale_diff_low); - -static void adjustTimeSlotLC(FIXP_DBL *ptrReal, - ENV_CALC_NRGS* nrgs, - UCHAR *ptrHarmIndex, - int lowSubbands, - int noSubbands, - int scale_change, - int noNoiseFlag, - int *ptrPhaseIndex); -static void adjustTimeSlotHQ(FIXP_DBL *ptrReal, - FIXP_DBL *ptrImag, - HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env, - ENV_CALC_NRGS* nrgs, - int lowSubbands, - int noSubbands, - int scale_change, - FIXP_SGL smooth_ratio, - int noNoiseFlag, - int filtBufferNoiseShift); - - -/*! - \brief Map sine flags from bitstream to QMF bands - - The bitstream carries only 1 sine flag per band and frame. - This function maps every sine flag from the bitstream to a specific QMF subband - and to a specific envelope where the sine shall start. - The result is stored in the vector sineMapped which contains one entry per - QMF subband. The value of an entry specifies the envelope where a sine - shall start. A value of #MAX_ENVELOPES indicates that no sine is present - in the subband. - The missing harmonics flags from the previous frame (harmFlagsPrev) determine - if a sine starts at the beginning of the frame or at the transient position. - Additionally, the flags in harmFlagsPrev are being updated by this function - for the next frame. -*/ -static void mapSineFlags(UCHAR *freqBandTable, /*!< Band borders (there's only 1 flag per band) */ - int nSfb, /*!< Number of bands in the table */ - UCHAR *addHarmonics, /*!< vector with 1 flag per sfb */ - int *harmFlagsPrev, /*!< Packed 'addHarmonics' */ - int tranEnv, /*!< Transient position */ - SCHAR *sineMapped) /*!< Resulting vector of sine start positions for each QMF band */ - -{ - int i; - int lowSubband2 = freqBandTable[0]<<1; - int bitcount = 0; - int oldflags = *harmFlagsPrev; - int newflags = 0; - - /* - Format of harmFlagsPrev: - - first word = flags for highest 16 sfb bands in use - second word = flags for next lower 16 sfb bands (if present) - third word = flags for lowest 16 sfb bands (if present) - - Up to MAX_FREQ_COEFFS sfb bands can be flagged for a sign. - The lowest bit of the first word corresponds to the _highest_ sfb band in use. - This is ensures that each flag is mapped to the same QMF band even after a - change of the crossover-frequency. - */ - - - /* Reset the output vector first */ - FDKmemset(sineMapped, MAX_ENVELOPES,MAX_FREQ_COEFFS); /* MAX_ENVELOPES means 'no sine' */ - - freqBandTable += nSfb; - addHarmonics += nSfb-1; - - for (i=nSfb; i!=0; i--) { - int ui = *freqBandTable--; /* Upper limit of the current scale factor band. */ - int li = *freqBandTable; /* Lower limit of the current scale factor band. */ - - if ( *addHarmonics-- ) { /* There is a sine in this band */ - - unsigned int mask = 1 << bitcount; - newflags |= mask; /* Set flag */ - - /* - If there was a sine in the last frame, let it continue from the first envelope on - else start at the transient position. - */ - sineMapped[(ui+li-lowSubband2) >> 1] = ( oldflags & mask ) ? 0 : tranEnv; - } - - if ((++bitcount == 16) || i==1) { - bitcount = 0; - *harmFlagsPrev++ = newflags; - oldflags = *harmFlagsPrev; /* Fetch 16 of the old flags */ - newflags = 0; - } - } -} - - -/*! - \brief Reduce gain-adjustment induced aliasing for real valued filterbank. -*/ -/*static*/ void -aliasingReduction(FIXP_DBL* degreeAlias, /*!< estimated aliasing for each QMF channel */ - ENV_CALC_NRGS* nrgs, - int* useAliasReduction, /*!< synthetic sine engergy for each subband, used as flag */ - int noSubbands) /*!< number of QMF channels to process */ -{ - FIXP_DBL* nrgGain = nrgs->nrgGain; /*!< subband gains to be modified */ - SCHAR* nrgGain_e = nrgs->nrgGain_e; /*!< subband gains to be modified (exponents) */ - FIXP_DBL* nrgEst = nrgs->nrgEst; /*!< subband energy before amplification */ - SCHAR* nrgEst_e = nrgs->nrgEst_e; /*!< subband energy before amplification (exponents) */ - int grouping = 0, index = 0, noGroups, k; - int groupVector[MAX_FREQ_COEFFS]; - - /* Calculate grouping*/ - for (k = 0; k < noSubbands-1; k++ ){ - if ( (degreeAlias[k + 1] != FL2FXCONST_DBL(0.0f)) && useAliasReduction[k] ) { - if(grouping==0){ - groupVector[index++] = k; - grouping = 1; - } - else{ - if(groupVector[index-1] + 3 == k){ - groupVector[index++] = k + 1; - grouping = 0; - } - } - } - else{ - if(grouping){ - if(useAliasReduction[k]) - groupVector[index++] = k + 1; - else - groupVector[index++] = k; - grouping = 0; - } - } - } - - if(grouping){ - groupVector[index++] = noSubbands; - } - noGroups = index >> 1; - - - /*Calculate new gain*/ - for (int group = 0; group < noGroups; group ++) { - FIXP_DBL nrgOrig = FL2FXCONST_DBL(0.0f); /* Original signal energy in current group of bands */ - SCHAR nrgOrig_e = 0; - FIXP_DBL nrgAmp = FL2FXCONST_DBL(0.0f); /* Amplified signal energy in group (using current gains) */ - SCHAR nrgAmp_e = 0; - FIXP_DBL nrgMod = FL2FXCONST_DBL(0.0f); /* Signal energy in group when applying modified gains */ - SCHAR nrgMod_e = 0; - FIXP_DBL groupGain; /* Total energy gain in group */ - SCHAR groupGain_e; - FIXP_DBL compensation; /* Compensation factor for the energy change when applying modified gains */ - SCHAR compensation_e; - - int startGroup = groupVector[2*group]; - int stopGroup = groupVector[2*group+1]; - - /* Calculate total energy in group before and after amplification with current gains: */ - for(k = startGroup; k < stopGroup; k++){ - /* Get original band energy */ - FIXP_DBL tmp = nrgEst[k]; - SCHAR tmp_e = nrgEst_e[k]; - - FDK_add_MantExp(tmp, tmp_e, nrgOrig, nrgOrig_e, &nrgOrig, &nrgOrig_e); - - /* Multiply band energy with current gain */ - tmp = fMult(tmp,nrgGain[k]); - tmp_e = tmp_e + nrgGain_e[k]; - - FDK_add_MantExp(tmp, tmp_e, nrgAmp, nrgAmp_e, &nrgAmp, &nrgAmp_e); - } - - /* Calculate total energy gain in group */ - FDK_divide_MantExp(nrgAmp, nrgAmp_e, - nrgOrig, nrgOrig_e, - &groupGain, &groupGain_e); - - for(k = startGroup; k < stopGroup; k++){ - FIXP_DBL tmp; - SCHAR tmp_e; - - FIXP_DBL alpha = degreeAlias[k]; - if (k < noSubbands - 1) { - if (degreeAlias[k + 1] > alpha) - alpha = degreeAlias[k + 1]; - } - - /* Modify gain depending on the degree of aliasing */ - FDK_add_MantExp( fMult(alpha,groupGain), groupGain_e, - fMult(/*FL2FXCONST_DBL(1.0f)*/ (FIXP_DBL)MAXVAL_DBL - alpha,nrgGain[k]), nrgGain_e[k], - &nrgGain[k], &nrgGain_e[k] ); - - /* Apply modified gain to original energy */ - tmp = fMult(nrgGain[k],nrgEst[k]); - tmp_e = nrgGain_e[k] + nrgEst_e[k]; - - /* Accumulate energy with modified gains applied */ - FDK_add_MantExp( tmp, tmp_e, - nrgMod, nrgMod_e, - &nrgMod, &nrgMod_e ); - } - - /* Calculate compensation factor to retain the energy of the amplified signal */ - FDK_divide_MantExp(nrgAmp, nrgAmp_e, - nrgMod, nrgMod_e, - &compensation, &compensation_e); - - /* Apply compensation factor to all gains of the group */ - for(k = startGroup; k < stopGroup; k++){ - nrgGain[k] = fMult(nrgGain[k],compensation); - nrgGain_e[k] = nrgGain_e[k] + compensation_e; - } - } -} - - - /* Convert headroom bits to exponent */ -#define SCALE2EXP(s) (15-(s)) -#define EXP2SCALE(e) (15-(e)) - -/*! - \brief Apply spectral envelope to subband samples - - This function is called from sbr_dec.cpp in each frame. - - To enhance accuracy and due to the usage of tables for squareroots and - inverse, some calculations are performed with the operands being split - into mantissa and exponent. The variable names in the source code carry - the suffixes _m and _e respectively. The control data - in #hFrameData containts envelope data which is represented by this format but - stored in single words. (See requantizeEnvelopeData() for details). This data - is unpacked within calculateSbrEnvelope() to follow the described suffix convention. - - The actual value (comparable to the corresponding float-variable in the - research-implementation) of a mantissa/exponent-pair can be calculated as - - \f$ value = value\_m * 2^{value\_e} \f$ - - All energies and noise levels decoded from the bitstream suit for an - original signal magnitude of \f$\pm 32768 \f$ rather than \f$ \pm 1\f$. Therefore, - the scale factor hb_scale passed into this function will be converted - to an 'input exponent' (#input_e), which fits the internal representation. - - Before the actual processing, an exponent #adj_e for resulting adjusted - samples is derived from the maximum reference energy. - - Then, for each envelope, the following steps are performed: - - \li Calculate energy in the signal to be adjusted. Depending on the the value of - #interpolFreq (interpolation mode), this is either done seperately - for each QMF-subband or for each SBR-band. - The resulting energies are stored in #nrgEst_m[#MAX_FREQ_COEFFS] (mantissas) - and #nrgEst_e[#MAX_FREQ_COEFFS] (exponents). - \li Calculate gain and noise level for each subband:
- \f$ gain = \sqrt{ \frac{nrgRef}{nrgEst} \cdot (1 - noiseRatio) } - \hspace{2cm} - noise = \sqrt{ nrgRef \cdot noiseRatio } - \f$
- where noiseRatio and nrgRef are extracted from the - bitstream and nrgEst is the subband energy before adjustment. - The resulting gains are stored in #nrgGain_m[#MAX_FREQ_COEFFS] - (mantissas) and #nrgGain_e[#MAX_FREQ_COEFFS] (exponents), the noise levels - are stored in #noiseLevel_m[#MAX_FREQ_COEFFS] and #noiseLevel_e[#MAX_FREQ_COEFFS] - (exponents). - The sine levels are stored in #nrgSine_m[#MAX_FREQ_COEFFS] - and #nrgSine_e[#MAX_FREQ_COEFFS]. - \li Noise limiting: The gain for each subband is limited both absolutely - and relatively compared to the total gain over all subbands. - \li Boost gain: Calculate and apply boost factor for each limiter band - in order to compensate for the energy loss imposed by the limiting. - \li Apply gains and add noise: The gains and noise levels are applied - to all timeslots of the current envelope. A short FIR-filter (length 4 - QMF-timeslots) can be used to smooth the sudden change at the envelope borders. - Each complex subband sample of the current timeslot is multiplied by the - smoothed gain, then random noise with the calculated level is added. - - \note - To reduce the stack size, some of the local arrays could be located within - the time output buffer. Of the 512 samples temporarily available there, - about half the size is already used by #SBR_FRAME_DATA. A pointer to the - remaining free memory could be supplied by an additional argument to calculateSbrEnvelope() - in sbr_dec: - - \par - \code - calculateSbrEnvelope (&hSbrDec->sbrScaleFactor, - &hSbrDec->SbrCalculateEnvelope, - hHeaderData, - hFrameData, - QmfBufferReal, - QmfBufferImag, - timeOutPtr + sizeof(SBR_FRAME_DATA)/sizeof(Float) + 1); - \endcode - - \par - Within calculateSbrEnvelope(), some pointers could be defined instead of the arrays - #nrgRef_m, #nrgRef_e, #nrgEst_m, #nrgEst_e, #noiseLevel_m: - - \par - \code - fract* nrgRef_m = timeOutPtr; - SCHAR* nrgRef_e = nrgRef_m + MAX_FREQ_COEFFS; - fract* nrgEst_m = nrgRef_e + MAX_FREQ_COEFFS; - SCHAR* nrgEst_e = nrgEst_m + MAX_FREQ_COEFFS; - fract* noiseLevel_m = nrgEst_e + MAX_FREQ_COEFFS; - \endcode - -
-*/ -void -calculateSbrEnvelope (QMF_SCALE_FACTOR *sbrScaleFactor, /*!< Scaling factors */ - HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env, /*!< Handle to struct filled by the create-function */ - HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ - HANDLE_SBR_FRAME_DATA hFrameData, /*!< Control data of current frame */ - FIXP_DBL **analysBufferReal, /*!< Real part of subband samples to be processed */ - FIXP_DBL **analysBufferImag, /*!< Imag part of subband samples to be processed */ - const int useLP, - FIXP_DBL *degreeAlias, /*!< Estimated aliasing for each QMF channel */ - const UINT flags, - const int frameErrorFlag - ) -{ - int c, i, j, envNoise = 0; - UCHAR* borders = hFrameData->frameInfo.borders; - - FIXP_SGL *noiseLevels = hFrameData->sbrNoiseFloorLevel; - HANDLE_FREQ_BAND_DATA hFreq = &hHeaderData->freqBandData; - - int lowSubband = hFreq->lowSubband; - int highSubband = hFreq->highSubband; - int noSubbands = highSubband - lowSubband; - - int noNoiseBands = hFreq->nNfb; - int no_cols = hHeaderData->numberTimeSlots * hHeaderData->timeStep; - UCHAR first_start = borders[0] * hHeaderData->timeStep; - - SCHAR sineMapped[MAX_FREQ_COEFFS]; - SCHAR ov_adj_e = SCALE2EXP(sbrScaleFactor->ov_hb_scale); - SCHAR adj_e = 0; - SCHAR output_e; - SCHAR final_e = 0; - - SCHAR maxGainLimit_e = (frameErrorFlag) ? MAX_GAIN_CONCEAL_EXP : MAX_GAIN_EXP; - - int useAliasReduction[64]; - UCHAR smooth_length = 0; - - FIXP_SGL * pIenv = hFrameData->iEnvelope; - - /* - Extract sine flags for all QMF bands - */ - mapSineFlags(hFreq->freqBandTable[1], - hFreq->nSfb[1], - hFrameData->addHarmonics, - h_sbr_cal_env->harmFlagsPrev, - hFrameData->frameInfo.tranEnv, - sineMapped); - - - /* - Scan for maximum in bufferd noise levels. - This is needed in case that we had strong noise in the previous frame - which is smoothed into the current frame. - The resulting exponent is used as start value for the maximum search - in reference energies - */ - if (!useLP) - adj_e = h_sbr_cal_env->filtBufferNoise_e - getScalefactor(h_sbr_cal_env->filtBufferNoise, noSubbands); - - /* - Scan for maximum reference energy to be able - to select appropriate values for adj_e and final_e. - */ - - for (i = 0; i < hFrameData->frameInfo.nEnvelopes; i++) { - INT maxSfbNrg_e = -FRACT_BITS+NRG_EXP_OFFSET; /* start value for maximum search */ - - /* Fetch frequency resolution for current envelope: */ - for (j=hFreq->nSfb[hFrameData->frameInfo.freqRes[i]]; j!=0; j--) { - maxSfbNrg_e = fixMax(maxSfbNrg_e,(INT)((LONG)(*pIenv++) & MASK_E)); - } - maxSfbNrg_e -= NRG_EXP_OFFSET; - - /* Energy -> magnitude (sqrt halfens exponent) */ - maxSfbNrg_e = (maxSfbNrg_e+1) >> 1; /* +1 to go safe (round to next higher int) */ - - /* Some safety margin is needed for 2 reasons: - - The signal energy is not equally spread over all subband samples in - a specific sfb of an envelope (Nrg could be too high by a factor of - envWidth * sfbWidth) - - Smoothing can smear high gains of the previous envelope into the current - */ - maxSfbNrg_e += 6; - - if (borders[i] < hHeaderData->numberTimeSlots) - /* This envelope affects timeslots that belong to the output frame */ - adj_e = (maxSfbNrg_e > adj_e) ? maxSfbNrg_e : adj_e; - - if (borders[i+1] > hHeaderData->numberTimeSlots) - /* This envelope affects timeslots after the output frame */ - final_e = (maxSfbNrg_e > final_e) ? maxSfbNrg_e : final_e; - - } - - /* - Calculate adjustment factors and apply them for every envelope. - */ - pIenv = hFrameData->iEnvelope; - - for (i = 0; i < hFrameData->frameInfo.nEnvelopes; i++) { - - int k, noNoiseFlag; - SCHAR noise_e, input_e = SCALE2EXP(sbrScaleFactor->hb_scale); - C_ALLOC_SCRATCH_START(pNrgs, ENV_CALC_NRGS, 1); - - /* - Helper variables. - */ - UCHAR start_pos = hHeaderData->timeStep * borders[i]; /* Start-position in time (subband sample) for current envelope. */ - UCHAR stop_pos = hHeaderData->timeStep * borders[i+1]; /* Stop-position in time (subband sample) for current envelope. */ - UCHAR freq_res = hFrameData->frameInfo.freqRes[i]; /* Frequency resolution for current envelope. */ - - - /* Always do fully initialize the temporary energy table. This prevents negative energies and extreme gain factors in - cases where the number of limiter bands exceeds the number of subbands. The latter can be caused by undetected bit - errors and is tested by some streams from the certification set. */ - FDKmemclear(pNrgs, sizeof(ENV_CALC_NRGS)); - - /* If the start-pos of the current envelope equals the stop pos of the current - noise envelope, increase the pointer (i.e. choose the next noise-floor).*/ - if (borders[i] == hFrameData->frameInfo.bordersNoise[envNoise+1]){ - noiseLevels += noNoiseBands; /* The noise floor data is stored in a row [noiseFloor1 noiseFloor2...].*/ - envNoise++; - } - - if(i==hFrameData->frameInfo.tranEnv || i==h_sbr_cal_env->prevTranEnv) /* attack */ - { - noNoiseFlag = 1; - if (!useLP) - smooth_length = 0; /* No smoothing on attacks! */ - } - else { - noNoiseFlag = 0; - if (!useLP) - smooth_length = (1 - hHeaderData->bs_data.smoothingLength) << 2; /* can become either 0 or 4 */ - } - - - /* - Energy estimation in transposed highband. - */ - if (hHeaderData->bs_data.interpolFreq) - calcNrgPerSubband(analysBufferReal, - (useLP) ? NULL : analysBufferImag, - lowSubband, highSubband, - start_pos, stop_pos, - input_e, - pNrgs->nrgEst, - pNrgs->nrgEst_e); - else - calcNrgPerSfb(analysBufferReal, - (useLP) ? NULL : analysBufferImag, - hFreq->nSfb[freq_res], - hFreq->freqBandTable[freq_res], - start_pos, stop_pos, - input_e, - pNrgs->nrgEst, - pNrgs->nrgEst_e); - - /* - Calculate subband gains - */ - { - UCHAR * table = hFreq->freqBandTable[freq_res]; - UCHAR * pUiNoise = &hFreq->freqBandTableNoise[1]; /*! Upper limit of the current noise floor band. */ - - FIXP_SGL * pNoiseLevels = noiseLevels; - - FIXP_DBL tmpNoise = FX_SGL2FX_DBL((FIXP_SGL)((LONG)(*pNoiseLevels) & MASK_M)); - SCHAR tmpNoise_e = (UCHAR)((LONG)(*pNoiseLevels++) & MASK_E) - NOISE_EXP_OFFSET; - - int cc = 0; - c = 0; - for (j = 0; j < hFreq->nSfb[freq_res]; j++) { - - FIXP_DBL refNrg = FX_SGL2FX_DBL((FIXP_SGL)((LONG)(*pIenv) & MASK_M)); - SCHAR refNrg_e = (SCHAR)((LONG)(*pIenv) & MASK_E) - NRG_EXP_OFFSET; - - UCHAR sinePresentFlag = 0; - int li = table[j]; - int ui = table[j+1]; - - for (k=li; k= sineMapped[cc]); - cc++; - } - - for (k=li; k= *pUiNoise) { - tmpNoise = FX_SGL2FX_DBL((FIXP_SGL)((LONG)(*pNoiseLevels) & MASK_M)); - tmpNoise_e = (SCHAR)((LONG)(*pNoiseLevels++) & MASK_E) - NOISE_EXP_OFFSET; - - pUiNoise++; - } - - FDK_ASSERT(k >= lowSubband); - - if (useLP) - useAliasReduction[k-lowSubband] = !sinePresentFlag; - - pNrgs->nrgSine[c] = FL2FXCONST_DBL(0.0f); - pNrgs->nrgSine_e[c] = 0; - - calcSubbandGain(refNrg, refNrg_e, pNrgs, c, - tmpNoise, tmpNoise_e, - sinePresentFlag, i >= sineMapped[c], - noNoiseFlag); - - pNrgs->nrgRef[c] = refNrg; - pNrgs->nrgRef_e[c] = refNrg_e; - - c++; - } - pIenv++; - } - } - - /* - Noise limiting - */ - - for (c = 0; c < hFreq->noLimiterBands; c++) { - - FIXP_DBL sumRef, boostGain, maxGain; - FIXP_DBL accu = FL2FXCONST_DBL(0.0f); - SCHAR sumRef_e, boostGain_e, maxGain_e, accu_e = 0; - - calcAvgGain(pNrgs, - hFreq->limiterBandTable[c], hFreq->limiterBandTable[c+1], - &sumRef, &sumRef_e, - &maxGain, &maxGain_e); - - /* Multiply maxGain with limiterGain: */ - maxGain = fMult(maxGain, FDK_sbrDecoder_sbr_limGains_m[hHeaderData->bs_data.limiterGains]); - maxGain_e += FDK_sbrDecoder_sbr_limGains_e[hHeaderData->bs_data.limiterGains]; - - /* Scale mantissa of MaxGain into range between 0.5 and 1: */ - if (maxGain == FL2FXCONST_DBL(0.0f)) - maxGain_e = -FRACT_BITS; - else { - SCHAR charTemp = CountLeadingBits(maxGain); - maxGain_e -= charTemp; - maxGain <<= (int)charTemp; - } - - if (maxGain_e >= maxGainLimit_e) { /* upper limit (e.g. 96 dB) */ - maxGain = FL2FXCONST_DBL(0.5f); - maxGain_e = maxGainLimit_e; - } - - - /* Every subband gain is compared to the scaled "average gain" - and limited if necessary: */ - for (k = hFreq->limiterBandTable[c]; k < hFreq->limiterBandTable[c+1]; k++) { - if ( (pNrgs->nrgGain_e[k] > maxGain_e) || (pNrgs->nrgGain_e[k] == maxGain_e && pNrgs->nrgGain[k]>maxGain) ) { - - FIXP_DBL noiseAmp; - SCHAR noiseAmp_e; - - FDK_divide_MantExp(maxGain, maxGain_e, pNrgs->nrgGain[k], pNrgs->nrgGain_e[k], &noiseAmp, &noiseAmp_e); - pNrgs->noiseLevel[k] = fMult(pNrgs->noiseLevel[k],noiseAmp); - pNrgs->noiseLevel_e[k] += noiseAmp_e; - pNrgs->nrgGain[k] = maxGain; - pNrgs->nrgGain_e[k] = maxGain_e; - } - } - - /* -- Boost gain - Calculate and apply boost factor for each limiter band: - 1. Check how much energy would be present when using the limited gain - 2. Calculate boost factor by comparison with reference energy - 3. Apply boost factor to compensate for the energy loss due to limiting - */ - for (k = hFreq->limiterBandTable[c]; k < hFreq->limiterBandTable[c + 1]; k++) { - - /* 1.a Add energy of adjusted signal (using preliminary gain) */ - FIXP_DBL tmp = fMult(pNrgs->nrgGain[k],pNrgs->nrgEst[k]); - SCHAR tmp_e = pNrgs->nrgGain_e[k] + pNrgs->nrgEst_e[k]; - FDK_add_MantExp(tmp, tmp_e, accu, accu_e, &accu, &accu_e); - - /* 1.b Add sine energy (if present) */ - if(pNrgs->nrgSine[k] != FL2FXCONST_DBL(0.0f)) { - FDK_add_MantExp(pNrgs->nrgSine[k], pNrgs->nrgSine_e[k], accu, accu_e, &accu, &accu_e); - } - else { - /* 1.c Add noise energy (if present) */ - if(noNoiseFlag == 0) { - FDK_add_MantExp(pNrgs->noiseLevel[k], pNrgs->noiseLevel_e[k], accu, accu_e, &accu, &accu_e); - } - } - } - - /* 2.a Calculate ratio of wanted energy and accumulated energy */ - if (accu == (FIXP_DBL)0) { /* If divisor is 0, limit quotient to +4 dB */ - boostGain = FL2FXCONST_DBL(0.6279716f); - boostGain_e = 2; - } else { - INT div_e; - boostGain = fDivNorm(sumRef, accu, &div_e); - boostGain_e = sumRef_e - accu_e + div_e; - } - - - /* 2.b Result too high? --> Limit the boost factor to +4 dB */ - if((boostGain_e > 3) || - (boostGain_e == 2 && boostGain > FL2FXCONST_DBL(0.6279716f)) || - (boostGain_e == 3 && boostGain > FL2FXCONST_DBL(0.3139858f)) ) - { - boostGain = FL2FXCONST_DBL(0.6279716f); - boostGain_e = 2; - } - /* 3. Multiply all signal components with the boost factor */ - for (k = hFreq->limiterBandTable[c]; k < hFreq->limiterBandTable[c + 1]; k++) { - pNrgs->nrgGain[k] = fMultDiv2(pNrgs->nrgGain[k],boostGain); - pNrgs->nrgGain_e[k] = pNrgs->nrgGain_e[k] + boostGain_e + 1; - - pNrgs->nrgSine[k] = fMultDiv2(pNrgs->nrgSine[k],boostGain); - pNrgs->nrgSine_e[k] = pNrgs->nrgSine_e[k] + boostGain_e + 1; - - pNrgs->noiseLevel[k] = fMultDiv2(pNrgs->noiseLevel[k],boostGain); - pNrgs->noiseLevel_e[k] = pNrgs->noiseLevel_e[k] + boostGain_e + 1; - } - } - /* End of noise limiting */ - - if (useLP) - aliasingReduction(degreeAlias+lowSubband, - pNrgs, - useAliasReduction, - noSubbands); - - /* For the timeslots within the range for the output frame, - use the same scale for the noise levels. - Drawback: If the envelope exceeds the frame border, the noise levels - will have to be rescaled later to fit final_e of - the gain-values. - */ - noise_e = (start_pos < no_cols) ? adj_e : final_e; - - /* - Convert energies to amplitude levels - */ - for (k=0; knrgSine[k], &pNrgs->nrgSine_e[k], &noise_e); - FDK_sqrt_MantExp(&pNrgs->nrgGain[k], &pNrgs->nrgGain_e[k], &pNrgs->nrgGain_e[k]); - FDK_sqrt_MantExp(&pNrgs->noiseLevel[k], &pNrgs->noiseLevel_e[k], &noise_e); - } - - - - /* - Apply calculated gains and adaptive noise - */ - - /* assembleHfSignals() */ - { - int scale_change, sc_change; - FIXP_SGL smooth_ratio; - int filtBufferNoiseShift=0; - - /* Initialize smoothing buffers with the first valid values */ - if (h_sbr_cal_env->startUp) - { - if (!useLP) { - h_sbr_cal_env->filtBufferNoise_e = noise_e; - - FDKmemcpy(h_sbr_cal_env->filtBuffer_e, pNrgs->nrgGain_e, noSubbands*sizeof(SCHAR)); - FDKmemcpy(h_sbr_cal_env->filtBufferNoise, pNrgs->noiseLevel, noSubbands*sizeof(FIXP_DBL)); - FDKmemcpy(h_sbr_cal_env->filtBuffer, pNrgs->nrgGain, noSubbands*sizeof(FIXP_DBL)); - - } - h_sbr_cal_env->startUp = 0; - } - - if (!useLP) { - - equalizeFiltBufferExp(h_sbr_cal_env->filtBuffer, /* buffered */ - h_sbr_cal_env->filtBuffer_e, /* buffered */ - pNrgs->nrgGain, /* current */ - pNrgs->nrgGain_e, /* current */ - noSubbands); - - /* Adapt exponent of buffered noise levels to the current exponent - so they can easily be smoothed */ - if((h_sbr_cal_env->filtBufferNoise_e - noise_e)>=0) { - int shift = fixMin(DFRACT_BITS-1,(int)(h_sbr_cal_env->filtBufferNoise_e - noise_e)); - for (k=0; kfiltBufferNoise[k] <<= shift; - } - else { - int shift = fixMin(DFRACT_BITS-1,-(int)(h_sbr_cal_env->filtBufferNoise_e - noise_e)); - for (k=0; kfiltBufferNoise[k] >>= shift; - } - - h_sbr_cal_env->filtBufferNoise_e = noise_e; - } - - /* find best scaling! */ - scale_change = -(DFRACT_BITS-1); - for(k=0;knrgGain_e[k]); - } - sc_change = (start_posnrgGain_e[k] + (sc_change-1); - pNrgs->nrgGain[k] >>= sc; - pNrgs->nrgGain_e[k] += sc; - } - - if (!useLP) { - for(k=0;kfiltBuffer_e[k] + (sc_change-1); - h_sbr_cal_env->filtBuffer[k] >>= sc; - } - } - - for (j = start_pos; j < stop_pos; j++) - { - /* This timeslot is located within the first part of the processing buffer - and will be fed into the QMF-synthesis for the current frame. - adj_e - input_e - This timeslot will not yet be fed into the QMF so we do not care - about the adj_e. - sc_change = final_e - input_e - */ - if ( (j==no_cols) && (start_posfiltBufferNoise[k] will be applied in function adjustTimeSlotHQ() */ - if (shift>=0) { - shift = fixMin(DFRACT_BITS-1,shift); - for (k=0; knrgSine[k] <<= shift; - pNrgs->noiseLevel[k] <<= shift; - /* - if (!useLP) - h_sbr_cal_env->filtBufferNoise[k] <<= shift; - */ - } - } - else { - shift = fixMin(DFRACT_BITS-1,-shift); - for (k=0; knrgSine[k] >>= shift; - pNrgs->noiseLevel[k] >>= shift; - /* - if (!useLP) - h_sbr_cal_env->filtBufferNoise[k] >>= shift; - */ - } - } - - /* update noise scaling */ - noise_e = final_e; - if (!useLP) - h_sbr_cal_env->filtBufferNoise_e = noise_e; /* scaling value unused! */ - - /* update gain buffer*/ - sc_change -= (final_e - input_e); - - if (sc_change<0) { - for(k=0;knrgGain[k] >>= -sc_change; - pNrgs->nrgGain_e[k] += -sc_change; - } - if (!useLP) { - for(k=0;kfiltBuffer[k] >>= -sc_change; - } - } - } else { - scale_change+=sc_change; - } - - } // if - - if (!useLP) { - - /* Prevent the smoothing filter from running on constant levels */ - if (j-start_pos < smooth_length) - smooth_ratio = FDK_sbrDecoder_sbr_smoothFilter[j-start_pos]; - else - smooth_ratio = FL2FXCONST_SGL(0.0f); - - adjustTimeSlotHQ(&analysBufferReal[j][lowSubband], - &analysBufferImag[j][lowSubband], - h_sbr_cal_env, - pNrgs, - lowSubband, - noSubbands, - scale_change, - smooth_ratio, - noNoiseFlag, - filtBufferNoiseShift); - } - else - { - if (flags & SBRDEC_ELD_GRID) { - adjustTimeSlot_EldGrid(&analysBufferReal[j][lowSubband], - pNrgs, - &h_sbr_cal_env->harmIndex, - lowSubband, - noSubbands, - scale_change, - noNoiseFlag, - &h_sbr_cal_env->phaseIndex, - EXP2SCALE(adj_e) - sbrScaleFactor->lb_scale); - } else - { - adjustTimeSlotLC(&analysBufferReal[j][lowSubband], - pNrgs, - &h_sbr_cal_env->harmIndex, - lowSubband, - noSubbands, - scale_change, - noNoiseFlag, - &h_sbr_cal_env->phaseIndex); - } - } - } // for - - if (!useLP) { - /* Update time-smoothing-buffers for gains and noise levels - The gains and the noise values of the current envelope are copied into the buffer. - This has to be done at the end of each envelope as the values are required for - a smooth transition to the next envelope. */ - FDKmemcpy(h_sbr_cal_env->filtBuffer, pNrgs->nrgGain, noSubbands*sizeof(FIXP_DBL)); - FDKmemcpy(h_sbr_cal_env->filtBuffer_e, pNrgs->nrgGain_e, noSubbands*sizeof(SCHAR)); - FDKmemcpy(h_sbr_cal_env->filtBufferNoise, pNrgs->noiseLevel, noSubbands*sizeof(FIXP_DBL)); - } - - } - C_ALLOC_SCRATCH_END(pNrgs, ENV_CALC_NRGS, 1); - } - - /* Rescale output samples */ - { - FIXP_DBL maxVal; - int ov_reserve, reserve; - - /* Determine headroom in old adjusted samples */ - maxVal = maxSubbandSample( analysBufferReal, - (useLP) ? NULL : analysBufferImag, - lowSubband, - highSubband, - 0, - first_start); - - ov_reserve = fNorm(maxVal); - - /* Determine headroom in new adjusted samples */ - maxVal = maxSubbandSample( analysBufferReal, - (useLP) ? NULL : analysBufferImag, - lowSubband, - highSubband, - first_start, - no_cols); - - reserve = fNorm(maxVal); - - /* Determine common output exponent */ - if (ov_adj_e - ov_reserve > adj_e - reserve ) /* set output_e to the maximum */ - output_e = ov_adj_e - ov_reserve; - else - output_e = adj_e - reserve; - - /* Rescale old samples */ - rescaleSubbandSamples( analysBufferReal, - (useLP) ? NULL : analysBufferImag, - lowSubband, highSubband, - 0, first_start, - ov_adj_e - output_e); - - /* Rescale new samples */ - rescaleSubbandSamples( analysBufferReal, - (useLP) ? NULL : analysBufferImag, - lowSubband, highSubband, - first_start, no_cols, - adj_e - output_e); - } - - /* Update hb_scale */ - sbrScaleFactor->hb_scale = EXP2SCALE(output_e); - - /* Save the current final exponent for the next frame: */ - sbrScaleFactor->ov_hb_scale = EXP2SCALE(final_e); - - - /* We need to remeber to the next frame that the transient - will occur in the first envelope (if tranEnv == nEnvelopes). */ - if(hFrameData->frameInfo.tranEnv == hFrameData->frameInfo.nEnvelopes) - h_sbr_cal_env->prevTranEnv = 0; - else - h_sbr_cal_env->prevTranEnv = -1; - -} - - -/*! - \brief Create envelope instance - - Must be called once for each channel before calculateSbrEnvelope() can be used. - - \return errorCode, 0 if successful -*/ -SBR_ERROR -createSbrEnvelopeCalc (HANDLE_SBR_CALCULATE_ENVELOPE hs, /*!< pointer to envelope instance */ - HANDLE_SBR_HEADER_DATA hHeaderData, /*!< static SBR control data, initialized with defaults */ - const int chan, /*!< Channel for which to assign buffers */ - const UINT flags) -{ - SBR_ERROR err = SBRDEC_OK; - int i; - - /* Clear previous missing harmonics flags */ - for (i=0; i<(MAX_FREQ_COEFFS+15)>>4; i++) { - hs->harmFlagsPrev[i] = 0; - } - hs->harmIndex = 0; - - /* - Setup pointers for time smoothing. - The buffer itself will be initialized later triggered by the startUp-flag. - */ - hs->prevTranEnv = -1; - - - /* initialization */ - resetSbrEnvelopeCalc(hs); - - if (chan==0) { /* do this only once */ - err = resetFreqBandTables(hHeaderData, flags); - } - - return err; -} - -/*! - \brief Create envelope instance - - Must be called once for each channel before calculateSbrEnvelope() can be used. - - \return errorCode, 0 if successful -*/ -int -deleteSbrEnvelopeCalc (HANDLE_SBR_CALCULATE_ENVELOPE hs) -{ - return 0; -} - - -/*! - \brief Reset envelope instance - - This function must be called for each channel on a change of configuration. - Note that resetFreqBandTables should also be called in this case. - - \return errorCode, 0 if successful -*/ -void -resetSbrEnvelopeCalc (HANDLE_SBR_CALCULATE_ENVELOPE hCalEnv) /*!< pointer to envelope instance */ -{ - hCalEnv->phaseIndex = 0; - - /* Noise exponent needs to be reset because the output exponent for the next frame depends on it */ - hCalEnv->filtBufferNoise_e = 0; - - hCalEnv->startUp = 1; -} - - -/*! - \brief Equalize exponents of the buffered gain values and the new ones - - After equalization of exponents, the FIR-filter addition for smoothing - can be performed. - This function is called once for each envelope before adjusting. -*/ -static void equalizeFiltBufferExp(FIXP_DBL *filtBuffer, /*!< bufferd gains */ - SCHAR *filtBuffer_e, /*!< exponents of bufferd gains */ - FIXP_DBL *nrgGain, /*!< gains for current envelope */ - SCHAR *nrgGain_e, /*!< exponents of gains for current envelope */ - int subbands) /*!< Number of QMF subbands */ -{ - int band; - int diff; - - for (band=0; band0) { - filtBuffer[band] >>= diff; /* Compensate for the scale change by shifting the mantissa. */ - filtBuffer_e[band] += diff; /* New gain is bigger, use its exponent */ - } - else if (diff<0) { - /* The buffered gains seem to be larger, but maybe there - are some unused bits left in the mantissa */ - - int reserve = CntLeadingZeros(fixp_abs(filtBuffer[band]))-1; - - if ((-diff) <= reserve) { - /* There is enough space in the buffered mantissa so - that we can take the new exponent as common. - */ - filtBuffer[band] <<= (-diff); - filtBuffer_e[band] += diff; /* becomes equal to *ptrNewExp */ - } - else { - filtBuffer[band] <<= reserve; /* Shift the mantissa as far as possible: */ - filtBuffer_e[band] -= reserve; /* Compensate in the exponent: */ - - /* For the remaining difference, change the new gain value */ - diff = fixMin(-(reserve + diff),DFRACT_BITS-1); - nrgGain[band] >>= diff; - nrgGain_e[band] += diff; - } - } - } -} - -/*! - \brief Shift left the mantissas of all subband samples - in the giventime and frequency range by the specified number of bits. - - This function is used to rescale the audio data in the overlap buffer - which has already been envelope adjusted with the last frame. -*/ -void rescaleSubbandSamples(FIXP_DBL ** re, /*!< Real part of input and output subband samples */ - FIXP_DBL ** im, /*!< Imaginary part of input and output subband samples */ - int lowSubband, /*!< Begin of frequency range to process */ - int highSubband, /*!< End of frequency range to process */ - int start_pos, /*!< Begin of time rage (QMF-timeslot) */ - int next_pos, /*!< End of time rage (QMF-timeslot) */ - int shift) /*!< number of bits to shift */ -{ - int width = highSubband-lowSubband; - - if ( (width > 0) && (shift!=0) ) { - if (im!=NULL) { - for (int l=start_pos; l 0 ) { - if (im!=NULL) - { - for (int l=start_pos; l>(DFRACT_BITS-1))); - maxVal |= (FIXP_DBL)((LONG)(tmp2)^((LONG)tmp2>>(DFRACT_BITS-1))); - } while(--k!=0); -#endif - } - } else - { - for (int l=start_pos; l>(DFRACT_BITS-1))); - }while(--k!=0); - } - } - } - - return(maxVal); -} - -#define SHIFT_BEFORE_SQUARE (3) /* (7/2) */ -/*!< - If the accumulator does not provide enough overflow bits or - does not provide a high dynamic range, the below energy calculation - requires an additional shift operation for each sample. - On the other hand, doing the shift allows using a single-precision - multiplication for the square (at least 16bit x 16bit). - For even values of OVRFLW_BITS (0, 2, 4, 6), saturated arithmetic - is required for the energy accumulation. - Theoretically, the sample-squares can sum up to a value of 76, - requiring 7 overflow bits. However since such situations are *very* - rare, accu can be limited to 64. - In case native saturated arithmetic is not available, overflows - can be prevented by replacing the above #define by - #define SHIFT_BEFORE_SQUARE ((8 - OVRFLW_BITS) / 2) - which will result in slightly reduced accuracy. -*/ - -/*! - \brief Estimates the mean energy of each filter-bank channel for the - duration of the current envelope - - This function is used when interpolFreq is true. -*/ -static void calcNrgPerSubband(FIXP_DBL **analysBufferReal, /*!< Real part of subband samples */ - FIXP_DBL **analysBufferImag, /*!< Imaginary part of subband samples */ - int lowSubband, /*!< Begin of the SBR frequency range */ - int highSubband, /*!< High end of the SBR frequency range */ - int start_pos, /*!< First QMF-slot of current envelope */ - int next_pos, /*!< Last QMF-slot of current envelope + 1 */ - SCHAR frameExp, /*!< Common exponent for all input samples */ - FIXP_DBL *nrgEst, /*!< resulting Energy (0..1) */ - SCHAR *nrgEst_e ) /*!< Exponent of resulting Energy */ -{ - FIXP_SGL invWidth; - SCHAR preShift; - SCHAR shift; - FIXP_DBL sum; - int k,l; - - /* Divide by width of envelope later: */ - invWidth = FX_DBL2FX_SGL(GetInvInt(next_pos - start_pos)); - /* The common exponent needs to be doubled because all mantissas are squared: */ - frameExp = frameExp << 1; - - for (k=lowSubband; k>(DFRACT_BITS-1))); - bufferReal[l] = analysBufferReal[l][k]; - maxVal |= (FIXP_DBL)((LONG)(bufferReal[l])^((LONG)bufferReal[l]>>(DFRACT_BITS-1))); - } - } - else - { - for (l=start_pos;l>(DFRACT_BITS-1))); - } - } - - if (maxVal!=FL2FXCONST_DBL(0.f)) { - - - /* If the accu does not provide enough overflow bits, we cannot - shift the samples up to the limit. - Instead, keep up to 3 free bits in each sample, i.e. up to - 6 bits after calculation of square. - Please note the comment on saturated arithmetic above! - */ - FIXP_DBL accu = FL2FXCONST_DBL(0.0f); - preShift = CntLeadingZeros(maxVal)-1; - preShift -= SHIFT_BEFORE_SQUARE; - - if (preShift>=0) { - if (analysBufferImag!=NULL) { - for (l=start_pos; l> (int)negpreShift; - FIXP_DBL temp2 = bufferImag[l] >> (int)negpreShift; - accu = fPow2AddDiv2(accu, temp1); - accu = fPow2AddDiv2(accu, temp2); - } - } else - { - for (l=start_pos; l> (int)negpreShift; - accu = fPow2AddDiv2(accu, temp); - } - } - } - accu <<= 1; - - /* Convert double precision to Mantissa/Exponent: */ - shift = fNorm(accu); - sum = accu << (int)shift; - - /* Divide by width of envelope and apply frame scale: */ - *nrgEst++ = fMult(sum, invWidth); - shift += 2 * preShift; - if (analysBufferImag!=NULL) - *nrgEst_e++ = frameExp - shift; - else - *nrgEst_e++ = frameExp - shift + 1; /* +1 due to missing imag. part */ - } /* maxVal!=0 */ - else { - - /* Prevent a zero-mantissa-number from being misinterpreted - due to its exponent. */ - *nrgEst++ = FL2FXCONST_DBL(0.0f); - *nrgEst_e++ = 0; - } - } -} - -/*! - \brief Estimates the mean energy of each Scale factor band for the - duration of the current envelope. - - This function is used when interpolFreq is false. -*/ -static void calcNrgPerSfb(FIXP_DBL **analysBufferReal, /*!< Real part of subband samples */ - FIXP_DBL **analysBufferImag, /*!< Imaginary part of subband samples */ - int nSfb, /*!< Number of scale factor bands */ - UCHAR *freqBandTable, /*!< First Subband for each Sfb */ - int start_pos, /*!< First QMF-slot of current envelope */ - int next_pos, /*!< Last QMF-slot of current envelope + 1 */ - SCHAR input_e, /*!< Common exponent for all input samples */ - FIXP_DBL *nrgEst, /*!< resulting Energy (0..1) */ - SCHAR *nrgEst_e ) /*!< Exponent of resulting Energy */ -{ - FIXP_SGL invWidth; - FIXP_DBL temp; - SCHAR preShift; - SCHAR shift, sum_e; - FIXP_DBL sum; - - int j,k,l,li,ui; - FIXP_DBL sumAll, sumLine; /* Single precision would be sufficient, - but overflow bits are required for accumulation */ - - /* Divide by width of envelope later: */ - invWidth = FX_DBL2FX_SGL(GetInvInt(next_pos - start_pos)); - /* The common exponent needs to be doubled because all mantissas are squared: */ - input_e = input_e << 1; - - for(j=0; j=0) { - for (l=start_pos; l> -(int)preShift; - sumLine += fPow2Div2(temp); - temp = analysBufferImag[l][k] >> -(int)preShift; - sumLine += fPow2Div2(temp); - } - } - } else - { - if (preShift>=0) { - for (l=start_pos; l> -(int)preShift; - sumLine += fPow2Div2(temp); - } - } - } - - /* The number of QMF-channels per SBR bands may be up to 15. - Shift right to avoid overflows in sum over all channels. */ - sumLine = sumLine >> (4-1); - sumAll += sumLine; - } - - /* Convert double precision to Mantissa/Exponent: */ - shift = fNorm(sumAll); - sum = sumAll << (int)shift; - - /* Divide by width of envelope: */ - sum = fMult(sum,invWidth); - - /* Divide by width of Sfb: */ - sum = fMult(sum, FX_DBL2FX_SGL(GetInvInt(ui-li))); - - /* Set all Subband energies in the Sfb to the average energy: */ - if (analysBufferImag!=NULL) - sum_e = input_e + 4 - shift; /* -4 to compensate right-shift */ - else - sum_e = input_e + 4 + 1 - shift; /* -4 to compensate right-shift; +1 due to missing imag. part */ - - sum_e -= 2 * preShift; - } /* maxVal!=0 */ - else { - - /* Prevent a zero-mantissa-number from being misinterpreted - due to its exponent. */ - sum = FL2FXCONST_DBL(0.0f); - sum_e = 0; - } - - for (k=li; knrgEst[i]; /*!< Energy in transposed signal */ - SCHAR nrgEst_e = nrgs->nrgEst_e[i]; /*!< Energy in transposed signal (exponent) */ - FIXP_DBL *ptrNrgGain = &nrgs->nrgGain[i]; /*!< Resulting energy gain */ - SCHAR *ptrNrgGain_e = &nrgs->nrgGain_e[i]; /*!< Resulting energy gain (exponent) */ - FIXP_DBL *ptrNoiseLevel = &nrgs->noiseLevel[i]; /*!< Resulting absolute noise energy */ - SCHAR *ptrNoiseLevel_e = &nrgs->noiseLevel_e[i]; /*!< Resulting absolute noise energy (exponent) */ - FIXP_DBL *ptrNrgSine = &nrgs->nrgSine[i]; /*!< Additional sine energy */ - SCHAR *ptrNrgSine_e = &nrgs->nrgSine_e[i]; /*!< Additional sine energy (exponent) */ - - FIXP_DBL a, b, c; - SCHAR a_e, b_e, c_e; - - /* - This addition of 1 prevents divisions by zero in the reference code. - For very small energies in nrgEst, it prevents the gains from becoming - very high which could cause some trouble due to the smoothing. - */ - b_e = (int)(nrgEst_e - 1); - if (b_e>=0) { - nrgEst = (FL2FXCONST_DBL(0.5f) >> (INT)fixMin(b_e+1,DFRACT_BITS-1)) + (nrgEst >> 1); - nrgEst_e += 1; /* shift by 1 bit to avoid overflow */ - - } else { - nrgEst = (nrgEst >> (INT)(fixMin(-b_e+1,DFRACT_BITS-1))) + (FL2FXCONST_DBL(0.5f) >> 1); - nrgEst_e = 2; /* shift by 1 bit to avoid overflow */ - } - - /* A = NrgRef * TmpNoise */ - a = fMult(nrgRef,tmpNoise); - a_e = nrgRef_e + tmpNoise_e; - - /* B = 1 + TmpNoise */ - b_e = (int)(tmpNoise_e - 1); - if (b_e>=0) { - b = (FL2FXCONST_DBL(0.5f) >> (INT)fixMin(b_e+1,DFRACT_BITS-1)) + (tmpNoise >> 1); - b_e = tmpNoise_e + 1; /* shift by 1 bit to avoid overflow */ - } else { - b = (tmpNoise >> (INT)(fixMin(-b_e+1,DFRACT_BITS-1))) + (FL2FXCONST_DBL(0.5f) >> 1); - b_e = 2; /* shift by 1 bit to avoid overflow */ - } - - /* noiseLevel = A / B = (NrgRef * TmpNoise) / (1 + TmpNoise) */ - FDK_divide_MantExp( a, a_e, - b, b_e, - ptrNoiseLevel, ptrNoiseLevel_e); - - if (sinePresentFlag) { - - /* C = (1 + TmpNoise) * NrgEst */ - c = fMult(b,nrgEst); - c_e = b_e + nrgEst_e; - - /* gain = A / C = (NrgRef * TmpNoise) / (1 + TmpNoise) * NrgEst */ - FDK_divide_MantExp( a, a_e, - c, c_e, - ptrNrgGain, ptrNrgGain_e); - - if (sineMapped) { - - /* sineLevel = nrgRef/ (1 + TmpNoise) */ - FDK_divide_MantExp( nrgRef, nrgRef_e, - b, b_e, - ptrNrgSine, ptrNrgSine_e); - } - } - else { - if (noNoiseFlag) { - /* B = NrgEst */ - b = nrgEst; - b_e = nrgEst_e; - } - else { - /* B = NrgEst * (1 + TmpNoise) */ - b = fMult(b,nrgEst); - b_e = b_e + nrgEst_e; - } - - - /* gain = nrgRef / B */ - FDK_divide_MantExp( nrgRef, nrgRef_e, - b, b_e, - ptrNrgGain, ptrNrgGain_e); - } -} - - -/*! - \brief Calculate "average gain" for the specified subband range. - - This is rather a gain of the average magnitude than the average - of gains! - The result is used as a relative limit for all gains within the - current "limiter band" (a certain frequency range). -*/ -static void calcAvgGain(ENV_CALC_NRGS* nrgs, - int lowSubband, /*!< Begin of the limiter band */ - int highSubband, /*!< High end of the limiter band */ - FIXP_DBL *ptrSumRef, - SCHAR *ptrSumRef_e, - FIXP_DBL *ptrAvgGain, /*!< Resulting overall gain (mantissa) */ - SCHAR *ptrAvgGain_e) /*!< Resulting overall gain (exponent) */ -{ - FIXP_DBL *nrgRef = nrgs->nrgRef; /*!< Reference Energy according to envelope data */ - SCHAR *nrgRef_e = nrgs->nrgRef_e; /*!< Reference Energy according to envelope data (exponent) */ - FIXP_DBL *nrgEst = nrgs->nrgEst; /*!< Energy in transposed signal */ - SCHAR *nrgEst_e = nrgs->nrgEst_e; /*!< Energy in transposed signal (exponent) */ - - FIXP_DBL sumRef = 1; - FIXP_DBL sumEst = 1; - SCHAR sumRef_e = -FRACT_BITS; - SCHAR sumEst_e = -FRACT_BITS; - int k; - - for (k=lowSubband; knrgGain; /*!< Gains of current envelope */ - FIXP_DBL *pNoiseLevel = nrgs->noiseLevel; /*!< Noise levels of current envelope */ - FIXP_DBL *pSineLevel = nrgs->nrgSine; /*!< Sine levels */ - - int phaseIndex = *ptrPhaseIndex; - UCHAR harmIndex = *ptrHarmIndex; - - static const INT harmonicPhase [2][4] = { - { 1, 0, -1, 0}, - { 0, 1, 0, -1} - }; - - static const FIXP_DBL harmonicPhaseX [2][4] = { - { FL2FXCONST_DBL(2.0*1.245183154539139e-001), FL2FXCONST_DBL(2.0*-1.123767859325028e-001), FL2FXCONST_DBL(2.0*-1.245183154539139e-001), FL2FXCONST_DBL(2.0* 1.123767859325028e-001) }, - { FL2FXCONST_DBL(2.0*1.245183154539139e-001), FL2FXCONST_DBL(2.0* 1.123767859325028e-001), FL2FXCONST_DBL(2.0*-1.245183154539139e-001), FL2FXCONST_DBL(2.0*-1.123767859325028e-001) } - }; - - for (k=0; k < noSubbands; k++) { - - phaseIndex = (phaseIndex + 1) & (SBR_NF_NO_RANDOM_VAL - 1); - - if( (pSineLevel[0] != FL2FXCONST_DBL(0.0f)) || (noNoiseFlag == 1) ){ - sbNoise = FL2FXCONST_DBL(0.0f); - } else { - sbNoise = pNoiseLevel[0]; - } - - signalReal = fMultDiv2(*ptrReal,*pGain) << ((int)scale_change); - - signalReal += (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[phaseIndex][0], sbNoise)<<4); - - signalReal += pSineLevel[0] * harmonicPhase[0][harmIndex]; - - *ptrReal = signalReal; - - if (k == 0) { - *(ptrReal-1) += scaleValue(fMultDiv2(harmonicPhaseX[lowSubband&1][harmIndex], pSineLevel[0]), -scale_diff_low) ; - if (k < noSubbands - 1) { - *(ptrReal) += fMultDiv2(pSineLevel[1], harmonicPhaseX[(lowSubband+1)&1][harmIndex]); - } - } - if (k > 0 && k < noSubbands - 1 && tone_count < 16) { - *(ptrReal) += fMultDiv2(pSineLevel[- 1], harmonicPhaseX [(lowSubband+k)&1] [harmIndex]); - *(ptrReal) += fMultDiv2(pSineLevel[+ 1], harmonicPhaseX [(lowSubband+k+1)&1][harmIndex]); - } - if (k == noSubbands - 1 && tone_count < 16) { - if (k > 0) { - *(ptrReal) += fMultDiv2(pSineLevel[- 1], harmonicPhaseX [(lowSubband+k)&1][harmIndex]); - } - if (k + lowSubband + 1< 63) { - *(ptrReal+1) += fMultDiv2(pSineLevel[0], harmonicPhaseX[(lowSubband+k+1)&1][harmIndex]); - } - } - - if(pSineLevel[0] != FL2FXCONST_DBL(0.0f)){ - tone_count++; - } - ptrReal++; - pNoiseLevel++; - pGain++; - pSineLevel++; - } - - *ptrHarmIndex = (harmIndex + 1) & 3; - *ptrPhaseIndex = phaseIndex & (SBR_NF_NO_RANDOM_VAL - 1); -} - -/*! - \brief Amplify one timeslot of the signal with the calculated gains - and add the noisefloor. -*/ - -static void adjustTimeSlotLC(FIXP_DBL *ptrReal, /*!< Subband samples to be adjusted, real part */ - ENV_CALC_NRGS* nrgs, - UCHAR *ptrHarmIndex, /*!< Harmonic index */ - int lowSubband, /*!< Lowest QMF-channel in the currently used SBR range. */ - int noSubbands, /*!< Number of QMF subbands */ - int scale_change, /*!< Number of bits to shift adjusted samples */ - int noNoiseFlag, /*!< Flag to suppress noise addition */ - int *ptrPhaseIndex) /*!< Start index to random number array */ -{ - FIXP_DBL *pGain = nrgs->nrgGain; /*!< Gains of current envelope */ - FIXP_DBL *pNoiseLevel = nrgs->noiseLevel; /*!< Noise levels of current envelope */ - FIXP_DBL *pSineLevel = nrgs->nrgSine; /*!< Sine levels */ - - int k; - int index = *ptrPhaseIndex; - UCHAR harmIndex = *ptrHarmIndex; - UCHAR freqInvFlag = (lowSubband & 1); - FIXP_DBL signalReal, sineLevel, sineLevelNext, sineLevelPrev; - int tone_count = 0; - int sineSign = 1; - - #define C1 ((FIXP_SGL)FL2FXCONST_SGL(2.f*0.00815f)) - #define C1_CLDFB ((FIXP_SGL)FL2FXCONST_SGL(2.f*0.16773f)) - - /* - First pass for k=0 pulled out of the loop: - */ - - index = (index + 1) & (SBR_NF_NO_RANDOM_VAL - 1); - - /* - The next multiplication constitutes the actual envelope adjustment - of the signal and should be carried out with full accuracy - (supplying #FRACT_BITS valid bits). - */ - signalReal = fMultDiv2(*ptrReal,*pGain++) << ((int)scale_change); - sineLevel = *pSineLevel++; - sineLevelNext = (noSubbands > 1) ? pSineLevel[0] : FL2FXCONST_DBL(0.0f); - - if (sineLevel!=FL2FXCONST_DBL(0.0f)) tone_count++; - else if (!noNoiseFlag) - /* Add noisefloor to the amplified signal */ - signalReal += (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], pNoiseLevel[0])<<4); - - { - if (!(harmIndex&0x1)) { - /* harmIndex 0,2 */ - signalReal += (harmIndex&0x2) ? -sineLevel : sineLevel; - *ptrReal++ = signalReal; - } - else { - /* harmIndex 1,3 in combination with freqInvFlag */ - int shift = (int) (scale_change+1); - shift = (shift>=0) ? fixMin(DFRACT_BITS-1,shift) : fixMax(-(DFRACT_BITS-1),shift); - - FIXP_DBL tmp1 = (shift>=0) ? ( fMultDiv2(C1, sineLevel) >> shift ) - : ( fMultDiv2(C1, sineLevel) << (-shift) ); - FIXP_DBL tmp2 = fMultDiv2(C1, sineLevelNext); - - - /* save switch and compare operations and reduce to XOR statement */ - if ( ((harmIndex>>1)&0x1)^freqInvFlag) { - *(ptrReal-1) += tmp1; - signalReal -= tmp2; - } else { - *(ptrReal-1) -= tmp1; - signalReal += tmp2; - } - *ptrReal++ = signalReal; - freqInvFlag = !freqInvFlag; - } - } - - pNoiseLevel++; - - if ( noSubbands > 2 ) { - if (!(harmIndex&0x1)) { - /* harmIndex 0,2 */ - if(!harmIndex) - { - sineSign = 0; - } - - for (k=noSubbands-2; k!=0; k--) { - FIXP_DBL sinelevel = *pSineLevel++; - index++; - if (((signalReal = (sineSign ? -sinelevel : sinelevel)) == FL2FXCONST_DBL(0.0f)) && !noNoiseFlag) - { - /* Add noisefloor to the amplified signal */ - index &= (SBR_NF_NO_RANDOM_VAL - 1); - signalReal += (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], pNoiseLevel[0])<<4); - } - - /* The next multiplication constitutes the actual envelope adjustment of the signal. */ - signalReal += fMultDiv2(*ptrReal,*pGain++) << ((int)scale_change); - - pNoiseLevel++; - *ptrReal++ = signalReal; - } /* for ... */ - } - else { - /* harmIndex 1,3 in combination with freqInvFlag */ - if (harmIndex==1) freqInvFlag = !freqInvFlag; - - for (k=noSubbands-2; k!=0; k--) { - index++; - /* The next multiplication constitutes the actual envelope adjustment of the signal. */ - signalReal = fMultDiv2(*ptrReal,*pGain++) << ((int)scale_change); - - if (*pSineLevel++!=FL2FXCONST_DBL(0.0f)) tone_count++; - else if (!noNoiseFlag) { - /* Add noisefloor to the amplified signal */ - index &= (SBR_NF_NO_RANDOM_VAL - 1); - signalReal += (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], pNoiseLevel[0])<<4); - } - - pNoiseLevel++; - - if (tone_count <= 16) { - FIXP_DBL addSine = fMultDiv2((pSineLevel[-2] - pSineLevel[0]), C1); - signalReal += (freqInvFlag) ? (-addSine) : (addSine); - } - - *ptrReal++ = signalReal; - freqInvFlag = !freqInvFlag; - } /* for ... */ - } - } - - if (noSubbands > -1) { - index++; - /* The next multiplication constitutes the actual envelope adjustment of the signal. */ - signalReal = fMultDiv2(*ptrReal,*pGain) << ((int)scale_change); - sineLevelPrev = fMultDiv2(pSineLevel[-1],FL2FX_SGL(0.0163f)); - sineLevel = pSineLevel[0]; - - if (pSineLevel[0]!=FL2FXCONST_DBL(0.0f)) tone_count++; - else if (!noNoiseFlag) { - /* Add noisefloor to the amplified signal */ - index &= (SBR_NF_NO_RANDOM_VAL - 1); - signalReal = signalReal + (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], pNoiseLevel[0])<<4); - } - - if (!(harmIndex&0x1)) { - /* harmIndex 0,2 */ - *ptrReal = signalReal + ( (sineSign) ? -sineLevel : sineLevel); - } - else { - /* harmIndex 1,3 in combination with freqInvFlag */ - if(tone_count <= 16){ - if (freqInvFlag) { - *ptrReal++ = signalReal - sineLevelPrev; - if (noSubbands + lowSubband < 63) - *ptrReal = *ptrReal + fMultDiv2(C1, sineLevel); - } - else { - *ptrReal++ = signalReal + sineLevelPrev; - if (noSubbands + lowSubband < 63) - *ptrReal = *ptrReal - fMultDiv2(C1, sineLevel); - } - } - else *ptrReal = signalReal; - } - } - *ptrHarmIndex = (harmIndex + 1) & 3; - *ptrPhaseIndex = index & (SBR_NF_NO_RANDOM_VAL - 1); -} -static void adjustTimeSlotHQ( - FIXP_DBL *RESTRICT ptrReal, /*!< Subband samples to be adjusted, real part */ - FIXP_DBL *RESTRICT ptrImag, /*!< Subband samples to be adjusted, imag part */ - HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env, - ENV_CALC_NRGS* nrgs, - int lowSubband, /*!< Lowest QMF-channel in the currently used SBR range. */ - int noSubbands, /*!< Number of QMF subbands */ - int scale_change, /*!< Number of bits to shift adjusted samples */ - FIXP_SGL smooth_ratio, /*!< Impact of last envelope */ - int noNoiseFlag, /*!< Start index to random number array */ - int filtBufferNoiseShift) /*!< Shift factor of filtBufferNoise */ -{ - - FIXP_DBL *RESTRICT gain = nrgs->nrgGain; /*!< Gains of current envelope */ - FIXP_DBL *RESTRICT noiseLevel = nrgs->noiseLevel; /*!< Noise levels of current envelope */ - FIXP_DBL *RESTRICT pSineLevel = nrgs->nrgSine; /*!< Sine levels */ - - FIXP_DBL *RESTRICT filtBuffer = h_sbr_cal_env->filtBuffer; /*!< Gains of last envelope */ - FIXP_DBL *RESTRICT filtBufferNoise = h_sbr_cal_env->filtBufferNoise; /*!< Noise levels of last envelope */ - UCHAR *RESTRICT ptrHarmIndex =&h_sbr_cal_env->harmIndex; /*!< Harmonic index */ - int *RESTRICT ptrPhaseIndex =&h_sbr_cal_env->phaseIndex; /*!< Start index to random number array */ - - int k; - FIXP_DBL signalReal, signalImag; - FIXP_DBL noiseReal, noiseImag; - FIXP_DBL smoothedGain, smoothedNoise; - FIXP_SGL direct_ratio = /*FL2FXCONST_SGL(1.0f) */ (FIXP_SGL)MAXVAL_SGL - smooth_ratio; - int index = *ptrPhaseIndex; - UCHAR harmIndex = *ptrHarmIndex; - int freqInvFlag = (lowSubband & 1); - FIXP_DBL sineLevel; - int shift; - - *ptrPhaseIndex = (index+noSubbands) & (SBR_NF_NO_RANDOM_VAL - 1); - *ptrHarmIndex = (harmIndex + 1) & 3; - - /* - Possible optimization: - smooth_ratio and harmIndex stay constant during the loop. - It might be faster to include a separate loop in each path. - - the check for smooth_ratio is now outside the loop and the workload - of the whole function decreased by about 20 % - */ - - filtBufferNoiseShift += 1; /* due to later use of fMultDiv2 instead of fMult */ - if (filtBufferNoiseShift<0) - shift = fixMin(DFRACT_BITS-1,-filtBufferNoiseShift); - else - shift = fixMin(DFRACT_BITS-1, filtBufferNoiseShift); - - if (smooth_ratio > FL2FXCONST_SGL(0.0f)) { - - for (k=0; k>shift) + - fMult(direct_ratio,noiseLevel[k]); - } - else { - smoothedNoise = (fMultDiv2(smooth_ratio,filtBufferNoise[k])< scale factor of 2 */ - temp = fMultNorm(oct_m, FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4_DBL[limiterBands], &temp_e); - - /* overall scale factor of temp ist addition of scalefactors from log2 calculation, - limiter bands scalefactor (2) and limiter bands multiplication */ - temp_e += oct_e + 2; - - /* div can be a maximum of 64 (k2 = 64 and kx = 1) - -> oct can be a maximum of 6 - -> temp can be a maximum of 18 (as limiterBandsPerOctoave is a maximum factor of 3) - -> we need a scale factor of 5 for comparisson - */ - if (temp >> (5 - temp_e) < FL2FXCONST_DBL (0.49f) >> 5) { - - if (workLimiterBandTable[hiLimIndex] == workLimiterBandTable[loLimIndex]) { - workLimiterBandTable[hiLimIndex] = highSubband; - nBands--; - hiLimIndex++; - continue; - } - isPatchBorder[0] = isPatchBorder[1] = 0; - for (k = 0; k <= noPatches; k++) { - if (workLimiterBandTable[hiLimIndex] == patchBorders[k]) { - isPatchBorder[1] = 1; - break; - } - } - if (!isPatchBorder[1]) { - workLimiterBandTable[hiLimIndex] = highSubband; - nBands--; - hiLimIndex++; - continue; - } - for (k = 0; k <= noPatches; k++) { - if (workLimiterBandTable[loLimIndex] == patchBorders[k]) { - isPatchBorder[0] = 1; - break; - } - } - if (!isPatchBorder[0]) { - workLimiterBandTable[loLimIndex] = highSubband; - nBands--; - } - } - loLimIndex = hiLimIndex; - hiLimIndex++; - - } - shellsort(workLimiterBandTable, tempNoLim + 1); - - /* Test if algorithm exceeded maximum allowed limiterbands */ - if( nBands > MAX_NUM_LIMITERS || nBands <= 0) { - return SBRDEC_UNSUPPORTED_CONFIG; - } - - /* Copy limiterbands from working buffer into final destination */ - for (k = 0; k <= nBands; k++) { - limiterBandTable[k] = workLimiterBandTable[k]; - } - } - *noLimiterBands = nBands; - - return SBRDEC_OK; -} - diff --git a/libSBRdec/src/env_calc.h b/libSBRdec/src/env_calc.h deleted file mode 100644 index 8154166..0000000 --- a/libSBRdec/src/env_calc.h +++ /dev/null @@ -1,165 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief Envelope calculation prototypes -*/ -#ifndef __ENV_CALC_H -#define __ENV_CALC_H - -#include "sbrdecoder.h" -#include "env_extr.h" /* for HANDLE_SBR_HEADER_DATA */ -#include "sbr_scale.h" - - -typedef struct -{ - FIXP_DBL filtBuffer[MAX_FREQ_COEFFS]; /*!< previous gains (required for smoothing) */ - FIXP_DBL filtBufferNoise[MAX_FREQ_COEFFS]; /*!< previous noise levels (required for smoothing) */ - SCHAR filtBuffer_e[MAX_FREQ_COEFFS]; /*!< Exponents of previous gains */ - SCHAR filtBufferNoise_e; /*!< Common exponent of previous noise levels */ - - int startUp; /*!< flag to signal initial conditions in buffers */ - int phaseIndex; /*!< Index for randomPase array */ - int prevTranEnv; /*!< The transient envelope of the previous frame. */ - - int harmFlagsPrev[(MAX_FREQ_COEFFS+15)/16]; - /*!< Words with 16 flags each indicating where a sine was added in the previous frame.*/ - UCHAR harmIndex; /*!< Current phase of synthetic sine */ - -} -SBR_CALCULATE_ENVELOPE; - -typedef SBR_CALCULATE_ENVELOPE *HANDLE_SBR_CALCULATE_ENVELOPE; - - - -void -calculateSbrEnvelope (QMF_SCALE_FACTOR *sbrScaleFactor, - HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env, - HANDLE_SBR_HEADER_DATA hHeaderData, - HANDLE_SBR_FRAME_DATA hFrameData, - FIXP_DBL **analysBufferReal, - FIXP_DBL **analysBufferImag, /*!< Imag part of subband samples to be processed */ - const int useLP, - FIXP_DBL *degreeAlias, /*!< Estimated aliasing for each QMF channel */ - const UINT flags, - const int frameErrorFlag - ); - -SBR_ERROR -createSbrEnvelopeCalc (HANDLE_SBR_CALCULATE_ENVELOPE hSbrCalculateEnvelope, - HANDLE_SBR_HEADER_DATA hHeaderData, - const int chan, - const UINT flags); - -int -deleteSbrEnvelopeCalc (HANDLE_SBR_CALCULATE_ENVELOPE hSbrCalculateEnvelope); - -void -resetSbrEnvelopeCalc (HANDLE_SBR_CALCULATE_ENVELOPE hCalEnv); - -SBR_ERROR -ResetLimiterBands ( UCHAR *limiterBandTable, - UCHAR *noLimiterBands, - UCHAR *freqBandTable, - int noFreqBands, - const PATCH_PARAM *patchParam, - int noPatches, - int limiterBands); - -void rescaleSubbandSamples( FIXP_DBL ** re, - FIXP_DBL ** im, - int lowSubband, int noSubbands, - int start_pos, int next_pos, - int shift); - -FIXP_DBL maxSubbandSample( FIXP_DBL ** analysBufferReal_m, - FIXP_DBL ** analysBufferImag_m, - int lowSubband, - int highSubband, - int start_pos, - int stop_pos); - -#endif // __ENV_CALC_H diff --git a/libSBRdec/src/env_dec.cpp b/libSBRdec/src/env_dec.cpp deleted file mode 100644 index c65c169..0000000 --- a/libSBRdec/src/env_dec.cpp +++ /dev/null @@ -1,852 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief envelope decoding - This module provides envelope decoding and error concealment algorithms. The main - entry point is decodeSbrData(). - - \sa decodeSbrData(),\ref documentationOverview -*/ - -#include "env_dec.h" - -#include "env_extr.h" -#include "transcendent.h" - -#include "genericStds.h" - - -static void decodeEnvelope (HANDLE_SBR_HEADER_DATA hHeaderData, - HANDLE_SBR_FRAME_DATA h_sbr_data, - HANDLE_SBR_PREV_FRAME_DATA h_prev_data, - HANDLE_SBR_PREV_FRAME_DATA h_prev_data_otherChannel); -static void sbr_envelope_unmapping (HANDLE_SBR_HEADER_DATA hHeaderData, - HANDLE_SBR_FRAME_DATA h_data_left, - HANDLE_SBR_FRAME_DATA h_data_right); -static void requantizeEnvelopeData (HANDLE_SBR_FRAME_DATA h_sbr_data, - int ampResolution); -static void deltaToLinearPcmEnvelopeDecoding (HANDLE_SBR_HEADER_DATA hHeaderData, - HANDLE_SBR_FRAME_DATA h_sbr_data, - HANDLE_SBR_PREV_FRAME_DATA h_prev_data); -static void decodeNoiseFloorlevels (HANDLE_SBR_HEADER_DATA hHeaderData, - HANDLE_SBR_FRAME_DATA h_sbr_data, - HANDLE_SBR_PREV_FRAME_DATA h_prev_data); -static void timeCompensateFirstEnvelope (HANDLE_SBR_HEADER_DATA hHeaderData, - HANDLE_SBR_FRAME_DATA h_sbr_data, - HANDLE_SBR_PREV_FRAME_DATA h_prev_data); -static int checkEnvelopeData (HANDLE_SBR_HEADER_DATA hHeaderData, - HANDLE_SBR_FRAME_DATA h_sbr_data, - HANDLE_SBR_PREV_FRAME_DATA h_prev_data); - - - -#define SBR_ENERGY_PAN_OFFSET (12 << ENV_EXP_FRACT) -#define SBR_MAX_ENERGY (35 << ENV_EXP_FRACT) - -#define DECAY ( 1 << ENV_EXP_FRACT) - -#if ENV_EXP_FRACT -#define DECAY_COUPLING ( 1 << (ENV_EXP_FRACT-1) ) /*!< corresponds to a value of 0.5 */ -#else -#define DECAY_COUPLING 1 /*!< If the energy data is not shifted, use 1 instead of 0.5 */ -#endif - - -/*! - \brief Convert table index -*/ -static int indexLow2High(int offset, /*!< mapping factor */ - int index, /*!< index to scalefactor band */ - int res) /*!< frequency resolution */ -{ - if(res == 0) - { - if (offset >= 0) - { - if (index < offset) - return(index); - else - return(2*index - offset); - } - else - { - offset = -offset; - if (index < offset) - return(2*index+index); - else - return(2*index + offset); - } - } - else - return(index); -} - - -/*! - \brief Update previous envelope value for delta-coding - - The current envelope values needs to be stored for delta-coding - in the next frame. The stored envelope is always represented with - the high frequency resolution. If the current envelope uses the - low frequency resolution, the energy value will be mapped to the - corresponding high-res bands. -*/ -static void mapLowResEnergyVal(FIXP_SGL currVal, /*!< current energy value */ - FIXP_SGL* prevData,/*!< pointer to previous data vector */ - int offset, /*!< mapping factor */ - int index, /*!< index to scalefactor band */ - int res) /*!< frequeny resolution */ -{ - if(res == 0) - { - if (offset >= 0) - { - if(index < offset) - prevData[index] = currVal; - else - { - prevData[2*index - offset] = currVal; - prevData[2*index+1 - offset] = currVal; - } - } - else - { - offset = -offset; - if (index < offset) - { - prevData[3*index] = currVal; - prevData[3*index+1] = currVal; - prevData[3*index+2] = currVal; - } - else - { - prevData[2*index + offset] = currVal; - prevData[2*index + 1 + offset] = currVal; - } - } - } - else - prevData[index] = currVal; -} - - - -/*! - \brief Convert raw envelope and noisefloor data to energy levels - - This function is being called by sbrDecoder_ParseElement() and provides two important algorithms: - - First the function decodes envelopes and noise floor levels as described in requantizeEnvelopeData() - and sbr_envelope_unmapping(). The function also implements concealment algorithms in case there are errors - within the sbr data. For both operations fractional arithmetic is used. - Therefore you might encounter different output values on your target - system compared to the reference implementation. -*/ -void -decodeSbrData (HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ - HANDLE_SBR_FRAME_DATA h_data_left, /*!< pointer to left channel frame data */ - HANDLE_SBR_PREV_FRAME_DATA h_prev_data_left, /*!< pointer to left channel previous frame data */ - HANDLE_SBR_FRAME_DATA h_data_right, /*!< pointer to right channel frame data */ - HANDLE_SBR_PREV_FRAME_DATA h_prev_data_right)/*!< pointer to right channel previous frame data */ -{ - FIXP_SGL tempSfbNrgPrev[MAX_FREQ_COEFFS]; - int errLeft; - - /* Save previous energy values to be able to reuse them later for concealment. */ - FDKmemcpy (tempSfbNrgPrev, h_prev_data_left->sfb_nrg_prev, MAX_FREQ_COEFFS * sizeof(FIXP_SGL)); - - decodeEnvelope (hHeaderData, h_data_left, h_prev_data_left, h_prev_data_right); - decodeNoiseFloorlevels (hHeaderData, h_data_left, h_prev_data_left); - - if(h_data_right != NULL) { - errLeft = hHeaderData->frameErrorFlag; - decodeEnvelope (hHeaderData, h_data_right, h_prev_data_right, h_prev_data_left); - decodeNoiseFloorlevels (hHeaderData, h_data_right, h_prev_data_right); - - if (!errLeft && hHeaderData->frameErrorFlag) { - /* If an error occurs in the right channel where the left channel seemed ok, - we apply concealment also on the left channel. This ensures that the coupling - modes of both channels match and that we have the same number of envelopes in - coupling mode. - However, as the left channel has already been processed before, the resulting - energy levels are not the same as if the left channel had been concealed - during the first call of decodeEnvelope(). - */ - /* Restore previous energy values for concealment, because the values have been - overwritten by the first call of decodeEnvelope(). */ - FDKmemcpy (h_prev_data_left->sfb_nrg_prev, tempSfbNrgPrev, MAX_FREQ_COEFFS * sizeof(FIXP_SGL)); - /* Do concealment */ - decodeEnvelope (hHeaderData, h_data_left, h_prev_data_left, h_prev_data_right); - } - - if (h_data_left->coupling) { - sbr_envelope_unmapping (hHeaderData, h_data_left, h_data_right); - } - } - - /* Display the data for debugging: */ -} - - -/*! - \brief Convert from coupled channels to independent L/R data -*/ -static void -sbr_envelope_unmapping (HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ - HANDLE_SBR_FRAME_DATA h_data_left, /*!< pointer to left channel */ - HANDLE_SBR_FRAME_DATA h_data_right) /*!< pointer to right channel */ -{ - int i; - FIXP_SGL tempL_m, tempR_m, tempRplus1_m, newL_m, newR_m; - SCHAR tempL_e, tempR_e, tempRplus1_e, newL_e, newR_e; - - - /* 1. Unmap (already dequantized) coupled envelope energies */ - - for (i = 0; i < h_data_left->nScaleFactors; i++) { - tempR_m = (FIXP_SGL)((LONG)h_data_right->iEnvelope[i] & MASK_M); - tempR_e = (SCHAR)((LONG)h_data_right->iEnvelope[i] & MASK_E); - - tempR_e -= (18 + NRG_EXP_OFFSET); /* -18 = ld(UNMAPPING_SCALE / h_data_right->nChannels) */ - tempL_m = (FIXP_SGL)((LONG)h_data_left->iEnvelope[i] & MASK_M); - tempL_e = (SCHAR)((LONG)h_data_left->iEnvelope[i] & MASK_E); - - tempL_e -= NRG_EXP_OFFSET; - - /* Calculate tempRight+1 */ - FDK_add_MantExp( tempR_m, tempR_e, - FL2FXCONST_SGL(0.5f), 1, /* 1.0 */ - &tempRplus1_m, &tempRplus1_e); - - FDK_divide_MantExp( tempL_m, tempL_e+1, /* 2 * tempLeft */ - tempRplus1_m, tempRplus1_e, - &newR_m, &newR_e ); - - if (newR_m >= ((FIXP_SGL)MAXVAL_SGL - ROUNDING)) { - newR_m >>= 1; - newR_e += 1; - } - - newL_m = FX_DBL2FX_SGL(fMult(tempR_m,newR_m)); - newL_e = tempR_e + newR_e; - - h_data_right->iEnvelope[i] = ((FIXP_SGL)((SHORT)(FIXP_SGL)(newR_m + ROUNDING) & MASK_M)) + - (FIXP_SGL)((SHORT)(FIXP_SGL)(newR_e + NRG_EXP_OFFSET) & MASK_E); - h_data_left->iEnvelope[i] = ((FIXP_SGL)((SHORT)(FIXP_SGL)(newL_m + ROUNDING) & MASK_M)) + - (FIXP_SGL)((SHORT)(FIXP_SGL)(newL_e + NRG_EXP_OFFSET) & MASK_E); - } - - /* 2. Dequantize and unmap coupled noise floor levels */ - - for (i = 0; i < hHeaderData->freqBandData.nNfb * h_data_left->frameInfo.nNoiseEnvelopes; i++) { - - tempL_e = (SCHAR)(6 - (LONG)h_data_left->sbrNoiseFloorLevel[i]); - tempR_e = (SCHAR)((LONG)h_data_right->sbrNoiseFloorLevel[i] - 12) /*SBR_ENERGY_PAN_OFFSET*/; - - /* Calculate tempR+1 */ - FDK_add_MantExp( FL2FXCONST_SGL(0.5f), 1+tempR_e, /* tempR */ - FL2FXCONST_SGL(0.5f), 1, /* 1.0 */ - &tempRplus1_m, &tempRplus1_e); - - /* Calculate 2*tempLeft/(tempR+1) */ - FDK_divide_MantExp( FL2FXCONST_SGL(0.5f), tempL_e+2, /* 2 * tempLeft */ - tempRplus1_m, tempRplus1_e, - &newR_m, &newR_e ); - - /* if (newR_m >= ((FIXP_SGL)MAXVAL_SGL - ROUNDING)) { - newR_m >>= 1; - newR_e += 1; - } */ - - /* L = tempR * R */ - newL_m = newR_m; - newL_e = newR_e + tempR_e; - h_data_right->sbrNoiseFloorLevel[i] = ((FIXP_SGL)((SHORT)(FIXP_SGL)(newR_m + ROUNDING) & MASK_M)) + - (FIXP_SGL)((SHORT)(FIXP_SGL)(newR_e + NOISE_EXP_OFFSET) & MASK_E); - h_data_left->sbrNoiseFloorLevel[i] = ((FIXP_SGL)((SHORT)(FIXP_SGL)(newL_m + ROUNDING) & MASK_M)) + - (FIXP_SGL)((SHORT)(FIXP_SGL)(newL_e + NOISE_EXP_OFFSET) & MASK_E); - } -} - - -/*! - \brief Simple alternative to the real SBR concealment - - If the real frameInfo is not available due to a frame loss, a replacement will - be constructed with 1 envelope spanning the whole frame (FIX-FIX). - The delta-coded energies are set to negative values, resulting in a fade-down. - In case of coupling, the balance-channel will move towards the center. -*/ -static void -leanSbrConcealment(HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ - HANDLE_SBR_FRAME_DATA h_sbr_data, /*!< pointer to current data */ - HANDLE_SBR_PREV_FRAME_DATA h_prev_data /*!< pointer to data of last frame */ - ) -{ - FIXP_SGL target; /* targeted level for sfb_nrg_prev during fade-down */ - FIXP_SGL step; /* speed of fade */ - int i; - - int currentStartPos = FDKmax(0, h_prev_data->stopPos - hHeaderData->numberTimeSlots); - int currentStopPos = hHeaderData->numberTimeSlots; - - - /* Use some settings of the previous frame */ - h_sbr_data->ampResolutionCurrentFrame = h_prev_data->ampRes; - h_sbr_data->coupling = h_prev_data->coupling; - for(i=0;isbr_invf_mode[i] = h_prev_data->sbr_invf_mode[i]; - - /* Generate concealing control data */ - - h_sbr_data->frameInfo.nEnvelopes = 1; - h_sbr_data->frameInfo.borders[0] = currentStartPos; - h_sbr_data->frameInfo.borders[1] = currentStopPos; - h_sbr_data->frameInfo.freqRes[0] = 1; - h_sbr_data->frameInfo.tranEnv = -1; /* no transient */ - h_sbr_data->frameInfo.nNoiseEnvelopes = 1; - h_sbr_data->frameInfo.bordersNoise[0] = currentStartPos; - h_sbr_data->frameInfo.bordersNoise[1] = currentStopPos; - - h_sbr_data->nScaleFactors = hHeaderData->freqBandData.nSfb[1]; - - /* Generate fake envelope data */ - - h_sbr_data->domain_vec[0] = 1; - - if (h_sbr_data->coupling == COUPLING_BAL) { - target = (FIXP_SGL)SBR_ENERGY_PAN_OFFSET; - step = (FIXP_SGL)DECAY_COUPLING; - } - else { - target = FL2FXCONST_SGL(0.0f); - step = (FIXP_SGL)DECAY; - } - if (hHeaderData->bs_info.ampResolution == 0) { - target <<= 1; - step <<= 1; - } - - for (i=0; i < h_sbr_data->nScaleFactors; i++) { - if (h_prev_data->sfb_nrg_prev[i] > target) - h_sbr_data->iEnvelope[i] = -step; - else - h_sbr_data->iEnvelope[i] = step; - } - - /* Noisefloor levels are always cleared ... */ - - h_sbr_data->domain_vec_noise[0] = 1; - for (i=0; i < hHeaderData->freqBandData.nNfb; i++) - h_sbr_data->sbrNoiseFloorLevel[i] = FL2FXCONST_SGL(0.0f); - - /* ... and so are the sines */ - FDKmemclear(h_sbr_data->addHarmonics, MAX_FREQ_COEFFS); -} - - -/*! - \brief Build reference energies and noise levels from bitstream elements -*/ -static void -decodeEnvelope (HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ - HANDLE_SBR_FRAME_DATA h_sbr_data, /*!< pointer to current data */ - HANDLE_SBR_PREV_FRAME_DATA h_prev_data, /*!< pointer to data of last frame */ - HANDLE_SBR_PREV_FRAME_DATA otherChannel /*!< other channel's last frame data */ - ) -{ - int i; - int fFrameError = hHeaderData->frameErrorFlag; - FIXP_SGL tempSfbNrgPrev[MAX_FREQ_COEFFS]; - - if (!fFrameError) { - /* - To avoid distortions after bad frames, set the error flag if delta coding in time occurs. - However, SBR can take a little longer to come up again. - */ - if ( h_prev_data->frameErrorFlag ) { - if (h_sbr_data->domain_vec[0] != 0) { - fFrameError = 1; - } - } else { - /* Check that the previous stop position and the current start position match. - (Could be done in checkFrameInfo(), but the previous frame data is not available there) */ - if ( h_sbr_data->frameInfo.borders[0] != h_prev_data->stopPos - hHeaderData->numberTimeSlots ) { - /* Both the previous as well as the current frame are flagged to be ok, but they do not match! */ - if (h_sbr_data->domain_vec[0] == 1) { - /* Prefer concealment over delta-time coding between the mismatching frames */ - fFrameError = 1; - } - else { - /* Close the gap in time by triggering timeCompensateFirstEnvelope() */ - fFrameError = 1; - } - } - } - } - - - if (fFrameError) /* Error is detected */ - { - leanSbrConcealment(hHeaderData, - h_sbr_data, - h_prev_data); - - /* decode the envelope data to linear PCM */ - deltaToLinearPcmEnvelopeDecoding (hHeaderData, h_sbr_data, h_prev_data); - } - else /*Do a temporary dummy decoding and check that the envelope values are within limits */ - { - if (h_prev_data->frameErrorFlag) { - timeCompensateFirstEnvelope (hHeaderData, h_sbr_data, h_prev_data); - if (h_sbr_data->coupling != h_prev_data->coupling) { - /* - Coupling mode has changed during concealment. - The stored energy levels need to be converted. - */ - for (i = 0; i < hHeaderData->freqBandData.nSfb[1]; i++) { - /* Former Level-Channel will be used for both channels */ - if (h_prev_data->coupling == COUPLING_BAL) - h_prev_data->sfb_nrg_prev[i] = otherChannel->sfb_nrg_prev[i]; - /* Former L/R will be combined as the new Level-Channel */ - else if (h_sbr_data->coupling == COUPLING_LEVEL) - h_prev_data->sfb_nrg_prev[i] = (h_prev_data->sfb_nrg_prev[i] + otherChannel->sfb_nrg_prev[i]) >> 1; - else if (h_sbr_data->coupling == COUPLING_BAL) - h_prev_data->sfb_nrg_prev[i] = (FIXP_SGL)SBR_ENERGY_PAN_OFFSET; - } - } - } - FDKmemcpy (tempSfbNrgPrev, h_prev_data->sfb_nrg_prev, - MAX_FREQ_COEFFS * sizeof (FIXP_SGL)); - - deltaToLinearPcmEnvelopeDecoding (hHeaderData, h_sbr_data, h_prev_data); - - fFrameError = checkEnvelopeData (hHeaderData, h_sbr_data, h_prev_data); - - if (fFrameError) - { - hHeaderData->frameErrorFlag = 1; - FDKmemcpy (h_prev_data->sfb_nrg_prev, tempSfbNrgPrev, - MAX_FREQ_COEFFS * sizeof (FIXP_SGL)); - decodeEnvelope (hHeaderData, h_sbr_data, h_prev_data, otherChannel); - return; - } - } - - requantizeEnvelopeData (h_sbr_data, h_sbr_data->ampResolutionCurrentFrame); - - hHeaderData->frameErrorFlag = fFrameError; -} - - -/*! - \brief Verify that envelope energies are within the allowed range - \return 0 if all is fine, 1 if an envelope value was too high -*/ -static int -checkEnvelopeData (HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ - HANDLE_SBR_FRAME_DATA h_sbr_data, /*!< pointer to current data */ - HANDLE_SBR_PREV_FRAME_DATA h_prev_data /*!< pointer to data of last frame */ - ) -{ - FIXP_SGL *iEnvelope = h_sbr_data->iEnvelope; - FIXP_SGL *sfb_nrg_prev = h_prev_data->sfb_nrg_prev; - int i = 0, errorFlag = 0; - FIXP_SGL sbr_max_energy = - (h_sbr_data->ampResolutionCurrentFrame == 1) ? SBR_MAX_ENERGY : (SBR_MAX_ENERGY << 1); - - /* - Range check for current energies - */ - for (i = 0; i < h_sbr_data->nScaleFactors; i++) { - if (iEnvelope[i] > sbr_max_energy) { - errorFlag = 1; - } - if (iEnvelope[i] < FL2FXCONST_SGL(0.0f)) { - errorFlag = 1; - /* iEnvelope[i] = FL2FXCONST_SGL(0.0f); */ - } - } - - /* - Range check for previous energies - */ - for (i = 0; i < hHeaderData->freqBandData.nSfb[1]; i++) { - sfb_nrg_prev[i] = fixMax(sfb_nrg_prev[i], FL2FXCONST_SGL(0.0f)); - sfb_nrg_prev[i] = fixMin(sfb_nrg_prev[i], sbr_max_energy); - } - - return (errorFlag); -} - - -/*! - \brief Verify that the noise levels are within the allowed range - - The function is equivalent to checkEnvelopeData(). - When the noise-levels are being decoded, it is already too late for - concealment. Therefore the noise levels are simply limited here. -*/ -static void -limitNoiseLevels(HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ - HANDLE_SBR_FRAME_DATA h_sbr_data) /*!< pointer to current data */ -{ - int i; - int nNfb = hHeaderData->freqBandData.nNfb; - - /* - Set range limits. The exact values depend on the coupling mode. - However this limitation is primarily intended to avoid unlimited - accumulation of the delta-coded noise levels. - */ - #define lowerLimit ((FIXP_SGL)0) /* lowerLimit actually refers to the _highest_ noise energy */ - #define upperLimit ((FIXP_SGL)35) /* upperLimit actually refers to the _lowest_ noise energy */ - - /* - Range check for current noise levels - */ - for (i = 0; i < h_sbr_data->frameInfo.nNoiseEnvelopes * nNfb; i++) { - h_sbr_data->sbrNoiseFloorLevel[i] = fixMin(h_sbr_data->sbrNoiseFloorLevel[i], upperLimit); - h_sbr_data->sbrNoiseFloorLevel[i] = fixMax(h_sbr_data->sbrNoiseFloorLevel[i], lowerLimit); - } -} - - -/*! - \brief Compensate for the wrong timing that might occur after a frame error. -*/ -static void -timeCompensateFirstEnvelope (HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ - HANDLE_SBR_FRAME_DATA h_sbr_data, /*!< pointer to actual data */ - HANDLE_SBR_PREV_FRAME_DATA h_prev_data) /*!< pointer to data of last frame */ -{ - int i, nScalefactors; - FRAME_INFO *pFrameInfo = &h_sbr_data->frameInfo; - UCHAR *nSfb = hHeaderData->freqBandData.nSfb; - int estimatedStartPos = h_prev_data->stopPos - hHeaderData->numberTimeSlots; - int refLen, newLen, shift; - FIXP_SGL deltaExp; - - /* Original length of first envelope according to bitstream */ - refLen = pFrameInfo->borders[1] - pFrameInfo->borders[0]; - /* Corrected length of first envelope (concealing can make the first envelope longer) */ - newLen = pFrameInfo->borders[1] - estimatedStartPos; - - if (newLen <= 0) { - /* An envelope length of <= 0 would not work, so we don't use it. - May occur if the previous frame was flagged bad due to a mismatch - of the old and new frame infos. */ - newLen = refLen; - estimatedStartPos = pFrameInfo->borders[0]; - } - - deltaExp = FDK_getNumOctavesDiv8(newLen, refLen); - - /* Shift by -3 to rescale ld-table, ampRes-1 to enable coarser steps */ - shift = (FRACT_BITS - 1 - ENV_EXP_FRACT - 1 + h_sbr_data->ampResolutionCurrentFrame - 3); - deltaExp = deltaExp >> shift; - pFrameInfo->borders[0] = estimatedStartPos; - pFrameInfo->bordersNoise[0] = estimatedStartPos; - - if (h_sbr_data->coupling != COUPLING_BAL) { - nScalefactors = (pFrameInfo->freqRes[0]) ? nSfb[1] : nSfb[0]; - - for (i = 0; i < nScalefactors; i++) - h_sbr_data->iEnvelope[i] = h_sbr_data->iEnvelope[i] + deltaExp; - } -} - - - -/*! - \brief Convert each envelope value from logarithmic to linear domain - - Energy levels are transmitted in powers of 2, i.e. only the exponent - is extracted from the bitstream. - Therefore, normally only integer exponents can occur. However during - fading (in case of a corrupt bitstream), a fractional part can also - occur. The data in the array iEnvelope is shifted left by ENV_EXP_FRACT - compared to an integer representation so that numbers smaller than 1 - can be represented. - - This function calculates a mantissa corresponding to the fractional - part of the exponent for each reference energy. The array iEnvelope - is converted in place to save memory. Input and output data must - be interpreted differently, as shown in the below figure: - - \image html EnvelopeData.png - - The data is then used in calculateSbrEnvelope(). -*/ -static void -requantizeEnvelopeData (HANDLE_SBR_FRAME_DATA h_sbr_data, int ampResolution) -{ - int i; - FIXP_SGL mantissa; - int ampShift = 1 - ampResolution; - int exponent; - - /* In case that ENV_EXP_FRACT is changed to something else but 0 or 8, - the initialization of this array has to be adapted! - */ -#if ENV_EXP_FRACT - static const FIXP_SGL pow2[ENV_EXP_FRACT] = - { - FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 1))), /* 0.7071 */ - FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 2))), /* 0.5946 */ - FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 3))), - FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 4))), - FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 5))), - FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 6))), - FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 7))), - FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 8))) /* 0.5013 */ - }; - - int bit, mask; -#endif - - for (i = 0; i < h_sbr_data->nScaleFactors; i++) { - exponent = (LONG)h_sbr_data->iEnvelope[i]; - -#if ENV_EXP_FRACT - - exponent = exponent >> ampShift; - mantissa = 0.5f; - - /* Amplify mantissa according to the fractional part of the - exponent (result will be between 0.500000 and 0.999999) - */ - mask = 1; /* begin with lowest bit of exponent */ - - for ( bit=ENV_EXP_FRACT-1; bit>=0; bit-- ) { - if (exponent & mask) { - /* The current bit of the exponent is set, - multiply mantissa with the corresponding factor: */ - mantissa = (FIXP_SGL)( (mantissa * pow2[bit]) << 1); - } - /* Advance to next bit */ - mask = mask << 1; - } - - /* Make integer part of exponent right aligned */ - exponent = exponent >> ENV_EXP_FRACT; - -#else - /* In case of the high amplitude resolution, 1 bit of the exponent gets lost by the shift. - This will be compensated by a mantissa of 0.5*sqrt(2) instead of 0.5 if that bit is 1. */ - mantissa = (exponent & ampShift) ? FL2FXCONST_SGL(0.707106781186548f) : FL2FXCONST_SGL(0.5f); - exponent = exponent >> ampShift; -#endif - - /* - Mantissa was set to 0.5 (instead of 1.0, therefore increase exponent by 1). - Multiply by L=nChannels=64 by increasing exponent by another 6. - => Increase exponent by 7 - */ - exponent += 7 + NRG_EXP_OFFSET; - - /* Combine mantissa and exponent and write back the result */ - h_sbr_data->iEnvelope[i] = (FIXP_SGL)(((LONG)mantissa & MASK_M) | (exponent & MASK_E)); - - } -} - - -/*! - \brief Build new reference energies from old ones and delta coded data -*/ -static void -deltaToLinearPcmEnvelopeDecoding (HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ - HANDLE_SBR_FRAME_DATA h_sbr_data, /*!< pointer to current data */ - HANDLE_SBR_PREV_FRAME_DATA h_prev_data) /*!< pointer to previous data */ -{ - int i, domain, no_of_bands, band, freqRes; - - FIXP_SGL *sfb_nrg_prev = h_prev_data->sfb_nrg_prev; - FIXP_SGL *ptr_nrg = h_sbr_data->iEnvelope; - - int offset = 2 * hHeaderData->freqBandData.nSfb[0] - hHeaderData->freqBandData.nSfb[1]; - - for (i = 0; i < h_sbr_data->frameInfo.nEnvelopes; i++) { - domain = h_sbr_data->domain_vec[i]; - freqRes = h_sbr_data->frameInfo.freqRes[i]; - - FDK_ASSERT(freqRes >= 0 && freqRes <= 1); - - no_of_bands = hHeaderData->freqBandData.nSfb[freqRes]; - - FDK_ASSERT(no_of_bands < (64)); - - if (domain == 0) - { - mapLowResEnergyVal(*ptr_nrg, sfb_nrg_prev, offset, 0, freqRes); - ptr_nrg++; - for (band = 1; band < no_of_bands; band++) - { - *ptr_nrg = *ptr_nrg + *(ptr_nrg-1); - mapLowResEnergyVal(*ptr_nrg, sfb_nrg_prev, offset, band, freqRes); - ptr_nrg++; - } - } - else - { - for (band = 0; band < no_of_bands; band++) - { - *ptr_nrg = *ptr_nrg + sfb_nrg_prev[indexLow2High(offset, band, freqRes)]; - mapLowResEnergyVal(*ptr_nrg, sfb_nrg_prev, offset, band, freqRes); - ptr_nrg++; - } - } - } -} - - -/*! - \brief Build new noise levels from old ones and delta coded data -*/ -static void -decodeNoiseFloorlevels (HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ - HANDLE_SBR_FRAME_DATA h_sbr_data, /*!< pointer to current data */ - HANDLE_SBR_PREV_FRAME_DATA h_prev_data) /*!< pointer to previous data */ -{ - int i; - int nNfb = hHeaderData->freqBandData.nNfb; - int nNoiseFloorEnvelopes = h_sbr_data->frameInfo.nNoiseEnvelopes; - - /* Decode first noise envelope */ - - if (h_sbr_data->domain_vec_noise[0] == 0) { - FIXP_SGL noiseLevel = h_sbr_data->sbrNoiseFloorLevel[0]; - for (i = 1; i < nNfb; i++) { - noiseLevel += h_sbr_data->sbrNoiseFloorLevel[i]; - h_sbr_data->sbrNoiseFloorLevel[i] = noiseLevel; - } - } - else { - for (i = 0; i < nNfb; i++) { - h_sbr_data->sbrNoiseFloorLevel[i] += h_prev_data->prevNoiseLevel[i]; - } - } - - /* If present, decode the second noise envelope - Note: nNoiseFloorEnvelopes can only be 1 or 2 */ - - if (nNoiseFloorEnvelopes > 1) { - if (h_sbr_data->domain_vec_noise[1] == 0) { - FIXP_SGL noiseLevel = h_sbr_data->sbrNoiseFloorLevel[nNfb]; - for (i = nNfb + 1; i < 2*nNfb; i++) { - noiseLevel += h_sbr_data->sbrNoiseFloorLevel[i]; - h_sbr_data->sbrNoiseFloorLevel[i] = noiseLevel; - } - } - else { - for (i = 0; i < nNfb; i++) { - h_sbr_data->sbrNoiseFloorLevel[i + nNfb] += h_sbr_data->sbrNoiseFloorLevel[i]; - } - } - } - - limitNoiseLevels(hHeaderData, h_sbr_data); - - /* Update prevNoiseLevel with the last noise envelope */ - for (i = 0; i < nNfb; i++) - h_prev_data->prevNoiseLevel[i] = h_sbr_data->sbrNoiseFloorLevel[i + nNfb*(nNoiseFloorEnvelopes-1)]; - - - /* Requantize the noise floor levels in COUPLING_OFF-mode */ - if (!h_sbr_data->coupling) { - int nf_e; - - for (i = 0; i < nNoiseFloorEnvelopes*nNfb; i++) { - nf_e = 6 - (LONG)h_sbr_data->sbrNoiseFloorLevel[i] + 1 + NOISE_EXP_OFFSET; - /* +1 to compensate for a mantissa of 0.5 instead of 1.0 */ - - h_sbr_data->sbrNoiseFloorLevel[i] = - (FIXP_SGL)( ((LONG)FL2FXCONST_SGL(0.5f)) + /* mantissa */ - (nf_e & MASK_E) ); /* exponent */ - - } - } -} diff --git a/libSBRdec/src/env_dec.h b/libSBRdec/src/env_dec.h deleted file mode 100644 index 6f6dae3..0000000 --- a/libSBRdec/src/env_dec.h +++ /dev/null @@ -1,101 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief Envelope decoding -*/ -#ifndef __ENV_DEC_H -#define __ENV_DEC_H - -#include "sbrdecoder.h" -#include "env_extr.h" - -void decodeSbrData (HANDLE_SBR_HEADER_DATA hHeaderData, - HANDLE_SBR_FRAME_DATA h_data_left, - HANDLE_SBR_PREV_FRAME_DATA h_prev_data_left, - HANDLE_SBR_FRAME_DATA h_data_right, - HANDLE_SBR_PREV_FRAME_DATA h_prev_data_right); - - -#endif diff --git a/libSBRdec/src/env_extr.cpp b/libSBRdec/src/env_extr.cpp deleted file mode 100644 index 4d53a13..0000000 --- a/libSBRdec/src/env_extr.cpp +++ /dev/null @@ -1,1398 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief Envelope extraction - The functions provided by this module are mostly called by applySBR(). After it is - determined that there is valid SBR data, sbrGetHeaderData() might be called if the current - SBR data contains an \ref SBR_HEADER_ELEMENT as opposed to a \ref SBR_STANDARD_ELEMENT. This function - may return various error codes as defined in #SBR_HEADER_STATUS . Most importantly it returns HEADER_RESET when decoder - settings need to be recalculated according to the SBR specifications. In that case applySBR() - will initiatite the required re-configuration. - - The header data is stored in a #SBR_HEADER_DATA structure. - - The actual SBR data for the current frame is decoded into SBR_FRAME_DATA stuctures by sbrGetChannelPairElement() - [for stereo streams] and sbrGetSingleChannelElement() [for mono streams]. There is no fractional arithmetic involved. - - Once the information is extracted, the data needs to be further prepared before the actual decoding process. - This is done in decodeSbrData(). - - \sa Description of buffer management in applySBR(). \ref documentationOverview - -

About the SBR data format:

- - Each frame includes SBR data (side chain information), and can be either the \ref SBR_HEADER_ELEMENT or the \ref SBR_STANDARD_ELEMENT. - Parts of the data can be protected by a CRC checksum. - - \anchor SBR_HEADER_ELEMENT

The SBR_HEADER_ELEMENT

- - The SBR_HEADER_ELEMENT can be transmitted with every frame, however, it typically is send every second or so. It contains fundamental - information such as SBR sampling frequency and frequency range as well as control signals that do not require frequent changes. It also - includes the \ref SBR_STANDARD_ELEMENT. - - Depending on the changes between the information in a current SBR_HEADER_ELEMENT and the previous SBR_HEADER_ELEMENT, the SBR decoder might need - to be reset and reconfigured (e.g. new tables need to be calculated). - - \anchor SBR_STANDARD_ELEMENT

The SBR_STANDARD_ELEMENT

- - This data can be subdivided into "side info" and "raw data", where side info is defined as signals needed to decode the raw data - and some decoder tuning signals. Raw data is referred to as PCM and Huffman coded envelope and noise floor estimates. The side info also - includes information about the time-frequency grid for the current frame. - - \sa \ref documentationOverview -*/ - -#include "env_extr.h" - -#include "sbr_ram.h" -#include "sbr_rom.h" -#include "huff_dec.h" - - -#include "psbitdec.h" - -#define DRM_PARAMETRIC_STEREO 0 -#define EXTENSION_ID_PS_CODING 2 - - -static int extractFrameInfo (HANDLE_FDK_BITSTREAM hBs, - HANDLE_SBR_HEADER_DATA hHeaderData, - HANDLE_SBR_FRAME_DATA h_frame_data, - const UINT nrOfChannels, - const UINT flags - ); - - -static int sbrGetEnvelope (HANDLE_SBR_HEADER_DATA hHeaderData, - HANDLE_SBR_FRAME_DATA h_frame_data, - HANDLE_FDK_BITSTREAM hBs, - const UINT flags); - -static void sbrGetDirectionControlData (HANDLE_SBR_FRAME_DATA hFrameData, - HANDLE_FDK_BITSTREAM hBs); - -static void sbrGetNoiseFloorData (HANDLE_SBR_HEADER_DATA hHeaderData, - HANDLE_SBR_FRAME_DATA h_frame_data, - HANDLE_FDK_BITSTREAM hBs); - -static int checkFrameInfo (FRAME_INFO *pFrameInfo, int numberOfTimeSlots, int overlap, int timeStep); - -SBR_ERROR -initHeaderData ( - HANDLE_SBR_HEADER_DATA hHeaderData, - const int sampleRateIn, - const int sampleRateOut, - const int samplesPerFrame, - const UINT flags - ) -{ - HANDLE_FREQ_BAND_DATA hFreq = &hHeaderData->freqBandData; - SBR_ERROR sbrError = SBRDEC_OK; - int numAnalysisBands; - - if ( sampleRateIn == sampleRateOut ) { - hHeaderData->sbrProcSmplRate = sampleRateOut<<1; - numAnalysisBands = 32; - } else { - hHeaderData->sbrProcSmplRate = sampleRateOut; - if ( (sampleRateOut>>1) == sampleRateIn) { - /* 1:2 */ - numAnalysisBands = 32; - } else if ( (sampleRateOut>>2) == sampleRateIn ) { - /* 1:4 */ - numAnalysisBands = 32; - } else if ( (sampleRateOut*3)>>3 == (sampleRateIn*8)>>3 ) { - /* 3:8, 3/4 core frame length */ - numAnalysisBands = 24; - } else { - sbrError = SBRDEC_UNSUPPORTED_CONFIG; - goto bail; - } - } - - /* Fill in default values first */ - hHeaderData->syncState = SBR_NOT_INITIALIZED; - hHeaderData->status = 0; - hHeaderData->frameErrorFlag = 0; - - hHeaderData->bs_info.ampResolution = 1; - hHeaderData->bs_info.xover_band = 0; - hHeaderData->bs_info.sbr_preprocessing = 0; - - hHeaderData->bs_data.startFreq = 5; - hHeaderData->bs_data.stopFreq = 0; - hHeaderData->bs_data.freqScale = 2; - hHeaderData->bs_data.alterScale = 1; - hHeaderData->bs_data.noise_bands = 2; - hHeaderData->bs_data.limiterBands = 2; - hHeaderData->bs_data.limiterGains = 2; - hHeaderData->bs_data.interpolFreq = 1; - hHeaderData->bs_data.smoothingLength = 1; - - hHeaderData->timeStep = (flags & SBRDEC_ELD_GRID) ? 1 : 2; - - /* Setup pointers to frequency band tables */ - hFreq->freqBandTable[0] = hFreq->freqBandTableLo; - hFreq->freqBandTable[1] = hFreq->freqBandTableHi; - - /* Patch some entries */ - if (sampleRateOut > 24000) { /* Trigger an error if SBR is going to be processed without */ - hHeaderData->bs_data.startFreq = 7; /* having read these frequency values from bit stream before. */ - hHeaderData->bs_data.stopFreq = 3; - } - - /* One SBR timeslot corresponds to the amount of samples equal to the amount of analysis bands, divided by the timestep. */ - hHeaderData->numberTimeSlots = (samplesPerFrame/numAnalysisBands) >> (hHeaderData->timeStep - 1); - if (hHeaderData->numberTimeSlots > (16)) { - sbrError = SBRDEC_UNSUPPORTED_CONFIG; - } - - hHeaderData->numberOfAnalysisBands = numAnalysisBands; - -bail: - return sbrError; -} - - -/*! - \brief Initialize the SBR_PREV_FRAME_DATA struct -*/ -void -initSbrPrevFrameData (HANDLE_SBR_PREV_FRAME_DATA h_prev_data, /*!< handle to struct SBR_PREV_FRAME_DATA */ - int timeSlots) /*!< Framelength in SBR-timeslots */ -{ - int i; - - /* Set previous energy and noise levels to 0 for the case - that decoding starts in the middle of a bitstream */ - for (i=0; i < MAX_FREQ_COEFFS; i++) - h_prev_data->sfb_nrg_prev[i] = (FIXP_DBL)0; - for (i=0; i < MAX_NOISE_COEFFS; i++) - h_prev_data->prevNoiseLevel[i] = (FIXP_DBL)0; - for (i=0; i < MAX_INVF_BANDS; i++) - h_prev_data->sbr_invf_mode[i] = INVF_OFF; - - h_prev_data->stopPos = timeSlots; - h_prev_data->coupling = COUPLING_OFF; - h_prev_data->ampRes = 0; -} - - -/*! - \brief Read header data from bitstream - - \return error status - 0 if ok -*/ -SBR_HEADER_STATUS -sbrGetHeaderData (HANDLE_SBR_HEADER_DATA hHeaderData, - HANDLE_FDK_BITSTREAM hBs, - const UINT flags, - const int fIsSbrData) -{ - SBR_HEADER_DATA_BS *pBsData; - SBR_HEADER_DATA_BS lastHeader; - SBR_HEADER_DATA_BS_INFO lastInfo; - int headerExtra1=0, headerExtra2=0; - - /* Copy SBR bit stream header to temporary header */ - lastHeader = hHeaderData->bs_data; - lastInfo = hHeaderData->bs_info; - - /* Read new header from bitstream */ - { - pBsData = &hHeaderData->bs_data; - } - - { - hHeaderData->bs_info.ampResolution = FDKreadBits (hBs, 1); - } - - pBsData->startFreq = FDKreadBits (hBs, 4); - pBsData->stopFreq = FDKreadBits (hBs, 4); - - { - hHeaderData->bs_info.xover_band = FDKreadBits (hBs, 3); - FDKreadBits (hBs, 2); - } - - headerExtra1 = FDKreadBits (hBs, 1); - headerExtra2 = FDKreadBits (hBs, 1); - - /* Handle extra header information */ - if( headerExtra1) - { - pBsData->freqScale = FDKreadBits (hBs, 2); - pBsData->alterScale = FDKreadBits (hBs, 1); - pBsData->noise_bands = FDKreadBits (hBs, 2); - } - else { - pBsData->freqScale = 2; - pBsData->alterScale = 1; - pBsData->noise_bands = 2; - } - - if (headerExtra2) { - pBsData->limiterBands = FDKreadBits (hBs, 2); - pBsData->limiterGains = FDKreadBits (hBs, 2); - pBsData->interpolFreq = FDKreadBits (hBs, 1); - pBsData->smoothingLength = FDKreadBits (hBs, 1); - } - else { - pBsData->limiterBands = 2; - pBsData->limiterGains = 2; - pBsData->interpolFreq = 1; - pBsData->smoothingLength = 1; - } - - /* Look for new settings. IEC 14496-3, 4.6.18.3.1 */ - if(hHeaderData->syncState < SBR_HEADER || - lastHeader.startFreq != pBsData->startFreq || - lastHeader.stopFreq != pBsData->stopFreq || - lastHeader.freqScale != pBsData->freqScale || - lastHeader.alterScale != pBsData->alterScale || - lastHeader.noise_bands != pBsData->noise_bands || - lastInfo.xover_band != hHeaderData->bs_info.xover_band) { - return HEADER_RESET; /* New settings */ - } - - return HEADER_OK; -} - -/*! - \brief Get missing harmonics parameters (only used for AAC+SBR) - - \return error status - 0 if ok -*/ -int -sbrGetSyntheticCodedData(HANDLE_SBR_HEADER_DATA hHeaderData, - HANDLE_SBR_FRAME_DATA hFrameData, - HANDLE_FDK_BITSTREAM hBs) -{ - int i, bitsRead = 0; - - int flag = FDKreadBits(hBs,1); - bitsRead++; - - if(flag){ - for(i=0;ifreqBandData.nSfb[1];i++){ - hFrameData->addHarmonics[i] = FDKreadBits (hBs, 1 ); - bitsRead++; - } - } - else { - for(i=0; iaddHarmonics[i] = 0; - } - return(bitsRead); -} - -/*! - \brief Reads extension data from the bitstream - - The bitstream format allows up to 4 kinds of extended data element. - Extended data may contain several elements, each identified by a 2-bit-ID. - So far, no extended data elements are defined hence the first 2 parameters - are unused. The data should be skipped in order to update the number - of read bits for the consistency check in applySBR(). -*/ -static int extractExtendedData( - HANDLE_SBR_HEADER_DATA hHeaderData, /*!< handle to SBR header */ - HANDLE_FDK_BITSTREAM hBs /*!< Handle to the bit buffer */ - ,HANDLE_PS_DEC hParametricStereoDec /*!< Parametric Stereo Decoder */ - ) { - INT nBitsLeft; - int extended_data; - int i, frameOk = 1; - - - extended_data = FDKreadBits(hBs, 1); - - if (extended_data) { - int cnt; - int bPsRead = 0; - - cnt = FDKreadBits(hBs, 4); - if (cnt == (1<<4)-1) - cnt += FDKreadBits(hBs, 8); - - - nBitsLeft = 8 * cnt; - - /* sanity check for cnt */ - if (nBitsLeft > (INT)FDKgetValidBits(hBs)) { - /* limit nBitsLeft */ - nBitsLeft = (INT)FDKgetValidBits(hBs); - /* set frame error */ - frameOk = 0; - } - - while (nBitsLeft > 7) { - int extension_id = FDKreadBits(hBs, 2); - nBitsLeft -= 2; - - switch(extension_id) { - - - - case EXTENSION_ID_PS_CODING: - - /* Read PS data from bitstream */ - - if (hParametricStereoDec != NULL) { - if(bPsRead && !hParametricStereoDec->bsData[hParametricStereoDec->bsReadSlot].mpeg.bPsHeaderValid) { - cnt = nBitsLeft >> 3; /* number of remaining bytes */ - for (i=0; i> 3; /* number of remaining bytes */ - for (i=0; icoupling = COUPLING_OFF; - - { - /* Reserved bits */ - if (FDKreadBits(hBs, 1)) { /* bs_data_extra */ - FDKreadBits(hBs, 4); - if (flags & SBRDEC_SYNTAX_SCAL) { - FDKreadBits(hBs, 4); - } - } - } - - if (flags & SBRDEC_SYNTAX_SCAL) { - FDKreadBits (hBs, 1); /* bs_coupling */ - } - - /* - Grid control - */ - if ( !extractFrameInfo ( hBs, hHeaderData, hFrameData, 1, flags) ) - return 0; - - if ( !checkFrameInfo (&hFrameData->frameInfo, hHeaderData->numberTimeSlots, overlap, hHeaderData->timeStep) ) - return 0; - - - /* - Fetch domain vectors (time or frequency direction for delta-coding) - */ - sbrGetDirectionControlData (hFrameData, hBs); - - for (i=0; ifreqBandData.nInvfBands; i++) { - hFrameData->sbr_invf_mode[i] = - (INVF_MODE) FDKreadBits (hBs, 2); - } - - - - /* raw data */ - if ( !sbrGetEnvelope (hHeaderData, hFrameData, hBs, flags) ) - return 0; - - - sbrGetNoiseFloorData (hHeaderData, hFrameData, hBs); - - sbrGetSyntheticCodedData(hHeaderData, hFrameData, hBs); - - { - /* sbr extended data */ - if (! extractExtendedData( - hHeaderData, - hBs - ,hParametricStereoDec - )) { - return 0; - } - } - - return 1; -} - - - -/*! - \brief Read bitstream elements of a channel pair - \return SbrFrameOK -*/ -int -sbrGetChannelPairElement (HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ - HANDLE_SBR_FRAME_DATA hFrameDataLeft, /*!< Dynamic control data for first channel */ - HANDLE_SBR_FRAME_DATA hFrameDataRight,/*!< Dynamic control data for second channel */ - HANDLE_FDK_BITSTREAM hBs, /*!< handle to struct BIT_BUF */ - const UINT flags, - const int overlap ) -{ - int i, bit; - - - /* Reserved bits */ - if (FDKreadBits(hBs, 1)) { /* bs_data_extra */ - FDKreadBits(hBs, 4); - FDKreadBits(hBs, 4); - } - - /* Read coupling flag */ - bit = FDKreadBits (hBs, 1); - - if (bit) { - hFrameDataLeft->coupling = COUPLING_LEVEL; - hFrameDataRight->coupling = COUPLING_BAL; - } - else { - hFrameDataLeft->coupling = COUPLING_OFF; - hFrameDataRight->coupling = COUPLING_OFF; - } - - - /* - Grid control - */ - if ( !extractFrameInfo (hBs, hHeaderData, hFrameDataLeft, 2, flags) ) - return 0; - - if ( !checkFrameInfo (&hFrameDataLeft->frameInfo, hHeaderData->numberTimeSlots, overlap, hHeaderData->timeStep) ) - return 0; - - if (hFrameDataLeft->coupling) { - FDKmemcpy (&hFrameDataRight->frameInfo, &hFrameDataLeft->frameInfo, sizeof(FRAME_INFO)); - hFrameDataRight->ampResolutionCurrentFrame = hFrameDataLeft->ampResolutionCurrentFrame; - } - else { - if ( !extractFrameInfo (hBs, hHeaderData, hFrameDataRight, 2, flags) ) - return 0; - - if ( !checkFrameInfo (&hFrameDataRight->frameInfo, hHeaderData->numberTimeSlots, overlap, hHeaderData->timeStep) ) - return 0; - } - - /* - Fetch domain vectors (time or frequency direction for delta-coding) - */ - sbrGetDirectionControlData (hFrameDataLeft, hBs); - sbrGetDirectionControlData (hFrameDataRight, hBs); - - for (i=0; ifreqBandData.nInvfBands; i++) { - hFrameDataLeft->sbr_invf_mode[i] = (INVF_MODE) FDKreadBits (hBs, 2); - } - - if (hFrameDataLeft->coupling) { - for (i=0; ifreqBandData.nInvfBands; i++) { - hFrameDataRight->sbr_invf_mode[i] = hFrameDataLeft->sbr_invf_mode[i]; - } - - - if ( !sbrGetEnvelope (hHeaderData, hFrameDataLeft, hBs, flags) ) { - return 0; - } - - sbrGetNoiseFloorData (hHeaderData, hFrameDataLeft, hBs); - - if ( !sbrGetEnvelope (hHeaderData, hFrameDataRight, hBs, flags) ) { - return 0; - } - } - else { - - for (i=0; ifreqBandData.nInvfBands; i++) { - hFrameDataRight->sbr_invf_mode[i] = (INVF_MODE) FDKreadBits (hBs, 2); - } - - - - if ( !sbrGetEnvelope (hHeaderData, hFrameDataLeft, hBs, flags) ) - return 0; - - if ( !sbrGetEnvelope (hHeaderData, hFrameDataRight, hBs, flags) ) - return 0; - - sbrGetNoiseFloorData (hHeaderData, hFrameDataLeft, hBs); - - } - sbrGetNoiseFloorData (hHeaderData, hFrameDataRight, hBs); - - sbrGetSyntheticCodedData(hHeaderData, hFrameDataLeft, hBs); - sbrGetSyntheticCodedData(hHeaderData, hFrameDataRight, hBs); - - { - if (! extractExtendedData( - hHeaderData, - hBs - ,NULL - ) ) { - return 0; - } - } - - return 1; -} - - - - -/*! - \brief Read direction control data from bitstream -*/ -void -sbrGetDirectionControlData (HANDLE_SBR_FRAME_DATA h_frame_data, /*!< handle to struct SBR_FRAME_DATA */ - HANDLE_FDK_BITSTREAM hBs) /*!< handle to struct BIT_BUF */ -{ - int i; - - for (i = 0; i < h_frame_data->frameInfo.nEnvelopes; i++) { - h_frame_data->domain_vec[i] = FDKreadBits (hBs, 1); - } - - for (i = 0; i < h_frame_data->frameInfo.nNoiseEnvelopes; i++) { - h_frame_data->domain_vec_noise[i] = FDKreadBits (hBs, 1); - } -} - - - -/*! - \brief Read noise-floor-level data from bitstream -*/ -void -sbrGetNoiseFloorData (HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ - HANDLE_SBR_FRAME_DATA h_frame_data, /*!< handle to struct SBR_FRAME_DATA */ - HANDLE_FDK_BITSTREAM hBs) /*!< handle to struct BIT_BUF */ -{ - int i,j; - int delta; - COUPLING_MODE coupling; - int noNoiseBands = hHeaderData->freqBandData.nNfb; - - Huffman hcb_noiseF; - Huffman hcb_noise; - int envDataTableCompFactor; - - coupling = h_frame_data->coupling; - - - /* - Select huffman codebook depending on coupling mode - */ - if (coupling == COUPLING_BAL) { - hcb_noise = (Huffman)&FDK_sbrDecoder_sbr_huffBook_NoiseBalance11T; - hcb_noiseF = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvBalance11F; /* "sbr_huffBook_NoiseBalance11F" */ - envDataTableCompFactor = 1; - } - else { - hcb_noise = (Huffman)&FDK_sbrDecoder_sbr_huffBook_NoiseLevel11T; - hcb_noiseF = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvLevel11F; /* "sbr_huffBook_NoiseLevel11F" */ - envDataTableCompFactor = 0; - } - - /* - Read raw noise-envelope data - */ - for (i=0; iframeInfo.nNoiseEnvelopes; i++) { - - - if (h_frame_data->domain_vec_noise[i] == 0) { - if (coupling == COUPLING_BAL) { - h_frame_data->sbrNoiseFloorLevel[i*noNoiseBands] = - (FIXP_SGL) (((int)FDKreadBits (hBs, 5)) << envDataTableCompFactor); - } - else { - h_frame_data->sbrNoiseFloorLevel[i*noNoiseBands] = - (FIXP_SGL) (int)FDKreadBits (hBs, 5); - } - - for (j = 1; j < noNoiseBands; j++) { - delta = DecodeHuffmanCW(hcb_noiseF, hBs); - h_frame_data->sbrNoiseFloorLevel[i*noNoiseBands+j] = (FIXP_SGL) (delta << envDataTableCompFactor); - } - } - else { - for (j = 0; j < noNoiseBands; j++) { - delta = DecodeHuffmanCW(hcb_noise, hBs); - h_frame_data->sbrNoiseFloorLevel[i*noNoiseBands+j] = (FIXP_SGL) (delta << envDataTableCompFactor); - } - } - } -} - - -/*! - \brief Read envelope data from bitstream -*/ -static int -sbrGetEnvelope (HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ - HANDLE_SBR_FRAME_DATA h_frame_data, /*!< handle to struct SBR_FRAME_DATA */ - HANDLE_FDK_BITSTREAM hBs, /*!< handle to struct BIT_BUF */ - const UINT flags) -{ - int i, j; - UCHAR no_band[MAX_ENVELOPES]; - int delta = 0; - int offset = 0; - COUPLING_MODE coupling = h_frame_data->coupling; - int ampRes = hHeaderData->bs_info.ampResolution; - int nEnvelopes = h_frame_data->frameInfo.nEnvelopes; - int envDataTableCompFactor; - int start_bits, start_bits_balance; - Huffman hcb_t, hcb_f; - - h_frame_data->nScaleFactors = 0; - - if ( (h_frame_data->frameInfo.frameClass == 0) && (nEnvelopes == 1) ) { - if (flags & SBRDEC_ELD_GRID) - ampRes = h_frame_data->ampResolutionCurrentFrame; - else - ampRes = 0; - } - h_frame_data->ampResolutionCurrentFrame = ampRes; - - /* - Set number of bits for first value depending on amplitude resolution - */ - if(ampRes == 1) - { - start_bits = 6; - start_bits_balance = 5; - } - else - { - start_bits = 7; - start_bits_balance = 6; - } - - /* - Calculate number of values for each envelope and alltogether - */ - for (i = 0; i < nEnvelopes; i++) { - no_band[i] = hHeaderData->freqBandData.nSfb[h_frame_data->frameInfo.freqRes[i]]; - h_frame_data->nScaleFactors += no_band[i]; - } - if (h_frame_data->nScaleFactors > MAX_NUM_ENVELOPE_VALUES) - return 0; - - /* - Select Huffman codebook depending on coupling mode and amplitude resolution - */ - if (coupling == COUPLING_BAL) { - envDataTableCompFactor = 1; - if (ampRes == 0) { - hcb_t = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvBalance10T; - hcb_f = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvBalance10F; - } - else { - hcb_t = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvBalance11T; - hcb_f = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvBalance11F; - } - } - else { - envDataTableCompFactor = 0; - if (ampRes == 0) { - hcb_t = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvLevel10T; - hcb_f = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvLevel10F; - } - else { - hcb_t = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvLevel11T; - hcb_f = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvLevel11F; - } - } - - /* - Now read raw envelope data - */ - for (j = 0, offset = 0; j < nEnvelopes; j++) { - - - if (h_frame_data->domain_vec[j] == 0) { - if (coupling == COUPLING_BAL) { - h_frame_data->iEnvelope[offset] = - (FIXP_SGL) (( (int)FDKreadBits(hBs, start_bits_balance)) << envDataTableCompFactor); - } - else { - h_frame_data->iEnvelope[offset] = - (FIXP_SGL) (int)FDKreadBits (hBs, start_bits); - } - } - - for (i = (1 - h_frame_data->domain_vec[j]); i < no_band[j]; i++) { - - if (h_frame_data->domain_vec[j] == 0) { - delta = DecodeHuffmanCW(hcb_f, hBs); - } - else { - delta = DecodeHuffmanCW(hcb_t, hBs); - } - - h_frame_data->iEnvelope[offset + i] = (FIXP_SGL) (delta << envDataTableCompFactor); - } - offset += no_band[j]; - } - -#if ENV_EXP_FRACT - /* Convert from int to scaled fract (ENV_EXP_FRACT bits for the fractional part) */ - for (i = 0; i < h_frame_data->nScaleFactors; i++) { - h_frame_data->iEnvelope[i] <<= ENV_EXP_FRACT; - } -#endif - - return 1; -} - - -//static const FRAME_INFO v_frame_info1_8 = { 0, 1, {0, 8}, {1}, -1, 1, {0, 8} }; -static const FRAME_INFO v_frame_info2_8 = { 0, 2, {0, 4, 8}, {1, 1}, -1, 2, {0, 4, 8} }; -static const FRAME_INFO v_frame_info4_8 = { 0, 4, {0, 2, 4, 6, 8}, {1, 1, 1, 1}, -1, 2, {0, 4, 8} }; - -/***************************************************************************/ -/*! - \brief Generates frame info for FIXFIXonly frame class used for low delay version - - \return nothing - ****************************************************************************/ - static void generateFixFixOnly ( FRAME_INFO *hSbrFrameInfo, - int tranPosInternal, - int numberTimeSlots - ) -{ - int nEnv, i, tranIdx; - const int *pTable; - - switch (numberTimeSlots) { - case 8: - pTable = FDK_sbrDecoder_envelopeTable_8[tranPosInternal]; - break; - case 15: - pTable = FDK_sbrDecoder_envelopeTable_15[tranPosInternal]; - break; - case 16: - pTable = FDK_sbrDecoder_envelopeTable_16[tranPosInternal]; - break; - default: - FDK_ASSERT(0); - /* in case assertion checks are disabled, force a definite memory fault at first access */ - pTable = NULL; - break; - } - - /* look number of envelopes in table */ - nEnv = pTable[0]; - /* look up envelope distribution in table */ - for (i=1; iborders[i] = pTable[i+2]; - /* open and close frame border */ - hSbrFrameInfo->borders[0] = 0; - hSbrFrameInfo->borders[nEnv] = numberTimeSlots; - hSbrFrameInfo->nEnvelopes = nEnv; - - /* transient idx */ - tranIdx = hSbrFrameInfo->tranEnv = pTable[1]; - - /* add noise floors */ - hSbrFrameInfo->bordersNoise[0] = 0; - hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[tranIdx?tranIdx:1]; - hSbrFrameInfo->bordersNoise[2] = numberTimeSlots; - /* nEnv is always > 1, so nNoiseEnvelopes is always 2 (IEC 14496-3 4.6.19.3.2) */ - hSbrFrameInfo->nNoiseEnvelopes = 2; -} - -/*! - \brief Extracts LowDelaySBR control data from the bitstream. - - \return zero for bitstream error, one for correct. -*/ -static int -extractLowDelayGrid (HANDLE_FDK_BITSTREAM hBitBuf, /*!< bitbuffer handle */ - HANDLE_SBR_HEADER_DATA hHeaderData, - HANDLE_SBR_FRAME_DATA h_frame_data, /*!< contains the FRAME_INFO struct to be filled */ - int timeSlots - ) -{ - FRAME_INFO * pFrameInfo = &h_frame_data->frameInfo; - INT numberTimeSlots = hHeaderData->numberTimeSlots; - INT temp = 0, k; - - /* FIXFIXonly framing case */ - h_frame_data->frameInfo.frameClass = 0; - - /* get the transient position from the bitstream */ - switch (timeSlots){ - case 8: - /* 3bit transient position (temp={0;..;7}) */ - temp = FDKreadBits( hBitBuf, 3); - break; - - case 16: - case 15: - /* 4bit transient position (temp={0;..;15}) */ - temp = FDKreadBits( hBitBuf, 4); - break; - - default: - return 0; - } - - /* calculate borders according to the transient position */ - generateFixFixOnly ( pFrameInfo, - temp, - numberTimeSlots - ); - - /* decode freq res: */ - for (k = 0; k < pFrameInfo->nEnvelopes; k++) { - pFrameInfo->freqRes[k] = (UCHAR) FDKreadBits (hBitBuf, 1); /* f = F [1 bits] */ - } - - - return 1; -} - -/*! - \brief Extract the frame information (structure FRAME_INFO) from the bitstream - \return Zero for bitstream error, one for correct. -*/ -int -extractFrameInfo ( HANDLE_FDK_BITSTREAM hBs, /*!< bitbuffer handle */ - HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ - HANDLE_SBR_FRAME_DATA h_frame_data, /*!< pointer to memory where the frame-info will be stored */ - const UINT nrOfChannels, - const UINT flags - ) -{ - FRAME_INFO * pFrameInfo = &h_frame_data->frameInfo; - int numberTimeSlots = hHeaderData->numberTimeSlots; - int pointer_bits = 0, nEnv = 0, b = 0, border, i, n = 0, - k, p, aL, aR, nL, nR, - temp = 0, staticFreqRes; - UCHAR frameClass; - - if (flags & SBRDEC_ELD_GRID) { - /* CODEC_AACLD (LD+SBR) only uses the normal 0 Grid for non-transient Frames and the LowDelayGrid for transient Frames */ - frameClass = FDKreadBits (hBs, 1); /* frameClass = [1 bit] */ - if ( frameClass == 1 ) { - /* if frameClass == 1, extract LowDelaySbrGrid, otherwise extract normal SBR-Grid for FIXIFX */ - /* extract the AACLD-Sbr-Grid */ - pFrameInfo->frameClass = frameClass; - extractLowDelayGrid (hBs, hHeaderData, h_frame_data, numberTimeSlots); - return 1; - } - } else - { - frameClass = FDKreadBits (hBs, 2); /* frameClass = C [2 bits] */ - } - - - switch (frameClass) { - case 0: - temp = FDKreadBits (hBs, 2); /* E [2 bits ] */ - nEnv = (int) (1 << temp); /* E -> e */ - - if ((flags & SBRDEC_ELD_GRID) && (nEnv == 1)) - h_frame_data->ampResolutionCurrentFrame = FDKreadBits( hBs, 1); /* new ELD Syntax 07-11-09 */ - - staticFreqRes = FDKreadBits (hBs, 1); - - { - if (nEnv > MAX_ENVELOPES_HEAAC) - return 0; - } - - b = nEnv + 1; - switch (nEnv) { - case 1: - switch (numberTimeSlots) { - case 15: - FDKmemcpy (pFrameInfo, &FDK_sbrDecoder_sbr_frame_info1_15, sizeof(FRAME_INFO)); - break; - case 16: - FDKmemcpy (pFrameInfo, &FDK_sbrDecoder_sbr_frame_info1_16, sizeof(FRAME_INFO)); - break; - default: - FDK_ASSERT(0); - } - break; - case 2: - switch (numberTimeSlots) { - case 15: - FDKmemcpy (pFrameInfo, &FDK_sbrDecoder_sbr_frame_info2_15, sizeof(FRAME_INFO)); - break; - case 16: - FDKmemcpy (pFrameInfo, &FDK_sbrDecoder_sbr_frame_info2_16, sizeof(FRAME_INFO)); - break; - default: - FDK_ASSERT(0); - } - break; - case 4: - switch (numberTimeSlots) { - case 15: - FDKmemcpy (pFrameInfo, &FDK_sbrDecoder_sbr_frame_info4_15, sizeof(FRAME_INFO)); - break; - case 16: - FDKmemcpy (pFrameInfo, &FDK_sbrDecoder_sbr_frame_info4_16, sizeof(FRAME_INFO)); - break; - default: - FDK_ASSERT(0); - } - break; - case 8: -#if (MAX_ENVELOPES >= 8) - switch (numberTimeSlots) { - case 15: - FDKmemcpy (pFrameInfo, &FDK_sbrDecoder_sbr_frame_info8_15, sizeof(FRAME_INFO)); - break; - case 16: - FDKmemcpy (pFrameInfo, &FDK_sbrDecoder_sbr_frame_info8_16, sizeof(FRAME_INFO)); - break; - default: - FDK_ASSERT(0); - } - break; -#else - return 0; -#endif - } - /* Apply correct freqRes (High is default) */ - if (!staticFreqRes) { - for (i = 0; i < nEnv ; i++) - pFrameInfo->freqRes[i] = 0; - } - - break; - case 1: - case 2: - temp = FDKreadBits (hBs, 2); /* A [2 bits] */ - - n = FDKreadBits (hBs, 2); /* n = N [2 bits] */ - - nEnv = n + 1; /* # envelopes */ - b = nEnv + 1; /* # borders */ - - break; - } - - switch (frameClass) { - case 1: - /* Decode borders: */ - pFrameInfo->borders[0] = 0; /* first border */ - border = temp + numberTimeSlots; /* A -> aR */ - i = b-1; /* frame info index for last border */ - pFrameInfo->borders[i] = border; /* last border */ - - for (k = 0; k < n; k++) { - temp = FDKreadBits (hBs, 2);/* R [2 bits] */ - border -= (2 * temp + 2); /* R -> r */ - pFrameInfo->borders[--i] = border; - } - - - /* Decode pointer: */ - pointer_bits = DFRACT_BITS - 1 - CountLeadingBits((FIXP_DBL)(n+1)); - p = FDKreadBits (hBs, pointer_bits); /* p = P [pointer_bits bits] */ - - if (p > n+1) - return 0; - - pFrameInfo->tranEnv = p ? n + 2 - p : -1; - - - /* Decode freq res: */ - for (k = n; k >= 0; k--) { - pFrameInfo->freqRes[k] = FDKreadBits (hBs, 1); /* f = F [1 bits] */ - } - - - /* Calculate noise floor middle border: */ - if (p == 0 || p == 1) - pFrameInfo->bordersNoise[1] = pFrameInfo->borders[n]; - else - pFrameInfo->bordersNoise[1] = pFrameInfo->borders[pFrameInfo->tranEnv]; - - break; - - case 2: - /* Decode borders: */ - border = temp; /* A -> aL */ - pFrameInfo->borders[0] = border; /* first border */ - - for (k = 1; k <= n; k++) { - temp = FDKreadBits (hBs, 2);/* R [2 bits] */ - border += (2 * temp + 2); /* R -> r */ - pFrameInfo->borders[k] = border; - } - pFrameInfo->borders[k] = numberTimeSlots; /* last border */ - - - /* Decode pointer: */ - pointer_bits = DFRACT_BITS - 1 - CountLeadingBits((FIXP_DBL)(n+1)); - p = FDKreadBits (hBs, pointer_bits); /* p = P [pointer_bits bits] */ - if (p > n+1) - return 0; - - if (p == 0 || p == 1) - pFrameInfo->tranEnv = -1; - else - pFrameInfo->tranEnv = p - 1; - - - - /* Decode freq res: */ - for (k = 0; k <= n; k++) { - pFrameInfo->freqRes[k] = FDKreadBits(hBs, 1); /* f = F [1 bits] */ - } - - - - /* Calculate noise floor middle border: */ - switch (p) { - case 0: - pFrameInfo->bordersNoise[1] = pFrameInfo->borders[1]; - break; - case 1: - pFrameInfo->bordersNoise[1] = pFrameInfo->borders[n]; - break; - default: - pFrameInfo->bordersNoise[1] = pFrameInfo->borders[pFrameInfo->tranEnv]; - break; - } - - break; - - case 3: - /* v_ctrlSignal = [frameClass,aL,aR,nL,nR,v_rL,v_rR,p,v_fLR]; */ - - aL = FDKreadBits (hBs, 2); /* AL [2 bits], AL -> aL */ - - aR = FDKreadBits (hBs, 2) + numberTimeSlots; /* AR [2 bits], AR -> aR */ - - nL = FDKreadBits (hBs, 2); /* nL = NL [2 bits] */ - - nR = FDKreadBits (hBs, 2); /* nR = NR [2 bits] */ - - - - /*------------------------------------------------------------------------- - Calculate help variables - --------------------------------------------------------------------------*/ - - /* general: */ - nEnv = nL + nR + 1; /* # envelopes */ - if (nEnv > MAX_ENVELOPES) - return 0; - b = nEnv + 1; /* # borders */ - - - - /*------------------------------------------------------------------------- - Decode envelopes - --------------------------------------------------------------------------*/ - - - /* L-borders: */ - border = aL; /* first border */ - pFrameInfo->borders[0] = border; - - for (k = 1; k <= nL; k++) { - temp = FDKreadBits (hBs, 2);/* R [2 bits] */ - border += (2 * temp + 2); /* R -> r */ - pFrameInfo->borders[k] = border; - } - - - /* R-borders: */ - border = aR; /* last border */ - i = nEnv; - - pFrameInfo->borders[i] = border; - - for (k = 0; k < nR; k++) { - temp = FDKreadBits (hBs, 2);/* R [2 bits] */ - border -= (2 * temp + 2); /* R -> r */ - pFrameInfo->borders[--i] = border; - } - - - /* decode pointer: */ - pointer_bits = DFRACT_BITS - 1 - CountLeadingBits((FIXP_DBL)(nL+nR+1)); - p = FDKreadBits (hBs, pointer_bits); /* p = P [pointer_bits bits] */ - - if (p > nL+nR+1) - return 0; - - pFrameInfo->tranEnv = p ? b - p : -1; - - - - /* decode freq res: */ - for (k = 0; k < nEnv; k++) { - pFrameInfo->freqRes[k] = FDKreadBits(hBs, 1); /* f = F [1 bits] */ - } - - - - /*------------------------------------------------------------------------- - Decode noise floors - --------------------------------------------------------------------------*/ - pFrameInfo->bordersNoise[0] = aL; - - if (nEnv == 1) { - /* 1 noise floor envelope: */ - pFrameInfo->bordersNoise[1] = aR; - } - else { - /* 2 noise floor envelopes */ - if (p == 0 || p == 1) - pFrameInfo->bordersNoise[1] = pFrameInfo->borders[nEnv - 1]; - else - pFrameInfo->bordersNoise[1] = pFrameInfo->borders[pFrameInfo->tranEnv]; - pFrameInfo->bordersNoise[2] = aR; - } - break; - } - - - /* - Store number of envelopes, noise floor envelopes and frame class - */ - pFrameInfo->nEnvelopes = nEnv; - - if (nEnv == 1) - pFrameInfo->nNoiseEnvelopes = 1; - else - pFrameInfo->nNoiseEnvelopes = 2; - - pFrameInfo->frameClass = frameClass; - - if (pFrameInfo->frameClass == 2 || pFrameInfo->frameClass == 1) { - /* calculate noise floor first and last borders: */ - pFrameInfo->bordersNoise[0] = pFrameInfo->borders[0]; - pFrameInfo->bordersNoise[pFrameInfo->nNoiseEnvelopes] = pFrameInfo->borders[nEnv]; - } - - - return 1; -} - - -/*! - \brief Check if the frameInfo vector has reasonable values. - \return Zero for error, one for correct -*/ -static int -checkFrameInfo (FRAME_INFO * pFrameInfo, /*!< pointer to frameInfo */ - int numberOfTimeSlots, /*!< QMF time slots per frame */ - int overlap, /*!< Amount of overlap QMF time slots */ - int timeStep) /*!< QMF slots to SBR slots step factor */ -{ - int maxPos,i,j; - int startPos; - int stopPos; - int tranEnv; - int startPosNoise; - int stopPosNoise; - int nEnvelopes = pFrameInfo->nEnvelopes; - int nNoiseEnvelopes = pFrameInfo->nNoiseEnvelopes; - - if(nEnvelopes < 1 || nEnvelopes > MAX_ENVELOPES) - return 0; - - if(nNoiseEnvelopes > MAX_NOISE_ENVELOPES) - return 0; - - startPos = pFrameInfo->borders[0]; - stopPos = pFrameInfo->borders[nEnvelopes]; - tranEnv = pFrameInfo->tranEnv; - startPosNoise = pFrameInfo->bordersNoise[0]; - stopPosNoise = pFrameInfo->bordersNoise[nNoiseEnvelopes]; - - if (overlap < 0 || overlap > (6)) { - return 0; - } - if (timeStep < 1 || timeStep > 2) { - return 0; - } - maxPos = numberOfTimeSlots + (overlap/timeStep); - - /* Check that the start and stop positions of the frame are reasonable values. */ - if( (startPos < 0) || (startPos >= stopPos) ) - return 0; - if( startPos > maxPos-numberOfTimeSlots ) /* First env. must start in or directly after the overlap buffer */ - return 0; - if( stopPos < numberOfTimeSlots ) /* One complete frame must be ready for output after processing */ - return 0; - if(stopPos > maxPos) - return 0; - - /* Check that the start border for every envelope is strictly later in time */ - for(i=0;iborders[i] >= pFrameInfo->borders[i+1]) - return 0; - } - - /* Check that the envelope to be shortened is actually among the envelopes */ - if(tranEnv>nEnvelopes) - return 0; - - - /* Check the noise borders */ - if(nEnvelopes==1 && nNoiseEnvelopes>1) - return 0; - - if(startPos != startPosNoise || stopPos != stopPosNoise) - return 0; - - - /* Check that the start border for every noise-envelope is strictly later in time*/ - for(i=0; ibordersNoise[i] >= pFrameInfo->bordersNoise[i+1]) - return 0; - } - - /* Check that every noise border is the same as an envelope border*/ - for(i=0; ibordersNoise[i]; - - for(j=0; jborders[j] == startPosNoise) - break; - } - if(j==nEnvelopes) - return 0; - } - - return 1; -} diff --git a/libSBRdec/src/env_extr.h b/libSBRdec/src/env_extr.h deleted file mode 100644 index 0518ea9..0000000 --- a/libSBRdec/src/env_extr.h +++ /dev/null @@ -1,324 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief Envelope extraction prototypes -*/ - -#ifndef __ENVELOPE_EXTRACTION_H -#define __ENVELOPE_EXTRACTION_H - -#include "sbrdecoder.h" - -#include "FDK_bitstream.h" -#include "lpp_tran.h" - -#include "psdec.h" - -#define ENV_EXP_FRACT 0 -/*!< Shift raw envelope data to support fractional numbers. - Can be set to 8 instead of 0 to enhance accuracy during concealment. - This is not required for conformance and #requantizeEnvelopeData() will - become more expensive. -*/ - -#define EXP_BITS 6 -/*!< Size of exponent-part of a pseudo float envelope value (should be at least 6). - The remaining bits in each word are used for the mantissa (should be at least 10). - This format is used in the arrays iEnvelope[] and sbrNoiseFloorLevel[] - in the FRAME_DATA struct which must fit in a certain part of the output buffer - (See buffer management in sbr_dec.cpp). - Exponents and mantissas could also be stored in separate arrays. - Accessing the exponent or the mantissa would be simplified and the masks #MASK_E - resp. #MASK_M would no longer be required. -*/ - -#define MASK_M (((1 << (FRACT_BITS - EXP_BITS)) - 1) << EXP_BITS) /*!< Mask for extracting the mantissa of a pseudo float envelope value */ -#define MASK_E ((1 << EXP_BITS) - 1) /*!< Mask for extracting the exponent of a pseudo float envelope value */ - -#define SIGN_EXT ( ((SCHAR)-1) ^ MASK_E) /*!< a CHAR-constant with all bits above our sign-bit set */ -#define ROUNDING ( (FIXP_SGL)(1<<(EXP_BITS-1)) ) /*!< 0.5-offset for rounding the mantissa of a pseudo-float envelope value */ -#define NRG_EXP_OFFSET 16 /*!< Will be added to the reference energy's exponent to prevent negative numbers */ -#define NOISE_EXP_OFFSET 38 /*!< Will be added to the noise level exponent to prevent negative numbers */ - -typedef enum -{ - HEADER_NOT_PRESENT, - HEADER_ERROR, - HEADER_OK, - HEADER_RESET -} -SBR_HEADER_STATUS; - -typedef enum -{ - SBR_NOT_INITIALIZED = 0, - UPSAMPLING = 1, - SBR_HEADER = 2, - SBR_ACTIVE = 3 -} -SBR_SYNC_STATE; - - -typedef enum -{ - COUPLING_OFF = 0, - COUPLING_LEVEL, - COUPLING_BAL -} -COUPLING_MODE; - -typedef struct -{ - UCHAR nSfb[2]; /*!< Number of SBR-bands for low and high freq-resolution */ - UCHAR nNfb; /*!< Actual number of noise bands to read from the bitstream*/ - UCHAR numMaster; /*!< Number of SBR-bands in v_k_master */ - UCHAR lowSubband; /*!< QMF-band where SBR frequency range starts */ - UCHAR highSubband; /*!< QMF-band where SBR frequency range ends */ - UCHAR limiterBandTable[MAX_NUM_LIMITERS+1]; /*!< Limiter band table. */ - UCHAR noLimiterBands; /*!< Number of limiter bands. */ - UCHAR nInvfBands; /*!< Number of bands for inverse filtering */ - UCHAR *freqBandTable[2]; /*!< Pointers to freqBandTableLo and freqBandTableHi */ - UCHAR freqBandTableLo[MAX_FREQ_COEFFS/2+1]; - /*!< Mapping of SBR bands to QMF bands for low frequency resolution */ - UCHAR freqBandTableHi[MAX_FREQ_COEFFS+1]; - /*!< Mapping of SBR bands to QMF bands for high frequency resolution */ - UCHAR freqBandTableNoise[MAX_NOISE_COEFFS+1]; - /*!< Mapping of SBR noise bands to QMF bands */ - UCHAR v_k_master[MAX_FREQ_COEFFS+1]; - /*!< Master BandTable which freqBandTable is derived from */ -} -FREQ_BAND_DATA; - -typedef FREQ_BAND_DATA *HANDLE_FREQ_BAND_DATA; - -#define SBRDEC_ELD_GRID 1 -#define SBRDEC_SYNTAX_SCAL 2 -#define SBRDEC_SYNTAX_USAC 4 -#define SBRDEC_SYNTAX_RSVD50 8 -#define SBRDEC_LOW_POWER 16 /* Flag indicating that Low Power QMF mode shall be used. */ -#define SBRDEC_PS_DECODED 32 /* Flag indicating that PS was decoded and rendered. */ -#define SBRDEC_LD_MPS_QMF 512 /* Flag indicating that the LD-MPS QMF shall be used. */ -#define SBRDEC_SYNTAX_DRM 2048 /* Flag indicating that DRM30/DRM+ reverse syntax is being used. */ -#define SBRDEC_DOWNSAMPLE 8192 /* Flag indicating that the downsampling mode is used. */ -#define SBRDEC_FLUSH 16384 /* Flag is used to flush all elements in use. */ -#define SBRDEC_FORCE_RESET 32768 /* Flag is used to force a reset of all elements in use. */ - -#define SBRDEC_HDR_STAT_RESET 1 -#define SBRDEC_HDR_STAT_UPDATE 2 - -typedef struct { - UCHAR ampResolution; /*!< Amplitude resolution of envelope values (0: 1.5dB, 1: 3dB) */ - UCHAR xover_band; /*!< Start index in #v_k_master[] used for dynamic crossover frequency */ - UCHAR sbr_preprocessing; /*!< SBR prewhitening flag. */ -} SBR_HEADER_DATA_BS_INFO; - -typedef struct { - /* Changes in these variables causes a reset of the decoder */ - UCHAR startFreq; /*!< Index for SBR start frequency */ - UCHAR stopFreq; /*!< Index for SBR highest frequency */ - UCHAR freqScale; /*!< 0: linear scale, 1-3 logarithmic scales */ - UCHAR alterScale; /*!< Flag for coarser frequency resolution */ - UCHAR noise_bands; /*!< Noise bands per octave, read from bitstream*/ - - /* don't require reset */ - UCHAR limiterBands; /*!< Index for number of limiter bands per octave */ - UCHAR limiterGains; /*!< Index to select gain limit */ - UCHAR interpolFreq; /*!< Select gain calculation method (1: per QMF channel, 0: per SBR band) */ - UCHAR smoothingLength; /*!< Smoothing of gains over time (0: on 1: off) */ - -} SBR_HEADER_DATA_BS; - -typedef struct -{ - SBR_SYNC_STATE syncState; /*!< The current initialization status of the header */ - - UCHAR status; /*!< Flags field used for signaling a reset right before the processing starts and an update from config (e.g. ASC). */ - UCHAR frameErrorFlag; /*!< Frame data valid flag. CAUTION: This variable will be overwritten by the flag stored in the element structure. - This is necessary because of the frame delay. There it might happen that different slots use the same header. */ - UCHAR numberTimeSlots; /*!< AAC: 16,15 */ - UCHAR numberOfAnalysisBands; /*!< Number of QMF analysis bands */ - UCHAR timeStep; /*!< Time resolution of SBR in QMF-slots */ - UINT sbrProcSmplRate; /*!< SBR processing sampling frequency (!= OutputSamplingRate) - (always: CoreSamplingRate * UpSamplingFactor; even in single rate mode) */ - - SBR_HEADER_DATA_BS bs_data; /*!< current SBR header. */ - SBR_HEADER_DATA_BS_INFO bs_info; /*!< SBR info. */ - - FREQ_BAND_DATA freqBandData; /*!< Pointer to struct #FREQ_BAND_DATA */ -} -SBR_HEADER_DATA; - -typedef SBR_HEADER_DATA *HANDLE_SBR_HEADER_DATA; - - -typedef struct -{ - UCHAR frameClass; /*!< Select grid type */ - UCHAR nEnvelopes; /*!< Number of envelopes */ - UCHAR borders[MAX_ENVELOPES+1]; /*!< Envelope borders (in SBR-timeslots, e.g. mp3PRO: 0..11) */ - UCHAR freqRes[MAX_ENVELOPES]; /*!< Frequency resolution for each envelope (0=low, 1=high) */ - SCHAR tranEnv; /*!< Transient envelope, -1 if none */ - UCHAR nNoiseEnvelopes; /*!< Number of noise envelopes */ - UCHAR bordersNoise[MAX_NOISE_ENVELOPES+1];/*!< borders of noise envelopes */ -} -FRAME_INFO; - - -typedef struct -{ - FIXP_SGL sfb_nrg_prev[MAX_FREQ_COEFFS]; /*!< Previous envelope (required for differential-coded values) */ - FIXP_SGL prevNoiseLevel[MAX_NOISE_COEFFS]; /*!< Previous noise envelope (required for differential-coded values) */ - COUPLING_MODE coupling; /*!< Stereo-mode of previous frame */ - INVF_MODE sbr_invf_mode[MAX_INVF_BANDS]; /*!< Previous strength of filtering in transposer */ - UCHAR ampRes; /*!< Previous amplitude resolution (0: 1.5dB, 1: 3dB) */ - UCHAR stopPos; /*!< Position in time where last envelope ended */ - UCHAR frameErrorFlag; /*!< Previous frame status */ -} -SBR_PREV_FRAME_DATA; - -typedef SBR_PREV_FRAME_DATA *HANDLE_SBR_PREV_FRAME_DATA; - - -typedef struct -{ - int nScaleFactors; /*!< total number of scalefactors in frame */ - - FRAME_INFO frameInfo; /*!< time grid for current frame */ - UCHAR domain_vec[MAX_ENVELOPES]; /*!< Bitfield containing direction of delta-coding for each envelope (0:frequency, 1:time) */ - UCHAR domain_vec_noise[MAX_NOISE_ENVELOPES]; /*!< Same as above, but for noise envelopes */ - - INVF_MODE sbr_invf_mode[MAX_INVF_BANDS]; /*!< Strength of filtering in transposer */ - COUPLING_MODE coupling; /*!< Stereo-mode */ - int ampResolutionCurrentFrame; /*!< Amplitude resolution of envelope values (0: 1.5dB, 1: 3dB) */ - - UCHAR addHarmonics[MAX_FREQ_COEFFS]; /*!< Flags for synthetic sine addition */ - - FIXP_SGL iEnvelope[MAX_NUM_ENVELOPE_VALUES]; /*!< Envelope data */ - FIXP_SGL sbrNoiseFloorLevel[MAX_NUM_NOISE_VALUES]; /*!< Noise envelope data */ -} -SBR_FRAME_DATA; - -typedef SBR_FRAME_DATA *HANDLE_SBR_FRAME_DATA; - -void initSbrPrevFrameData (HANDLE_SBR_PREV_FRAME_DATA h_prev_data, - int timeSlots); - - -int sbrGetSingleChannelElement (HANDLE_SBR_HEADER_DATA hHeaderData, - HANDLE_SBR_FRAME_DATA hFrameData, - HANDLE_FDK_BITSTREAM hBitBuf, - HANDLE_PS_DEC hParametricStereoDec, - const UINT flags, - const int overlap - ); - -int sbrGetChannelPairElement (HANDLE_SBR_HEADER_DATA hHeaderData, - HANDLE_SBR_FRAME_DATA hFrameDataLeft, - HANDLE_SBR_FRAME_DATA hFrameDataRight, - HANDLE_FDK_BITSTREAM hBitBuf, - const UINT flags, - const int overlap); - -SBR_HEADER_STATUS -sbrGetHeaderData (HANDLE_SBR_HEADER_DATA headerData, - HANDLE_FDK_BITSTREAM hBitBuf, - const UINT flags, - const int fIsSbrData); - -/*! - \brief Initialize SBR header data - - Copy default values to the header data struct and patch some entries - depending on the core codec. -*/ -SBR_ERROR -initHeaderData ( - HANDLE_SBR_HEADER_DATA hHeaderData, - const int sampleRateIn, - const int sampleRateOut, - const int samplesPerFrame, - const UINT flags - ); -#endif diff --git a/libSBRdec/src/huff_dec.cpp b/libSBRdec/src/huff_dec.cpp deleted file mode 100644 index 31d686d..0000000 --- a/libSBRdec/src/huff_dec.cpp +++ /dev/null @@ -1,120 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief Huffman Decoder -*/ - -#include "huff_dec.h" - -/***************************************************************************/ -/*! - \brief Decodes one huffman code word - - Reads bits from the bitstream until a valid codeword is found. - The table entries are interpreted either as index to the next entry - or - if negative - as the codeword. - - \return decoded value - - \author - -****************************************************************************/ -int -DecodeHuffmanCW (Huffman h, /*!< pointer to huffman codebook table */ - HANDLE_FDK_BITSTREAM hBs) /*!< Handle to Bitbuffer */ -{ - SCHAR index = 0; - int value, bit; - - while (index >= 0) { - bit = FDKreadBits (hBs, 1); - index = h[index][bit]; - } - - value = index+64; /* Add offset */ - - - return value; -} diff --git a/libSBRdec/src/huff_dec.h b/libSBRdec/src/huff_dec.h deleted file mode 100644 index 5443658..0000000 --- a/libSBRdec/src/huff_dec.h +++ /dev/null @@ -1,100 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief Huffman Decoder -*/ -#ifndef __HUFF_DEC_H -#define __HUFF_DEC_H - -#include "sbrdecoder.h" -#include "FDK_bitstream.h" - -typedef const SCHAR (*Huffman)[2]; - -int -DecodeHuffmanCW (Huffman h, - HANDLE_FDK_BITSTREAM hBitBuf); - -#endif diff --git a/libSBRdec/src/lpp_tran.cpp b/libSBRdec/src/lpp_tran.cpp deleted file mode 100644 index 117e739..0000000 --- a/libSBRdec/src/lpp_tran.cpp +++ /dev/null @@ -1,986 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief Low Power Profile Transposer, - This module provides the transposer. The main entry point is lppTransposer(). The function generates - high frequency content by copying data from the low band (provided by core codec) into the high band. - This process is also referred to as "patching". The function also implements spectral whitening by means of - inverse filtering based on LPC coefficients. - - Together with the QMF filterbank the transposer can be tested using a supplied test program. See main_audio.cpp for details. - This module does use fractional arithmetic and the accuracy of the computations has an impact on the overall sound quality. - The module also needs to take into account the different scaling of spectral data. - - \sa lppTransposer(), main_audio.cpp, sbr_scale.h, \ref documentationOverview -*/ - -#include "lpp_tran.h" - -#include "sbr_ram.h" -#include "sbr_rom.h" - -#include "genericStds.h" -#include "autocorr2nd.h" - - - -#if defined(__arm__) -#include "arm/lpp_tran_arm.cpp" -#endif - - - -#define LPC_SCALE_FACTOR 2 - - -/*! - * - * \brief Get bandwidth expansion factor from filtering level - * - * Returns a filter parameter (bandwidth expansion factor) depending on - * the desired filtering level signalled in the bitstream. - * When switching the filtering level from LOW to OFF, an additional - * level is being inserted to achieve a smooth transition. - */ - -#ifndef FUNCTION_mapInvfMode -static FIXP_DBL -mapInvfMode (INVF_MODE mode, - INVF_MODE prevMode, - WHITENING_FACTORS whFactors) -{ - switch (mode) { - case INVF_LOW_LEVEL: - if(prevMode == INVF_OFF) - return whFactors.transitionLevel; - else - return whFactors.lowLevel; - - case INVF_MID_LEVEL: - return whFactors.midLevel; - - case INVF_HIGH_LEVEL: - return whFactors.highLevel; - - default: - if(prevMode == INVF_LOW_LEVEL) - return whFactors.transitionLevel; - else - return whFactors.off; - } -} -#endif /* #ifndef FUNCTION_mapInvfMode */ - -/*! - * - * \brief Perform inverse filtering level emphasis - * - * Retrieve bandwidth expansion factor and apply smoothing for each filter band - * - */ - -#ifndef FUNCTION_inverseFilteringLevelEmphasis -static void -inverseFilteringLevelEmphasis(HANDLE_SBR_LPP_TRANS hLppTrans,/*!< Handle of lpp transposer */ - UCHAR nInvfBands, /*!< Number of bands for inverse filtering */ - INVF_MODE *sbr_invf_mode, /*!< Current inverse filtering modes */ - INVF_MODE *sbr_invf_mode_prev, /*!< Previous inverse filtering modes */ - FIXP_DBL * bwVector /*!< Resulting filtering levels */ - ) -{ - for(int i = 0; i < nInvfBands; i++) { - FIXP_DBL accu; - FIXP_DBL bwTmp = mapInvfMode (sbr_invf_mode[i], - sbr_invf_mode_prev[i], - hLppTrans->pSettings->whFactors); - - if(bwTmp < hLppTrans->bwVectorOld[i]) { - accu = fMultDiv2(FL2FXCONST_DBL(0.75f),bwTmp) + - fMultDiv2(FL2FXCONST_DBL(0.25f),hLppTrans->bwVectorOld[i]); - } - else { - accu = fMultDiv2(FL2FXCONST_DBL(0.90625f),bwTmp) + - fMultDiv2(FL2FXCONST_DBL(0.09375f),hLppTrans->bwVectorOld[i]); - } - - if (accu < FL2FXCONST_DBL(0.015625f)>>1) - bwVector[i] = FL2FXCONST_DBL(0.0f); - else - bwVector[i] = fixMin(accu<<1,FL2FXCONST_DBL(0.99609375f)); - } -} -#endif /* #ifndef FUNCTION_inverseFilteringLevelEmphasis */ - -/* Resulting autocorrelation determinant exponent */ -#define ACDET_EXP (2*(DFRACT_BITS+sbrScaleFactor->lb_scale+10-ac.det_scale)) -#define AC_EXP (-sbrScaleFactor->lb_scale+LPC_SCALE_FACTOR) -#define ALPHA_EXP (-sbrScaleFactor->lb_scale+LPC_SCALE_FACTOR+1) -/* Resulting transposed QMF values exponent 16 bit normalized samplebits assumed. */ -#define QMFOUT_EXP ((SAMPLE_BITS-15)-sbrScaleFactor->lb_scale) - -/*! - * - * \brief Perform transposition by patching of subband samples. - * This function serves as the main entry point into the module. The function determines the areas for the - * patching process (these are the source range as well as the target range) and implements spectral whitening - * by means of inverse filtering. The function autoCorrelation2nd() is an auxiliary function for calculating the - * LPC coefficients for the filtering. The actual calculation of the LPC coefficients and the implementation - * of the filtering are done as part of lppTransposer(). - * - * Note that the filtering is done on all available QMF subsamples, whereas the patching is only done on those QMF - * subsamples that will be used in the next QMF synthesis. The filtering is also implemented before the patching - * includes further dependencies on parameters from the SBR data. - * - */ - -void lppTransposer (HANDLE_SBR_LPP_TRANS hLppTrans, /*!< Handle of lpp transposer */ - QMF_SCALE_FACTOR *sbrScaleFactor, /*!< Scaling factors */ - FIXP_DBL **qmfBufferReal, /*!< Pointer to pointer to real part of subband samples (source) */ - - FIXP_DBL *degreeAlias, /*!< Vector for results of aliasing estimation */ - FIXP_DBL **qmfBufferImag, /*!< Pointer to pointer to imaginary part of subband samples (source) */ - const int useLP, - const int timeStep, /*!< Time step of envelope */ - const int firstSlotOffs, /*!< Start position in time */ - const int lastSlotOffs, /*!< Number of overlap-slots into next frame */ - const int nInvfBands, /*!< Number of bands for inverse filtering */ - INVF_MODE *sbr_invf_mode, /*!< Current inverse filtering modes */ - INVF_MODE *sbr_invf_mode_prev /*!< Previous inverse filtering modes */ - ) -{ - INT bwIndex[MAX_NUM_PATCHES]; - FIXP_DBL bwVector[MAX_NUM_PATCHES]; /*!< pole moving factors */ - - int i; - int loBand, start, stop; - TRANSPOSER_SETTINGS *pSettings = hLppTrans->pSettings; - PATCH_PARAM *patchParam = pSettings->patchParam; - int patch; - - FIXP_SGL alphar[LPC_ORDER], a0r, a1r; - FIXP_SGL alphai[LPC_ORDER], a0i=0, a1i=0; - FIXP_SGL bw = FL2FXCONST_SGL(0.0f); - - int autoCorrLength; - - FIXP_DBL k1, k1_below=0, k1_below2=0; - - ACORR_COEFS ac; - int startSample; - int stopSample; - int stopSampleClear; - - int comLowBandScale; - int ovLowBandShift; - int lowBandShift; -/* int ovHighBandShift;*/ - int targetStopBand; - - - alphai[0] = FL2FXCONST_SGL(0.0f); - alphai[1] = FL2FXCONST_SGL(0.0f); - - - startSample = firstSlotOffs * timeStep; - stopSample = pSettings->nCols + lastSlotOffs * timeStep; - - - inverseFilteringLevelEmphasis(hLppTrans, nInvfBands, sbr_invf_mode, sbr_invf_mode_prev, bwVector); - - stopSampleClear = stopSample; - - autoCorrLength = pSettings->nCols + pSettings->overlap; - - /* Set upper subbands to zero: - This is required in case that the patches do not cover the complete highband - (because the last patch would be too short). - Possible optimization: Clearing bands up to usb would be sufficient here. */ - targetStopBand = patchParam[pSettings->noOfPatches-1].targetStartBand - + patchParam[pSettings->noOfPatches-1].numBandsInPatch; - - int memSize = ((64) - targetStopBand) * sizeof(FIXP_DBL); - - if (!useLP) { - for (i = startSample; i < stopSampleClear; i++) { - FDKmemclear(&qmfBufferReal[i][targetStopBand], memSize); - FDKmemclear(&qmfBufferImag[i][targetStopBand], memSize); - } - } else - for (i = startSample; i < stopSampleClear; i++) { - FDKmemclear(&qmfBufferReal[i][targetStopBand], memSize); - } - - /* init bwIndex for each patch */ - FDKmemclear(bwIndex, pSettings->noOfPatches*sizeof(INT)); - - /* - Calc common low band scale factor - */ - comLowBandScale = fixMin(sbrScaleFactor->ov_lb_scale,sbrScaleFactor->lb_scale); - - ovLowBandShift = sbrScaleFactor->ov_lb_scale - comLowBandScale; - lowBandShift = sbrScaleFactor->lb_scale - comLowBandScale; - /* ovHighBandShift = firstSlotOffs == 0 ? ovLowBandShift:0;*/ - - /* outer loop over bands to do analysis only once for each band */ - - if (!useLP) { - start = pSettings->lbStartPatching; - stop = pSettings->lbStopPatching; - } else - { - start = fixMax(1, pSettings->lbStartPatching - 2); - stop = patchParam[0].targetStartBand; - } - - - for ( loBand = start; loBand < stop; loBand++ ) { - - FIXP_DBL lowBandReal[(((1024)/(32))+(6))+LPC_ORDER]; - FIXP_DBL *plowBandReal = lowBandReal; - FIXP_DBL **pqmfBufferReal = qmfBufferReal; - FIXP_DBL lowBandImag[(((1024)/(32))+(6))+LPC_ORDER]; - FIXP_DBL *plowBandImag = lowBandImag; - FIXP_DBL **pqmfBufferImag = qmfBufferImag; - int resetLPCCoeffs=0; - int dynamicScale = DFRACT_BITS-1-LPC_SCALE_FACTOR; - int acDetScale = 0; /* scaling of autocorrelation determinant */ - - for(i=0;ilpcFilterStatesReal[i][loBand]; - if (!useLP) - *plowBandImag++ = hLppTrans->lpcFilterStatesImag[i][loBand]; - } - - /* - Take old slope length qmf slot source values out of (overlap)qmf buffer - */ - if (!useLP) { - for(i=0;inCols+pSettings->overlap;i++){ - *plowBandReal++ = (*pqmfBufferReal++)[loBand]; - *plowBandImag++ = (*pqmfBufferImag++)[loBand]; - } - } else - { - /* pSettings->overlap is always even */ - FDK_ASSERT((pSettings->overlap & 1) == 0); - - for(i=0;i<((pSettings->overlap+pSettings->nCols)>>1);i++) { - *plowBandReal++ = (*pqmfBufferReal++)[loBand]; - *plowBandReal++ = (*pqmfBufferReal++)[loBand]; - } - if (pSettings->nCols & 1) { - *plowBandReal++ = (*pqmfBufferReal++)[loBand]; - } - } - - /* - Determine dynamic scaling value. - */ - dynamicScale = fixMin(dynamicScale, getScalefactor(lowBandReal, LPC_ORDER+pSettings->overlap) + ovLowBandShift); - dynamicScale = fixMin(dynamicScale, getScalefactor(&lowBandReal[LPC_ORDER+pSettings->overlap], pSettings->nCols) + lowBandShift); - if (!useLP) { - dynamicScale = fixMin(dynamicScale, getScalefactor(lowBandImag, LPC_ORDER+pSettings->overlap) + ovLowBandShift); - dynamicScale = fixMin(dynamicScale, getScalefactor(&lowBandImag[LPC_ORDER+pSettings->overlap], pSettings->nCols) + lowBandShift); - } - dynamicScale = fixMax(0, dynamicScale-1); /* one additional bit headroom to prevent -1.0 */ - - /* - Scale temporal QMF buffer. - */ - scaleValues(&lowBandReal[0], LPC_ORDER+pSettings->overlap, dynamicScale-ovLowBandShift); - scaleValues(&lowBandReal[LPC_ORDER+pSettings->overlap], pSettings->nCols, dynamicScale-lowBandShift); - - if (!useLP) { - scaleValues(&lowBandImag[0], LPC_ORDER+pSettings->overlap, dynamicScale-ovLowBandShift); - scaleValues(&lowBandImag[LPC_ORDER+pSettings->overlap], pSettings->nCols, dynamicScale-lowBandShift); - } - - - if (!useLP) { - acDetScale += autoCorr2nd_cplx(&ac, lowBandReal+LPC_ORDER, lowBandImag+LPC_ORDER, autoCorrLength); - } - else - { - acDetScale += autoCorr2nd_real(&ac, lowBandReal+LPC_ORDER, autoCorrLength); - } - - /* Examine dynamic of determinant in autocorrelation. */ - acDetScale += 2*(comLowBandScale + dynamicScale); - acDetScale *= 2; /* two times reflection coefficent scaling */ - acDetScale += ac.det_scale; /* ac scaling of determinant */ - - /* In case of determinant < 10^-38, resetLPCCoeffs=1 has to be enforced. */ - if (acDetScale>126 ) { - resetLPCCoeffs = 1; - } - - - alphar[1] = FL2FXCONST_SGL(0.0f); - if (!useLP) - alphai[1] = FL2FXCONST_SGL(0.0f); - - if (ac.det != FL2FXCONST_DBL(0.0f)) { - FIXP_DBL tmp,absTmp,absDet; - - absDet = fixp_abs(ac.det); - - if (!useLP) { - tmp = ( fMultDiv2(ac.r01r,ac.r12r) >> (LPC_SCALE_FACTOR-1) ) - - ( (fMultDiv2(ac.r01i,ac.r12i) + fMultDiv2(ac.r02r,ac.r11r)) >> (LPC_SCALE_FACTOR-1) ); - } else - { - tmp = ( fMultDiv2(ac.r01r,ac.r12r) >> (LPC_SCALE_FACTOR-1) ) - - ( fMultDiv2(ac.r02r,ac.r11r) >> (LPC_SCALE_FACTOR-1) ); - } - absTmp = fixp_abs(tmp); - - /* - Quick check: is first filter coeff >= 1(4) - */ - { - INT scale; - FIXP_DBL result = fDivNorm(absTmp, absDet, &scale); - scale = scale+ac.det_scale; - - if ( (scale > 0) && (result >= (FIXP_DBL)MAXVAL_DBL>>scale) ) { - resetLPCCoeffs = 1; - } - else { - alphar[1] = FX_DBL2FX_SGL(scaleValue(result,scale)); - if((tmp> (LPC_SCALE_FACTOR-1) ) + - ( (fMultDiv2(ac.r01r,ac.r12i) - (FIXP_DBL)fMultDiv2(ac.r02i,ac.r11r)) >> (LPC_SCALE_FACTOR-1) ) ; - - absTmp = fixp_abs(tmp); - - /* - Quick check: is second filter coeff >= 1(4) - */ - { - INT scale; - FIXP_DBL result = fDivNorm(absTmp, absDet, &scale); - scale = scale+ac.det_scale; - - if ( (scale > 0) && (result >= /*FL2FXCONST_DBL(1.f)*/ (FIXP_DBL)MAXVAL_DBL>>scale) ) { - resetLPCCoeffs = 1; - } - else { - alphai[1] = FX_DBL2FX_SGL(scaleValue(result,scale)); - if((tmp=0 */ - FIXP_DBL tmp,absTmp; - - if (!useLP) { - tmp = (ac.r01r>>(LPC_SCALE_FACTOR+1)) + - (fMultDiv2(alphar[1],ac.r12r) + fMultDiv2(alphai[1],ac.r12i)); - } else - { - if(ac.r01r>=FL2FXCONST_DBL(0.0f)) - tmp = (ac.r01r>>(LPC_SCALE_FACTOR+1)) + fMultDiv2(alphar[1],ac.r12r); - else - tmp = -((-ac.r01r)>>(LPC_SCALE_FACTOR+1)) + fMultDiv2(alphar[1],ac.r12r); - } - - absTmp = fixp_abs(tmp); - - /* - Quick check: is first filter coeff >= 1(4) - */ - - if (absTmp >= (ac.r11r>>1)) { - resetLPCCoeffs=1; - } - else { - INT scale; - FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale); - alphar[0] = FX_DBL2FX_SGL(scaleValue(result,scale+1)); - - if((tmp>FL2FX_DBL(0.0f)) ^ (ac.r11r>(LPC_SCALE_FACTOR+1)) + - (fMultDiv2(alphai[1],ac.r12r) - fMultDiv2(alphar[1],ac.r12i)); - - absTmp = fixp_abs(tmp); - - /* - Quick check: is second filter coeff >= 1(4) - */ - if (absTmp >= (ac.r11r>>1)) { - resetLPCCoeffs=1; - } - else { - INT scale; - FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale); - alphai[0] = FX_DBL2FX_SGL(scaleValue(result,scale+1)); - if((tmp>FL2FX_DBL(0.0f)) ^ (ac.r11r= FL2FXCONST_DBL(0.5f) ) - resetLPCCoeffs=1; - if( (fMultDiv2(alphar[1],alphar[1]) + fMultDiv2(alphai[1],alphai[1])) >= FL2FXCONST_DBL(0.5f) ) - resetLPCCoeffs=1; - } - - if(resetLPCCoeffs){ - alphar[0] = FL2FXCONST_SGL(0.0f); - alphar[1] = FL2FXCONST_SGL(0.0f); - if (!useLP) - { - alphai[0] = FL2FXCONST_SGL(0.0f); - alphai[1] = FL2FXCONST_SGL(0.0f); - } - } - - if (useLP) - { - - /* Aliasing detection */ - if(ac.r11r==FL2FXCONST_DBL(0.0f)) { - k1 = FL2FXCONST_DBL(0.0f); - } - else { - if ( fixp_abs(ac.r01r) >= fixp_abs(ac.r11r) ) { - if ( fMultDiv2(ac.r01r,ac.r11r) < FL2FX_DBL(0.0f)) { - k1 = (FIXP_DBL)MAXVAL_DBL /*FL2FXCONST_SGL(1.0f)*/; - }else { - /* Since this value is squared later, it must not ever become -1.0f. */ - k1 = (FIXP_DBL)(MINVAL_DBL+1) /*FL2FXCONST_SGL(-1.0f)*/; - } - } - else { - INT scale; - FIXP_DBL result = fDivNorm(fixp_abs(ac.r01r), fixp_abs(ac.r11r), &scale); - k1 = scaleValue(result,scale); - - if(!((ac.r01r 1){ - /* Check if the gain should be locked */ - FIXP_DBL deg = /*FL2FXCONST_DBL(1.0f)*/ (FIXP_DBL)MAXVAL_DBL - fPow2(k1_below); - degreeAlias[loBand] = FL2FXCONST_DBL(0.0f); - if (((loBand & 1) == 0) && (k1 < FL2FXCONST_DBL(0.0f))){ - if (k1_below < FL2FXCONST_DBL(0.0f)) { /* 2-Ch Aliasing Detection */ - degreeAlias[loBand] = (FIXP_DBL)MAXVAL_DBL /*FL2FXCONST_DBL(1.0f)*/; - if ( k1_below2 > FL2FXCONST_DBL(0.0f) ) { /* 3-Ch Aliasing Detection */ - degreeAlias[loBand-1] = deg; - } - } - else if ( k1_below2 > FL2FXCONST_DBL(0.0f) ) { /* 3-Ch Aliasing Detection */ - degreeAlias[loBand] = deg; - } - } - if (((loBand & 1) == 1) && (k1 > FL2FXCONST_DBL(0.0f))){ - if (k1_below > FL2FXCONST_DBL(0.0f)) { /* 2-CH Aliasing Detection */ - degreeAlias[loBand] = (FIXP_DBL)MAXVAL_DBL /*FL2FXCONST_DBL(1.0f)*/; - if ( k1_below2 < FL2FXCONST_DBL(0.0f) ) { /* 3-CH Aliasing Detection */ - degreeAlias[loBand-1] = deg; - } - } - else if ( k1_below2 < FL2FXCONST_DBL(0.0f) ) { /* 3-CH Aliasing Detection */ - degreeAlias[loBand] = deg; - } - } - } - /* remember k1 values of the 2 QMF channels below the current channel */ - k1_below2 = k1_below; - k1_below = k1; - } - - patch = 0; - - while ( patch < pSettings->noOfPatches ) { /* inner loop over every patch */ - - int hiBand = loBand + patchParam[patch].targetBandOffs; - - if ( loBand < patchParam[patch].sourceStartBand - || loBand >= patchParam[patch].sourceStopBand - //|| hiBand >= hLppTrans->pSettings->noChannels - ) { - /* Lowband not in current patch - proceed */ - patch++; - continue; - } - - FDK_ASSERT( hiBand < (64) ); - - /* bwIndex[patch] is already initialized with value from previous band inside this patch */ - while (hiBand >= pSettings->bwBorders[bwIndex[patch]]) - bwIndex[patch]++; - - - /* - Filter Step 2: add the left slope with the current filter to the buffer - pure source values are already in there - */ - bw = FX_DBL2FX_SGL(bwVector[bwIndex[patch]]); - - a0r = FX_DBL2FX_SGL(fMult(bw,alphar[0])); /* Apply current bandwidth expansion factor */ - - - if (!useLP) - a0i = FX_DBL2FX_SGL(fMult(bw,alphai[0])); - bw = FX_DBL2FX_SGL(fPow2(bw)); - a1r = FX_DBL2FX_SGL(fMult(bw,alphar[1])); - if (!useLP) - a1i = FX_DBL2FX_SGL(fMult(bw,alphai[1])); - - - - /* - Filter Step 3: insert the middle part which won't be windowed - */ - - if ( bw <= FL2FXCONST_SGL(0.0f) ) { - if (!useLP) { - int descale = fixMin(DFRACT_BITS-1, (LPC_SCALE_FACTOR+dynamicScale)); - for(i = startSample; i < stopSample; i++ ) { - qmfBufferReal[i][hiBand] = lowBandReal[LPC_ORDER+i]>>descale; - qmfBufferImag[i][hiBand] = lowBandImag[LPC_ORDER+i]>>descale; - } - } else - { - int descale = fixMin(DFRACT_BITS-1, (LPC_SCALE_FACTOR+dynamicScale)); - for(i = startSample; i < stopSample; i++ ) { - qmfBufferReal[i][hiBand] = lowBandReal[LPC_ORDER+i]>>descale; - } - } - } - else { /* bw <= 0 */ - - if (!useLP) { - int descale = fixMin(DFRACT_BITS-1, (LPC_SCALE_FACTOR+dynamicScale)); -#ifdef FUNCTION_LPPTRANSPOSER_func1 - lppTransposer_func1(lowBandReal+LPC_ORDER+startSample,lowBandImag+LPC_ORDER+startSample, - qmfBufferReal+startSample,qmfBufferImag+startSample, - stopSample-startSample, (int) hiBand, - dynamicScale,descale, - a0r, a0i, a1r, a1i); -#else - for(i = startSample; i < stopSample; i++ ) { - FIXP_DBL accu1, accu2; - - accu1 = (fMultDiv2(a0r,lowBandReal[LPC_ORDER+i-1]) - fMultDiv2(a0i,lowBandImag[LPC_ORDER+i-1]) + - fMultDiv2(a1r,lowBandReal[LPC_ORDER+i-2]) - fMultDiv2(a1i,lowBandImag[LPC_ORDER+i-2]))>>dynamicScale; - accu2 = (fMultDiv2(a0i,lowBandReal[LPC_ORDER+i-1]) + fMultDiv2(a0r,lowBandImag[LPC_ORDER+i-1]) + - fMultDiv2(a1i,lowBandReal[LPC_ORDER+i-2]) + fMultDiv2(a1r,lowBandImag[LPC_ORDER+i-2]))>>dynamicScale; - - qmfBufferReal[i][hiBand] = (lowBandReal[LPC_ORDER+i]>>descale) + (accu1<<1); - qmfBufferImag[i][hiBand] = (lowBandImag[LPC_ORDER+i]>>descale) + (accu2<<1); - } -#endif - } else - { - int descale = fixMin(DFRACT_BITS-1, (LPC_SCALE_FACTOR+dynamicScale)); - - FDK_ASSERT(dynamicScale >= 0); - for(i = startSample; i < stopSample; i++ ) { - FIXP_DBL accu1; - - accu1 = (fMultDiv2(a0r,lowBandReal[LPC_ORDER+i-1]) + fMultDiv2(a1r,lowBandReal[LPC_ORDER+i-2]))>>dynamicScale; - - qmfBufferReal[i][hiBand] = (lowBandReal[LPC_ORDER+i]>>descale) + (accu1<<1); - } - } - } /* bw <= 0 */ - - patch++; - - } /* inner loop over patches */ - - /* - * store the unmodified filter coefficients if there is - * an overlapping envelope - *****************************************************************/ - - - } /* outer loop over bands (loBand) */ - - if (useLP) - { - for ( loBand = pSettings->lbStartPatching; loBand < pSettings->lbStopPatching; loBand++ ) { - patch = 0; - while ( patch < pSettings->noOfPatches ) { - - UCHAR hiBand = loBand + patchParam[patch].targetBandOffs; - - if ( loBand < patchParam[patch].sourceStartBand - || loBand >= patchParam[patch].sourceStopBand - || hiBand >= (64) /* Highband out of range (biterror) */ - ) { - /* Lowband not in current patch or highband out of range (might be caused by biterrors)- proceed */ - patch++; - continue; - } - - if(hiBand != patchParam[patch].targetStartBand) - degreeAlias[hiBand] = degreeAlias[loBand]; - - patch++; - } - }/* end for loop */ - } - - for (i = 0; i < nInvfBands; i++ ) { - hLppTrans->bwVectorOld[i] = bwVector[i]; - } - - /* - set high band scale factor - */ - sbrScaleFactor->hb_scale = comLowBandScale-(LPC_SCALE_FACTOR); - -} - -/*! - * - * \brief Initialize one low power transposer instance - * - * - */ -SBR_ERROR -createLppTransposer (HANDLE_SBR_LPP_TRANS hs, /*!< Handle of low power transposer */ - TRANSPOSER_SETTINGS *pSettings, /*!< Pointer to settings */ - const int highBandStartSb, /*!< ? */ - UCHAR *v_k_master, /*!< Master table */ - const int numMaster, /*!< Valid entries in master table */ - const int usb, /*!< Highband area stop subband */ - const int timeSlots, /*!< Number of time slots */ - const int nCols, /*!< Number of colums (codec qmf bank) */ - UCHAR *noiseBandTable, /*!< Mapping of SBR noise bands to QMF bands */ - const int noNoiseBands, /*!< Number of noise bands */ - UINT fs, /*!< Sample Frequency */ - const int chan, /*!< Channel number */ - const int overlap - ) -{ - /* FB inverse filtering settings */ - hs->pSettings = pSettings; - - pSettings->nCols = nCols; - pSettings->overlap = overlap; - - switch (timeSlots) { - - case 15: - case 16: - break; - - default: - return SBRDEC_UNSUPPORTED_CONFIG; /* Unimplemented */ - } - - if (chan==0) { - /* Init common data only once */ - hs->pSettings->nCols = nCols; - - return resetLppTransposer (hs, - highBandStartSb, - v_k_master, - numMaster, - noiseBandTable, - noNoiseBands, - usb, - fs); - } - return SBRDEC_OK; -} - - -static int findClosestEntry(UCHAR goalSb, UCHAR *v_k_master, UCHAR numMaster, UCHAR direction) -{ - int index; - - if( goalSb <= v_k_master[0] ) - return v_k_master[0]; - - if( goalSb >= v_k_master[numMaster] ) - return v_k_master[numMaster]; - - if(direction) { - index = 0; - while( v_k_master[index] < goalSb ) { - index++; - } - } else { - index = numMaster; - while( v_k_master[index] > goalSb ) { - index--; - } - } - - return v_k_master[index]; -} - - -/*! - * - * \brief Reset memory for one lpp transposer instance - * - * \return SBRDEC_OK on success, SBRDEC_UNSUPPORTED_CONFIG on error - */ -SBR_ERROR -resetLppTransposer (HANDLE_SBR_LPP_TRANS hLppTrans, /*!< Handle of lpp transposer */ - UCHAR highBandStartSb, /*!< High band area: start subband */ - UCHAR *v_k_master, /*!< Master table */ - UCHAR numMaster, /*!< Valid entries in master table */ - UCHAR *noiseBandTable, /*!< Mapping of SBR noise bands to QMF bands */ - UCHAR noNoiseBands, /*!< Number of noise bands */ - UCHAR usb, /*!< High band area: stop subband */ - UINT fs /*!< SBR output sampling frequency */ - ) -{ - TRANSPOSER_SETTINGS *pSettings = hLppTrans->pSettings; - PATCH_PARAM *patchParam = pSettings->patchParam; - - int i, patch; - int targetStopBand; - int sourceStartBand; - int patchDistance; - int numBandsInPatch; - - int lsb = v_k_master[0]; /* Start subband expressed in "non-critical" sampling terms*/ - int xoverOffset = highBandStartSb - lsb; /* Calculate distance in QMF bands between k0 and kx */ - int startFreqHz; - - int desiredBorder; - - usb = fixMin(usb, v_k_master[numMaster]); /* Avoid endless loops (compare with float code). */ - - /* - * Plausibility check - */ - - if ( lsb - SHIFT_START_SB < 4 ) { - return SBRDEC_UNSUPPORTED_CONFIG; - } - - - /* - * Initialize the patching parameter - */ - /* ISO/IEC 14496-3 (Figure 4.48): goalSb = round( 2.048e6 / fs ) */ - desiredBorder = (((2048000*2) / fs) + 1) >> 1; - - desiredBorder = findClosestEntry(desiredBorder, v_k_master, numMaster, 1); /* Adapt region to master-table */ - - /* First patch */ - sourceStartBand = SHIFT_START_SB + xoverOffset; - targetStopBand = lsb + xoverOffset; /* upperBand */ - - /* Even (odd) numbered channel must be patched to even (odd) numbered channel */ - patch = 0; - while(targetStopBand < usb) { - - /* Too many patches? - Allow MAX_NUM_PATCHES+1 patches here. - we need to check later again, since patch might be the highest patch - AND contain less than 3 bands => actual number of patches will be reduced by 1. - */ - if (patch > MAX_NUM_PATCHES) { - return SBRDEC_UNSUPPORTED_CONFIG; - } - - patchParam[patch].guardStartBand = targetStopBand; - patchParam[patch].targetStartBand = targetStopBand; - - numBandsInPatch = desiredBorder - targetStopBand; /* Get the desired range of the patch */ - - if ( numBandsInPatch >= lsb - sourceStartBand ) { - /* Desired number bands are not available -> patch whole source range */ - patchDistance = targetStopBand - sourceStartBand; /* Get the targetOffset */ - patchDistance = patchDistance & ~1; /* Rounding off odd numbers and make all even */ - numBandsInPatch = lsb - (targetStopBand - patchDistance); /* Update number of bands to be patched */ - numBandsInPatch = findClosestEntry(targetStopBand + numBandsInPatch, v_k_master, numMaster, 0) - - targetStopBand; /* Adapt region to master-table */ - } - - /* Desired number bands are available -> get the minimal even patching distance */ - patchDistance = numBandsInPatch + targetStopBand - lsb; /* Get minimal distance */ - patchDistance = (patchDistance + 1) & ~1; /* Rounding up odd numbers and make all even */ - - if (numBandsInPatch > 0) { - patchParam[patch].sourceStartBand = targetStopBand - patchDistance; - patchParam[patch].targetBandOffs = patchDistance; - patchParam[patch].numBandsInPatch = numBandsInPatch; - patchParam[patch].sourceStopBand = patchParam[patch].sourceStartBand + numBandsInPatch; - - targetStopBand += patchParam[patch].numBandsInPatch; - patch++; - } - - /* All patches but first */ - sourceStartBand = SHIFT_START_SB; - - /* Check if we are close to desiredBorder */ - if( desiredBorder - targetStopBand < 3) /* MPEG doc */ - { - desiredBorder = usb; - } - - } - - patch--; - - /* If highest patch contains less than three subband: skip it */ - if ( (patch>0) && (patchParam[patch].numBandsInPatch < 3) ) { - patch--; - targetStopBand = patchParam[patch].targetStartBand + patchParam[patch].numBandsInPatch; - } - - /* now check if we don't have one too many */ - if (patch >= MAX_NUM_PATCHES) { - return SBRDEC_UNSUPPORTED_CONFIG; - } - - pSettings->noOfPatches = patch + 1; - - /* Check lowest and highest source subband */ - pSettings->lbStartPatching = targetStopBand; - pSettings->lbStopPatching = 0; - for ( patch = 0; patch < pSettings->noOfPatches; patch++ ) { - pSettings->lbStartPatching = fixMin( pSettings->lbStartPatching, patchParam[patch].sourceStartBand ); - pSettings->lbStopPatching = fixMax( pSettings->lbStopPatching, patchParam[patch].sourceStopBand ); - } - - for(i = 0 ; i < noNoiseBands; i++){ - pSettings->bwBorders[i] = noiseBandTable[i+1]; - } - - /* - * Choose whitening factors - */ - - startFreqHz = ( (lsb + xoverOffset)*fs ) >> 7; /* Shift does a division by 2*(64) */ - - for( i = 1; i < NUM_WHFACTOR_TABLE_ENTRIES; i++ ) - { - if( startFreqHz < FDK_sbrDecoder_sbr_whFactorsIndex[i]) - break; - } - i--; - - pSettings->whFactors.off = FDK_sbrDecoder_sbr_whFactorsTable[i][0]; - pSettings->whFactors.transitionLevel = FDK_sbrDecoder_sbr_whFactorsTable[i][1]; - pSettings->whFactors.lowLevel = FDK_sbrDecoder_sbr_whFactorsTable[i][2]; - pSettings->whFactors.midLevel = FDK_sbrDecoder_sbr_whFactorsTable[i][3]; - pSettings->whFactors.highLevel = FDK_sbrDecoder_sbr_whFactorsTable[i][4]; - - return SBRDEC_OK; -} diff --git a/libSBRdec/src/lpp_tran.h b/libSBRdec/src/lpp_tran.h deleted file mode 100644 index 003a547..0000000 --- a/libSBRdec/src/lpp_tran.h +++ /dev/null @@ -1,242 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief Low Power Profile Transposer, -*/ - -#ifndef _LPP_TRANS_H -#define _LPP_TRANS_H - -#include "sbrdecoder.h" -#include "qmf.h" - -/* - Common -*/ -#define QMF_OUT_SCALE 8 - -/* - Env-Adjust -*/ -#define MAX_NOISE_ENVELOPES 2 -#define MAX_NOISE_COEFFS 5 -#define MAX_NUM_NOISE_VALUES (MAX_NOISE_ENVELOPES * MAX_NOISE_COEFFS) -#define MAX_NUM_LIMITERS 12 - -/* Set MAX_ENVELOPES to the largest value of all supported BSFORMATs - by overriding MAX_ENVELOPES in the correct order: */ -#define MAX_ENVELOPES_HEAAC 5 -#define MAX_ENVELOPES MAX_ENVELOPES_HEAAC - -#define MAX_FREQ_COEFFS 48 -#define MAX_FREQ_COEFFS_FS44100 35 -#define MAX_FREQ_COEFFS_FS48000 32 - - -#define MAX_NUM_ENVELOPE_VALUES (MAX_ENVELOPES * MAX_FREQ_COEFFS) - -#define MAX_GAIN_EXP 34 -/* Maximum gain will be sqrt(0.5 * 2^MAX_GAIN_EXP) - example: 34=99dB */ -#define MAX_GAIN_CONCEAL_EXP 1 -/* Maximum gain will be sqrt(0.5 * 2^MAX_GAIN_CONCEAL_EXP) in concealment case (0dB) */ - -/* - LPP Transposer -*/ -#define LPC_ORDER 2 - -#define MAX_INVF_BANDS MAX_NOISE_COEFFS - -#define MAX_NUM_PATCHES 6 -#define SHIFT_START_SB 1 /*!< lowest subband of source range */ - -typedef enum -{ - INVF_OFF = 0, - INVF_LOW_LEVEL, - INVF_MID_LEVEL, - INVF_HIGH_LEVEL, - INVF_SWITCHED /* not a real choice but used here to control behaviour */ -} -INVF_MODE; - - -/** parameter set for one single patch */ -typedef struct { - UCHAR sourceStartBand; /*!< first band in lowbands where to take the samples from */ - UCHAR sourceStopBand; /*!< first band in lowbands which is not included in the patch anymore */ - UCHAR guardStartBand; /*!< first band in highbands to be filled with zeros in order to - reduce interferences between patches */ - UCHAR targetStartBand; /*!< first band in highbands to be filled with whitened lowband signal */ - UCHAR targetBandOffs; /*!< difference between 'startTargetBand' and 'startSourceBand' */ - UCHAR numBandsInPatch; /*!< number of consecutive bands in this one patch */ -} PATCH_PARAM; - - -/** whitening factors for different levels of whitening - need to be initialized corresponding to crossover frequency */ -typedef struct { - FIXP_DBL off; /*!< bw factor for signal OFF */ - FIXP_DBL transitionLevel; - FIXP_DBL lowLevel; /*!< bw factor for signal LOW_LEVEL */ - FIXP_DBL midLevel; /*!< bw factor for signal MID_LEVEL */ - FIXP_DBL highLevel; /*!< bw factor for signal HIGH_LEVEL */ -} WHITENING_FACTORS; - - -/*! The transposer settings are calculated on a header reset and are shared by both channels. */ -typedef struct { - UCHAR nCols; /*!< number subsamples of a codec frame */ - UCHAR noOfPatches; /*!< number of patches */ - UCHAR lbStartPatching; /*!< first band of lowbands that will be patched */ - UCHAR lbStopPatching; /*!< first band that won't be patched anymore*/ - UCHAR bwBorders[MAX_NUM_NOISE_VALUES]; /*!< spectral bands with different inverse filtering levels */ - - PATCH_PARAM patchParam[MAX_NUM_PATCHES]; /*!< new parameter set for patching */ - WHITENING_FACTORS whFactors; /*!< the pole moving factors for certain whitening levels as indicated - in the bitstream depending on the crossover frequency */ - UCHAR overlap; /*!< Overlap size */ -} TRANSPOSER_SETTINGS; - - -typedef struct -{ - TRANSPOSER_SETTINGS *pSettings; /*!< Common settings for both channels */ - FIXP_DBL bwVectorOld[MAX_NUM_PATCHES]; /*!< pole moving factors of past frame */ - FIXP_DBL lpcFilterStatesReal[LPC_ORDER][(32)]; /*!< pointer array to save filter states */ - FIXP_DBL lpcFilterStatesImag[LPC_ORDER][(32)]; /*!< pointer array to save filter states */ -} -SBR_LPP_TRANS; - -typedef SBR_LPP_TRANS *HANDLE_SBR_LPP_TRANS; - - -void lppTransposer (HANDLE_SBR_LPP_TRANS hLppTrans, - QMF_SCALE_FACTOR *sbrScaleFactor, - FIXP_DBL **qmfBufferReal, - - FIXP_DBL *degreeAlias, - FIXP_DBL **qmfBufferImag, - const int useLP, - const int timeStep, - const int firstSlotOffset, - const int lastSlotOffset, - const int nInvfBands, - INVF_MODE *sbr_invf_mode, - INVF_MODE *sbr_invf_mode_prev - ); - - -SBR_ERROR -createLppTransposer (HANDLE_SBR_LPP_TRANS hLppTrans, - TRANSPOSER_SETTINGS *pSettings, - const int highBandStartSb, - UCHAR *v_k_master, - const int numMaster, - const int usb, - const int timeSlots, - const int nCols, - UCHAR *noiseBandTable, - const int noNoiseBands, - UINT fs, - const int chan, - const int overlap); - - -SBR_ERROR -resetLppTransposer (HANDLE_SBR_LPP_TRANS hLppTrans, - UCHAR highBandStartSb, - UCHAR *v_k_master, - UCHAR numMaster, - UCHAR *noiseBandTable, - UCHAR noNoiseBands, - UCHAR usb, - UINT fs); - - - -#endif /* _LPP_TRANS_H */ - diff --git a/libSBRdec/src/psbitdec.cpp b/libSBRdec/src/psbitdec.cpp deleted file mode 100644 index ec6e484..0000000 --- a/libSBRdec/src/psbitdec.cpp +++ /dev/null @@ -1,593 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -#include "psbitdec.h" - - -#include "sbr_rom.h" -#include "huff_dec.h" - -/* PS dec privat functions */ -SBR_ERROR ResetPsDec(HANDLE_PS_DEC h_ps_d); -void ResetPsDeCor (HANDLE_PS_DEC h_ps_d); - -/***************************************************************************/ -/*! - \brief huffman decoding by codebook table - - \return index of huffman codebook table - -****************************************************************************/ -static SCHAR -decode_huff_cw (Huffman h, /*!< pointer to huffman codebook table */ - HANDLE_FDK_BITSTREAM hBitBuf, /*!< Handle to Bitbuffer */ - int *length) /*!< length of huffman codeword (or NULL) */ -{ - UCHAR bit = 0; - SCHAR index = 0; - UCHAR bitCount = 0; - - while (index >= 0) { - bit = FDKreadBits (hBitBuf, 1); - bitCount++; - index = h[index][bit]; - } - if (length) { - *length = bitCount; - } - return( index+64 ); /* Add offset */ -} - -/***************************************************************************/ -/*! - \brief helper function - limiting of value to min/max values - - \return limited value - -****************************************************************************/ - -static SCHAR -limitMinMax(SCHAR i, - SCHAR min, - SCHAR max) -{ - if (imax) - return max; - else - return i; -} - -/***************************************************************************/ -/*! - \brief Decodes delta values in-place and updates - data buffers according to quantization classes. - - When delta coded in frequency the first element is deltacode from zero. - aIndex buffer is decoded from delta values to actual values. - - \return none - -****************************************************************************/ -static void -deltaDecodeArray(SCHAR enable, - SCHAR *aIndex, /*!< ICC/IID parameters */ - SCHAR *aPrevFrameIndex, /*!< ICC/IID parameters of previous frame */ - SCHAR DtDf, - UCHAR nrElements, /*!< as conveyed in bitstream */ - /*!< output array size: nrElements*stride */ - UCHAR stride, /*!< 1=dflt, 2=half freq. resolution */ - SCHAR minIdx, - SCHAR maxIdx) -{ - int i; - - /* Delta decode */ - if ( enable==1 ) { - if (DtDf == 0) { /* Delta coded in freq */ - aIndex[0] = 0 + aIndex[0]; - aIndex[0] = limitMinMax(aIndex[0],minIdx,maxIdx); - for (i = 1; i < nrElements; i++) { - aIndex[i] = aIndex[i-1] + aIndex[i]; - aIndex[i] = limitMinMax(aIndex[i],minIdx,maxIdx); - } - } - else { /* Delta time */ - for (i = 0; i < nrElements; i++) { - aIndex[i] = aPrevFrameIndex[i*stride] + aIndex[i]; - aIndex[i] = limitMinMax(aIndex[i],minIdx,maxIdx); - } - } - } - else { /* No data is sent, set index to zero */ - for (i = 0; i < nrElements; i++) { - aIndex[i] = 0; - } - } - if (stride==2) { - for (i=nrElements*stride-1; i>0; i--) { - aIndex[i] = aIndex[i>>1]; - } - } -} - -/***************************************************************************/ -/*! - \brief Mapping of ICC/IID parameters to 20 stereo bands - - \return none - -****************************************************************************/ -static void map34IndexTo20 (SCHAR *aIndex, /*!< decoded ICC/IID parameters */ - UCHAR noBins) /*!< number of stereo bands */ -{ - aIndex[0] = (2*aIndex[0]+aIndex[1])/3; - aIndex[1] = (aIndex[1]+2*aIndex[2])/3; - aIndex[2] = (2*aIndex[3]+aIndex[4])/3; - aIndex[3] = (aIndex[4]+2*aIndex[5])/3; - aIndex[4] = (aIndex[6]+aIndex[7])/2; - aIndex[5] = (aIndex[8]+aIndex[9])/2; - aIndex[6] = aIndex[10]; - aIndex[7] = aIndex[11]; - aIndex[8] = (aIndex[12]+aIndex[13])/2; - aIndex[9] = (aIndex[14]+aIndex[15])/2; - aIndex[10] = aIndex[16]; - /* For IPD/OPD it stops here */ - - if (noBins == NO_HI_RES_BINS) - { - aIndex[11] = aIndex[17]; - aIndex[12] = aIndex[18]; - aIndex[13] = aIndex[19]; - aIndex[14] = (aIndex[20]+aIndex[21])/2; - aIndex[15] = (aIndex[22]+aIndex[23])/2; - aIndex[16] = (aIndex[24]+aIndex[25])/2; - aIndex[17] = (aIndex[26]+aIndex[27])/2; - aIndex[18] = (aIndex[28]+aIndex[29]+aIndex[30]+aIndex[31])/4; - aIndex[19] = (aIndex[32]+aIndex[33])/2; - } -} - -/***************************************************************************/ -/*! - \brief Decodes delta coded IID, ICC, IPD and OPD indices - - \return PS processing flag. If set to 1 - -****************************************************************************/ -int -DecodePs( struct PS_DEC *h_ps_d, /*!< PS handle */ - const UCHAR frameError ) /*!< Flag telling that frame had errors */ -{ - MPEG_PS_BS_DATA *pBsData; - UCHAR gr, env; - int bPsHeaderValid, bPsDataAvail; - - /* Shortcuts to avoid deferencing and keep the code readable */ - pBsData = &h_ps_d->bsData[h_ps_d->processSlot].mpeg; - bPsHeaderValid = pBsData->bPsHeaderValid; - bPsDataAvail = (h_ps_d->bPsDataAvail[h_ps_d->processSlot] == ppt_mpeg) ? 1 : 0; - - /*************************************************************************************** - * Decide whether to process or to conceal PS data or not. */ - - if ( ( h_ps_d->psDecodedPrv && !frameError && !bPsDataAvail) - || (!h_ps_d->psDecodedPrv && (frameError || !bPsDataAvail || !bPsHeaderValid)) ) { - /* Don't apply PS processing. - * Declare current PS header and bitstream data invalid. */ - pBsData->bPsHeaderValid = 0; - h_ps_d->bPsDataAvail[h_ps_d->processSlot] = ppt_none; - return (0); - } - - if (frameError || !bPsHeaderValid) - { /* no new PS data available (e.g. frame loss) */ - /* => keep latest data constant (i.e. FIX with noEnv=0) */ - pBsData->noEnv = 0; - } - - /*************************************************************************************** - * Decode bitstream payload or prepare parameter for concealment: - */ - for (env=0; envnoEnv; env++) { - SCHAR *aPrevIidIndex; - SCHAR *aPrevIccIndex; - - UCHAR noIidSteps = pBsData->bFineIidQ?NO_IID_STEPS_FINE:NO_IID_STEPS; - - if (env==0) { - aPrevIidIndex = h_ps_d->specificTo.mpeg.aIidPrevFrameIndex; - aPrevIccIndex = h_ps_d->specificTo.mpeg.aIccPrevFrameIndex; - } - else { - aPrevIidIndex = pBsData->aaIidIndex[env-1]; - aPrevIccIndex = pBsData->aaIccIndex[env-1]; - } - - deltaDecodeArray(pBsData->bEnableIid, - pBsData->aaIidIndex[env], - aPrevIidIndex, - pBsData->abIidDtFlag[env], - FDK_sbrDecoder_aNoIidBins[pBsData->freqResIid], - (pBsData->freqResIid)?1:2, - -noIidSteps, - noIidSteps); - - deltaDecodeArray(pBsData->bEnableIcc, - pBsData->aaIccIndex[env], - aPrevIccIndex, - pBsData->abIccDtFlag[env], - FDK_sbrDecoder_aNoIccBins[pBsData->freqResIcc], - (pBsData->freqResIcc)?1:2, - 0, - NO_ICC_STEPS-1); - } /* for (env=0; envnoEnv; env++) */ - - /* handling of FIX noEnv=0 */ - if (pBsData->noEnv==0) { - /* set noEnv=1, keep last parameters or force 0 if not enabled */ - pBsData->noEnv = 1; - - if (pBsData->bEnableIid) { - for (gr = 0; gr < NO_HI_RES_IID_BINS; gr++) { - pBsData->aaIidIndex[pBsData->noEnv-1][gr] = - h_ps_d->specificTo.mpeg.aIidPrevFrameIndex[gr]; - } - } - else { - for (gr = 0; gr < NO_HI_RES_IID_BINS; gr++) { - pBsData->aaIidIndex[pBsData->noEnv-1][gr] = 0; - } - } - - if (pBsData->bEnableIcc) { - for (gr = 0; gr < NO_HI_RES_ICC_BINS; gr++) { - pBsData->aaIccIndex[pBsData->noEnv-1][gr] = - h_ps_d->specificTo.mpeg.aIccPrevFrameIndex[gr]; - } - } - else { - for (gr = 0; gr < NO_HI_RES_ICC_BINS; gr++) { - pBsData->aaIccIndex[pBsData->noEnv-1][gr] = 0; - } - } - } - - /* Update previous frame index buffers */ - for (gr = 0; gr < NO_HI_RES_IID_BINS; gr++) { - h_ps_d->specificTo.mpeg.aIidPrevFrameIndex[gr] = - pBsData->aaIidIndex[pBsData->noEnv-1][gr]; - } - for (gr = 0; gr < NO_HI_RES_ICC_BINS; gr++) { - h_ps_d->specificTo.mpeg.aIccPrevFrameIndex[gr] = - pBsData->aaIccIndex[pBsData->noEnv-1][gr]; - } - - /* PS data from bitstream (if avail) was decoded now */ - h_ps_d->bPsDataAvail[h_ps_d->processSlot] = ppt_none; - - /* handling of env borders for FIX & VAR */ - if (pBsData->bFrameClass == 0) { - /* FIX_BORDERS NoEnv=0,1,2,4 */ - pBsData->aEnvStartStop[0] = 0; - for (env=1; envnoEnv; env++) { - pBsData->aEnvStartStop[env] = - (env * h_ps_d->noSubSamples) / pBsData->noEnv; - } - pBsData->aEnvStartStop[pBsData->noEnv] = h_ps_d->noSubSamples; - /* 1024 (32 slots) env borders: 0, 8, 16, 24, 32 */ - /* 960 (30 slots) env borders: 0, 7, 15, 22, 30 */ - } - else { /* if (h_ps_d->bFrameClass == 0) */ - /* VAR_BORDERS NoEnv=1,2,3,4 */ - pBsData->aEnvStartStop[0] = 0; - - /* handle case aEnvStartStop[noEnv]aEnvStartStop[pBsData->noEnv] < h_ps_d->noSubSamples) { - for (gr = 0; gr < NO_HI_RES_IID_BINS; gr++) { - pBsData->aaIidIndex[pBsData->noEnv][gr] = - pBsData->aaIidIndex[pBsData->noEnv-1][gr]; - } - for (gr = 0; gr < NO_HI_RES_ICC_BINS; gr++) { - pBsData->aaIccIndex[pBsData->noEnv][gr] = - pBsData->aaIccIndex[pBsData->noEnv-1][gr]; - } - pBsData->noEnv++; - pBsData->aEnvStartStop[pBsData->noEnv] = h_ps_d->noSubSamples; - } - - /* enforce strictly monotonic increasing borders */ - for (env=1; envnoEnv; env++) { - UCHAR thr; - thr = (UCHAR)h_ps_d->noSubSamples - (pBsData->noEnv - env); - if (pBsData->aEnvStartStop[env] > thr) { - pBsData->aEnvStartStop[env] = thr; - } - else { - thr = pBsData->aEnvStartStop[env-1]+1; - if (pBsData->aEnvStartStop[env] < thr) { - pBsData->aEnvStartStop[env] = thr; - } - } - } - } /* if (h_ps_d->bFrameClass == 0) ... else */ - - /* copy data prior to possible 20<->34 in-place mapping */ - for (env=0; envnoEnv; env++) { - UCHAR i; - for (i=0; ispecificTo.mpeg.coef.aaIidIndexMapped[env][i] = pBsData->aaIidIndex[env][i]; - } - for (i=0; ispecificTo.mpeg.coef.aaIccIndexMapped[env][i] = pBsData->aaIccIndex[env][i]; - } - } - - - /* MPEG baseline PS */ - /* Baseline version of PS always uses the hybrid filter structure with 20 stereo bands. */ - /* If ICC/IID parameters for 34 stereo bands are decoded they have to be mapped to 20 */ - /* stereo bands. */ - /* Additionaly the IPD/OPD parameters won't be used. */ - - for (env=0; envnoEnv; env++) { - if (pBsData->freqResIid == 2) - map34IndexTo20 (h_ps_d->specificTo.mpeg.coef.aaIidIndexMapped[env], NO_HI_RES_IID_BINS); - if (pBsData->freqResIcc == 2) - map34IndexTo20 (h_ps_d->specificTo.mpeg.coef.aaIccIndexMapped[env], NO_HI_RES_ICC_BINS); - - /* IPD/OPD is disabled in baseline version and thus was removed here */ - } - - return (1); -} - - -/***************************************************************************/ -/*! - - \brief Reads parametric stereo data from bitstream - - \return - -****************************************************************************/ -unsigned int -ReadPsData (HANDLE_PS_DEC h_ps_d, /*!< handle to struct PS_DEC */ - HANDLE_FDK_BITSTREAM hBitBuf, /*!< handle to struct BIT_BUF */ - int nBitsLeft /*!< max number of bits available */ - ) -{ - MPEG_PS_BS_DATA *pBsData; - - UCHAR gr, env; - SCHAR dtFlag; - INT startbits; - Huffman CurrentTable; - SCHAR bEnableHeader; - - if (!h_ps_d) - return 0; - - pBsData = &h_ps_d->bsData[h_ps_d->bsReadSlot].mpeg; - - if (h_ps_d->bsReadSlot != h_ps_d->bsLastSlot) { - /* Copy last header data */ - FDKmemcpy(pBsData, &h_ps_d->bsData[h_ps_d->bsLastSlot].mpeg, sizeof(MPEG_PS_BS_DATA)); - } - - - startbits = (INT) FDKgetValidBits(hBitBuf); - - bEnableHeader = (SCHAR) FDKreadBits (hBitBuf, 1); - - /* Read header */ - if (bEnableHeader) { - pBsData->bPsHeaderValid = 1; - pBsData->bEnableIid = (UCHAR) FDKreadBits (hBitBuf, 1); - if (pBsData->bEnableIid) { - pBsData->modeIid = (UCHAR) FDKreadBits (hBitBuf, 3); - } - - pBsData->bEnableIcc = (UCHAR) FDKreadBits (hBitBuf, 1); - if (pBsData->bEnableIcc) { - pBsData->modeIcc = (UCHAR) FDKreadBits (hBitBuf, 3); - } - - pBsData->bEnableExt = (UCHAR) FDKreadBits (hBitBuf, 1); - } - - pBsData->bFrameClass = (UCHAR) FDKreadBits (hBitBuf, 1); - if (pBsData->bFrameClass == 0) { - /* FIX_BORDERS NoEnv=0,1,2,4 */ - pBsData->noEnv = FDK_sbrDecoder_aFixNoEnvDecode[(UCHAR) FDKreadBits (hBitBuf, 2)]; - /* all additional handling of env borders is now in DecodePs() */ - } - else { - /* VAR_BORDERS NoEnv=1,2,3,4 */ - pBsData->noEnv = 1+(UCHAR) FDKreadBits (hBitBuf, 2); - for (env=1; envnoEnv+1; env++) - pBsData->aEnvStartStop[env] = ((UCHAR) FDKreadBits (hBitBuf, 5)) + 1; - /* all additional handling of env borders is now in DecodePs() */ - } - - /* verify that IID & ICC modes (quant grid, freq res) are supported */ - if ((pBsData->modeIid > 5) || (pBsData->modeIcc > 5)) { - /* no useful PS data could be read from bitstream */ - h_ps_d->bPsDataAvail[h_ps_d->bsReadSlot] = ppt_none; - /* discard all remaining bits */ - nBitsLeft -= startbits - FDKgetValidBits(hBitBuf); - while (nBitsLeft > 0) { - int i = nBitsLeft; - if (i>8) { - i = 8; - } - FDKreadBits (hBitBuf, i); - nBitsLeft -= i; - } - return (startbits - FDKgetValidBits(hBitBuf)); - } - - if (pBsData->modeIid > 2){ - pBsData->freqResIid = pBsData->modeIid-3; - pBsData->bFineIidQ = 1; - } - else{ - pBsData->freqResIid = pBsData->modeIid; - pBsData->bFineIidQ = 0; - } - - if (pBsData->modeIcc > 2){ - pBsData->freqResIcc = pBsData->modeIcc-3; - } - else{ - pBsData->freqResIcc = pBsData->modeIcc; - } - - - /* Extract IID data */ - if (pBsData->bEnableIid) { - for (env=0; envnoEnv; env++) { - dtFlag = (SCHAR)FDKreadBits (hBitBuf, 1); - if (!dtFlag) - { - if (pBsData->bFineIidQ) - CurrentTable = (Huffman)&aBookPsIidFineFreqDecode; - else - CurrentTable = (Huffman)&aBookPsIidFreqDecode; - } - else - { - if (pBsData->bFineIidQ) - CurrentTable = (Huffman)&aBookPsIidFineTimeDecode; - else - CurrentTable = (Huffman)&aBookPsIidTimeDecode; - } - - for (gr = 0; gr < FDK_sbrDecoder_aNoIidBins[pBsData->freqResIid]; gr++) - pBsData->aaIidIndex[env][gr] = decode_huff_cw(CurrentTable,hBitBuf,NULL); - pBsData->abIidDtFlag[env] = dtFlag; - } - } - - /* Extract ICC data */ - if (pBsData->bEnableIcc) { - for (env=0; envnoEnv; env++) { - dtFlag = (SCHAR)FDKreadBits (hBitBuf, 1); - if (!dtFlag) - CurrentTable = (Huffman)&aBookPsIccFreqDecode; - else - CurrentTable = (Huffman)&aBookPsIccTimeDecode; - - for (gr = 0; gr < FDK_sbrDecoder_aNoIccBins[pBsData->freqResIcc]; gr++) - pBsData->aaIccIndex[env][gr] = decode_huff_cw(CurrentTable,hBitBuf,NULL); - pBsData->abIccDtFlag[env] = dtFlag; - } - } - - if (pBsData->bEnableExt) { - - /*! - Decoders that support only the baseline version of the PS tool are allowed - to ignore the IPD/OPD data, but according header data has to be parsed. - ISO/IEC 14496-3 Subpart 8 Annex 4 - */ - - int cnt = FDKreadBits(hBitBuf, PS_EXTENSION_SIZE_BITS); - if (cnt == (1<bPsDataAvail[h_ps_d->bsReadSlot] = ppt_mpeg; - - - - return (startbits - FDKgetValidBits(hBitBuf)); -} - diff --git a/libSBRdec/src/psbitdec.h b/libSBRdec/src/psbitdec.h deleted file mode 100644 index a2d4d6c..0000000 --- a/libSBRdec/src/psbitdec.h +++ /dev/null @@ -1,103 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -#ifndef __PSBITDEC_H -#define __PSBITDEC_H - -#include "sbrdecoder.h" - - -#include "psdec.h" - - -unsigned int -ReadPsData (struct PS_DEC *h_ps_d, - HANDLE_FDK_BITSTREAM hBs, - int nBitsLeft); - -int -DecodePs(struct PS_DEC *h_ps_d, - const UCHAR frameError); - - -#endif /* __PSBITDEC_H */ diff --git a/libSBRdec/src/psdec.cpp b/libSBRdec/src/psdec.cpp deleted file mode 100644 index 965917a..0000000 --- a/libSBRdec/src/psdec.cpp +++ /dev/null @@ -1,1414 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief parametric stereo decoder -*/ - -#include "psdec.h" - - - -#include "FDK_bitbuffer.h" -#include "psdec_hybrid.h" - -#include "sbr_rom.h" -#include "sbr_ram.h" - -#include "FDK_tools_rom.h" - -#include "genericStds.h" - -#include "FDK_trigFcts.h" - - -/********************************************************************/ -/* MLQUAL DEFINES */ -/********************************************************************/ - - #define FRACT_ZERO FRACT_BITS-1 -/********************************************************************/ - -SBR_ERROR ResetPsDec( HANDLE_PS_DEC h_ps_d ); - -void ResetPsDeCor( HANDLE_PS_DEC h_ps_d ); - - -/***** HELPERS *****/ - -static void assignTimeSlotsPS (FIXP_DBL *bufAdr, FIXP_DBL **bufPtr, const int numSlots, const int numChan); - - - -/*******************/ - -#define DIV3 FL2FXCONST_DBL(1.f/3.f) /* division 3.0 */ -#define DIV1_5 FL2FXCONST_DBL(2.f/3.f) /* division 1.5 */ - -/***************************************************************************/ -/*! - \brief Creates one instance of the PS_DEC struct - - \return Error info - -****************************************************************************/ -int -CreatePsDec( HANDLE_PS_DEC *h_PS_DEC, /*!< pointer to the module state */ - int aacSamplesPerFrame - ) -{ - SBR_ERROR errorInfo = SBRDEC_OK; - HANDLE_PS_DEC h_ps_d; - int i; - - if (*h_PS_DEC == NULL) { - /* Get ps dec ram */ - h_ps_d = GetRam_ps_dec(); - if (h_ps_d == NULL) { - errorInfo = SBRDEC_MEM_ALLOC_FAILED; - goto bail; - } - } else { - /* Reset an open instance */ - h_ps_d = *h_PS_DEC; - } - - /* initialisation */ - switch (aacSamplesPerFrame) { - case 960: - h_ps_d->noSubSamples = 30; /* col */ - break; - case 1024: - h_ps_d->noSubSamples = 32; /* col */ - break; - default: - h_ps_d->noSubSamples = -1; - break; - } - - if (h_ps_d->noSubSamples > MAX_NUM_COL - || h_ps_d->noSubSamples <= 0) - { - goto bail; - } - h_ps_d->noChannels = NO_QMF_CHANNELS; /* row */ - - h_ps_d->psDecodedPrv = 0; - h_ps_d->procFrameBased = -1; - for (i = 0; i < (1)+1; i++) { - h_ps_d->bPsDataAvail[i] = ppt_none; - } - - - for (i = 0; i < (1)+1; i++) { - FDKmemclear(&h_ps_d->bsData[i].mpeg, sizeof(MPEG_PS_BS_DATA)); - } - - errorInfo = ResetPsDec( h_ps_d ); - - if ( errorInfo != SBRDEC_OK ) - goto bail; - - ResetPsDeCor( h_ps_d ); - - *h_PS_DEC = h_ps_d; - - - - return 0; - -bail: - DeletePsDec(&h_ps_d); - - return -1; -} /*END CreatePsDec */ - -/***************************************************************************/ -/*! - \brief Delete one instance of the PS_DEC struct - - \return Error info - -****************************************************************************/ -int -DeletePsDec( HANDLE_PS_DEC *h_PS_DEC) /*!< pointer to the module state */ -{ - if (*h_PS_DEC == NULL) { - return -1; - } - - - FreeRam_ps_dec(h_PS_DEC); - - - return 0; -} /*END DeletePsDec */ - -/***************************************************************************/ -/*! - \brief resets some values of the PS handle to default states - - \return - -****************************************************************************/ -SBR_ERROR ResetPsDec( HANDLE_PS_DEC h_ps_d ) /*!< pointer to the module state */ -{ - SBR_ERROR errorInfo = SBRDEC_OK; - INT i; - - const UCHAR noQmfBandsInHybrid20 = 3; - /* const UCHAR noQmfBandsInHybrid34 = 5; */ - - const UCHAR aHybridResolution20[] = { HYBRID_8_CPLX, - HYBRID_2_REAL, - HYBRID_2_REAL }; - - h_ps_d->specificTo.mpeg.delayBufIndex = 0; - - /* explicitly init state variables to safe values (until first ps header arrives) */ - - h_ps_d->specificTo.mpeg.lastUsb = 0; - - h_ps_d->specificTo.mpeg.scaleFactorPsDelayBuffer = -(DFRACT_BITS-1); - - FDKmemclear(h_ps_d->specificTo.mpeg.aDelayBufIndexDelayQmf, (NO_QMF_CHANNELS-FIRST_DELAY_SB)*sizeof(UCHAR)); - h_ps_d->specificTo.mpeg.noSampleDelay = delayIndexQmf[0]; - - for (i=0 ; i < NO_SERIAL_ALLPASS_LINKS; i++) { - h_ps_d->specificTo.mpeg.aDelayRBufIndexSer[i] = 0; - } - - h_ps_d->specificTo.mpeg.pAaRealDelayBufferQmf[0] = h_ps_d->specificTo.mpeg.aaQmfDelayBufReal; - - assignTimeSlotsPS ( h_ps_d->specificTo.mpeg.pAaRealDelayBufferQmf[0] + (NO_QMF_CHANNELS-FIRST_DELAY_SB), - &h_ps_d->specificTo.mpeg.pAaRealDelayBufferQmf[1], - h_ps_d->specificTo.mpeg.noSampleDelay-1, - (NO_DELAY_BUFFER_BANDS-FIRST_DELAY_SB)); - - h_ps_d->specificTo.mpeg.pAaImagDelayBufferQmf[0] = h_ps_d->specificTo.mpeg.aaQmfDelayBufImag; - - assignTimeSlotsPS ( h_ps_d->specificTo.mpeg.pAaImagDelayBufferQmf[0] + (NO_QMF_CHANNELS-FIRST_DELAY_SB), - &h_ps_d->specificTo.mpeg.pAaImagDelayBufferQmf[1], - h_ps_d->specificTo.mpeg.noSampleDelay-1, - (NO_DELAY_BUFFER_BANDS-FIRST_DELAY_SB)); - - /* Hybrid Filter Bank 1 creation. */ - errorInfo = InitHybridFilterBank ( &h_ps_d->specificTo.mpeg.hybrid, - h_ps_d->noSubSamples, - noQmfBandsInHybrid20, - aHybridResolution20 ); - - for ( i = 0; i < NO_IID_GROUPS; i++ ) - { - h_ps_d->specificTo.mpeg.h11rPrev[i] = FL2FXCONST_DBL(0.5f); - h_ps_d->specificTo.mpeg.h12rPrev[i] = FL2FXCONST_DBL(0.5f); - } - - FDKmemclear( h_ps_d->specificTo.mpeg.h21rPrev, sizeof( h_ps_d->specificTo.mpeg.h21rPrev ) ); - FDKmemclear( h_ps_d->specificTo.mpeg.h22rPrev, sizeof( h_ps_d->specificTo.mpeg.h22rPrev ) ); - - return errorInfo; -} - -/***************************************************************************/ -/*! - \brief clear some buffers used in decorrelation process - - \return - -****************************************************************************/ -void ResetPsDeCor( HANDLE_PS_DEC h_ps_d ) /*!< pointer to the module state */ -{ - INT i; - - FDKmemclear(h_ps_d->specificTo.mpeg.aPeakDecayFastBin, NO_MID_RES_BINS*sizeof(FIXP_DBL)); - FDKmemclear(h_ps_d->specificTo.mpeg.aPrevNrgBin, NO_MID_RES_BINS*sizeof(FIXP_DBL)); - FDKmemclear(h_ps_d->specificTo.mpeg.aPrevPeakDiffBin, NO_MID_RES_BINS*sizeof(FIXP_DBL)); - FDKmemclear(h_ps_d->specificTo.mpeg.aPowerPrevScal, NO_MID_RES_BINS*sizeof(SCHAR)); - - for (i=0 ; i < FIRST_DELAY_SB ; i++) { - FDKmemclear(h_ps_d->specificTo.mpeg.aaaRealDelayRBufferSerQmf[i], NO_DELAY_LENGTH_VECTORS*sizeof(FIXP_DBL)); - FDKmemclear(h_ps_d->specificTo.mpeg.aaaImagDelayRBufferSerQmf[i], NO_DELAY_LENGTH_VECTORS*sizeof(FIXP_DBL)); - } - for (i=0 ; i < NO_SUB_QMF_CHANNELS ; i++) { - FDKmemclear(h_ps_d->specificTo.mpeg.aaaRealDelayRBufferSerSubQmf[i], NO_DELAY_LENGTH_VECTORS*sizeof(FIXP_DBL)); - FDKmemclear(h_ps_d->specificTo.mpeg.aaaImagDelayRBufferSerSubQmf[i], NO_DELAY_LENGTH_VECTORS*sizeof(FIXP_DBL)); - } - -} - -/*******************************************************************************/ - -/* slot based funcion prototypes */ - -static void deCorrelateSlotBased( HANDLE_PS_DEC h_ps_d, - - FIXP_DBL *mHybridRealLeft, - FIXP_DBL *mHybridImagLeft, - SCHAR sf_mHybridLeft, - - FIXP_DBL *rIntBufferLeft, - FIXP_DBL *iIntBufferLeft, - SCHAR sf_IntBuffer, - - FIXP_DBL *mHybridRealRight, - FIXP_DBL *mHybridImagRight, - - FIXP_DBL *rIntBufferRight, - FIXP_DBL *iIntBufferRight ); - -static void applySlotBasedRotation( HANDLE_PS_DEC h_ps_d, - - FIXP_DBL *mHybridRealLeft, - FIXP_DBL *mHybridImagLeft, - - FIXP_DBL *QmfLeftReal, - FIXP_DBL *QmfLeftImag, - - FIXP_DBL *mHybridRealRight, - FIXP_DBL *mHybridImagRight, - - FIXP_DBL *QmfRightReal, - FIXP_DBL *QmfRightImag - ); - - -/***************************************************************************/ -/*! - \brief Get scale factor for all ps delay buffer. - - \return - -****************************************************************************/ -static -int getScaleFactorPsStatesBuffer(HANDLE_PS_DEC h_ps_d) -{ - INT i; - int scale = DFRACT_BITS-1; - - for (i=0; ispecificTo.mpeg.hybrid.mQmfBufferRealSlot[i], NO_SUB_QMF_CHANNELS)); - scale = fMin(scale, getScalefactor(h_ps_d->specificTo.mpeg.hybrid.mQmfBufferImagSlot[i], NO_SUB_QMF_CHANNELS)); - } - - for (i=0; ispecificTo.mpeg.aaRealDelayBufferQmf[i], FIRST_DELAY_SB)); - scale = fMin(scale, getScalefactor(h_ps_d->specificTo.mpeg.aaImagDelayBufferQmf[i], FIRST_DELAY_SB)); - } - - for (i=0; ispecificTo.mpeg.aaRealDelayBufferSubQmf[i], NO_SUB_QMF_CHANNELS)); - scale = fMin(scale, getScalefactor(h_ps_d->specificTo.mpeg.aaImagDelayBufferSubQmf[i], NO_SUB_QMF_CHANNELS)); - } - - for (i=0; ispecificTo.mpeg.aaaRealDelayRBufferSerQmf[i], NO_DELAY_LENGTH_VECTORS)); - scale = fMin(scale, getScalefactor(h_ps_d->specificTo.mpeg.aaaImagDelayRBufferSerQmf[i], NO_DELAY_LENGTH_VECTORS)); - } - - for (i=0; ispecificTo.mpeg.aaaRealDelayRBufferSerSubQmf[i], NO_DELAY_LENGTH_VECTORS)); - scale = fMin(scale, getScalefactor(h_ps_d->specificTo.mpeg.aaaImagDelayRBufferSerSubQmf[i], NO_DELAY_LENGTH_VECTORS)); - } - - for (i=0; ispecificTo.mpeg.pAaRealDelayBufferQmf[i], len)); - scale = fMin(scale, getScalefactor(h_ps_d->specificTo.mpeg.pAaImagDelayBufferQmf[i], len)); - } - - return (scale); -} - -/***************************************************************************/ -/*! - \brief Rescale all ps delay buffer. - - \return - -****************************************************************************/ -static -void scalePsStatesBuffer(HANDLE_PS_DEC h_ps_d, - int scale) -{ - INT i; - - if (scale < 0) - scale = fixMax((INT)scale,(INT)-(DFRACT_BITS-1)); - else - scale = fixMin((INT)scale,(INT)DFRACT_BITS-1); - - for (i=0; ispecificTo.mpeg.hybrid.mQmfBufferRealSlot[i], NO_SUB_QMF_CHANNELS, scale ); - scaleValues( h_ps_d->specificTo.mpeg.hybrid.mQmfBufferImagSlot[i], NO_SUB_QMF_CHANNELS, scale ); - } - - for (i=0; ispecificTo.mpeg.aaRealDelayBufferQmf[i], FIRST_DELAY_SB, scale ); - scaleValues( h_ps_d->specificTo.mpeg.aaImagDelayBufferQmf[i], FIRST_DELAY_SB, scale ); - } - - for (i=0; ispecificTo.mpeg.aaRealDelayBufferSubQmf[i], NO_SUB_QMF_CHANNELS, scale ); - scaleValues( h_ps_d->specificTo.mpeg.aaImagDelayBufferSubQmf[i], NO_SUB_QMF_CHANNELS, scale ); - } - - for (i=0; ispecificTo.mpeg.aaaRealDelayRBufferSerQmf[i], NO_DELAY_LENGTH_VECTORS, scale ); - scaleValues( h_ps_d->specificTo.mpeg.aaaImagDelayRBufferSerQmf[i], NO_DELAY_LENGTH_VECTORS, scale ); - } - - for (i=0; ispecificTo.mpeg.aaaRealDelayRBufferSerSubQmf[i], NO_DELAY_LENGTH_VECTORS, scale ); - scaleValues( h_ps_d->specificTo.mpeg.aaaImagDelayRBufferSerSubQmf[i], NO_DELAY_LENGTH_VECTORS, scale ); - } - - for (i=0; ispecificTo.mpeg.pAaRealDelayBufferQmf[i], len, scale ); - scaleValues( h_ps_d->specificTo.mpeg.pAaImagDelayBufferQmf[i], len, scale ); - } - - scale <<= 1; - - scaleValues( h_ps_d->specificTo.mpeg.aPeakDecayFastBin, NO_MID_RES_BINS, scale ); - scaleValues( h_ps_d->specificTo.mpeg.aPrevPeakDiffBin, NO_MID_RES_BINS, scale ); - scaleValues( h_ps_d->specificTo.mpeg.aPrevNrgBin, NO_MID_RES_BINS, scale ); -} - -/***************************************************************************/ -/*! - \brief Scale input channel to the same scalefactor and rescale hybrid - filterbank values - - \return - -****************************************************************************/ - -void scalFilterBankValues( HANDLE_PS_DEC h_ps_d, - FIXP_DBL **fixpQmfReal, - FIXP_DBL **fixpQmfImag, - int lsb, - int scaleFactorLowBandSplitLow, - int scaleFactorLowBandSplitHigh, - SCHAR *scaleFactorLowBand_lb, - SCHAR *scaleFactorLowBand_hb, - int scaleFactorHighBands, - INT *scaleFactorHighBand, - INT noCols - ) -{ - INT maxScal; - - INT i; - - scaleFactorHighBands = -scaleFactorHighBands; - scaleFactorLowBandSplitLow = -scaleFactorLowBandSplitLow; - scaleFactorLowBandSplitHigh = -scaleFactorLowBandSplitHigh; - - /* get max scale factor */ - maxScal = fixMax(scaleFactorHighBands,fixMax(scaleFactorLowBandSplitLow, scaleFactorLowBandSplitHigh )); - - { - int headroom = getScaleFactorPsStatesBuffer(h_ps_d); - maxScal = fixMax(maxScal,(INT)(h_ps_d->specificTo.mpeg.scaleFactorPsDelayBuffer-headroom)); - maxScal += 1; - } - - /* scale whole left channel to the same scale factor */ - - /* low band ( overlap buffer ) */ - if ( maxScal != scaleFactorLowBandSplitLow ) { - INT scale = scaleFactorLowBandSplitLow - maxScal; - for ( i=0; i<(6); i++ ) { - scaleValues( fixpQmfReal[i], lsb, scale ); - scaleValues( fixpQmfImag[i], lsb, scale ); - } - } - /* low band ( current frame ) */ - if ( maxScal != scaleFactorLowBandSplitHigh ) { - INT scale = scaleFactorLowBandSplitHigh - maxScal; - /* for ( i=(6); i<(6)+MAX_NUM_COL; i++ ) { */ - for ( i=(6); i<(6)+noCols; i++ ) { - scaleValues( fixpQmfReal[i], lsb, scale ); - scaleValues( fixpQmfImag[i], lsb, scale ); - } - } - /* high band */ - if ( maxScal != scaleFactorHighBands ) { - INT scale = scaleFactorHighBands - maxScal; - /* for ( i=0; ispecificTo.mpeg.scaleFactorPsDelayBuffer ) - scalePsStatesBuffer(h_ps_d,(h_ps_d->specificTo.mpeg.scaleFactorPsDelayBuffer-maxScal)); - - h_ps_d->specificTo.mpeg.hybrid.sf_mQmfBuffer = maxScal; - h_ps_d->specificTo.mpeg.scaleFactorPsDelayBuffer = maxScal; - - *scaleFactorHighBand += maxScal - scaleFactorHighBands; - - h_ps_d->rescal = maxScal - scaleFactorLowBandSplitHigh; - h_ps_d->sf_IntBuffer = maxScal; - - *scaleFactorLowBand_lb += maxScal - scaleFactorLowBandSplitLow; - *scaleFactorLowBand_hb += maxScal - scaleFactorLowBandSplitHigh; -} - -void rescalFilterBankValues( HANDLE_PS_DEC h_ps_d, /* parametric stereo decoder handle */ - FIXP_DBL **QmfBufferReal, /* qmf filterbank values */ - FIXP_DBL **QmfBufferImag, /* qmf filterbank values */ - int lsb, /* sbr start subband */ - INT noCols) -{ - int i; - /* scale back 6 timeslots look ahead for hybrid filterbank to original value */ - for ( i=noCols; irescal ); - scaleValues( QmfBufferImag[i], lsb, h_ps_d->rescal ); - } -} - -/***************************************************************************/ -/*! - \brief Generate decorrelated side channel using allpass/delay - - \return - -****************************************************************************/ -static void -deCorrelateSlotBased( HANDLE_PS_DEC h_ps_d, /*!< pointer to the module state */ - - FIXP_DBL *mHybridRealLeft, /*!< left (mono) hybrid values real */ - FIXP_DBL *mHybridImagLeft, /*!< left (mono) hybrid values imag */ - SCHAR sf_mHybridLeft, /*!< scalefactor for left (mono) hybrid bands */ - - FIXP_DBL *rIntBufferLeft, /*!< real qmf bands left (mono) (38x64) */ - FIXP_DBL *iIntBufferLeft, /*!< real qmf bands left (mono) (38x64) */ - SCHAR sf_IntBuffer, /*!< scalefactor for all left and right qmf bands */ - - FIXP_DBL *mHybridRealRight, /*!< right (decorrelated) hybrid values real */ - FIXP_DBL *mHybridImagRight, /*!< right (decorrelated) hybrid values imag */ - - FIXP_DBL *rIntBufferRight, /*!< real qmf bands right (decorrelated) (38x64) */ - FIXP_DBL *iIntBufferRight ) /*!< real qmf bands right (decorrelated) (38x64) */ -{ - - INT i, m, sb, gr, bin; - - FIXP_DBL peakDiff, nrg, transRatio; - - FIXP_DBL *RESTRICT aaLeftReal; - FIXP_DBL *RESTRICT aaLeftImag; - - FIXP_DBL *RESTRICT aaRightReal; - FIXP_DBL *RESTRICT aaRightImag; - - FIXP_DBL *RESTRICT pRealDelayBuffer; - FIXP_DBL *RESTRICT pImagDelayBuffer; - - C_ALLOC_SCRATCH_START(aaPowerSlot, FIXP_DBL, NO_MID_RES_BINS); - C_ALLOC_SCRATCH_START(aaTransRatioSlot, FIXP_DBL, NO_MID_RES_BINS); - -/*! -
-   parameter index       qmf bands             hybrid bands
-  ----------------------------------------------------------------------------
-         0                   0                      0,7
-         1                   0                      1,6
-         2                   0                      2
-         3                   0                      3           HYBRID BANDS
-         4                   1                      9
-         5                   1                      8
-         6                   2                     10
-         7                   2                     11
-  ----------------------------------------------------------------------------
-         8                   3
-         9                   4
-        10                   5
-        11                   6
-        12                   7
-        13                   8
-        14                   9,10      (2 )                      QMF BANDS
-        15                   11 - 13   (3 )
-        16                   14 - 17   (4 )
-        17                   18 - 22   (5 )
-        18                   23 - 34   (12)
-        19                   35 - 63   (29)
-  ----------------------------------------------------------------------------
-
-*/ - - #define FLTR_SCALE 3 - - /* hybrid bands (parameter index 0 - 7) */ - aaLeftReal = mHybridRealLeft; - aaLeftImag = mHybridImagLeft; - - aaPowerSlot[0] = ( fMultAddDiv2( fMultDiv2(aaLeftReal[0], aaLeftReal[0]), aaLeftImag[0], aaLeftImag[0] ) >> FLTR_SCALE ) + - ( fMultAddDiv2( fMultDiv2(aaLeftReal[7], aaLeftReal[7]), aaLeftImag[7], aaLeftImag[7] ) >> FLTR_SCALE ); - - aaPowerSlot[1] = ( fMultAddDiv2( fMultDiv2(aaLeftReal[1], aaLeftReal[1]), aaLeftImag[1], aaLeftImag[1] ) >> FLTR_SCALE ) + - ( fMultAddDiv2( fMultDiv2(aaLeftReal[6], aaLeftReal[6]), aaLeftImag[6], aaLeftImag[6] ) >> FLTR_SCALE ); - - aaPowerSlot[2] = fMultAddDiv2( fMultDiv2(aaLeftReal[2], aaLeftReal[2]), aaLeftImag[2], aaLeftImag[2] ) >> FLTR_SCALE; - aaPowerSlot[3] = fMultAddDiv2( fMultDiv2(aaLeftReal[3], aaLeftReal[3]), aaLeftImag[3], aaLeftImag[3] ) >> FLTR_SCALE; - - aaPowerSlot[4] = fMultAddDiv2( fMultDiv2(aaLeftReal[9], aaLeftReal[9]), aaLeftImag[9], aaLeftImag[9] ) >> FLTR_SCALE; - aaPowerSlot[5] = fMultAddDiv2( fMultDiv2(aaLeftReal[8], aaLeftReal[8]), aaLeftImag[8], aaLeftImag[8] ) >> FLTR_SCALE; - - aaPowerSlot[6] = fMultAddDiv2( fMultDiv2(aaLeftReal[10], aaLeftReal[10]), aaLeftImag[10], aaLeftImag[10] ) >> FLTR_SCALE; - aaPowerSlot[7] = fMultAddDiv2( fMultDiv2(aaLeftReal[11], aaLeftReal[11]), aaLeftImag[11], aaLeftImag[11] ) >> FLTR_SCALE; - - /* qmf bands (parameter index 8 - 19) */ - for ( bin = 8; bin < NO_MID_RES_BINS; bin++ ) { - FIXP_DBL slotNrg = FL2FXCONST_DBL(0.f); - - for ( i = groupBorders20[bin+2]; i < groupBorders20[bin+3]; i++ ) { /* max loops: 29 */ - slotNrg += fMultAddDiv2 ( fMultDiv2(rIntBufferLeft[i], rIntBufferLeft[i]), iIntBufferLeft[i], iIntBufferLeft[i]) >> FLTR_SCALE; - } - aaPowerSlot[bin] = slotNrg; - - } - - - /* calculation of transient ratio */ - for (bin=0; bin < NO_MID_RES_BINS; bin++) { /* noBins = 20 ( BASELINE_PS ) */ - - h_ps_d->specificTo.mpeg.aPeakDecayFastBin[bin] = fMult( h_ps_d->specificTo.mpeg.aPeakDecayFastBin[bin], PEAK_DECAY_FACTOR ); - - if (h_ps_d->specificTo.mpeg.aPeakDecayFastBin[bin] < aaPowerSlot[bin]) { - h_ps_d->specificTo.mpeg.aPeakDecayFastBin[bin] = aaPowerSlot[bin]; - } - - /* calculate PSmoothPeakDecayDiffNrg */ - peakDiff = fMultAdd ( (h_ps_d->specificTo.mpeg.aPrevPeakDiffBin[bin]>>1), - INT_FILTER_COEFF, h_ps_d->specificTo.mpeg.aPeakDecayFastBin[bin] - aaPowerSlot[bin] - h_ps_d->specificTo.mpeg.aPrevPeakDiffBin[bin]); - - /* save peakDiff for the next frame */ - h_ps_d->specificTo.mpeg.aPrevPeakDiffBin[bin] = peakDiff; - - nrg = h_ps_d->specificTo.mpeg.aPrevNrgBin[bin] + fMult( INT_FILTER_COEFF, aaPowerSlot[bin] - h_ps_d->specificTo.mpeg.aPrevNrgBin[bin] ); - - /* Negative energies don't exist. But sometimes they appear due to rounding. */ - - nrg = fixMax(nrg,FL2FXCONST_DBL(0.f)); - - /* save nrg for the next frame */ - h_ps_d->specificTo.mpeg.aPrevNrgBin[bin] = nrg; - - nrg = fMult( nrg, TRANSIENT_IMPACT_FACTOR ); - - /* save transient impact factor */ - if ( peakDiff <= nrg || peakDiff == FL2FXCONST_DBL(0.0) ) { - aaTransRatioSlot[bin] = (FIXP_DBL)MAXVAL_DBL /* FL2FXCONST_DBL(1.0f)*/; - } - else if ( nrg <= FL2FXCONST_DBL(0.0f) ) { - aaTransRatioSlot[bin] = FL2FXCONST_DBL(0.f); - } - else { - /* scale to denominator */ - INT scale_left = fixMax(0, CntLeadingZeros(peakDiff) - 1); - aaTransRatioSlot[bin] = schur_div( nrg<specificTo.mpeg.delayBufIndex; /* set delay indices */ - - pRealDelayBuffer = h_ps_d->specificTo.mpeg.aaRealDelayBufferSubQmf[TempDelay]; - pImagDelayBuffer = h_ps_d->specificTo.mpeg.aaImagDelayBufferSubQmf[TempDelay]; - - aaLeftReal = mHybridRealLeft; - aaLeftImag = mHybridImagLeft; - aaRightReal = mHybridRealRight; - aaRightImag = mHybridImagRight; - - /************************/ - /* ICC groups : 0 - 9 */ - /************************/ - - /* gr = ICC groups */ - for (gr=0; gr < SUBQMF_GROUPS; gr++) { - - transRatio = aaTransRatioSlot[bins2groupMap20[gr]]; - - /* sb = subQMF/QMF subband */ - sb = groupBorders20[gr]; - - /* Update delay buffers, sample delay allpass = 2 */ - rTmp0 = pRealDelayBuffer[sb]; - iTmp0 = pImagDelayBuffer[sb]; - - pRealDelayBuffer[sb] = aaLeftReal[sb]; - pImagDelayBuffer[sb] = aaLeftImag[sb]; - - /* delay by fraction */ - cplxMultDiv2(&rR0, &iR0, rTmp0, iTmp0, aaFractDelayPhaseFactorReSubQmf20[sb], aaFractDelayPhaseFactorImSubQmf20[sb]); - rR0<<=1; - iR0<<=1; - - FIXP_DBL *pAaaRealDelayRBufferSerSubQmf = h_ps_d->specificTo.mpeg.aaaRealDelayRBufferSerSubQmf[sb]; - FIXP_DBL *pAaaImagDelayRBufferSerSubQmf = h_ps_d->specificTo.mpeg.aaaImagDelayRBufferSerSubQmf[sb]; - - for (m=0; mspecificTo.mpeg.aDelayRBufIndexSer[m]; - - /* get delayed values from according buffer : m(0)=3; m(1)=4; m(2)=5; */ - rTmp0 = pAaaRealDelayRBufferSerSubQmf[tmpDelayRSer]; - iTmp0 = pAaaImagDelayRBufferSerSubQmf[tmpDelayRSer]; - - /* delay by fraction */ - cplxMultDiv2(&rTmp, &iTmp, rTmp0, iTmp0, aaFractDelayPhaseFactorSerReSubQmf20[sb][m], aaFractDelayPhaseFactorSerImSubQmf20[sb][m]); - - rTmp = (rTmp - fMultDiv2(aAllpassLinkDecaySer[m], rR0)) << 1; - iTmp = (iTmp - fMultDiv2(aAllpassLinkDecaySer[m], iR0)) << 1; - - pAaaRealDelayRBufferSerSubQmf[tmpDelayRSer] = rR0 + fMult(aAllpassLinkDecaySer[m], rTmp); - pAaaImagDelayRBufferSerSubQmf[tmpDelayRSer] = iR0 + fMult(aAllpassLinkDecaySer[m], iTmp); - - rR0 = rTmp; - iR0 = iTmp; - - pAaaRealDelayRBufferSerSubQmf += aAllpassLinkDelaySer[m]; - pAaaImagDelayRBufferSerSubQmf += aAllpassLinkDelaySer[m]; - - } /* m */ - - /* duck if a past transient is found */ - aaRightReal[sb] = fMult(transRatio, rR0); - aaRightImag[sb] = fMult(transRatio, iR0); - - } /* gr */ - - - scaleValues( mHybridRealLeft, NO_SUB_QMF_CHANNELS, -SCAL_HEADROOM ); - scaleValues( mHybridImagLeft, NO_SUB_QMF_CHANNELS, -SCAL_HEADROOM ); - scaleValues( mHybridRealRight, NO_SUB_QMF_CHANNELS, -SCAL_HEADROOM ); - scaleValues( mHybridImagRight, NO_SUB_QMF_CHANNELS, -SCAL_HEADROOM ); - - - /************************/ - - aaLeftReal = rIntBufferLeft; - aaLeftImag = iIntBufferLeft; - aaRightReal = rIntBufferRight; - aaRightImag = iIntBufferRight; - - pRealDelayBuffer = h_ps_d->specificTo.mpeg.aaRealDelayBufferQmf[TempDelay]; - pImagDelayBuffer = h_ps_d->specificTo.mpeg.aaImagDelayBufferQmf[TempDelay]; - - /************************/ - /* ICC groups : 10 - 19 */ - /************************/ - - - /* gr = ICC groups */ - for (gr=SUBQMF_GROUPS; gr < NO_IID_GROUPS - NR_OF_DELAY_GROUPS; gr++) { - - transRatio = aaTransRatioSlot[bins2groupMap20[gr]]; - - /* sb = subQMF/QMF subband */ - for (sb = groupBorders20[gr]; sb < groupBorders20[gr+1]; sb++) { - FIXP_DBL resR, resI; - - /* decayScaleFactor = 1.0f + decay_cutoff * DECAY_SLOPE - DECAY_SLOPE * sb; DECAY_SLOPE = 0.05 */ - FIXP_DBL decayScaleFactor = decayScaleFactTable[sb]; - - /* Update delay buffers, sample delay allpass = 2 */ - rTmp0 = pRealDelayBuffer[sb]; - iTmp0 = pImagDelayBuffer[sb]; - - pRealDelayBuffer[sb] = aaLeftReal[sb]; - pImagDelayBuffer[sb] = aaLeftImag[sb]; - - /* delay by fraction */ - cplxMultDiv2(&rR0, &iR0, rTmp0, iTmp0, aaFractDelayPhaseFactorReQmf[sb], aaFractDelayPhaseFactorImQmf[sb]); - rR0<<=1; - iR0<<=1; - - resR = fMult(decayScaleFactor, rR0); - resI = fMult(decayScaleFactor, iR0); - - FIXP_DBL *pAaaRealDelayRBufferSerQmf = h_ps_d->specificTo.mpeg.aaaRealDelayRBufferSerQmf[sb]; - FIXP_DBL *pAaaImagDelayRBufferSerQmf = h_ps_d->specificTo.mpeg.aaaImagDelayRBufferSerQmf[sb]; - - for (m=0; mspecificTo.mpeg.aDelayRBufIndexSer[m]; - - /* get delayed values from according buffer : m(0)=3; m(1)=4; m(2)=5; */ - rTmp0 = pAaaRealDelayRBufferSerQmf[tmpDelayRSer]; - iTmp0 = pAaaImagDelayRBufferSerQmf[tmpDelayRSer]; - - /* delay by fraction */ - cplxMultDiv2(&rTmp, &iTmp, rTmp0, iTmp0, aaFractDelayPhaseFactorSerReQmf[sb][m], aaFractDelayPhaseFactorSerImQmf[sb][m]); - - rTmp = (rTmp - fMultDiv2(aAllpassLinkDecaySer[m], resR))<<1; - iTmp = (iTmp - fMultDiv2(aAllpassLinkDecaySer[m], resI))<<1; - - resR = fMult(decayScaleFactor, rTmp); - resI = fMult(decayScaleFactor, iTmp); - - pAaaRealDelayRBufferSerQmf[tmpDelayRSer] = rR0 + fMult(aAllpassLinkDecaySer[m], resR); - pAaaImagDelayRBufferSerQmf[tmpDelayRSer] = iR0 + fMult(aAllpassLinkDecaySer[m], resI); - - rR0 = rTmp; - iR0 = iTmp; - - pAaaRealDelayRBufferSerQmf += aAllpassLinkDelaySer[m]; - pAaaImagDelayRBufferSerQmf += aAllpassLinkDelaySer[m]; - - } /* m */ - - /* duck if a past transient is found */ - aaRightReal[sb] = fMult(transRatio, rR0); - aaRightImag[sb] = fMult(transRatio, iR0); - - } /* sb */ - } /* gr */ - - /************************/ - /* ICC groups : 20, 21 */ - /************************/ - - - /* gr = ICC groups */ - for (gr=DELAY_GROUP_OFFSET; gr < NO_IID_GROUPS; gr++) { - - INT sbStart = groupBorders20[gr]; - INT sbStop = groupBorders20[gr+1]; - - UCHAR *pDelayBufIdx = &h_ps_d->specificTo.mpeg.aDelayBufIndexDelayQmf[sbStart-FIRST_DELAY_SB]; - - transRatio = aaTransRatioSlot[bins2groupMap20[gr]]; - - /* sb = subQMF/QMF subband */ - for (sb = sbStart; sb < sbStop; sb++) { - - /* Update delay buffers */ - rR0 = h_ps_d->specificTo.mpeg.pAaRealDelayBufferQmf[*pDelayBufIdx][sb-FIRST_DELAY_SB]; - iR0 = h_ps_d->specificTo.mpeg.pAaImagDelayBufferQmf[*pDelayBufIdx][sb-FIRST_DELAY_SB]; - - h_ps_d->specificTo.mpeg.pAaRealDelayBufferQmf[*pDelayBufIdx][sb-FIRST_DELAY_SB] = aaLeftReal[sb]; - h_ps_d->specificTo.mpeg.pAaImagDelayBufferQmf[*pDelayBufIdx][sb-FIRST_DELAY_SB] = aaLeftImag[sb]; - - /* duck if a past transient is found */ - aaRightReal[sb] = fMult(transRatio, rR0); - aaRightImag[sb] = fMult(transRatio, iR0); - - if (++(*pDelayBufIdx) >= delayIndexQmf[sb]) { - *pDelayBufIdx = 0; - } - pDelayBufIdx++; - - } /* sb */ - } /* gr */ - - - /* Update delay buffer index */ - if (++h_ps_d->specificTo.mpeg.delayBufIndex >= NO_SAMPLE_DELAY_ALLPASS) - h_ps_d->specificTo.mpeg.delayBufIndex = 0; - - for (m=0; mspecificTo.mpeg.aDelayRBufIndexSer[m] >= aAllpassLinkDelaySer[m]) - h_ps_d->specificTo.mpeg.aDelayRBufIndexSer[m] = 0; - } - - - scaleValues( &rIntBufferLeft[NO_QMF_BANDS_HYBRID20], NO_QMF_CHANNELS-NO_QMF_BANDS_HYBRID20, -SCAL_HEADROOM ); - scaleValues( &iIntBufferLeft[NO_QMF_BANDS_HYBRID20], NO_QMF_CHANNELS-NO_QMF_BANDS_HYBRID20, -SCAL_HEADROOM ); - scaleValues( &rIntBufferRight[NO_QMF_BANDS_HYBRID20], NO_QMF_CHANNELS-NO_QMF_BANDS_HYBRID20, -SCAL_HEADROOM ); - scaleValues( &iIntBufferRight[NO_QMF_BANDS_HYBRID20], NO_QMF_CHANNELS-NO_QMF_BANDS_HYBRID20, -SCAL_HEADROOM ); - - /* free memory on scratch */ - C_ALLOC_SCRATCH_END(aaTransRatioSlot, FIXP_DBL, NO_MID_RES_BINS); - C_ALLOC_SCRATCH_END(aaPowerSlot, FIXP_DBL, NO_MID_RES_BINS); -} - - -void initSlotBasedRotation( HANDLE_PS_DEC h_ps_d, /*!< pointer to the module state */ - int env, - int usb - ) { - - INT group = 0; - INT bin = 0; - INT noIidSteps; - -/* const UCHAR *pQuantizedIIDs;*/ - - FIXP_SGL invL; - FIXP_DBL ScaleL, ScaleR; - FIXP_DBL Alpha, Beta; - FIXP_DBL h11r, h12r, h21r, h22r; - - const FIXP_DBL *PScaleFactors; - - /* Overwrite old values in delay buffers when upper subband is higher than in last frame */ - if (env == 0) { - - if ((usb > h_ps_d->specificTo.mpeg.lastUsb) && h_ps_d->specificTo.mpeg.lastUsb) { - - INT i,k,length; - - for (i=h_ps_d->specificTo.mpeg.lastUsb ; i < FIRST_DELAY_SB; i++) { - FDKmemclear(h_ps_d->specificTo.mpeg.aaaRealDelayRBufferSerQmf[i], NO_DELAY_LENGTH_VECTORS*sizeof(FIXP_DBL)); - FDKmemclear(h_ps_d->specificTo.mpeg.aaaImagDelayRBufferSerQmf[i], NO_DELAY_LENGTH_VECTORS*sizeof(FIXP_DBL)); - } - - for (k=0 ; kspecificTo.mpeg.pAaRealDelayBufferQmf[k], FIRST_DELAY_SB*sizeof(FIXP_DBL)); - } - length = (usb-FIRST_DELAY_SB)*sizeof(FIXP_DBL); - if(length>0) { - FDKmemclear(h_ps_d->specificTo.mpeg.pAaRealDelayBufferQmf[0], length); - FDKmemclear(h_ps_d->specificTo.mpeg.pAaImagDelayBufferQmf[0], length); - } - length = (fixMin(NO_DELAY_BUFFER_BANDS,(INT)usb)-FIRST_DELAY_SB)*sizeof(FIXP_DBL); - if(length>0) { - for (k=1 ; k < h_ps_d->specificTo.mpeg.noSampleDelay; k++) { - FDKmemclear(h_ps_d->specificTo.mpeg.pAaRealDelayBufferQmf[k], length); - FDKmemclear(h_ps_d->specificTo.mpeg.pAaImagDelayBufferQmf[k], length); - } - } - } - h_ps_d->specificTo.mpeg.lastUsb = usb; - } /* env == 0 */ - - if (h_ps_d->bsData[h_ps_d->processSlot].mpeg.bFineIidQ) - { - PScaleFactors = ScaleFactorsFine; /* values are shiftet right by one */ - noIidSteps = NO_IID_STEPS_FINE; - /*pQuantizedIIDs = quantizedIIDsFine;*/ - } - - else - { - PScaleFactors = ScaleFactors; /* values are shiftet right by one */ - noIidSteps = NO_IID_STEPS; - /*pQuantizedIIDs = quantizedIIDs;*/ - } - - - /* dequantize and decode */ - for ( group = 0; group < NO_IID_GROUPS; group++ ) { - - bin = bins2groupMap20[group]; - - /*! -

type 'A' rotation

- mixing procedure R_a, used in baseline version
- - Scale-factor vectors c1 and c2 are precalculated in initPsTables () and stored in - scaleFactors[] and scaleFactorsFine[] = pScaleFactors []. - From the linearized IID parameters (intensity differences), two scale factors are - calculated. They are used to obtain the coefficients h11... h22. - */ - - /* ScaleR and ScaleL are scaled by 1 shift right */ - - ScaleR = PScaleFactors[noIidSteps + h_ps_d->specificTo.mpeg.coef.aaIidIndexMapped[env][bin]]; - ScaleL = PScaleFactors[noIidSteps - h_ps_d->specificTo.mpeg.coef.aaIidIndexMapped[env][bin]]; - - Beta = fMult (fMult( Alphas[h_ps_d->specificTo.mpeg.coef.aaIccIndexMapped[env][bin]], ( ScaleR - ScaleL )), FIXP_SQRT05); - Alpha = Alphas[h_ps_d->specificTo.mpeg.coef.aaIccIndexMapped[env][bin]]>>1; - - /* Alpha and Beta are now both scaled by 2 shifts right */ - - /* calculate the coefficients h11... h22 from scale-factors and ICC parameters */ - - /* h values are scaled by 1 shift right */ - { - FIXP_DBL trigData[4]; - - inline_fixp_cos_sin(Beta + Alpha, Beta - Alpha, 2, trigData); - h11r = fMult( ScaleL, trigData[0]); - h12r = fMult( ScaleR, trigData[2]); - h21r = fMult( ScaleL, trigData[1]); - h22r = fMult( ScaleR, trigData[3]); - } - /*****************************************************************************************/ - /* Interpolation of the matrices H11... H22: */ - /* */ - /* H11(k,n) = H11(k,n[e]) + (n-n[e]) * (H11(k,n[e+1] - H11(k,n[e])) / (n[e+1] - n[e]) */ - /* ... */ - /*****************************************************************************************/ - - /* invL = 1/(length of envelope) */ - invL = FX_DBL2FX_SGL(GetInvInt(h_ps_d->bsData[h_ps_d->processSlot].mpeg.aEnvStartStop[env + 1] - h_ps_d->bsData[h_ps_d->processSlot].mpeg.aEnvStartStop[env])); - - h_ps_d->specificTo.mpeg.coef.H11r[group] = h_ps_d->specificTo.mpeg.h11rPrev[group]; - h_ps_d->specificTo.mpeg.coef.H12r[group] = h_ps_d->specificTo.mpeg.h12rPrev[group]; - h_ps_d->specificTo.mpeg.coef.H21r[group] = h_ps_d->specificTo.mpeg.h21rPrev[group]; - h_ps_d->specificTo.mpeg.coef.H22r[group] = h_ps_d->specificTo.mpeg.h22rPrev[group]; - - h_ps_d->specificTo.mpeg.coef.DeltaH11r[group] = fMult ( h11r - h_ps_d->specificTo.mpeg.coef.H11r[group], invL ); - h_ps_d->specificTo.mpeg.coef.DeltaH12r[group] = fMult ( h12r - h_ps_d->specificTo.mpeg.coef.H12r[group], invL ); - h_ps_d->specificTo.mpeg.coef.DeltaH21r[group] = fMult ( h21r - h_ps_d->specificTo.mpeg.coef.H21r[group], invL ); - h_ps_d->specificTo.mpeg.coef.DeltaH22r[group] = fMult ( h22r - h_ps_d->specificTo.mpeg.coef.H22r[group], invL ); - - /* update prev coefficients for interpolation in next envelope */ - - h_ps_d->specificTo.mpeg.h11rPrev[group] = h11r; - h_ps_d->specificTo.mpeg.h12rPrev[group] = h12r; - h_ps_d->specificTo.mpeg.h21rPrev[group] = h21r; - h_ps_d->specificTo.mpeg.h22rPrev[group] = h22r; - - } /* group loop */ -} - - -static void applySlotBasedRotation( HANDLE_PS_DEC h_ps_d, /*!< pointer to the module state */ - - FIXP_DBL *mHybridRealLeft, /*!< hybrid values real left */ - FIXP_DBL *mHybridImagLeft, /*!< hybrid values imag left */ - - FIXP_DBL *QmfLeftReal, /*!< real bands left qmf channel */ - FIXP_DBL *QmfLeftImag, /*!< imag bands left qmf channel */ - - FIXP_DBL *mHybridRealRight, /*!< hybrid values real right */ - FIXP_DBL *mHybridImagRight, /*!< hybrid values imag right */ - - FIXP_DBL *QmfRightReal, /*!< real bands right qmf channel */ - FIXP_DBL *QmfRightImag /*!< imag bands right qmf channel */ - ) -{ - INT group; - INT subband; - - FIXP_DBL *RESTRICT HybrLeftReal; - FIXP_DBL *RESTRICT HybrLeftImag; - FIXP_DBL *RESTRICT HybrRightReal; - FIXP_DBL *RESTRICT HybrRightImag; - - FIXP_DBL tmpLeft, tmpRight; - - - /**********************************************************************************************/ - /*! -

Mapping

- - The number of stereo bands that is actually used depends on the number of availble - parameters for IID and ICC: - 
-   nr. of IID para.| nr. of ICC para. | nr. of Stereo bands
-   ----------------|------------------|-------------------
-     10,20         |     10,20        |        20
-     10,20         |     34           |        34
-     34            |     10,20        |        34
-     34            |     34           |        34
-
- In the case the number of parameters for IIS and ICC differs from the number of stereo - bands, a mapping from the lower number to the higher number of parameters is applied. - Index mapping of IID and ICC parameters is already done in psbitdec.cpp. Further mapping is - not needed here in baseline version. - **********************************************************************************************/ - - /************************************************************************************************/ - /*! -

Mixing

- - To generate the QMF subband signals for the subband samples n = n[e]+1 ,,, n_[e+1] the - parameters at position n[e] and n[e+1] are required as well as the subband domain signals - s_k(n) and d_k(n) for n = n[e]+1... n_[e+1]. n[e] represents the start position for - envelope e. The border positions n[e] are handled in DecodePS(). - - The stereo sub subband signals are constructed as: - 
-  l_k(n) = H11(k,n) s_k(n) + H21(k,n) d_k(n)
-  r_k(n) = H21(k,n) s_k(n) + H22(k,n) d_k(n)
-
- In order to obtain the matrices H11(k,n)... H22 (k,n), the vectors h11(b)... h22(b) need to - be calculated first (b: parameter index). Depending on ICC mode either mixing procedure R_a - or R_b is used for that. For both procedures, the parameters for parameter position n[e+1] - is used. - ************************************************************************************************/ - - - /************************************************************************************************/ - /*! -

Phase parameters

- With disabled phase parameters (which is the case in baseline version), the H-matrices are - just calculated by: - - 
-  H11(k,n[e+1] = h11(b(k))
-  (...)
-  b(k): parameter index according to mapping table
-
- -

Processing of the samples in the sub subbands

- this loop includes the interpolation of the coefficients Hxx - ************************************************************************************************/ - - - /* loop thru all groups ... */ - HybrLeftReal = mHybridRealLeft; - HybrLeftImag = mHybridImagLeft; - HybrRightReal = mHybridRealRight; - HybrRightImag = mHybridImagRight; - - /******************************************************/ - /* construct stereo sub subband signals according to: */ - /* */ - /* l_k(n) = H11(k,n) s_k(n) + H21(k,n) d_k(n) */ - /* r_k(n) = H12(k,n) s_k(n) + H22(k,n) d_k(n) */ - /******************************************************/ - for ( group = 0; group < SUBQMF_GROUPS; group++ ) { - - h_ps_d->specificTo.mpeg.coef.H11r[group] += h_ps_d->specificTo.mpeg.coef.DeltaH11r[group]; - h_ps_d->specificTo.mpeg.coef.H12r[group] += h_ps_d->specificTo.mpeg.coef.DeltaH12r[group]; - h_ps_d->specificTo.mpeg.coef.H21r[group] += h_ps_d->specificTo.mpeg.coef.DeltaH21r[group]; - h_ps_d->specificTo.mpeg.coef.H22r[group] += h_ps_d->specificTo.mpeg.coef.DeltaH22r[group]; - - subband = groupBorders20[group]; - - tmpLeft = fMultAddDiv2( fMultDiv2(h_ps_d->specificTo.mpeg.coef.H11r[group], HybrLeftReal[subband]), h_ps_d->specificTo.mpeg.coef.H21r[group], HybrRightReal[subband]); - tmpRight = fMultAddDiv2( fMultDiv2(h_ps_d->specificTo.mpeg.coef.H12r[group], HybrLeftReal[subband]), h_ps_d->specificTo.mpeg.coef.H22r[group], HybrRightReal[subband]); - HybrLeftReal [subband] = tmpLeft<<1; - HybrRightReal[subband] = tmpRight<<1; - - tmpLeft = fMultAdd( fMultDiv2(h_ps_d->specificTo.mpeg.coef.H11r[group], HybrLeftImag[subband]), h_ps_d->specificTo.mpeg.coef.H21r[group], HybrRightImag[subband]); - tmpRight = fMultAdd( fMultDiv2(h_ps_d->specificTo.mpeg.coef.H12r[group], HybrLeftImag[subband]), h_ps_d->specificTo.mpeg.coef.H22r[group], HybrRightImag[subband]); - HybrLeftImag [subband] = tmpLeft; - HybrRightImag[subband] = tmpRight; - } - - /* continue in the qmf buffers */ - HybrLeftReal = QmfLeftReal; - HybrLeftImag = QmfLeftImag; - HybrRightReal = QmfRightReal; - HybrRightImag = QmfRightImag; - - for (; group < NO_IID_GROUPS; group++ ) { - - h_ps_d->specificTo.mpeg.coef.H11r[group] += h_ps_d->specificTo.mpeg.coef.DeltaH11r[group]; - h_ps_d->specificTo.mpeg.coef.H12r[group] += h_ps_d->specificTo.mpeg.coef.DeltaH12r[group]; - h_ps_d->specificTo.mpeg.coef.H21r[group] += h_ps_d->specificTo.mpeg.coef.DeltaH21r[group]; - h_ps_d->specificTo.mpeg.coef.H22r[group] += h_ps_d->specificTo.mpeg.coef.DeltaH22r[group]; - - for ( subband = groupBorders20[group]; subband < groupBorders20[group + 1]; subband++ ) - { - tmpLeft = fMultAdd( fMultDiv2(h_ps_d->specificTo.mpeg.coef.H11r[group], HybrLeftReal[subband]), h_ps_d->specificTo.mpeg.coef.H21r[group], HybrRightReal[subband]); - tmpRight = fMultAdd( fMultDiv2(h_ps_d->specificTo.mpeg.coef.H12r[group], HybrLeftReal[subband]), h_ps_d->specificTo.mpeg.coef.H22r[group], HybrRightReal[subband]); - HybrLeftReal [subband] = tmpLeft; - HybrRightReal[subband] = tmpRight; - - tmpLeft = fMultAdd( fMultDiv2(h_ps_d->specificTo.mpeg.coef.H11r[group], HybrLeftImag[subband]), h_ps_d->specificTo.mpeg.coef.H21r[group], HybrRightImag[subband]); - tmpRight = fMultAdd( fMultDiv2(h_ps_d->specificTo.mpeg.coef.H12r[group], HybrLeftImag[subband]), h_ps_d->specificTo.mpeg.coef.H22r[group], HybrRightImag[subband]); - HybrLeftImag [subband] = tmpLeft; - HybrRightImag[subband] = tmpRight; - - } /* subband */ - } -} - - -/***************************************************************************/ -/*! - \brief Applies IID, ICC, IPD and OPD parameters to the current frame. - - \return none - -****************************************************************************/ -void -ApplyPsSlot( HANDLE_PS_DEC h_ps_d, /*!< handle PS_DEC*/ - FIXP_DBL **rIntBufferLeft, /*!< real bands left qmf channel (38x64) */ - FIXP_DBL **iIntBufferLeft, /*!< imag bands left qmf channel (38x64) */ - FIXP_DBL *rIntBufferRight, /*!< real bands right qmf channel (38x64) */ - FIXP_DBL *iIntBufferRight /*!< imag bands right qmf channel (38x64) */ - ) -{ - - /*! - The 64-band QMF representation of the monaural signal generated by the SBR tool - is used as input of the PS tool. After the PS processing, the outputs of the left - and right hybrid synthesis filterbanks are used to generate the stereo output - signal. - - 
-
-             -------------            ----------            -------------
-            | Hybrid      | M_n[k,m] |          | L_n[k,m] | Hybrid      | l[n]
-   m[n] --->| analysis    |--------->|          |--------->| synthesis   |----->
-            | filter bank |          |          |          | filter bank |
-             -------------           | Stereo   |           -------------
-                   |                 | recon-   |
-                   |                 | stuction |
-                  \|/                |          |
-             -------------           |          |
-            | De-         | D_n[k,m] |          |
-            | correlation |--------->|          |
-             -------------           |          |           -------------
-                                     |          | R_n[k,m] | Hybrid      | r[n]
-                                     |          |--------->| synthesis   |----->
-   IID, ICC ------------------------>|          |          | filter bank |
-  (IPD, OPD)                          ----------            -------------
-
-  m[n]:      QMF represantation of the mono input
-  M_n[k,m]:  (sub-)sub-band domain signals of the mono input
-  D_n[k,m]:  decorrelated (sub-)sub-band domain signals
-  L_n[k,m]:  (sub-)sub-band domain signals of the left output
-  R_n[k,m]:  (sub-)sub-band domain signals of the right output
-  l[n],r[n]: left/right output signals
-
-
- */ - - /* get temporary hybrid qmf values of one timeslot */ - C_ALLOC_SCRATCH_START(hybridRealLeft, FIXP_DBL, NO_SUB_QMF_CHANNELS); - C_ALLOC_SCRATCH_START(hybridImagLeft, FIXP_DBL, NO_SUB_QMF_CHANNELS); - C_ALLOC_SCRATCH_START(hybridRealRight, FIXP_DBL, NO_SUB_QMF_CHANNELS); - C_ALLOC_SCRATCH_START(hybridImagRight, FIXP_DBL, NO_SUB_QMF_CHANNELS); - - SCHAR sf_IntBuffer = h_ps_d->sf_IntBuffer; - - /* clear workbuffer */ - FDKmemclear(hybridRealLeft, NO_SUB_QMF_CHANNELS*sizeof(FIXP_DBL)); - FDKmemclear(hybridImagLeft, NO_SUB_QMF_CHANNELS*sizeof(FIXP_DBL)); - FDKmemclear(hybridRealRight, NO_SUB_QMF_CHANNELS*sizeof(FIXP_DBL)); - FDKmemclear(hybridImagRight, NO_SUB_QMF_CHANNELS*sizeof(FIXP_DBL)); - - - /*! - Hybrid analysis filterbank: - The lower 3 (5) of the 64 QMF subbands are further split to provide better frequency resolution. - for PS processing. - For the 10 and 20 stereo bands configuration, the QMF band H_0(w) is split - up into 8 (sub-) sub-bands and the QMF bands H_1(w) and H_2(w) are spit into 2 (sub-) - 4th. (See figures 8.20 and 8.22 of ISO/IEC 14496-3:2001/FDAM 2:2004(E) ) - */ - - - if (h_ps_d->procFrameBased == 1) /* If we have switched from frame to slot based processing */ - { /* fill hybrid delay buffer. */ - h_ps_d->procFrameBased = 0; - - fillHybridDelayLine( rIntBufferLeft, - iIntBufferLeft, - hybridRealLeft, - hybridImagLeft, - hybridRealRight, - hybridImagRight, - &h_ps_d->specificTo.mpeg.hybrid ); - } - - slotBasedHybridAnalysis ( rIntBufferLeft[HYBRID_FILTER_DELAY], /* qmf filterbank values */ - iIntBufferLeft[HYBRID_FILTER_DELAY], /* qmf filterbank values */ - hybridRealLeft, /* hybrid filterbank values */ - hybridImagLeft, /* hybrid filterbank values */ - &h_ps_d->specificTo.mpeg.hybrid); /* hybrid filterbank handle */ - - - SCHAR hybridScal = h_ps_d->specificTo.mpeg.hybrid.sf_mQmfBuffer; - - - /*! - Decorrelation: - By means of all-pass filtering and delaying, the (sub-)sub-band samples s_k(n) are - converted into de-correlated (sub-)sub-band samples d_k(n). - - k: frequency in hybrid spectrum - - n: time index - */ - - deCorrelateSlotBased( h_ps_d, /* parametric stereo decoder handle */ - hybridRealLeft, /* left hybrid time slot */ - hybridImagLeft, - hybridScal, /* scale factor of left hybrid time slot */ - rIntBufferLeft[0], /* left qmf time slot */ - iIntBufferLeft[0], - sf_IntBuffer, /* scale factor of left and right qmf time slot */ - hybridRealRight, /* right hybrid time slot */ - hybridImagRight, - rIntBufferRight, /* right qmf time slot */ - iIntBufferRight ); - - - - /*! - Stereo Processing: - The sets of (sub-)sub-band samples s_k(n) and d_k(n) are processed according to - the stereo cues which are defined per stereo band. - */ - - - applySlotBasedRotation( h_ps_d, /* parametric stereo decoder handle */ - hybridRealLeft, /* left hybrid time slot */ - hybridImagLeft, - rIntBufferLeft[0], /* left qmf time slot */ - iIntBufferLeft[0], - hybridRealRight, /* right hybrid time slot */ - hybridImagRight, - rIntBufferRight, /* right qmf time slot */ - iIntBufferRight ); - - - - - /*! - Hybrid synthesis filterbank: - The stereo processed hybrid subband signals l_k(n) and r_k(n) are fed into the hybrid synthesis - filterbanks which are identical to the 64 complex synthesis filterbank of the SBR tool. The - input to the filterbank are slots of 64 QMF samples. For each slot the filterbank outputs one - block of 64 samples of one reconstructed stereo channel. The hybrid synthesis filterbank is - computed seperatly for the left and right channel. - */ - - - /* left channel */ - slotBasedHybridSynthesis ( hybridRealLeft, /* one timeslot of hybrid filterbank values */ - hybridImagLeft, - rIntBufferLeft[0], /* one timeslot of qmf filterbank values */ - iIntBufferLeft[0], - &h_ps_d->specificTo.mpeg.hybrid ); /* hybrid filterbank handle */ - - /* right channel */ - slotBasedHybridSynthesis ( hybridRealRight, /* one timeslot of hybrid filterbank values */ - hybridImagRight, - rIntBufferRight, /* one timeslot of qmf filterbank values */ - iIntBufferRight, - &h_ps_d->specificTo.mpeg.hybrid ); /* hybrid filterbank handle */ - - - - - - - - /* free temporary hybrid qmf values of one timeslot */ - C_ALLOC_SCRATCH_END(hybridImagRight, FIXP_DBL, NO_SUB_QMF_CHANNELS); - C_ALLOC_SCRATCH_END(hybridRealRight, FIXP_DBL, NO_SUB_QMF_CHANNELS); - C_ALLOC_SCRATCH_END(hybridImagLeft, FIXP_DBL, NO_SUB_QMF_CHANNELS); - C_ALLOC_SCRATCH_END(hybridRealLeft, FIXP_DBL, NO_SUB_QMF_CHANNELS); - -}/* END ApplyPsSlot */ - - -/***************************************************************************/ -/*! - - \brief assigns timeslots to an array - - \return - -****************************************************************************/ - -static void assignTimeSlotsPS (FIXP_DBL *bufAdr, - FIXP_DBL **bufPtr, - const int numSlots, - const int numChan) -{ - FIXP_DBL *ptr; - int slot; - ptr = bufAdr; - for(slot=0; slot < numSlots; slot++) { - bufPtr [slot] = ptr; - ptr += numChan; - } -} - diff --git a/libSBRdec/src/psdec.h b/libSBRdec/src/psdec.h deleted file mode 100644 index 3dbc76d..0000000 --- a/libSBRdec/src/psdec.h +++ /dev/null @@ -1,352 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief Sbr decoder -*/ -#ifndef __PSDEC_H -#define __PSDEC_H - -#include "sbrdecoder.h" - - - -/* This PS decoder implements the baseline version. So it always uses the */ -/* hybrid filter structure for 20 stereo bands and does not implemet IPD/OPD */ -/* synthesis. The baseline version has to support the complete PS bitstream */ -/* syntax. But IPD/OPD data is ignored and set to 0. If 34 stereo band config */ -/* is used in the bitstream for IIS/ICC the decoded parameters are mapped to */ -/* 20 stereo bands. */ - - -#include "FDK_bitstream.h" - -#include "psdec_hybrid.h" - -#define SCAL_HEADROOM ( 2 ) - -#define PS_EXTENSION_SIZE_BITS ( 4 ) -#define PS_EXTENSION_ESC_COUNT_BITS ( 8 ) - -#define NO_QMF_CHANNELS ( 64 ) -#define MAX_NUM_COL ( 32 ) - - - #define NO_QMF_BANDS_HYBRID20 ( 3 ) - #define NO_SUB_QMF_CHANNELS ( 12 ) - - #define NRG_INT_COEFF ( 0.75f ) - #define INT_FILTER_COEFF (FL2FXCONST_DBL( 1.0f - NRG_INT_COEFF )) - #define PEAK_DECAY_FACTOR (FL2FXCONST_DBL( 0.765928338364649f )) - #define TRANSIENT_IMPACT_FACTOR (FL2FXCONST_DBL( 2.0 / 3.0 )) - - #define NO_SERIAL_ALLPASS_LINKS ( 3 ) - #define MAX_NO_PS_ENV ( 4 + 1 ) /* +1 needed for VAR_BORDER */ - - #define MAX_DELAY_BUFFER_SIZE ( 14 ) - #define NO_DELAY_BUFFER_BANDS ( 35 ) - - #define NO_HI_RES_BINS ( 34 ) - #define NO_MID_RES_BINS ( 20 ) - #define NO_LOW_RES_BINS ( 10 ) - - #define FIRST_DELAY_SB ( 23 ) - #define NO_SAMPLE_DELAY_ALLPASS ( 2 ) - #define NO_DELAY_LENGTH_VECTORS ( 12 ) /* d(m): d(0)=3 + d(1)=4 + d(2)=5 */ - - #define NO_HI_RES_IID_BINS ( NO_HI_RES_BINS ) - #define NO_HI_RES_ICC_BINS ( NO_HI_RES_BINS ) - - #define NO_MID_RES_IID_BINS ( NO_MID_RES_BINS ) - #define NO_MID_RES_ICC_BINS ( NO_MID_RES_BINS ) - - #define NO_LOW_RES_IID_BINS ( NO_LOW_RES_BINS ) - #define NO_LOW_RES_ICC_BINS ( NO_LOW_RES_BINS ) - - #define SUBQMF_GROUPS ( 10 ) - #define QMF_GROUPS ( 12 ) - - #define SUBQMF_GROUPS_HI_RES ( 32 ) - #define QMF_GROUPS_HI_RES ( 18 ) - - #define NO_IID_GROUPS ( SUBQMF_GROUPS + QMF_GROUPS ) - #define NO_IID_GROUPS_HI_RES ( SUBQMF_GROUPS_HI_RES + QMF_GROUPS_HI_RES ) - - #define NO_IID_STEPS ( 7 ) /* 1 .. + 7 */ - #define NO_IID_STEPS_FINE ( 15 ) /* 1 .. +15 */ - #define NO_ICC_STEPS ( 8 ) /* 0 .. + 7 */ - - #define NO_IID_LEVELS ( 2 * NO_IID_STEPS + 1 ) /* - 7 .. + 7 */ - #define NO_IID_LEVELS_FINE ( 2 * NO_IID_STEPS_FINE + 1 ) /* -15 .. +15 */ - #define NO_ICC_LEVELS ( NO_ICC_STEPS ) /* 0 .. + 7 */ - - #define FIXP_SQRT05 ((FIXP_DBL)0x5a827980) /* 1/SQRT2 */ - - struct PS_DEC_COEFFICIENTS { - - FIXP_DBL H11r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */ - FIXP_DBL H12r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */ - FIXP_DBL H21r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */ - FIXP_DBL H22r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */ - - FIXP_DBL DeltaH11r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */ - FIXP_DBL DeltaH12r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */ - FIXP_DBL DeltaH21r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */ - FIXP_DBL DeltaH22r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */ - - SCHAR aaIidIndexMapped[MAX_NO_PS_ENV][NO_HI_RES_IID_BINS]; /*!< The mapped IID index for all envelopes and all IID bins */ - SCHAR aaIccIndexMapped[MAX_NO_PS_ENV][NO_HI_RES_ICC_BINS]; /*!< The mapped ICC index for all envelopes and all ICC bins */ - - }; - - - - -typedef enum { - ppt_none = 0, - ppt_mpeg = 1, - ppt_drm = 2 -} PS_PAYLOAD_TYPE; - - -typedef struct { - UCHAR bPsHeaderValid; /*!< set if new header is available from bitstream */ - - UCHAR bEnableIid; /*!< One bit denoting the presence of IID parameters */ - UCHAR bEnableIcc; /*!< One bit denoting the presence of ICC parameters */ - UCHAR bEnableExt; /*!< The PS extension layer is enabled using the enable_ext bit. - If it is set to %1 the IPD and OPD parameters are sent. - If it is disabled, i.e. %0, the extension layer is skipped. */ - - UCHAR modeIid; /*!< The configuration of IID parameters (number of bands and - quantisation grid, iid_quant) is determined by iid_mode. */ - UCHAR modeIcc; /*!< The configuration of Inter-channel Coherence parameters - (number of bands and quantisation grid) is determined by - icc_mode. */ - - UCHAR freqResIid; /*!< 0=low, 1=mid or 2=high frequency resolution for iid */ - UCHAR freqResIcc; /*!< 0=low, 1=mid or 2=high frequency resolution for icc */ - - UCHAR bFineIidQ; /*!< Use fine Iid quantisation. */ - - UCHAR bFrameClass; /*!< The frame_class bit determines whether the parameter - positions of the current frame are uniformly spaced - accross the frame or they are defined using the positions - described by border_position. */ - - UCHAR noEnv; /*!< The number of envelopes per frame */ - UCHAR aEnvStartStop[MAX_NO_PS_ENV+1]; /*!< In case of variable parameter spacing the parameter - positions are determined by border_position */ - - SCHAR abIidDtFlag[MAX_NO_PS_ENV]; /*!< Deltacoding time/freq flag for IID, 0 => freq */ - SCHAR abIccDtFlag[MAX_NO_PS_ENV]; /*!< Deltacoding time/freq flag for ICC, 0 => freq */ - - SCHAR aaIidIndex[MAX_NO_PS_ENV][NO_HI_RES_IID_BINS]; /*!< The IID index for all envelopes and all IID bins */ - SCHAR aaIccIndex[MAX_NO_PS_ENV][NO_HI_RES_ICC_BINS]; /*!< The ICC index for all envelopes and all ICC bins */ - -} MPEG_PS_BS_DATA; - - - -struct PS_DEC { - - SCHAR noSubSamples; - SCHAR noChannels; - - SCHAR procFrameBased; /*!< Helper to detected switching from frame based to slot based - processing */ - - PS_PAYLOAD_TYPE bPsDataAvail[(1)+1]; /*!< set if new data available from bitstream */ - UCHAR psDecodedPrv; /*!< set if PS has been processed in the last frame */ - - /* helpers for frame delay line */ - UCHAR bsLastSlot; /*!< Index of last read slot. */ - UCHAR bsReadSlot; /*!< Index of current read slot for additional delay. */ - UCHAR processSlot; /*!< Index of current slot for processing (need for add. delay). */ - - - INT rescal; - INT sf_IntBuffer; - - union { /* Bitstream data */ - MPEG_PS_BS_DATA mpeg; /*!< Struct containing all MPEG specific PS data from bitstream. */ - } bsData[(1)+1]; - - shouldBeUnion { /* Static data */ - struct { - SCHAR aIidPrevFrameIndex[NO_HI_RES_IID_BINS]; /*!< The IID index for previous frame */ - SCHAR aIccPrevFrameIndex[NO_HI_RES_ICC_BINS]; /*!< The ICC index for previous frame */ - - UCHAR delayBufIndex; /*!< Pointer to where the latest sample is in buffer */ - UCHAR noSampleDelay; /*!< How many QMF samples delay is used. */ - UCHAR lastUsb; /*!< uppermost WMF delay band of last frame */ - - UCHAR aDelayRBufIndexSer[NO_SERIAL_ALLPASS_LINKS]; /*!< Delay buffer for reverb filter */ - UCHAR aDelayBufIndexDelayQmf[NO_QMF_CHANNELS-FIRST_DELAY_SB]; /*!< Delay buffer for ICC group 20 & 21 */ - - SCHAR scaleFactorPsDelayBuffer; /*!< Scale factor for ps delay buffer */ - - /* hybrid filter bank delay lines */ - FIXP_DBL aaQmfDelayBufReal[(NO_QMF_CHANNELS-FIRST_DELAY_SB) + (MAX_DELAY_BUFFER_SIZE-1)*(NO_DELAY_BUFFER_BANDS-FIRST_DELAY_SB)]; - FIXP_DBL aaQmfDelayBufImag[(NO_QMF_CHANNELS-FIRST_DELAY_SB) + (MAX_DELAY_BUFFER_SIZE-1)*(NO_DELAY_BUFFER_BANDS-FIRST_DELAY_SB)]; - - FIXP_DBL *pAaRealDelayBufferQmf[MAX_DELAY_BUFFER_SIZE]; /*!< Real part delay buffer */ - FIXP_DBL *pAaImagDelayBufferQmf[MAX_DELAY_BUFFER_SIZE]; /*!< Imaginary part delay buffer */ - - FIXP_DBL aaRealDelayBufferQmf[NO_SAMPLE_DELAY_ALLPASS][FIRST_DELAY_SB]; /*!< Real part delay buffer */ - FIXP_DBL aaImagDelayBufferQmf[NO_SAMPLE_DELAY_ALLPASS][FIRST_DELAY_SB]; /*!< Imaginary part delay buffer*/ - - FIXP_DBL aaRealDelayBufferSubQmf[NO_SAMPLE_DELAY_ALLPASS][NO_SUB_QMF_CHANNELS]; /*!< Real part delay buffer */ - FIXP_DBL aaImagDelayBufferSubQmf[NO_SAMPLE_DELAY_ALLPASS][NO_SUB_QMF_CHANNELS]; /*!< Imaginary part delay buffer */ - - FIXP_DBL aaaRealDelayRBufferSerQmf[FIRST_DELAY_SB][NO_DELAY_LENGTH_VECTORS]; /*!< Real part delay buffer */ - FIXP_DBL aaaImagDelayRBufferSerQmf[FIRST_DELAY_SB][NO_DELAY_LENGTH_VECTORS]; /*!< Imaginary part delay buffer */ - - FIXP_DBL aaaRealDelayRBufferSerSubQmf[NO_SUB_QMF_CHANNELS][NO_DELAY_LENGTH_VECTORS]; /*!< Real part delay buffer */ - FIXP_DBL aaaImagDelayRBufferSerSubQmf[NO_SUB_QMF_CHANNELS][NO_DELAY_LENGTH_VECTORS]; /*!< Imaginary part delay buffer */ - - HYBRID hybrid; /*!< hybrid filter bank struct 1 or 2. */ - - FIXP_DBL aPrevNrgBin[NO_MID_RES_BINS]; /*!< energy of previous frame */ - FIXP_DBL aPrevPeakDiffBin[NO_MID_RES_BINS]; /*!< peak difference of previous frame */ - FIXP_DBL aPeakDecayFastBin[NO_MID_RES_BINS]; /*!< Saved max. peak decay value per bin */ - SCHAR aPowerPrevScal[NO_MID_RES_BINS]; /*!< Last power value (each bin) of previous frame */ - - FIXP_DBL h11rPrev[NO_IID_GROUPS]; /*!< previous calculated h(xy) coefficients */ - FIXP_DBL h12rPrev[NO_IID_GROUPS]; /*!< previous calculated h(xy) coefficients */ - FIXP_DBL h21rPrev[NO_IID_GROUPS]; /*!< previous calculated h(xy) coefficients */ - FIXP_DBL h22rPrev[NO_IID_GROUPS]; /*!< previous calculated h(xy) coefficients */ - - PS_DEC_COEFFICIENTS coef; /*!< temporal coefficients (reusable scratch memory) */ - - } mpeg; - - } specificTo; - - -}; - -typedef struct PS_DEC *HANDLE_PS_DEC; - - -int CreatePsDec(HANDLE_PS_DEC *h_PS_DEC, int aacSamplesPerFrame); - -int DeletePsDec(HANDLE_PS_DEC *h_PS_DEC); - -void -scalFilterBankValues( HANDLE_PS_DEC h_ps_d, /* parametric stereo decoder handle */ - FIXP_DBL **fixpQmfReal, /* qmf filterbank values */ - FIXP_DBL **fixpQmfImag, /* qmf filterbank values */ - int lsb, /* sbr start subband */ - int scaleFactorLowBandSplitLow, - int scaleFactorLowBandSplitHigh, - SCHAR *scaleFactorLowBand_lb, - SCHAR *scaleFactorLowBand_hb, - int scaleFactorHighBands, - INT *scaleFactorHighBand, - INT noCols); - -void -rescalFilterBankValues( HANDLE_PS_DEC h_ps_d, /* parametric stereo decoder handle */ - FIXP_DBL **QmfBufferReal, /* qmf filterbank values */ - FIXP_DBL **QmfBufferImag, /* qmf filterbank values */ - int lsb, /* sbr start subband */ - INT noCols); - - -void -initSlotBasedRotation( HANDLE_PS_DEC h_ps_d, - int env, - int usb); - -void -ApplyPsSlot( HANDLE_PS_DEC h_ps_d, /* parametric stereo decoder handle */ - FIXP_DBL **rIntBufferLeft, /* real values of left qmf timeslot */ - FIXP_DBL **iIntBufferLeft, /* imag values of left qmf timeslot */ - FIXP_DBL *rIntBufferRight, /* real values of right qmf timeslot */ - FIXP_DBL *iIntBufferRight); /* imag values of right qmf timeslot */ - - - -#endif /* __PSDEC_H */ diff --git a/libSBRdec/src/psdec_hybrid.cpp b/libSBRdec/src/psdec_hybrid.cpp deleted file mode 100644 index cbd0e92..0000000 --- a/libSBRdec/src/psdec_hybrid.cpp +++ /dev/null @@ -1,652 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -#include "psdec_hybrid.h" - - -#include "fft.h" -#include "sbr_ram.h" - -#include "FDK_tools_rom.h" -#include "sbr_rom.h" - -/******************************************************************************* - Functionname: InitHybridFilterBank - ******************************************************************************* - - Description: Init one instance of HANDLE_HYBRID stuct - - Arguments: - - Return: none - -*******************************************************************************/ - - -SBR_ERROR -InitHybridFilterBank ( HANDLE_HYBRID hs, /*!< Handle to HYBRID struct. */ - SCHAR frameSize, /*!< Framesize (in Qmf súbband samples). */ - SCHAR noBands, /*!< Number of Qmf bands for hybrid filtering. */ - const UCHAR *pResolution ) /*!< Resolution in Qmf bands (length noBands). */ -{ - SCHAR i; - UCHAR maxNoChannels = 0; - - for (i = 0; i < noBands; i++) { - hs->pResolution[i] = pResolution[i]; - if(pResolution[i] > maxNoChannels) - maxNoChannels = pResolution[i]; - } - - hs->nQmfBands = noBands; - hs->frameSize = frameSize; - hs->qmfBufferMove = HYBRID_FILTER_LENGTH - 1; - - hs->sf_mQmfBuffer = 0; - - return SBRDEC_OK; -} - -/******************************************************************************* - Functionname: dualChannelFiltering - ******************************************************************************* - - Description: fast 2-channel real-valued filtering with 6-tap delay. - - Arguments: - - Return: none - -*******************************************************************************/ - -/*! -2 channel filter -
-   Filter Coefs:
-   0.0,
-   0.01899487526049,
-   0.0,
-   -0.07293139167538,
-   0.0,
-   0.30596630545168,
-   0.5,
-   0.30596630545168,
-   0.0,
-   -0.07293139167538,
-   0.0,
-   0.01899487526049,
-   0.0
-
-
-   Filter design:
-   h[q,n] = g[n] * cos(2pi/2 * q * (n-6) );  n = 0..12,  q = 0,1;
-
-   ->  h[0,n] = g[n] * 1;
-   ->  h[1,n] = g[n] * pow(-1,n);
-
-*/ - -static void slotBasedDualChannelFiltering( const FIXP_DBL *pQmfReal, - const FIXP_DBL *pQmfImag, - - FIXP_DBL *mHybridReal, - FIXP_DBL *mHybridImag) -{ - - FIXP_DBL t1, t3, t5, t6; - - /* symmetric filter coefficients */ - - /* you don't have to shift the result after fMult because of p2_13_20 <= 0.5 */ - t1 = fMultDiv2(p2_13_20[1] , ( (pQmfReal[1] >> 1) + (pQmfReal[11] >> 1))); - t3 = fMultDiv2(p2_13_20[3] , ( (pQmfReal[3] >> 1) + (pQmfReal[ 9] >> 1))); - t5 = fMultDiv2(p2_13_20[5] , ( (pQmfReal[5] >> 1) + (pQmfReal[ 7] >> 1))); - t6 = fMultDiv2(p2_13_20[6] , (pQmfReal[6] >> 1) ); - - mHybridReal[0] = (t1 + t3 + t5 + t6) << 2; - mHybridReal[1] = (- t1 - t3 - t5 + t6) << 2; - - t1 = fMultDiv2(p2_13_20[1] , ( (pQmfImag[1] >> 1) + (pQmfImag[11] >> 1))); - t3 = fMultDiv2(p2_13_20[3] , ( (pQmfImag[3] >> 1) + (pQmfImag[ 9] >> 1))); - t5 = fMultDiv2(p2_13_20[5] , ( (pQmfImag[5] >> 1) + (pQmfImag[ 7] >> 1))); - t6 = fMultDiv2(p2_13_20[6] , pQmfImag[6] >> 1 ); - - mHybridImag[0] = (t1 + t3 + t5 + t6) << 2; - mHybridImag[1] = (- t1 - t3 - t5 + t6) << 2; -} - - -/******************************************************************************* - Functionname: eightChannelFiltering - ******************************************************************************* - - Description: fast 8-channel complex-valued filtering with 6-tap delay. - - Arguments: - - Return: none - -*******************************************************************************/ -/*! - 8 channel filter - - Implementation using a FFT of length 8 -
-   prototype filter coefficients:
-   0.00746082949812   0.02270420949825   0.04546865930473   0.07266113929591   0.09885108575264   0.11793710567217
-   0.125
-   0.11793710567217   0.09885108575264   0.07266113929591   0.04546865930473   0.02270420949825   0.00746082949812
-
-   Filter design:
-   N = 13; Q = 8;
-   h[q,n]       = g[n] * exp(j * 2 * pi / Q * (q + .5) * (n - 6));  n = 0..(N-1),  q = 0..(Q-1);
-
-   Time Signal:   x[t];
-   Filter Bank Output
-   y[q,t] = conv(x[t],h[q,t]) = conv(h[q,t],x[t]) = sum(x[k] * h[q, t - k] ) = sum(h[q, k] * x[t - k] ); k = 0..(N-1);
-
-   y[q,t] =   x[t - 12]*h[q, 12]  +  x[t - 11]*h[q, 11]  +  x[t - 10]*h[q, 10]  +  x[t -  9]*h[q,  9]
-           +  x[t -  8]*h[q,  8]  +  x[t -  7]*h[q,  7]
-           +  x[t -  6]*h[q,  6]
-           +  x[t -  5]*h[q,  5]  +  x[t -  4]*h[q,  4]
-           +  x[t -  3]*h[q,  3]  +  x[t -  2]*h[q,  2]  +  x[t -  1]*h[q,  1]  +  x[t -  0]*h[q,  0];
-
-   h'[q, n] = h[q,(N-1)-n] = g[n] * exp(j * 2 * pi / Q * (q + .5) * (6 - n));  n = 0..(N-1),  q = 0..(Q-1);
-
-   y[q,t] =   x[t - 12]*h'[q,  0]  +  x[t - 11]*h'[q,  1]  +  x[t - 10]*h'[q,  2]  +  x[t -  9]*h'[q,  3]
-           +  x[t -  8]*h'[q,  4]  +  x[t -  7]*h'[q,  5]
-           +  x[t -  6]*h'[q,  6]
-           +  x[t -  5]*h'[q,  7]  +  x[t -  4]*h'[q,  8]
-           +  x[t -  3]*h'[q,  9]  +  x[t -  2]*h'[q, 10]  +  x[t -  1]*h'[q, 11]  +  x[t -  0]*h'[q, 12];
-
-   Try to split off FFT Modulation Term:
-   FFT(x[t], q) = sum(x[t+k]*exp(-j*2*pi/N *q * k))
-                                           c                                           m
-   Step 1:  h'[q,n] = g[n] * ( exp(j * 2 * pi / 8 * .5 * (6 - n)) ) * ( exp (j * 2 * pi / 8 * q * (6 - n)) );
-
-    h'[q,n] = g[n] *c[n] * m[q,n]; (see above)
-    c[n]    = exp( j * 2 * pi / 8 * .5 * (6 - n) );
-    m[q,n]  = exp( j * 2 * pi / 8 *  q * (6 - n) );
-
-    y[q,t] = x[t -  0]*g[0]*c[0]*m[q,0]  +  x[t -  1]*g[1]*c[ 1]*m[q, 1]  + ...
-             ...                         +  x[t - 12]*g[2]*c[12]*m[q,12];
-
-                                                                              |
-    n                   m                            *exp(-j*2*pi)            |   n'                   fft
--------------------------------------------------------------------------------------------------------------------------
-    0       exp( j * 2 * pi / 8 * q * 6) ->  exp(-j * 2 * pi / 8 * q * 2)     |   2         exp(-j * 2 * pi / 8 * q * 0)
-    1       exp( j * 2 * pi / 8 * q * 5) ->  exp(-j * 2 * pi / 8 * q * 3)     |   3         exp(-j * 2 * pi / 8 * q * 1)
-    2       exp( j * 2 * pi / 8 * q * 4) ->  exp(-j * 2 * pi / 8 * q * 4)     |   4         exp(-j * 2 * pi / 8 * q * 2)
-    3       exp( j * 2 * pi / 8 * q * 3) ->  exp(-j * 2 * pi / 8 * q * 5)     |   5         exp(-j * 2 * pi / 8 * q * 3)
-    4       exp( j * 2 * pi / 8 * q * 2) ->  exp(-j * 2 * pi / 8 * q * 6)     |   6         exp(-j * 2 * pi / 8 * q * 4)
-    5       exp( j * 2 * pi / 8 * q * 1) ->  exp(-j * 2 * pi / 8 * q * 7)     |   7         exp(-j * 2 * pi / 8 * q * 5)
-    6       exp( j * 2 * pi / 8 * q * 0)                                      |   0         exp(-j * 2 * pi / 8 * q * 6)
-    7       exp(-j * 2 * pi / 8 * q * 1)                                      |   1         exp(-j * 2 * pi / 8 * q * 7)
-    8       exp(-j * 2 * pi / 8 * q * 2)                                      |   2
-    9       exp(-j * 2 * pi / 8 * q * 3)                                      |   3
-    10      exp(-j * 2 * pi / 8 * q * 4)                                      |   4
-    11      exp(-j * 2 * pi / 8 * q * 5)                                      |   5
-    12      exp(-j * 2 * pi / 8 * q * 6)                                      |   6
-
-
-    now use fft modulation coefficients
-    m[6]  =       = fft[0]
-    m[7]  =       = fft[1]
-    m[8]  = m[ 0] = fft[2]
-    m[9]  = m[ 1] = fft[3]
-    m[10] = m[ 2] = fft[4]
-    m[11] = m[ 3] = fft[5]
-    m[12] = m[ 4] = fft[6]
-            m[ 5] = fft[7]
-
-    y[q,t] = (                       x[t- 6]*g[ 6]*c[ 6] ) * fft[q,0]  +
-             (                       x[t- 7]*g[ 7]*c[ 7] ) * fft[q,1]  +
-             ( x[t- 0]*g[ 0]*c[ 0] + x[t- 8]*g[ 8]*c[ 8] ) * fft[q,2]  +
-             ( x[t- 1]*g[ 1]*c[ 1] + x[t- 9]*g[ 9]*c[ 9] ) * fft[q,3]  +
-             ( x[t- 2]*g[ 2]*c[ 2] + x[t-10]*g[10]*c[10] ) * fft[q,4]  +
-             ( x[t- 3]*g[ 3]*c[ 3] + x[t-11]*g[11]*c[11] ) * fft[q,5]  +
-             ( x[t- 4]*g[ 4]*c[ 4] + x[t-12]*g[12]*c[12] ) * fft[q,6]  +
-             ( x[t- 5]*g[ 5]*c[ 5]                       ) * fft[q,7];
-
-    pre twiddle factors c[n] = exp(j * 2 * pi / 8 * .5 * (6 - n));
-    n                c]           |  n                c[n]         |  n                c[n]
----------------------------------------------------------------------------------------------------
-    0       exp( j * 6 * pi / 8)  |  1       exp( j * 5 * pi / 8)  |  2       exp( j * 4 * pi / 8)
-    3       exp( j * 3 * pi / 8)  |  4       exp( j * 2 * pi / 8)  |  5       exp( j * 1 * pi / 8)
-    6       exp( j * 0 * pi / 8)  |  7       exp(-j * 1 * pi / 8)  |  8       exp(-j * 2 * pi / 8)
-    9       exp(-j * 3 * pi / 8)  | 10       exp(-j * 4 * pi / 8)  | 11       exp(-j * 5 * pi / 8)
-   12       exp(-j * 6 * pi / 8)  |                                |
-
-*/ - -/* defining rotation factors for *ChannelFiltering */ - -#define cos0Pi FL2FXCONST_DBL( 1.f) -#define sin0Pi FL2FXCONST_DBL( 0.f) - -#define cos1Pi FL2FXCONST_DBL(-1.f) -#define sin1Pi FL2FXCONST_DBL( 0.f) - -#define cos1Pi_2 FL2FXCONST_DBL( 0.f) -#define sin1Pi_2 FL2FXCONST_DBL( 1.f) - -#define cos1Pi_3 FL2FXCONST_DBL( 0.5f) -#define sin1Pi_3 FL2FXCONST_DBL( 0.86602540378444f) - -#define cos0Pi_4 cos0Pi -#define cos1Pi_4 FL2FXCONST_DBL(0.70710678118655f) -#define cos2Pi_4 cos1Pi_2 -#define cos3Pi_4 (-cos1Pi_4) -#define cos4Pi_4 (-cos0Pi_4) -#define cos5Pi_4 cos3Pi_4 -#define cos6Pi_4 cos2Pi_4 - -#define sin0Pi_4 sin0Pi -#define sin1Pi_4 FL2FXCONST_DBL(0.70710678118655f) -#define sin2Pi_4 sin1Pi_2 -#define sin3Pi_4 sin1Pi_4 -#define sin4Pi_4 sin0Pi_4 -#define sin5Pi_4 (-sin3Pi_4) -#define sin6Pi_4 (-sin2Pi_4) - -#define cos0Pi_8 cos0Pi -#define cos1Pi_8 FL2FXCONST_DBL(0.92387953251129f) -#define cos2Pi_8 cos1Pi_4 -#define cos3Pi_8 FL2FXCONST_DBL(0.38268343236509f) -#define cos4Pi_8 cos2Pi_4 -#define cos5Pi_8 (-cos3Pi_8) -#define cos6Pi_8 (-cos2Pi_8) - -#define sin0Pi_8 sin0Pi -#define sin1Pi_8 cos3Pi_8 -#define sin2Pi_8 sin1Pi_4 -#define sin3Pi_8 cos1Pi_8 -#define sin4Pi_8 sin2Pi_4 -#define sin5Pi_8 sin3Pi_8 -#define sin6Pi_8 sin1Pi_4 - -#if defined(ARCH_PREFER_MULT_32x16) - #define FIXP_HYB FIXP_SGL - #define FIXP_CAST FX_DBL2FX_SGL -#else - #define FIXP_HYB FIXP_DBL - #define FIXP_CAST -#endif - -static const FIXP_HYB cr[13] = -{ - FIXP_CAST(cos6Pi_8), FIXP_CAST(cos5Pi_8), FIXP_CAST(cos4Pi_8), - FIXP_CAST(cos3Pi_8), FIXP_CAST(cos2Pi_8), FIXP_CAST(cos1Pi_8), - FIXP_CAST(cos0Pi_8), - FIXP_CAST(cos1Pi_8), FIXP_CAST(cos2Pi_8), FIXP_CAST(cos3Pi_8), - FIXP_CAST(cos4Pi_8), FIXP_CAST(cos5Pi_8), FIXP_CAST(cos6Pi_8) -}; - -static const FIXP_HYB ci[13] = -{ - FIXP_CAST( sin6Pi_8), FIXP_CAST( sin5Pi_8), FIXP_CAST( sin4Pi_8), - FIXP_CAST( sin3Pi_8), FIXP_CAST( sin2Pi_8), FIXP_CAST( sin1Pi_8), - FIXP_CAST( sin0Pi_8) , - FIXP_CAST(-sin1Pi_8), FIXP_CAST(-sin2Pi_8), FIXP_CAST(-sin3Pi_8), - FIXP_CAST(-sin4Pi_8), FIXP_CAST(-sin5Pi_8), FIXP_CAST(-sin6Pi_8) -}; - -static void slotBasedEightChannelFiltering( const FIXP_DBL *pQmfReal, - const FIXP_DBL *pQmfImag, - - FIXP_DBL *mHybridReal, - FIXP_DBL *mHybridImag) -{ - - int bin; - FIXP_DBL _fft[128 + ALIGNMENT_DEFAULT - 1]; - FIXP_DBL *fft = (FIXP_DBL *)ALIGN_PTR(_fft); - -#if defined(ARCH_PREFER_MULT_32x16) - const FIXP_SGL *p = p8_13_20; /* BASELINE_PS */ -#else - const FIXP_DBL *p = p8_13_20; /* BASELINE_PS */ -#endif - - /* pre twiddeling */ - - /* x*(a*b + c*d) = fMultDiv2(x, fMultAddDiv2(fMultDiv2(a, b), c, d)) */ - /* x*(a*b - c*d) = fMultDiv2(x, fMultSubDiv2(fMultDiv2(a, b), c, d)) */ - FIXP_DBL accu1, accu2, accu3, accu4; - - #define TWIDDLE_1(n_0,n_1,n_2) \ - cplxMultDiv2(&accu1, &accu2, pQmfReal[n_0], pQmfImag[n_0], cr[n_0], ci[n_0]); \ - accu1 = fMultDiv2(p[n_0], accu1); \ - accu2 = fMultDiv2(p[n_0], accu2); \ - cplxMultDiv2(&accu3, &accu4, pQmfReal[n_1], pQmfImag[n_1], cr[n_1], ci[n_1]); \ - accu3 = fMultDiv2(p[n_1], accu3); \ - accu4 = fMultDiv2(p[n_1], accu4); \ - fft[FIXP_FFT_IDX_R(n_2)] = accu1 + accu3; \ - fft[FIXP_FFT_IDX_I(n_2)] = accu2 + accu4; - - #define TWIDDLE_0(n_0,n_1) \ - cplxMultDiv2(&accu1, &accu2, pQmfReal[n_0], pQmfImag[n_0], cr[n_0], ci[n_0]); \ - fft[FIXP_FFT_IDX_R(n_1)] = fMultDiv2(p[n_0], accu1); \ - fft[FIXP_FFT_IDX_I(n_1)] = fMultDiv2(p[n_0], accu2); - - TWIDDLE_0( 6, 0) - TWIDDLE_0( 7, 1) - - TWIDDLE_1( 0, 8, 2) - TWIDDLE_1( 1, 9, 3) - TWIDDLE_1( 2,10, 4) - TWIDDLE_1( 3,11, 5) - TWIDDLE_1( 4,12, 6) - - TWIDDLE_0( 5, 7) - - fft_8 (fft); - - /* resort fft data into output array*/ - for(bin=0; bin<8;bin++ ) { - mHybridReal[bin] = fft[FIXP_FFT_IDX_R(bin)] << 4; - mHybridImag[bin] = fft[FIXP_FFT_IDX_I(bin)] << 4; - } -} - - -/******************************************************************************* - Functionname: fillHybridDelayLine - ******************************************************************************* - - Description: The delay line of the hybrid filter is filled and copied from - left to right. - - Return: none - -*******************************************************************************/ - -void -fillHybridDelayLine( FIXP_DBL **fixpQmfReal, /*!< Qmf real Values */ - FIXP_DBL **fixpQmfImag, /*!< Qmf imag Values */ - FIXP_DBL fixpHybridLeftR[12], /*!< Hybrid real Values left channel */ - FIXP_DBL fixpHybridLeftI[12], /*!< Hybrid imag Values left channel */ - FIXP_DBL fixpHybridRightR[12], /*!< Hybrid real Values right channel */ - FIXP_DBL fixpHybridRightI[12], /*!< Hybrid imag Values right channel */ - HANDLE_HYBRID hHybrid ) -{ - int i; - - for (i = 0; i < HYBRID_FILTER_DELAY; i++) { - slotBasedHybridAnalysis ( fixpQmfReal[i], - fixpQmfReal[i], - fixpHybridLeftR, - fixpHybridLeftI, - hHybrid ); - } - - FDKmemcpy(fixpHybridRightR, fixpHybridLeftR, sizeof(FIXP_DBL)*NO_SUB_QMF_CHANNELS); - FDKmemcpy(fixpHybridRightI, fixpHybridLeftI, sizeof(FIXP_DBL)*NO_SUB_QMF_CHANNELS); -} - - -/******************************************************************************* - Functionname: slotBasedHybridAnalysis - ******************************************************************************* - - Description: The lower QMF subbands are further split to provide better - frequency resolution for PS processing. - - Return: none - -*******************************************************************************/ - - -void -slotBasedHybridAnalysis ( FIXP_DBL *fixpQmfReal, /*!< Qmf real Values */ - FIXP_DBL *fixpQmfImag, /*!< Qmf imag Values */ - - FIXP_DBL fixpHybridReal[12], /*!< Hybrid real Values */ - FIXP_DBL fixpHybridImag[12], /*!< Hybrid imag Values */ - - HANDLE_HYBRID hHybrid) -{ - int k, band; - HYBRID_RES hybridRes; - int chOffset = 0; - - C_ALLOC_SCRATCH_START(pTempRealSlot, FIXP_DBL, 4*HYBRID_FILTER_LENGTH); - - FIXP_DBL *pTempImagSlot = pTempRealSlot + HYBRID_FILTER_LENGTH; - FIXP_DBL *pWorkRealSlot = pTempImagSlot + HYBRID_FILTER_LENGTH; - FIXP_DBL *pWorkImagSlot = pWorkRealSlot + HYBRID_FILTER_LENGTH; - - /*! - Hybrid filtering is applied to the first hHybrid->nQmfBands QMF bands (3 when 10 or 20 stereo bands - are used, 5 when 34 stereo bands are used). For the remaining QMF bands a delay would be necessary. - But there is no need to implement a delay because there is a look-ahead of HYBRID_FILTER_DELAY = 6 - QMF samples in the low-band buffer. - */ - - for(band = 0; band < hHybrid->nQmfBands; band++) { - - /* get hybrid resolution per qmf band */ - /* in case of baseline ps 10/20 band stereo mode : */ - /* */ - /* qmfBand[0] : 8 ( HYBRID_8_CPLX ) */ - /* qmfBand[1] : 2 ( HYBRID_2_REAL ) */ - /* qmfBand[2] : 2 ( HYBRID_2_REAL ) */ - /* */ - /* (split the 3 lower qmf band to 12 hybrid bands) */ - - hybridRes = (HYBRID_RES)hHybrid->pResolution[band]; - - FDKmemcpy(pWorkRealSlot, hHybrid->mQmfBufferRealSlot[band], hHybrid->qmfBufferMove * sizeof(FIXP_DBL)); - FDKmemcpy(pWorkImagSlot, hHybrid->mQmfBufferImagSlot[band], hHybrid->qmfBufferMove * sizeof(FIXP_DBL)); - - pWorkRealSlot[hHybrid->qmfBufferMove] = fixpQmfReal[band]; - pWorkImagSlot[hHybrid->qmfBufferMove] = fixpQmfImag[band]; - - FDKmemcpy(hHybrid->mQmfBufferRealSlot[band], pWorkRealSlot + 1, hHybrid->qmfBufferMove * sizeof(FIXP_DBL)); - FDKmemcpy(hHybrid->mQmfBufferImagSlot[band], pWorkImagSlot + 1, hHybrid->qmfBufferMove * sizeof(FIXP_DBL)); - - if (fixpQmfReal) { - - /* actual filtering only if output signal requested */ - switch( hybridRes ) { - - /* HYBRID_2_REAL & HYBRID_8_CPLX are only needful for baseline ps */ - case HYBRID_2_REAL: - - slotBasedDualChannelFiltering( pWorkRealSlot, - pWorkImagSlot, - pTempRealSlot, - pTempImagSlot); - break; - - case HYBRID_8_CPLX: - - slotBasedEightChannelFiltering( pWorkRealSlot, - pWorkImagSlot, - pTempRealSlot, - pTempImagSlot); - break; - - default: - FDK_ASSERT(0); - } - - for(k = 0; k < (SCHAR)hybridRes; k++) { - fixpHybridReal [chOffset + k] = pTempRealSlot[k]; - fixpHybridImag [chOffset + k] = pTempImagSlot[k]; - } - chOffset += hybridRes; - } /* if (mHybridReal) */ - } - - /* group hybrid channels 3+4 -> 3 and 2+5 -> 2 */ - fixpHybridReal[3] += fixpHybridReal[4]; - fixpHybridImag[3] += fixpHybridImag[4]; - fixpHybridReal[4] = (FIXP_DBL)0; - fixpHybridImag[4] = (FIXP_DBL)0; - - fixpHybridReal[2] += fixpHybridReal[5]; - fixpHybridImag[2] += fixpHybridImag[5]; - fixpHybridReal[5] = (FIXP_DBL)0; - fixpHybridImag[5] = (FIXP_DBL)0; - - /* free memory on scratch */ - C_ALLOC_SCRATCH_END(pTempRealSlot, FIXP_DBL, 4*HYBRID_FILTER_LENGTH); - -} - - -/******************************************************************************* - Functionname: slotBasedHybridSynthesis - ******************************************************************************* - - Description: The coefficients offering higher resolution for the lower QMF - channel are simply added prior to the synthesis with the 54 - subbands QMF. - - Arguments: - - Return: none - -*******************************************************************************/ - -/*! 
-      l,r0(n) ---\
-      l,r1(n) ---- + --\
-      l,r2(n) ---/      \
-                         + --> F0(w)
-      l,r3(n) ---\      /
-      l,r4(n) ---- + --/
-      l,r5(n) ---/
-
-
-      l,r6(n) ---\
-                  + ---------> F1(w)
-      l,r7(n) ---/
-
-
-      l,r8(n) ---\
-                  + ---------> F2(w)
-      l,r9(n) ---/
-
-
- Hybrid QMF synthesis filterbank for the 10 and 20 stereo-bands configurations. The - coefficients offering higher resolution for the lower QMF channel are simply added - prior to the synthesis with the 54 subbands QMF. - - [see ISO/IEC 14496-3:2001/FDAM 2:2004(E) - Page 52] -*/ - - -void -slotBasedHybridSynthesis ( FIXP_DBL *fixpHybridReal, /*!< Hybrid real Values */ - FIXP_DBL *fixpHybridImag, /*!< Hybrid imag Values */ - FIXP_DBL *fixpQmfReal, /*!< Qmf real Values */ - FIXP_DBL *fixpQmfImag, /*!< Qmf imag Values */ - HANDLE_HYBRID hHybrid ) /*!< Handle to HYBRID struct. */ -{ - int k, band; - - HYBRID_RES hybridRes; - int chOffset = 0; - - for(band = 0; band < hHybrid->nQmfBands; band++) { - - FIXP_DBL qmfReal = FL2FXCONST_DBL(0.f); - FIXP_DBL qmfImag = FL2FXCONST_DBL(0.f); - hybridRes = (HYBRID_RES)hHybrid->pResolution[band]; - - for(k = 0; k < (SCHAR)hybridRes; k++) { - qmfReal += fixpHybridReal[chOffset + k]; - qmfImag += fixpHybridImag[chOffset + k]; - } - - fixpQmfReal[band] = qmfReal; - fixpQmfImag[band] = qmfImag; - - chOffset += hybridRes; - } -} - - - diff --git a/libSBRdec/src/psdec_hybrid.h b/libSBRdec/src/psdec_hybrid.h deleted file mode 100644 index fcf9e3e..0000000 --- a/libSBRdec/src/psdec_hybrid.h +++ /dev/null @@ -1,165 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -#ifndef __HYBRID_H -#define __HYBRID_H - -#include "sbrdecoder.h" - - -#define HYBRID_FILTER_LENGTH 13 -#define HYBRID_FILTER_DELAY 6 - - -#define FAST_FILTER2 -#define FAST_FILTER4 -#define FAST_FILTER8 -#define FAST_FILTER12 - -#define FFT_IDX_R(a) (2*a) -#define FFT_IDX_I(a) (2*a+1) - -#define FIXP_FFT_IDX_R(a) (a<<1) -#define FIXP_FFT_IDX_I(a) ((a<<1) + 1) - - -typedef enum { - - HYBRID_2_REAL = 2, - HYBRID_4_CPLX = 4, - HYBRID_8_CPLX = 8, - HYBRID_12_CPLX = 12 - -} HYBRID_RES; - -typedef struct -{ - SCHAR nQmfBands; - SCHAR frameSize; - SCHAR qmfBufferMove; - - UCHAR pResolution[3]; - - FIXP_DBL mQmfBufferRealSlot[3][HYBRID_FILTER_LENGTH]; /**< Stores old Qmf samples. */ - FIXP_DBL mQmfBufferImagSlot[3][HYBRID_FILTER_LENGTH]; - SCHAR sf_mQmfBuffer; - -} HYBRID; - -typedef HYBRID *HANDLE_HYBRID; - -void -fillHybridDelayLine( FIXP_DBL **fixpQmfReal, - FIXP_DBL **fixpQmfImag, - FIXP_DBL fixpHybridLeftR[12], - FIXP_DBL fixpHybridLeftI[12], - FIXP_DBL fixpHybridRightR[12], - FIXP_DBL fixpHybridRightI[12], - HANDLE_HYBRID hHybrid ); - -void -slotBasedHybridAnalysis ( FIXP_DBL *fixpQmfReal, - FIXP_DBL *fixpQmfImag, - - FIXP_DBL *fixpHybridReal, - FIXP_DBL *fixpHybridImag, - - HANDLE_HYBRID hHybrid); - - -void -slotBasedHybridSynthesis ( FIXP_DBL *fixpHybridReal, - FIXP_DBL *fixpHybridImag, - - FIXP_DBL *fixpQmfReal, - FIXP_DBL *fixpQmfImag, - - HANDLE_HYBRID hHybrid ); - -SBR_ERROR InitHybridFilterBank ( HANDLE_HYBRID hHybrid, - SCHAR frameSize, - SCHAR noBands, - const UCHAR *pResolution ); - - -#endif /* __HYBRID_H */ diff --git a/libSBRdec/src/sbr_crc.cpp b/libSBRdec/src/sbr_crc.cpp deleted file mode 100644 index a495f10..0000000 --- a/libSBRdec/src/sbr_crc.cpp +++ /dev/null @@ -1,183 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief CRC check coutines -*/ - -#include "sbr_crc.h" - -#include "FDK_bitstream.h" -#include "transcendent.h" - -#define MAXCRCSTEP 16 -#define MAXCRCSTEP_LD 4 - -/*! - \brief crc calculation -*/ -static ULONG -calcCRC (HANDLE_CRC hCrcBuf, ULONG bValue, int nBits) -{ - int i; - ULONG bMask = (1UL << (nBits - 1)); - - for (i = 0; i < nBits; i++, bMask >>= 1) { - USHORT flag = (hCrcBuf->crcState & hCrcBuf->crcMask) ? 1 : 0; - USHORT flag1 = (bMask & bValue) ? 1 : 0; - - flag ^= flag1; - hCrcBuf->crcState <<= 1; - if (flag) - hCrcBuf->crcState ^= hCrcBuf->crcPoly; - } - - return (hCrcBuf->crcState); -} - - -/*! - \brief crc -*/ -static int -getCrc (HANDLE_FDK_BITSTREAM hBs, ULONG NrBits) -{ - int i; - CRC_BUFFER CrcBuf; - - CrcBuf.crcState = SBR_CRC_START; - CrcBuf.crcPoly = SBR_CRC_POLY; - CrcBuf.crcMask = SBR_CRC_MASK; - - int CrcStep = NrBits>>MAXCRCSTEP_LD; - - int CrcNrBitsRest = (NrBits - CrcStep * MAXCRCSTEP); - ULONG bValue; - - for (i = 0; i < CrcStep; i++) { - bValue = FDKreadBits (hBs, MAXCRCSTEP); - calcCRC (&CrcBuf, bValue, MAXCRCSTEP); - } - - bValue = FDKreadBits (hBs, CrcNrBitsRest); - calcCRC (&CrcBuf, bValue, CrcNrBitsRest); - - return (CrcBuf.crcState & SBR_CRC_RANGE); - -} - - -/*! - \brief crc interface - \return 1: CRC OK, 0: CRC check failure -*/ -int -SbrCrcCheck (HANDLE_FDK_BITSTREAM hBs, /*!< handle to bit-buffer */ - LONG NrBits) /*!< max. CRC length */ -{ - int crcResult = 1; - ULONG NrCrcBits; - ULONG crcCheckResult; - LONG NrBitsAvailable; - ULONG crcCheckSum; - - crcCheckSum = FDKreadBits (hBs, 10); - - NrBitsAvailable = FDKgetValidBits(hBs); - if (NrBitsAvailable <= 0){ - return 0; - } - - NrCrcBits = fixMin ((INT)NrBits, (INT)NrBitsAvailable); - - crcCheckResult = getCrc (hBs, NrCrcBits); - FDKpushBack(hBs, (NrBitsAvailable - FDKgetValidBits(hBs)) ); - - - if (crcCheckResult != crcCheckSum) { - crcResult = 0; - } - - return (crcResult); -} diff --git a/libSBRdec/src/sbr_crc.h b/libSBRdec/src/sbr_crc.h deleted file mode 100644 index 30b8329..0000000 --- a/libSBRdec/src/sbr_crc.h +++ /dev/null @@ -1,123 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief CRC checking routines -*/ -#ifndef __SBR_CRC_H -#define __SBR_CRC_H - -#include "sbrdecoder.h" - -#include "FDK_bitstream.h" - -/* some useful crc polynoms: - -crc5: x^5+x^4+x^2+x^1+1 -crc6: x^6+x^5+x^3+x^2+x+1 -crc7: x^7+x^6+x^2+1 -crc8: x^8+x^2+x+x+1 -*/ - -/* default SBR CRC */ /* G(x) = x^10 + x^9 + x^5 + x^4 + x + 1 */ -#define SBR_CRC_POLY 0x0233 -#define SBR_CRC_MASK 0x0200 -#define SBR_CRC_START 0x0000 -#define SBR_CRC_RANGE 0x03FF - -typedef struct -{ - USHORT crcState; - USHORT crcMask; - USHORT crcPoly; -} -CRC_BUFFER; - -typedef CRC_BUFFER *HANDLE_CRC; - -int SbrCrcCheck (HANDLE_FDK_BITSTREAM hBitBuf, - LONG NrCrcBits); - - -#endif diff --git a/libSBRdec/src/sbr_deb.cpp b/libSBRdec/src/sbr_deb.cpp deleted file mode 100644 index 9baff2e..0000000 --- a/libSBRdec/src/sbr_deb.cpp +++ /dev/null @@ -1,90 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief Print selected debug messages -*/ - -#include "sbr_deb.h" - diff --git a/libSBRdec/src/sbr_deb.h b/libSBRdec/src/sbr_deb.h deleted file mode 100644 index cb954ba..0000000 --- a/libSBRdec/src/sbr_deb.h +++ /dev/null @@ -1,94 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief Debugging aids -*/ - -#ifndef __SBR_DEB_H -#define __SBR_DEB_H - -#include "sbrdecoder.h" - -#endif diff --git a/libSBRdec/src/sbr_dec.cpp b/libSBRdec/src/sbr_dec.cpp deleted file mode 100644 index 76009ba..0000000 --- a/libSBRdec/src/sbr_dec.cpp +++ /dev/null @@ -1,1102 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief Sbr decoder - This module provides the actual decoder implementation. The SBR data (side information) is already - decoded. Only three functions are provided: - - \li 1.) createSbrDec(): One time initialization - \li 2.) resetSbrDec(): Called by sbr_Apply() when the information contained in an SBR_HEADER_ELEMENT requires a reset - and recalculation of important SBR structures. - \li 3.) sbr_dec(): The actual decoder. Calls the different tools such as filterbanks, lppTransposer(), and calculateSbrEnvelope() - [the envelope adjuster]. - - \sa sbr_dec(), \ref documentationOverview -*/ - -#include "sbr_dec.h" - -#include "sbr_ram.h" -#include "env_extr.h" -#include "env_calc.h" -#include "scale.h" - -#include "genericStds.h" - -#include "sbrdec_drc.h" - - - -static void assignLcTimeSlots( HANDLE_SBR_DEC hSbrDec, /*!< handle to Decoder channel */ - FIXP_DBL **QmfBufferReal, - int noCols ) -{ - int slot, i; - FIXP_DBL *ptr; - - /* Number of QMF timeslots in the overlap buffer: */ - ptr = hSbrDec->pSbrOverlapBuffer; - for(slot=0; slotLppTrans.pSettings->overlap; slot++) { - QmfBufferReal[slot] = ptr; ptr += (64); - } - - /* Assign timeslots to Workbuffer1 */ - ptr = hSbrDec->WorkBuffer1; - for(i=0; i> 1) + hSbrDec->LppTrans.pSettings->overlap; - int totCols = noCols + hSbrDec->LppTrans.pSettings->overlap; - - /* Number of QMF timeslots in the overlap buffer: */ - ptr = hSbrDec->pSbrOverlapBuffer; - for(slot=0; slotLppTrans.pSettings->overlap; slot++) { - QmfBufferReal[slot] = ptr; ptr += (64); - QmfBufferImag[slot] = ptr; ptr += (64); - } - - /* Assign first half of timeslots to Workbuffer1 */ - ptr = hSbrDec->WorkBuffer1; - for(; slotWorkBuffer2; - for(; slotuseLP = useLP; - if (useLP) { - hSbrDec->SynthesisQMF.flags |= QMF_FLAG_LP; - hSbrDec->AnalysiscQMF.flags |= QMF_FLAG_LP; - } else { - hSbrDec->SynthesisQMF.flags &= ~QMF_FLAG_LP; - hSbrDec->AnalysiscQMF.flags &= ~QMF_FLAG_LP; - } - if (!useLP) - assignHqTimeSlots( hSbrDec, hSbrDec->QmfBufferReal, hSbrDec->QmfBufferImag, noCols ); - else - { - assignLcTimeSlots( hSbrDec, hSbrDec->QmfBufferReal, noCols ); - } -} - -static void changeQmfType( HANDLE_SBR_DEC hSbrDec, /*!< handle to Decoder channel */ - int useLdTimeAlign ) -{ - UINT synQmfFlags = hSbrDec->SynthesisQMF.flags; - UINT anaQmfFlags = hSbrDec->AnalysiscQMF.flags; - int resetSynQmf = 0; - int resetAnaQmf = 0; - - /* assign qmf type */ - if (useLdTimeAlign) { - if (synQmfFlags & QMF_FLAG_CLDFB) { - /* change the type to MPSLD */ - synQmfFlags &= ~QMF_FLAG_CLDFB; - synQmfFlags |= QMF_FLAG_MPSLDFB; - resetSynQmf = 1; - } - if (anaQmfFlags & QMF_FLAG_CLDFB) { - /* change the type to MPSLD */ - anaQmfFlags &= ~QMF_FLAG_CLDFB; - anaQmfFlags |= QMF_FLAG_MPSLDFB; - resetAnaQmf = 1; - } - } else { - if (synQmfFlags & QMF_FLAG_MPSLDFB) { - /* change the type to CLDFB */ - synQmfFlags &= ~QMF_FLAG_MPSLDFB; - synQmfFlags |= QMF_FLAG_CLDFB; - resetSynQmf = 1; - } - if (anaQmfFlags & QMF_FLAG_MPSLDFB) { - /* change the type to CLDFB */ - anaQmfFlags &= ~QMF_FLAG_MPSLDFB; - anaQmfFlags |= QMF_FLAG_CLDFB; - resetAnaQmf = 1; - } - } - - if (resetAnaQmf) { - QMF_FILTER_BANK prvAnaQmf; - int qmfErr; - - /* Store current configuration */ - FDKmemcpy(&prvAnaQmf, &hSbrDec->AnalysiscQMF, sizeof(QMF_FILTER_BANK)); - - /* Reset analysis QMF */ - qmfErr = qmfInitAnalysisFilterBank ( - &hSbrDec->AnalysiscQMF, - hSbrDec->anaQmfStates, - hSbrDec->AnalysiscQMF.no_col, - hSbrDec->AnalysiscQMF.lsb, - hSbrDec->AnalysiscQMF.usb, - hSbrDec->AnalysiscQMF.no_channels, - anaQmfFlags | QMF_FLAG_KEEP_STATES - ); - - if (qmfErr != 0) { - /* Restore old configuration of analysis QMF */ - FDKmemcpy(&hSbrDec->AnalysiscQMF, &prvAnaQmf, sizeof(QMF_FILTER_BANK)); - } - } - - if (resetSynQmf) { - QMF_FILTER_BANK prvSynQmf; - int qmfErr; - - /* Store current configuration */ - FDKmemcpy(&prvSynQmf, &hSbrDec->SynthesisQMF, sizeof(QMF_FILTER_BANK)); - - /* Reset synthesis QMF */ - qmfErr = qmfInitSynthesisFilterBank ( - &hSbrDec->SynthesisQMF, - hSbrDec->pSynQmfStates, - hSbrDec->SynthesisQMF.no_col, - hSbrDec->SynthesisQMF.lsb, - hSbrDec->SynthesisQMF.usb, - hSbrDec->SynthesisQMF.no_channels, - synQmfFlags | QMF_FLAG_KEEP_STATES - ); - - if (qmfErr != 0) { - /* Restore old configuration of synthesis QMF */ - FDKmemcpy(&hSbrDec->SynthesisQMF, &prvSynQmf, sizeof(QMF_FILTER_BANK)); - } - } -} - - -/*! - \brief SBR decoder core function for one channel - - \image html BufferMgmtDetailed-1632.png - - Besides the filter states of the QMF filter bank and the LPC-states of - the LPP-Transposer, processing is mainly based on four buffers: - #timeIn, #timeOut, #WorkBuffer2 and #OverlapBuffer. The #WorkBuffer2 - is reused for all channels and might be used by the core decoder, a - static overlap buffer is required for each channel. Du to in-place - processing, #timeIn and #timeOut point to identical locations. - - The spectral data is organized in so-called slots, each slot - containing 64 bands of complex data. The number of slots per frame is - dependend on the frame size. For mp3PRO, there are 18 slots per frame - and 6 slots per #OverlapBuffer. It is not necessary to have the slots - in located consecutive address ranges. - - To optimize memory usage and to minimize the number of memory - accesses, the memory management is organized as follows (Slot numbers - based on mp3PRO): - - 1.) Input time domain signal is located in #timeIn, the last slots - (0..5) of the spectral data of the previous frame are located in the - #OverlapBuffer. In addition, #frameData of the current frame resides - in the upper part of #timeIn. - - 2.) During the cplxAnalysisQmfFiltering(), 32 samples from #timeIn are transformed - into a slot of up to 32 complex spectral low band values at a - time. The first spectral slot -- nr. 6 -- is written at slot number - zero of #WorkBuffer2. #WorkBuffer2 will be completely filled with - spectral data. - - 3.) LPP-Transposition in lppTransposer() is processed on 24 slots. During the - transposition, the high band part of the spectral data is replicated - based on the low band data. - - Envelope Adjustment is processed on the high band part of the spectral - data only by calculateSbrEnvelope(). - - 4.) The cplxSynthesisQmfFiltering() creates 64 time domain samples out - of a slot of 64 complex spectral values at a time. The first 6 slots - in #timeOut are filled from the results of spectral slots 0..5 in the - #OverlapBuffer. The consecutive slots in timeOut are now filled with - the results of spectral slots 6..17. - - 5.) The preprocessed slots 18..23 have to be stored in the - #OverlapBuffer. - -*/ - -void -sbr_dec ( HANDLE_SBR_DEC hSbrDec, /*!< handle to Decoder channel */ - INT_PCM *timeIn, /*!< pointer to input time signal */ - INT_PCM *timeOut, /*!< pointer to output time signal */ - HANDLE_SBR_DEC hSbrDecRight, /*!< handle to Decoder channel right */ - INT_PCM *timeOutRight, /*!< pointer to output time signal */ - const int strideIn, /*!< Time data traversal strideIn */ - const int strideOut, /*!< Time data traversal strideOut */ - HANDLE_SBR_HEADER_DATA hHeaderData,/*!< Static control data */ - HANDLE_SBR_FRAME_DATA hFrameData, /*!< Control data of current frame */ - HANDLE_SBR_PREV_FRAME_DATA hPrevFrameData, /*!< Some control data of last frame */ - const int applyProcessing, /*!< Flag for SBR operation */ - HANDLE_PS_DEC h_ps_d, - const UINT flags, - const int codecFrameSize - ) -{ - int i, slot, reserve; - int saveLbScale; - int ov_len; - int lastSlotOffs; - FIXP_DBL maxVal; - - /* 1+1/3 frames of spectral data: */ - FIXP_DBL **QmfBufferReal = hSbrDec->QmfBufferReal; - FIXP_DBL **QmfBufferImag = hSbrDec->QmfBufferImag; - - /* Number of QMF timeslots in the overlap buffer: */ - ov_len = hSbrDec->LppTrans.pSettings->overlap; - - /* Number of QMF slots per frame */ - int noCols = hHeaderData->numberTimeSlots * hHeaderData->timeStep; - - /* assign qmf time slots */ - if ( ((flags & SBRDEC_LOW_POWER ) ? 1 : 0) != ((hSbrDec->SynthesisQMF.flags & QMF_FLAG_LP) ? 1 : 0) ) { - assignTimeSlots( hSbrDec, hHeaderData->numberTimeSlots * hHeaderData->timeStep, flags & SBRDEC_LOW_POWER); - } - - if (flags & SBRDEC_ELD_GRID) { - /* Choose the right low delay filter bank */ - changeQmfType( hSbrDec, (flags & SBRDEC_LD_MPS_QMF) ? 1 : 0 ); - - /* If the LD-MPS QMF is not available delay the signal by (96-48*ldSbrSamplingRate) - * samples according to ISO/IEC 14496-3:2009/FDAM 2:2010(E) chapter 4.5.2.13. */ - if ( (flags & SBRDEC_LD_MPS_QMF) - && (hSbrDec->AnalysiscQMF.flags & QMF_FLAG_CLDFB) ) - { - INT_PCM *pDlyBuf = hSbrDec->coreDelayBuf; /* DLYBUF */ - int smpl, delay = 96 >> (!(flags & SBRDEC_DOWNSAMPLE) ? 1 : 0); - /* Create TMPBUF */ - C_AALLOC_SCRATCH_START(pcmTemp, INT_PCM, (96)); - /* Copy delay samples from INBUF to TMPBUF */ - for (smpl = 0; smpl < delay; smpl += 1) { - pcmTemp[smpl] = timeIn[(codecFrameSize-delay+smpl)*strideIn]; - } - /* Move input signal remainder to the very end of INBUF */ - for (smpl = (codecFrameSize-delay-1)*strideIn; smpl >= 0; smpl -= strideIn) { - timeIn[smpl+delay] = timeIn[smpl]; - } - /* Copy delayed samples from last frame from DLYBUF to the very beginning of INBUF */ - for (smpl = 0; smpl < delay; smpl += 1) { - timeIn[smpl*strideIn] = pDlyBuf[smpl]; - } - /* Copy TMPBUF to DLYBUF */ - FDKmemcpy(pDlyBuf, pcmTemp, delay*sizeof(INT_PCM)); - /* Destory TMPBUF */ - C_AALLOC_SCRATCH_END(pcmTemp, INT_PCM, (96)); - } - } - - /* - low band codec signal subband filtering - */ - - { - C_AALLOC_SCRATCH_START(qmfTemp, FIXP_DBL, 2*(64)); - - qmfAnalysisFiltering( &hSbrDec->AnalysiscQMF, - QmfBufferReal + ov_len, - QmfBufferImag + ov_len, - &hSbrDec->sbrScaleFactor, - timeIn, - strideIn, - qmfTemp - ); - - C_AALLOC_SCRATCH_END(qmfTemp, FIXP_DBL, 2*(64)); - } - - /* - Clear upper half of spectrum - */ - { - int nAnalysisBands = hHeaderData->numberOfAnalysisBands; - - if (! (flags & SBRDEC_LOW_POWER)) { - for (slot = ov_len; slot < noCols+ov_len; slot++) { - FDKmemclear(&QmfBufferReal[slot][nAnalysisBands],((64)-nAnalysisBands)*sizeof(FIXP_DBL)); - FDKmemclear(&QmfBufferImag[slot][nAnalysisBands],((64)-nAnalysisBands)*sizeof(FIXP_DBL)); - } - } else - for (slot = ov_len; slot < noCols+ov_len; slot++) { - FDKmemclear(&QmfBufferReal[slot][nAnalysisBands],((64)-nAnalysisBands)*sizeof(FIXP_DBL)); - } - } - - - - /* - Shift spectral data left to gain accuracy in transposer and adjustor - */ - maxVal = maxSubbandSample( QmfBufferReal, - (flags & SBRDEC_LOW_POWER) ? NULL : QmfBufferImag, - 0, - hSbrDec->AnalysiscQMF.lsb, - ov_len, - noCols+ov_len ); - - reserve = fixMax(0,CntLeadingZeros(maxVal)-1) ; - reserve = fixMin(reserve,DFRACT_BITS-1-hSbrDec->sbrScaleFactor.lb_scale); - - /* If all data is zero, lb_scale could become too large */ - rescaleSubbandSamples( QmfBufferReal, - (flags & SBRDEC_LOW_POWER) ? NULL : QmfBufferImag, - 0, - hSbrDec->AnalysiscQMF.lsb, - ov_len, - noCols+ov_len, - reserve); - - hSbrDec->sbrScaleFactor.lb_scale += reserve; - - /* - save low band scale, wavecoding or parametric stereo may modify it - */ - saveLbScale = hSbrDec->sbrScaleFactor.lb_scale; - - - if (applyProcessing) - { - UCHAR * borders = hFrameData->frameInfo.borders; - lastSlotOffs = borders[hFrameData->frameInfo.nEnvelopes] - hHeaderData->numberTimeSlots; - - FIXP_DBL degreeAlias[(64)]; - - /* The transposer will override most values in degreeAlias[]. - The array needs to be cleared at least from lowSubband to highSubband before. */ - if (flags & SBRDEC_LOW_POWER) - FDKmemclear(°reeAlias[hHeaderData->freqBandData.lowSubband], (hHeaderData->freqBandData.highSubband-hHeaderData->freqBandData.lowSubband)*sizeof(FIXP_DBL)); - - /* - Inverse filtering of lowband and transposition into the SBR-frequency range - */ - - lppTransposer ( &hSbrDec->LppTrans, - &hSbrDec->sbrScaleFactor, - QmfBufferReal, - degreeAlias, // only used if useLP = 1 - QmfBufferImag, - flags & SBRDEC_LOW_POWER, - hHeaderData->timeStep, - borders[0], - lastSlotOffs, - hHeaderData->freqBandData.nInvfBands, - hFrameData->sbr_invf_mode, - hPrevFrameData->sbr_invf_mode ); - - - - - - /* - Adjust envelope of current frame. - */ - - calculateSbrEnvelope (&hSbrDec->sbrScaleFactor, - &hSbrDec->SbrCalculateEnvelope, - hHeaderData, - hFrameData, - QmfBufferReal, - QmfBufferImag, - flags & SBRDEC_LOW_POWER, - - degreeAlias, - flags, - (hHeaderData->frameErrorFlag || hPrevFrameData->frameErrorFlag)); - - - /* - Update hPrevFrameData (to be used in the next frame) - */ - for (i=0; ifreqBandData.nInvfBands; i++) { - hPrevFrameData->sbr_invf_mode[i] = hFrameData->sbr_invf_mode[i]; - } - hPrevFrameData->coupling = hFrameData->coupling; - hPrevFrameData->stopPos = borders[hFrameData->frameInfo.nEnvelopes]; - hPrevFrameData->ampRes = hFrameData->ampResolutionCurrentFrame; - } - else { - /* Reset hb_scale if no highband is present, because hb_scale is considered in the QMF-synthesis */ - hSbrDec->sbrScaleFactor.hb_scale = saveLbScale; - } - - - for (i=0; iLppTrans.lpcFilterStatesReal[i], QmfBufferReal[noCols-LPC_ORDER+i], hSbrDec->AnalysiscQMF.lsb*sizeof(FIXP_DBL)); - FDKmemcpy(hSbrDec->LppTrans.lpcFilterStatesImag[i], QmfBufferImag[noCols-LPC_ORDER+i], hSbrDec->AnalysiscQMF.lsb*sizeof(FIXP_DBL)); - } else - FDKmemcpy(hSbrDec->LppTrans.lpcFilterStatesReal[i], QmfBufferReal[noCols-LPC_ORDER+i], hSbrDec->AnalysiscQMF.lsb*sizeof(FIXP_DBL)); - } - - /* - Synthesis subband filtering. - */ - - if ( ! (flags & SBRDEC_PS_DECODED) ) { - - { - int outScalefactor = 0; - - if (h_ps_d != NULL) { - h_ps_d->procFrameBased = 1; /* we here do frame based processing */ - } - - - sbrDecoder_drcApply(&hSbrDec->sbrDrcChannel, - QmfBufferReal, - (flags & SBRDEC_LOW_POWER) ? NULL : QmfBufferImag, - hSbrDec->SynthesisQMF.no_col, - &outScalefactor - ); - - - - qmfChangeOutScalefactor(&hSbrDec->SynthesisQMF, outScalefactor ); - - { - C_AALLOC_SCRATCH_START(qmfTemp, FIXP_DBL, 2*(64)); - - qmfSynthesisFiltering( &hSbrDec->SynthesisQMF, - QmfBufferReal, - (flags & SBRDEC_LOW_POWER) ? NULL : QmfBufferImag, - &hSbrDec->sbrScaleFactor, - hSbrDec->LppTrans.pSettings->overlap, - timeOut, - strideOut, - qmfTemp); - - C_AALLOC_SCRATCH_END(qmfTemp, FIXP_DBL, 2*(64)); - } - - } - - } else { /* (flags & SBRDEC_PS_DECODED) */ - INT i, sdiff, outScalefactor, scaleFactorLowBand, scaleFactorHighBand; - SCHAR scaleFactorLowBand_ov, scaleFactorLowBand_no_ov; - - HANDLE_QMF_FILTER_BANK synQmf = &hSbrDec->SynthesisQMF; - HANDLE_QMF_FILTER_BANK synQmfRight = &hSbrDecRight->SynthesisQMF; - - /* adapt scaling */ - sdiff = hSbrDec->sbrScaleFactor.lb_scale - reserve; /* Scaling difference */ - scaleFactorHighBand = sdiff - hSbrDec->sbrScaleFactor.hb_scale; /* Scale of current high band */ - scaleFactorLowBand_ov = sdiff - hSbrDec->sbrScaleFactor.ov_lb_scale; /* Scale of low band overlapping QMF data */ - scaleFactorLowBand_no_ov = sdiff - hSbrDec->sbrScaleFactor.lb_scale; /* Scale of low band current QMF data */ - outScalefactor = 0; /* Initial output scale */ - - if (h_ps_d->procFrameBased == 1) /* If we have switched from frame to slot based processing copy filter states */ - { /* procFrameBased will be unset later */ - /* copy filter states from left to right */ - FDKmemcpy(synQmfRight->FilterStates, synQmf->FilterStates, ((640)-(64))*sizeof(FIXP_QSS)); - } - - /* scale ALL qmf vales ( real and imag ) of mono / left channel to the - same scale factor ( ov_lb_sf, lb_sf and hq_sf ) */ - scalFilterBankValues( h_ps_d, /* parametric stereo decoder handle */ - QmfBufferReal, /* qmf filterbank values */ - QmfBufferImag, /* qmf filterbank values */ - synQmf->lsb, /* sbr start subband */ - hSbrDec->sbrScaleFactor.ov_lb_scale, - hSbrDec->sbrScaleFactor.lb_scale, - &scaleFactorLowBand_ov, /* adapt scaling values */ - &scaleFactorLowBand_no_ov, /* adapt scaling values */ - hSbrDec->sbrScaleFactor.hb_scale, /* current frame ( highband ) */ - &scaleFactorHighBand, - synQmf->no_col); - - /* use the same synthese qmf values for left and right channel */ - synQmfRight->no_col = synQmf->no_col; - synQmfRight->lsb = synQmf->lsb; - synQmfRight->usb = synQmf->usb; - - int env=0; - - outScalefactor += (SCAL_HEADROOM+1); /* psDiffScale! */ - - { - C_AALLOC_SCRATCH_START(pWorkBuffer, FIXP_DBL, 2*(64)); - - int maxShift = 0; - - if (hSbrDec->sbrDrcChannel.enable != 0) { - if (hSbrDec->sbrDrcChannel.prevFact_exp > maxShift) { - maxShift = hSbrDec->sbrDrcChannel.prevFact_exp; - } - if (hSbrDec->sbrDrcChannel.currFact_exp > maxShift) { - maxShift = hSbrDec->sbrDrcChannel.currFact_exp; - } - if (hSbrDec->sbrDrcChannel.nextFact_exp > maxShift) { - maxShift = hSbrDec->sbrDrcChannel.nextFact_exp; - } - } - - /* copy DRC data to right channel (with PS both channels use the same DRC gains) */ - FDKmemcpy(&hSbrDecRight->sbrDrcChannel, &hSbrDec->sbrDrcChannel, sizeof(SBRDEC_DRC_CHANNEL)); - - for (i = 0; i < synQmf->no_col; i++) { /* ----- no_col loop ----- */ - - INT outScalefactorR, outScalefactorL; - outScalefactorR = outScalefactorL = outScalefactor; - - /* qmf timeslot of right channel */ - FIXP_DBL* rQmfReal = pWorkBuffer; - FIXP_DBL* rQmfImag = pWorkBuffer + 64; - - - { - if ( i == h_ps_d->bsData[h_ps_d->processSlot].mpeg.aEnvStartStop[env] ) { - initSlotBasedRotation( h_ps_d, env, hHeaderData->freqBandData.highSubband ); - env++; - } - - ApplyPsSlot( h_ps_d, /* parametric stereo decoder handle */ - (QmfBufferReal + i), /* one timeslot of left/mono channel */ - (QmfBufferImag + i), /* one timeslot of left/mono channel */ - rQmfReal, /* one timeslot or right channel */ - rQmfImag); /* one timeslot or right channel */ - } - - - scaleFactorLowBand = (i<(6)) ? scaleFactorLowBand_ov : scaleFactorLowBand_no_ov; - - - sbrDecoder_drcApplySlot ( /* right channel */ - &hSbrDecRight->sbrDrcChannel, - rQmfReal, - rQmfImag, - i, - synQmfRight->no_col, - maxShift - ); - - outScalefactorR += maxShift; - - sbrDecoder_drcApplySlot ( /* left channel */ - &hSbrDec->sbrDrcChannel, - *(QmfBufferReal + i), - *(QmfBufferImag + i), - i, - synQmf->no_col, - maxShift - ); - - outScalefactorL += maxShift; - - - /* scale filter states for left and right channel */ - qmfChangeOutScalefactor( synQmf, outScalefactorL ); - qmfChangeOutScalefactor( synQmfRight, outScalefactorR ); - - { - - qmfSynthesisFilteringSlot( synQmfRight, - rQmfReal, /* QMF real buffer */ - rQmfImag, /* QMF imag buffer */ - scaleFactorLowBand, - scaleFactorHighBand, - timeOutRight+(i*synQmf->no_channels*strideOut), - strideOut, - pWorkBuffer); - - qmfSynthesisFilteringSlot( synQmf, - *(QmfBufferReal + i), /* QMF real buffer */ - *(QmfBufferImag + i), /* QMF imag buffer */ - scaleFactorLowBand, - scaleFactorHighBand, - timeOut+(i*synQmf->no_channels*strideOut), - strideOut, - pWorkBuffer); - - } - } /* no_col loop i */ - - /* scale back (6) timeslots look ahead for hybrid filterbank to original value */ - rescalFilterBankValues( h_ps_d, - QmfBufferReal, - QmfBufferImag, - synQmf->lsb, - synQmf->no_col ); - - C_AALLOC_SCRATCH_END(pWorkBuffer, FIXP_DBL, 2*(64)); - } - } - - sbrDecoder_drcUpdateChannel( &hSbrDec->sbrDrcChannel ); - - - /* - Update overlap buffer - Even bands above usb are copied to avoid outdated spectral data in case - the stop frequency raises. - */ - - if (hSbrDec->LppTrans.pSettings->overlap > 0) - { - if (! (flags & SBRDEC_LOW_POWER)) { - for ( i=0; iLppTrans.pSettings->overlap; i++ ) { - FDKmemcpy(QmfBufferReal[i], QmfBufferReal[i+noCols], (64)*sizeof(FIXP_DBL)); - FDKmemcpy(QmfBufferImag[i], QmfBufferImag[i+noCols], (64)*sizeof(FIXP_DBL)); - } - } else - for ( i=0; iLppTrans.pSettings->overlap; i++ ) { - FDKmemcpy(QmfBufferReal[i], QmfBufferReal[i+noCols], (64)*sizeof(FIXP_DBL)); - } - } - - hSbrDec->sbrScaleFactor.ov_lb_scale = saveLbScale; - - /* Save current frame status */ - hPrevFrameData->frameErrorFlag = hHeaderData->frameErrorFlag; - -} // sbr_dec() - - -/*! - \brief Creates sbr decoder structure - \return errorCode, 0 if successful -*/ -SBR_ERROR -createSbrDec (SBR_CHANNEL * hSbrChannel, - HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ - TRANSPOSER_SETTINGS *pSettings, - const int downsampleFac, /*!< Downsampling factor */ - const UINT qmfFlags, /*!< flags -> 1: HQ/LP selector, 2: CLDFB */ - const UINT flags, - const int overlap, - int chan) /*!< Channel for which to assign buffers etc. */ - -{ - SBR_ERROR err = SBRDEC_OK; - int timeSlots = hHeaderData->numberTimeSlots; /* Number of SBR slots per frame */ - int noCols = timeSlots * hHeaderData->timeStep; /* Number of QMF slots per frame */ - HANDLE_SBR_DEC hs = &(hSbrChannel->SbrDec); - - /* Initialize scale factors */ - hs->sbrScaleFactor.ov_lb_scale = 0; - hs->sbrScaleFactor.ov_hb_scale = 0; - hs->sbrScaleFactor.hb_scale = 0; - - - /* - create envelope calculator - */ - err = createSbrEnvelopeCalc (&hs->SbrCalculateEnvelope, - hHeaderData, - chan, - flags); - if (err != SBRDEC_OK) { - return err; - } - - /* - create QMF filter banks - */ - { - int qmfErr; - /* Adapted QMF analysis post-twiddles for down-sampled HQ SBR */ - const UINT downSampledFlag = (flags & SBRDEC_DOWNSAMPLE) ? QMF_FLAG_DOWNSAMPLED : 0; - - qmfErr = qmfInitAnalysisFilterBank ( - &hs->AnalysiscQMF, - hs->anaQmfStates, - noCols, - hHeaderData->freqBandData.lowSubband, - hHeaderData->freqBandData.highSubband, - hHeaderData->numberOfAnalysisBands, - (qmfFlags & (~QMF_FLAG_KEEP_STATES)) | downSampledFlag - ); - if (qmfErr != 0) { - return SBRDEC_UNSUPPORTED_CONFIG; - } - } - if (hs->pSynQmfStates == NULL) { - hs->pSynQmfStates = GetRam_sbr_QmfStatesSynthesis(chan); - if (hs->pSynQmfStates == NULL) - return SBRDEC_MEM_ALLOC_FAILED; - } - - { - int qmfErr; - - qmfErr = qmfInitSynthesisFilterBank ( - &hs->SynthesisQMF, - hs->pSynQmfStates, - noCols, - hHeaderData->freqBandData.lowSubband, - hHeaderData->freqBandData.highSubband, - (64) / downsampleFac, - qmfFlags & (~QMF_FLAG_KEEP_STATES) - ); - - if (qmfErr != 0) { - return SBRDEC_UNSUPPORTED_CONFIG; - } - } - initSbrPrevFrameData (&hSbrChannel->prevFrameData, timeSlots); - - /* - create transposer - */ - err = createLppTransposer (&hs->LppTrans, - pSettings, - hHeaderData->freqBandData.lowSubband, - hHeaderData->freqBandData.v_k_master, - hHeaderData->freqBandData.numMaster, - hs->SynthesisQMF.usb, - timeSlots, - hs->AnalysiscQMF.no_col, - hHeaderData->freqBandData.freqBandTableNoise, - hHeaderData->freqBandData.nNfb, - hHeaderData->sbrProcSmplRate, - chan, - overlap ); - if (err != SBRDEC_OK) { - return err; - } - - /* The CLDFB does not have overlap */ - if ((qmfFlags & QMF_FLAG_CLDFB) == 0) { - if (hs->pSbrOverlapBuffer == NULL) { - hs->pSbrOverlapBuffer = GetRam_sbr_OverlapBuffer(chan); - if (hs->pSbrOverlapBuffer == NULL) { - return SBRDEC_MEM_ALLOC_FAILED; - } - } else { - /* Clear overlap buffer */ - FDKmemclear( hs->pSbrOverlapBuffer, - sizeof(FIXP_DBL) * 2 * (6) * (64) - ); - } - } - - /* Clear input delay line */ - FDKmemclear(hs->coreDelayBuf, (96)*sizeof(INT_PCM)); - - /* assign qmf time slots */ - assignTimeSlots( &hSbrChannel->SbrDec, hHeaderData->numberTimeSlots * hHeaderData->timeStep, qmfFlags & QMF_FLAG_LP); - - return err; -} - -/*! - \brief Delete sbr decoder structure - \return errorCode, 0 if successful -*/ -int -deleteSbrDec (SBR_CHANNEL * hSbrChannel) -{ - HANDLE_SBR_DEC hs = &hSbrChannel->SbrDec; - - deleteSbrEnvelopeCalc (&hs->SbrCalculateEnvelope); - - /* delete QMF filter states */ - if (hs->pSynQmfStates != NULL) { - FreeRam_sbr_QmfStatesSynthesis(&hs->pSynQmfStates); - } - - - if (hs->pSbrOverlapBuffer != NULL) { - FreeRam_sbr_OverlapBuffer(&hs->pSbrOverlapBuffer); - } - - return 0; -} - - -/*! - \brief resets sbr decoder structure - \return errorCode, 0 if successful -*/ -SBR_ERROR -resetSbrDec (HANDLE_SBR_DEC hSbrDec, - HANDLE_SBR_HEADER_DATA hHeaderData, - HANDLE_SBR_PREV_FRAME_DATA hPrevFrameData, - const int useLP, - const int downsampleFac - ) -{ - SBR_ERROR sbrError = SBRDEC_OK; - - int old_lsb = hSbrDec->SynthesisQMF.lsb; - int new_lsb = hHeaderData->freqBandData.lowSubband; - int l, startBand, stopBand, startSlot, size; - - int source_scale, target_scale, delta_scale, target_lsb, target_usb, reserve; - FIXP_DBL maxVal; - - /* overlapBuffer point to first (6) slots */ - FIXP_DBL **OverlapBufferReal = hSbrDec->QmfBufferReal; - FIXP_DBL **OverlapBufferImag = hSbrDec->QmfBufferImag; - - if (!hSbrDec->LppTrans.pSettings) { - return SBRDEC_NOT_INITIALIZED; - } - - /* assign qmf time slots */ - assignTimeSlots( hSbrDec, hHeaderData->numberTimeSlots * hHeaderData->timeStep, useLP); - - - - resetSbrEnvelopeCalc (&hSbrDec->SbrCalculateEnvelope); - - hSbrDec->SynthesisQMF.lsb = hHeaderData->freqBandData.lowSubband; - hSbrDec->SynthesisQMF.usb = fixMin((INT)hSbrDec->SynthesisQMF.no_channels, (INT)hHeaderData->freqBandData.highSubband); - - hSbrDec->AnalysiscQMF.lsb = hSbrDec->SynthesisQMF.lsb; - hSbrDec->AnalysiscQMF.usb = hSbrDec->SynthesisQMF.usb; - - - /* - The following initialization of spectral data in the overlap buffer - is required for dynamic x-over or a change of the start-freq for 2 reasons: - - 1. If the lowband gets _wider_, unadjusted data would remain - - 2. If the lowband becomes _smaller_, the highest bands of the old lowband - must be cleared because the whitening would be affected - */ - startBand = old_lsb; - stopBand = new_lsb; - startSlot = hHeaderData->timeStep * (hPrevFrameData->stopPos - hHeaderData->numberTimeSlots); - size = fixMax(0,stopBand-startBand); - - /* keep already adjusted data in the x-over-area */ - if (!useLP) { - for (l=startSlot; lLppTrans.pSettings->overlap; l++) { - FDKmemclear(&OverlapBufferReal[l][startBand], size*sizeof(FIXP_DBL)); - FDKmemclear(&OverlapBufferImag[l][startBand], size*sizeof(FIXP_DBL)); - } - } else - for (l=startSlot; lLppTrans.pSettings->overlap ; l++) { - FDKmemclear(&OverlapBufferReal[l][startBand], size*sizeof(FIXP_DBL)); - } - - - /* - reset LPC filter states - */ - startBand = fixMin(old_lsb,new_lsb); - stopBand = fixMax(old_lsb,new_lsb); - size = fixMax(0,stopBand-startBand); - - FDKmemclear(&hSbrDec->LppTrans.lpcFilterStatesReal[0][startBand], size*sizeof(FIXP_DBL)); - FDKmemclear(&hSbrDec->LppTrans.lpcFilterStatesReal[1][startBand], size*sizeof(FIXP_DBL)); - if (!useLP) { - FDKmemclear(&hSbrDec->LppTrans.lpcFilterStatesImag[0][startBand], size*sizeof(FIXP_DBL)); - FDKmemclear(&hSbrDec->LppTrans.lpcFilterStatesImag[1][startBand], size*sizeof(FIXP_DBL)); - } - - - /* - Rescale already processed spectral data between old and new x-over frequency. - This must be done because of the separate scalefactors for lowband and highband. - */ - startBand = fixMin(old_lsb,new_lsb); - stopBand = fixMax(old_lsb,new_lsb); - - if (new_lsb > old_lsb) { - /* The x-over-area was part of the highband before and will now belong to the lowband */ - source_scale = hSbrDec->sbrScaleFactor.ov_hb_scale; - target_scale = hSbrDec->sbrScaleFactor.ov_lb_scale; - target_lsb = 0; - target_usb = old_lsb; - } - else { - /* The x-over-area was part of the lowband before and will now belong to the highband */ - source_scale = hSbrDec->sbrScaleFactor.ov_lb_scale; - target_scale = hSbrDec->sbrScaleFactor.ov_hb_scale; - /* jdr: The values old_lsb and old_usb might be wrong because the previous frame might have been "upsamling". */ - target_lsb = hSbrDec->SynthesisQMF.lsb; - target_usb = hSbrDec->SynthesisQMF.usb; - } - - /* Shift left all samples of the x-over-area as much as possible - An unnecessary coarse scale could cause ov_lb_scale or ov_hb_scale to be - adapted and the accuracy in the next frame would seriously suffer! */ - - maxVal = maxSubbandSample( OverlapBufferReal, - (useLP) ? NULL : OverlapBufferImag, - startBand, - stopBand, - 0, - startSlot); - - reserve = CntLeadingZeros(maxVal)-1; - reserve = fixMin(reserve,DFRACT_BITS-1-source_scale); - - rescaleSubbandSamples( OverlapBufferReal, - (useLP) ? NULL : OverlapBufferImag, - startBand, - stopBand, - 0, - startSlot, - reserve); - source_scale += reserve; - - delta_scale = target_scale - source_scale; - - if (delta_scale > 0) { /* x-over-area is dominant */ - delta_scale = -delta_scale; - startBand = target_lsb; - stopBand = target_usb; - - if (new_lsb > old_lsb) { - /* The lowband has to be rescaled */ - hSbrDec->sbrScaleFactor.ov_lb_scale = source_scale; - } - else { - /* The highband has be be rescaled */ - hSbrDec->sbrScaleFactor.ov_hb_scale = source_scale; - } - } - - FDK_ASSERT(startBand <= stopBand); - - if (!useLP) { - for (l=0; lLppTrans, - hHeaderData->freqBandData.lowSubband, - hHeaderData->freqBandData.v_k_master, - hHeaderData->freqBandData.numMaster, - hHeaderData->freqBandData.freqBandTableNoise, - hHeaderData->freqBandData.nNfb, - hHeaderData->freqBandData.highSubband, - hHeaderData->sbrProcSmplRate); - if (sbrError != SBRDEC_OK) - return sbrError; - - sbrError = ResetLimiterBands ( hHeaderData->freqBandData.limiterBandTable, - &hHeaderData->freqBandData.noLimiterBands, - hHeaderData->freqBandData.freqBandTable[0], - hHeaderData->freqBandData.nSfb[0], - hSbrDec->LppTrans.pSettings->patchParam, - hSbrDec->LppTrans.pSettings->noOfPatches, - hHeaderData->bs_data.limiterBands); - - - return sbrError; -} diff --git a/libSBRdec/src/sbr_dec.h b/libSBRdec/src/sbr_dec.h deleted file mode 100644 index edde637..0000000 --- a/libSBRdec/src/sbr_dec.h +++ /dev/null @@ -1,214 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief Sbr decoder -*/ -#ifndef __SBR_DEC_H -#define __SBR_DEC_H - -#include "sbrdecoder.h" - -#include "lpp_tran.h" -#include "qmf.h" -#include "env_calc.h" -#include "FDK_audio.h" - - -#include "sbrdec_drc.h" - -#define SACDEC_ALIGNMENT_FIX - -typedef struct -{ - QMF_FILTER_BANK AnalysiscQMF; - QMF_FILTER_BANK SynthesisQMF; - - SBR_CALCULATE_ENVELOPE SbrCalculateEnvelope; - SBR_LPP_TRANS LppTrans; - - QMF_SCALE_FACTOR sbrScaleFactor; - QMF_SCALE_FACTOR sbrScaleFactorRight; - - /*! Delayed spectral data needed for the dynamic framing of SBR. Not required in case of CLDFB */ - FIXP_DBL * pSbrOverlapBuffer; - - /* References to workbuffers */ - FIXP_DBL * WorkBuffer1; - FIXP_DBL * WorkBuffer2; - - /* Delayed time input signal needed to align CLDFD with LD-MPS QMF. */ - INT_PCM coreDelayBuf[(96)]; - - /* QMF filter states */ - FIXP_QAS anaQmfStates[(320)]; - FIXP_QSS * pSynQmfStates; - - /* Reference pointer arrays for QMF time slots, - mixed among overlap and current slots. */ - FIXP_DBL * QmfBufferReal[(((1024)/(32))+(6))]; - FIXP_DBL * QmfBufferImag[(((1024)/(32))+(6))]; - int useLP; - - /* QMF domain extension time slot reference pointer array */ - - SBRDEC_DRC_CHANNEL sbrDrcChannel; - -} SBR_DEC; - -typedef SBR_DEC *HANDLE_SBR_DEC; - - -typedef struct -{ - SBR_FRAME_DATA frameData[(1)+1]; - SBR_PREV_FRAME_DATA prevFrameData; - SBR_DEC SbrDec; -} -SBR_CHANNEL; - -typedef SBR_CHANNEL *HANDLE_SBR_CHANNEL; - -void -SbrDecodeAndProcess (HANDLE_SBR_DEC hSbrDec, - INT_PCM *timeIn, - HANDLE_SBR_HEADER_DATA hHeaderData, - HANDLE_SBR_FRAME_DATA hFrameData, - HANDLE_SBR_PREV_FRAME_DATA hPrevFrameData, - int applyProcessing, - int channelNr - , UCHAR useLP - ); - - -void -SbrConstructTimeOutput (HANDLE_SBR_DEC hSbrDec, /*!< handle to Decoder channel */ - INT_PCM *timeOut, /*!< pointer to output time signal */ - HANDLE_SBR_HEADER_DATA hHeaderData,/*!< Static control data */ - HANDLE_SBR_PREV_FRAME_DATA hPrevFrameData, /*!< Some control data of last frame */ - int channelNr - ,UCHAR useLP - ); - - -void -sbr_dec (HANDLE_SBR_DEC hSbrDec, /*!< handle to Decoder channel */ - INT_PCM *timeIn, /*!< pointer to input time signal */ - INT_PCM *timeOut, /*!< pointer to output time signal */ - HANDLE_SBR_DEC hSbrDecRight, /*!< handle to Decoder channel right */ - INT_PCM *timeOutRight, /*!< pointer to output time signal */ - const int strideIn, /*!< Time data traversal strideIn */ - const int strideOut, /*!< Time data traversal strideOut */ - HANDLE_SBR_HEADER_DATA hHeaderData,/*!< Static control data */ - HANDLE_SBR_FRAME_DATA hFrameData, /*!< Control data of current frame */ - HANDLE_SBR_PREV_FRAME_DATA hPrevFrameData, /*!< Some control data of last frame */ - const int applyProcessing, /*!< Flag for SBR operation */ - HANDLE_PS_DEC h_ps_d, - const UINT flags, - const int codecFrameSize - ); - - - -SBR_ERROR -createSbrDec (SBR_CHANNEL * hSbrChannel, - HANDLE_SBR_HEADER_DATA hHeaderData, - TRANSPOSER_SETTINGS *pSettings, - const int downsampleFac, - const UINT qmfFlags, - const UINT flags, - const int overlap, - int chan); - -int -deleteSbrDec (SBR_CHANNEL * hSbrChannel); - -SBR_ERROR -resetSbrDec (HANDLE_SBR_DEC hSbrDec, - HANDLE_SBR_HEADER_DATA hHeaderData, - HANDLE_SBR_PREV_FRAME_DATA hPrevFrameData, - const int useLP, - const int downsampleFac); - -#endif diff --git a/libSBRdec/src/sbr_ram.cpp b/libSBRdec/src/sbr_ram.cpp deleted file mode 100644 index c1c2499..0000000 --- a/libSBRdec/src/sbr_ram.cpp +++ /dev/null @@ -1,194 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief Memory layout - - - This module declares all static and dynamic memory spaces -*/ - -#include "sbr_ram.h" - - - - -#define WORKBUFFER1_TAG 0 -#define WORKBUFFER2_TAG 1 - -/*! - \name StaticSbrData - - Static memory areas, must not be overwritten in other sections of the decoder -*/ -/* @{ */ - -/*! SBR Decoder main structure */ -C_ALLOC_MEM(Ram_SbrDecoder, struct SBR_DECODER_INSTANCE, 1) -/*! SBR Decoder element data
- Dimension: (8) */ -C_ALLOC_MEM2(Ram_SbrDecElement, SBR_DECODER_ELEMENT, 1, (8)) -/*! SBR Decoder individual channel data
- Dimension: (8) */ -C_ALLOC_MEM2(Ram_SbrDecChannel, SBR_CHANNEL, 1, (8)+1) - -/*! Filter states for QMF-synthesis.
- Dimension: #(8) * (#QMF_FILTER_STATE_SYN_SIZE-#(64)) */ -C_AALLOC_MEM2_L(Ram_sbr_QmfStatesSynthesis, FIXP_QSS, (640)-(64), (8)+1, SECT_DATA_L1) - -/*! Delayed spectral data needed for the dynamic framing of SBR. - For mp3PRO, 1/3 of a frame is buffered (#(6) 6) */ -C_AALLOC_MEM2(Ram_sbr_OverlapBuffer, FIXP_DBL, 2 * (6) * (64), (8)+1) - -/*! Static Data of PS */ - -C_ALLOC_MEM(Ram_ps_dec, PS_DEC, 1) - - -/* @} */ - - -/*! - \name DynamicSbrData - - Dynamic memory areas, might be reused in other algorithm sections, - e.g. the core decoder -
- Depending on the mode set by DONT_USE_CORE_WORKBUFFER, workbuffers are - defined additionally to the CoreWorkbuffer. -
- The size of WorkBuffers is ((1024)/(32))*(64) = 2048. -
- WorkBuffer2 is a pointer to the CoreWorkBuffer wich is reused here in the SBR part. In case of - DONT_USE_CORE_WORKBUFFER, the CoreWorkbuffer is not used and the according - Workbuffer2 is defined locally in this file. -
- WorkBuffer1 is reused in the AAC core (-> aacdecoder.cpp, aac_ram.cpp) -
- - Use of WorkBuffers: - 
-
-    -------------------------------------------------------------
-    AAC core:
-
-      CoreWorkbuffer: spectral coefficients
-      WorkBuffer1:    CAacDecoderChannelInfo, CAacDecoderDynamicData
-
-    -------------------------------------------------------------
-    SBR part:
-      ----------------------------------------------
-      Low Power Mode (useLP=1 or LOW_POWER_SBR_ONLY), see assignLcTimeSlots()
-
-        SLOT_BASED_PROTOTYPE_SYN_FILTER
-
-        WorkBuffer1                                WorkBuffer2(=CoreWorkbuffer)
-         ________________                           ________________
-        | RealLeft       |                         | RealRight      |
-        |________________|                         |________________|
-
-      ----------------------------------------------
-      High Quality Mode (!LOW_POWER_SBR_ONLY and useLP=0), see assignHqTimeSlots()
-
-         SLOTBASED_PS
-
-         WorkBuffer1                                WorkBuffer2(=CoreWorkbuffer)
-         ________________                           ________________
-        | Real/Imag      |  interleaved            | Real/Imag      |  interleaved
-        |________________|  first half actual ch   |________________|  second half actual ch
-
-    -------------------------------------------------------------
-
-
- -*/ -/* @{ */ -C_ALLOC_MEM_OVERLAY(Ram_SbrDecWorkBuffer1, FIXP_DBL, ((1024)/(32))*(64), SECT_DATA_L1, WORKBUFFER1_TAG) -C_ALLOC_MEM_OVERLAY(Ram_SbrDecWorkBuffer2, FIXP_DBL, ((1024)/(32))*(64), SECT_DATA_L2, WORKBUFFER2_TAG) - -/* @} */ - - - - diff --git a/libSBRdec/src/sbr_ram.h b/libSBRdec/src/sbr_ram.h deleted file mode 100644 index 7ab5044..0000000 --- a/libSBRdec/src/sbr_ram.h +++ /dev/null @@ -1,159 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! -\file -\brief Memory layout - -*/ -#ifndef _SBR_RAM_H_ -#define _SBR_RAM_H_ - -#include "sbrdecoder.h" - -#include "env_extr.h" -#include "sbr_dec.h" - - - -#define SBRDEC_MAX_CH_PER_ELEMENT (2) - -typedef struct -{ - SBR_CHANNEL *pSbrChannel[SBRDEC_MAX_CH_PER_ELEMENT]; - TRANSPOSER_SETTINGS transposerSettings; /* Common transport settings for each individual channel of an element */ - HANDLE_FDK_BITSTREAM hBs; - - MP4_ELEMENT_ID elementID; /* Element ID set during initialization. Can be used for concealment */ - int nChannels; /* Number of elements output channels (=2 in case of PS) */ - - UCHAR frameErrorFlag[(1)+1]; /* Frame error status (for every slot in the delay line). - Will be copied into header at the very beginning of decodeElement() routine. */ - - UCHAR useFrameSlot; /* Index which defines which slot will be decoded/filled next (used with additional delay) */ - UCHAR useHeaderSlot[(1)+1]; /* Index array that provides the link between header and frame data - (important when processing with additional delay). */ -} SBR_DECODER_ELEMENT; - - -struct SBR_DECODER_INSTANCE -{ - SBR_DECODER_ELEMENT *pSbrElement[(8)]; - SBR_HEADER_DATA sbrHeader[(8)][(1)+1]; /* Sbr header for each individual channel of an element */ - - FIXP_DBL *workBuffer1; - FIXP_DBL *workBuffer2; - - HANDLE_PS_DEC hParametricStereoDec; - - /* Global parameters */ - AUDIO_OBJECT_TYPE coreCodec; /* AOT of core codec */ - int numSbrElements; - int numSbrChannels; - INT sampleRateIn; /* SBR decoder input sampling rate; might be different than the transposer input sampling rate. */ - INT sampleRateOut; /* Sampling rate of the SBR decoder output audio samples. */ - USHORT codecFrameSize; - UCHAR synDownsampleFac; - UCHAR numDelayFrames; /* The current number of additional delay frames used for processing. */ - UCHAR numFlushedFrames; /* The variable counts the number of frames which are flushed consecutively. */ - - UINT flags; - -}; - -H_ALLOC_MEM(Ram_SbrDecElement, SBR_DECODER_ELEMENT) -H_ALLOC_MEM(Ram_SbrDecChannel, SBR_CHANNEL) -H_ALLOC_MEM(Ram_SbrDecoder, struct SBR_DECODER_INSTANCE) - -H_ALLOC_MEM(Ram_sbr_QmfStatesSynthesis, FIXP_QSS) -H_ALLOC_MEM(Ram_sbr_OverlapBuffer, FIXP_DBL) - - -H_ALLOC_MEM(Ram_ps_dec, PS_DEC) - - -H_ALLOC_MEM_OVERLAY(Ram_SbrDecWorkBuffer1, FIXP_DBL) -H_ALLOC_MEM_OVERLAY(Ram_SbrDecWorkBuffer2, FIXP_DBL) - - -#endif /* _SBR_RAM_H_ */ diff --git a/libSBRdec/src/sbr_rom.cpp b/libSBRdec/src/sbr_rom.cpp deleted file mode 100644 index 4f2cc48..0000000 --- a/libSBRdec/src/sbr_rom.cpp +++ /dev/null @@ -1,1423 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief Definition of constant tables - - - This module contains most of the constant data that can be stored in ROM. -*/ - -#include "sbr_rom.h" - - - - -/*! - \name StartStopBands - \brief Start and stop subbands of the highband. - - k_o = startMin + offset[bs_start_freq]; - startMin = {3000,4000,5000} * (128/FS_sbr) / FS_sbr < 32Khz, 32Khz <= FS_sbr < 64KHz, 64KHz <= FS_sbr - The stop subband can also be calculated to save memory by defining #CALC_STOP_BAND. -*/ -//@{ -const UCHAR FDK_sbrDecoder_sbr_start_freq_16[16] = {16, 17, 18, 19, 20, 21, 22, 23, 24, 25, 26, 27, 28, 29, 30, 31}; -const UCHAR FDK_sbrDecoder_sbr_start_freq_22[16] = {12, 13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 23, 24, 26, 28, 30}; -const UCHAR FDK_sbrDecoder_sbr_start_freq_24[16] = {11, 13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 23, 25, 27, 29, 32}; -const UCHAR FDK_sbrDecoder_sbr_start_freq_32[16] = {10, 12, 14, 15, 16, 17, 18, 19, 20, 21, 22, 23, 25, 27, 29, 32}; -const UCHAR FDK_sbrDecoder_sbr_start_freq_40[16] = {12, 13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 24, 26, 28, 30, 32}; -const UCHAR FDK_sbrDecoder_sbr_start_freq_44[16] = { 8, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 21, 23, 25, 28, 32}; -const UCHAR FDK_sbrDecoder_sbr_start_freq_48[16] = { 7, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 20, 22, 24, 27, 31}; -const UCHAR FDK_sbrDecoder_sbr_start_freq_64[16] = { 6, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 19, 21, 23, 26, 30}; -const UCHAR FDK_sbrDecoder_sbr_start_freq_88[16] = { 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 16, 18, 20, 23, 27, 31}; -//@} - - -/*! - \name Whitening - \brief Coefficients for spectral whitening in the transposer -*/ -//@{ -/*! Assignment of whitening tuning depending on the crossover frequency */ -const USHORT FDK_sbrDecoder_sbr_whFactorsIndex[NUM_WHFACTOR_TABLE_ENTRIES] = { - 0, - 5000, - 6000, - 6500, - 7000, - 7500, - 8000, - 9000, - 10000 -}; - -/*! - \brief Whithening levels tuning table - - With the current tuning, there are some redundant entries: - - \li NUM_WHFACTOR_TABLE_ENTRIES can be reduced by 3, - \li the first coloumn can be eliminated. - -*/ -const FIXP_DBL FDK_sbrDecoder_sbr_whFactorsTable[NUM_WHFACTOR_TABLE_ENTRIES][6] = { - /* OFF_LEVEL, TRANSITION_LEVEL, LOW_LEVEL, MID_LEVEL, HIGH_LEVEL */ - { FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)}, /* < 5000 */ - { FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)}, /* 5000 < 6000 */ - { FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)}, /* 6000 < 6500 */ - { FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)}, /* 6500 < 7000 */ - { FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)}, /* 7000 < 7500 */ - { FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)}, /* 7500 < 8000 */ - { FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)}, /* 8000 < 9000 */ - { FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)}, /* 9000 < 10000 */ - { FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)}, /* > 10000 */ -}; - - -//@} - - -/*! - \name EnvAdj - \brief Constants and tables used for envelope adjustment -*/ -//@{ - -/*! Mantissas of gain limits */ -const FIXP_SGL FDK_sbrDecoder_sbr_limGains_m[4] = -{ - FL2FXCONST_SGL(0.5011932025f), /*!< -3 dB. Gain limit when limiterGains in frameData is 0 */ - FL2FXCONST_SGL(0.5f), /*!< 0 dB. Gain limit when limiterGains in frameData is 1 */ - FL2FXCONST_SGL(0.9976346258f), /*!< +3 dB. Gain limit when limiterGains in frameData is 2 */ - FL2FXCONST_SGL(0.6776263578f) /*!< Inf. Gain limit when limiterGains in frameData is 3 */ -}; - -/*! Exponents of gain limits */ -const UCHAR FDK_sbrDecoder_sbr_limGains_e[4] = -{ - 0, 1, 1, 67 -}; - -/*! Constants for calculating the number of limiter bands */ -const FIXP_SGL FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4[4] = -{ - FL2FXCONST_SGL(1.0f / 4.0f), - FL2FXCONST_SGL(1.2f / 4.0f), - FL2FXCONST_SGL(2.0f / 4.0f), - FL2FXCONST_SGL(3.0f / 4.0f) -}; - -/*! Constants for calculating the number of limiter bands */ -const FIXP_DBL FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4_DBL[4] = -{ - FL2FXCONST_DBL(1.0f / 4.0f), - FL2FXCONST_DBL(1.2f / 4.0f), - FL2FXCONST_DBL(2.0f / 4.0f), - FL2FXCONST_DBL(3.0f / 4.0f) -}; - -/*! Ratio of old gains and noise levels for the first 4 timeslots of an envelope */ -const FIXP_SGL FDK_sbrDecoder_sbr_smoothFilter[4] = { - FL2FXCONST_SGL(0.66666666666666f), - FL2FXCONST_SGL(0.36516383427084f), - FL2FXCONST_SGL(0.14699433520835f), - FL2FXCONST_SGL(0.03183050093751f) -}; - - -/*! Real and imaginary part of random noise which will be modulated - to the desired level. An accuracy of 13 bits is sufficient for these - random numbers. -*/ -const FIXP_SGL FDK_sbrDecoder_sbr_randomPhase[SBR_NF_NO_RANDOM_VAL][2] = { - { FL2FXCONST_SGL(-0.99948153278296f / 8.0), FL2FXCONST_SGL(-0.59483417516607f / 8.0) }, - { FL2FXCONST_SGL( 0.97113454393991f / 8.0), FL2FXCONST_SGL(-0.67528515225647f / 8.0) }, - { FL2FXCONST_SGL( 0.14130051758487f / 8.0), FL2FXCONST_SGL(-0.95090983575689f / 8.0) }, - { FL2FXCONST_SGL(-0.47005496701697f / 8.0), FL2FXCONST_SGL(-0.37340549728647f / 8.0) }, - { FL2FXCONST_SGL( 0.80705063769351f / 8.0), FL2FXCONST_SGL( 0.29653668284408f / 8.0) }, - { FL2FXCONST_SGL(-0.38981478896926f / 8.0), FL2FXCONST_SGL( 0.89572605717087f / 8.0) }, - { FL2FXCONST_SGL(-0.01053049862020f / 8.0), FL2FXCONST_SGL(-0.66959058036166f / 8.0) }, - { FL2FXCONST_SGL(-0.91266367957293f / 8.0), FL2FXCONST_SGL(-0.11522938140034f / 8.0) }, - { FL2FXCONST_SGL( 0.54840422910309f / 8.0), FL2FXCONST_SGL( 0.75221367176302f / 8.0) }, - { FL2FXCONST_SGL( 0.40009252867955f / 8.0), FL2FXCONST_SGL(-0.98929400334421f / 8.0) }, - { FL2FXCONST_SGL(-0.99867974711855f / 8.0), FL2FXCONST_SGL(-0.88147068645358f / 8.0) }, - { FL2FXCONST_SGL(-0.95531076805040f / 8.0), FL2FXCONST_SGL( 0.90908757154593f / 8.0) }, - { FL2FXCONST_SGL(-0.45725933317144f / 8.0), FL2FXCONST_SGL(-0.56716323646760f / 8.0) }, - { FL2FXCONST_SGL(-0.72929675029275f / 8.0), FL2FXCONST_SGL(-0.98008272727324f / 8.0) }, - { FL2FXCONST_SGL( 0.75622801399036f / 8.0), FL2FXCONST_SGL( 0.20950329995549f / 8.0) }, - { FL2FXCONST_SGL( 0.07069442601050f / 8.0), FL2FXCONST_SGL(-0.78247898470706f / 8.0) }, - { FL2FXCONST_SGL( 0.74496252926055f / 8.0), FL2FXCONST_SGL(-0.91169004445807f / 8.0) }, - { FL2FXCONST_SGL(-0.96440182703856f / 8.0), FL2FXCONST_SGL(-0.94739918296622f / 8.0) }, - { FL2FXCONST_SGL( 0.30424629369539f / 8.0), FL2FXCONST_SGL(-0.49438267012479f / 8.0) }, - { FL2FXCONST_SGL( 0.66565033746925f / 8.0), FL2FXCONST_SGL( 0.64652935542491f / 8.0) }, - { FL2FXCONST_SGL( 0.91697008020594f / 8.0), FL2FXCONST_SGL( 0.17514097332009f / 8.0) }, - { FL2FXCONST_SGL(-0.70774918760427f / 8.0), FL2FXCONST_SGL( 0.52548653416543f / 8.0) }, - { FL2FXCONST_SGL(-0.70051415345560f / 8.0), FL2FXCONST_SGL(-0.45340028808763f / 8.0) }, - { FL2FXCONST_SGL(-0.99496513054797f / 8.0), FL2FXCONST_SGL(-0.90071908066973f / 8.0) }, - { FL2FXCONST_SGL( 0.98164490790123f / 8.0), FL2FXCONST_SGL(-0.77463155528697f / 8.0) }, - { FL2FXCONST_SGL(-0.54671580548181f / 8.0), FL2FXCONST_SGL(-0.02570928536004f / 8.0) }, - { FL2FXCONST_SGL(-0.01689629065389f / 8.0), FL2FXCONST_SGL( 0.00287506445732f / 8.0) }, - { FL2FXCONST_SGL(-0.86110349531986f / 8.0), FL2FXCONST_SGL( 0.42548583726477f / 8.0) }, - { FL2FXCONST_SGL(-0.98892980586032f / 8.0), FL2FXCONST_SGL(-0.87881132267556f / 8.0) }, - { FL2FXCONST_SGL( 0.51756627678691f / 8.0), FL2FXCONST_SGL( 0.66926784710139f / 8.0) }, - { FL2FXCONST_SGL(-0.99635026409640f / 8.0), FL2FXCONST_SGL(-0.58107730574765f / 8.0) }, - { FL2FXCONST_SGL(-0.99969370862163f / 8.0), FL2FXCONST_SGL( 0.98369989360250f / 8.0) }, - { FL2FXCONST_SGL( 0.55266258627194f / 8.0), FL2FXCONST_SGL( 0.59449057465591f / 8.0) }, - { FL2FXCONST_SGL( 0.34581177741673f / 8.0), FL2FXCONST_SGL( 0.94879421061866f / 8.0) }, - { FL2FXCONST_SGL( 0.62664209577999f / 8.0), FL2FXCONST_SGL(-0.74402970906471f / 8.0) }, - { FL2FXCONST_SGL(-0.77149701404973f / 8.0), FL2FXCONST_SGL(-0.33883658042801f / 8.0) }, - { FL2FXCONST_SGL(-0.91592244254432f / 8.0), FL2FXCONST_SGL( 0.03687901376713f / 8.0) }, - { FL2FXCONST_SGL(-0.76285492357887f / 8.0), FL2FXCONST_SGL(-0.91371867919124f / 8.0) }, - { FL2FXCONST_SGL( 0.79788337195331f / 8.0), FL2FXCONST_SGL(-0.93180971199849f / 8.0) }, - { FL2FXCONST_SGL( 0.54473080610200f / 8.0), FL2FXCONST_SGL(-0.11919206037186f / 8.0) }, - { FL2FXCONST_SGL(-0.85639281671058f / 8.0), FL2FXCONST_SGL( 0.42429854760451f / 8.0) }, - { FL2FXCONST_SGL(-0.92882402971423f / 8.0), FL2FXCONST_SGL( 0.27871809078609f / 8.0) }, - { FL2FXCONST_SGL(-0.11708371046774f / 8.0), FL2FXCONST_SGL(-0.99800843444966f / 8.0) }, - { FL2FXCONST_SGL( 0.21356749817493f / 8.0), FL2FXCONST_SGL(-0.90716295627033f / 8.0) }, - { FL2FXCONST_SGL(-0.76191692573909f / 8.0), FL2FXCONST_SGL( 0.99768118356265f / 8.0) }, - { FL2FXCONST_SGL( 0.98111043100884f / 8.0), FL2FXCONST_SGL(-0.95854459734407f / 8.0) }, - { FL2FXCONST_SGL(-0.85913269895572f / 8.0), FL2FXCONST_SGL( 0.95766566168880f / 8.0) }, - { FL2FXCONST_SGL(-0.93307242253692f / 8.0), FL2FXCONST_SGL( 0.49431757696466f / 8.0) }, - { FL2FXCONST_SGL( 0.30485754879632f / 8.0), FL2FXCONST_SGL(-0.70540034357529f / 8.0) }, - { FL2FXCONST_SGL( 0.85289650925190f / 8.0), FL2FXCONST_SGL( 0.46766131791044f / 8.0) }, - { FL2FXCONST_SGL( 0.91328082618125f / 8.0), FL2FXCONST_SGL(-0.99839597361769f / 8.0) }, - { FL2FXCONST_SGL(-0.05890199924154f / 8.0), FL2FXCONST_SGL( 0.70741827819497f / 8.0) }, - { FL2FXCONST_SGL( 0.28398686150148f / 8.0), FL2FXCONST_SGL( 0.34633555702188f / 8.0) }, - { FL2FXCONST_SGL( 0.95258164539612f / 8.0), FL2FXCONST_SGL(-0.54893416026939f / 8.0) }, - { FL2FXCONST_SGL(-0.78566324168507f / 8.0), FL2FXCONST_SGL(-0.75568541079691f / 8.0) }, - { FL2FXCONST_SGL(-0.95789495447877f / 8.0), FL2FXCONST_SGL(-0.20423194696966f / 8.0) }, - { FL2FXCONST_SGL( 0.82411158711197f / 8.0), FL2FXCONST_SGL( 0.96654618432562f / 8.0) }, - { FL2FXCONST_SGL(-0.65185446735885f / 8.0), FL2FXCONST_SGL(-0.88734990773289f / 8.0) }, - { FL2FXCONST_SGL(-0.93643603134666f / 8.0), FL2FXCONST_SGL( 0.99870790442385f / 8.0) }, - { FL2FXCONST_SGL( 0.91427159529618f / 8.0), FL2FXCONST_SGL(-0.98290505544444f / 8.0) }, - { FL2FXCONST_SGL(-0.70395684036886f / 8.0), FL2FXCONST_SGL( 0.58796798221039f / 8.0) }, - { FL2FXCONST_SGL( 0.00563771969365f / 8.0), FL2FXCONST_SGL( 0.61768196727244f / 8.0) }, - { FL2FXCONST_SGL( 0.89065051931895f / 8.0), FL2FXCONST_SGL( 0.52783352697585f / 8.0) }, - { FL2FXCONST_SGL(-0.68683707712762f / 8.0), FL2FXCONST_SGL( 0.80806944710339f / 8.0) }, - { FL2FXCONST_SGL( 0.72165342518718f / 8.0), FL2FXCONST_SGL(-0.69259857349564f / 8.0) }, - { FL2FXCONST_SGL(-0.62928247730667f / 8.0), FL2FXCONST_SGL( 0.13627037407335f / 8.0) }, - { FL2FXCONST_SGL( 0.29938434065514f / 8.0), FL2FXCONST_SGL(-0.46051329682246f / 8.0) }, - { FL2FXCONST_SGL(-0.91781958879280f / 8.0), FL2FXCONST_SGL(-0.74012716684186f / 8.0) }, - { FL2FXCONST_SGL( 0.99298717043688f / 8.0), FL2FXCONST_SGL( 0.40816610075661f / 8.0) }, - { FL2FXCONST_SGL( 0.82368298622748f / 8.0), FL2FXCONST_SGL(-0.74036047190173f / 8.0) }, - { FL2FXCONST_SGL(-0.98512833386833f / 8.0), FL2FXCONST_SGL(-0.99972330709594f / 8.0) }, - { FL2FXCONST_SGL(-0.95915368242257f / 8.0), FL2FXCONST_SGL(-0.99237800466040f / 8.0) }, - { FL2FXCONST_SGL(-0.21411126572790f / 8.0), FL2FXCONST_SGL(-0.93424819052545f / 8.0) }, - { FL2FXCONST_SGL(-0.68821476106884f / 8.0), FL2FXCONST_SGL(-0.26892306315457f / 8.0) }, - { FL2FXCONST_SGL( 0.91851997982317f / 8.0), FL2FXCONST_SGL( 0.09358228901785f / 8.0) }, - { FL2FXCONST_SGL(-0.96062769559127f / 8.0), FL2FXCONST_SGL( 0.36099095133739f / 8.0) }, - { FL2FXCONST_SGL( 0.51646184922287f / 8.0), FL2FXCONST_SGL(-0.71373332873917f / 8.0) }, - { FL2FXCONST_SGL( 0.61130721139669f / 8.0), FL2FXCONST_SGL( 0.46950141175917f / 8.0) }, - { FL2FXCONST_SGL( 0.47336129371299f / 8.0), FL2FXCONST_SGL(-0.27333178296162f / 8.0) }, - { FL2FXCONST_SGL( 0.90998308703519f / 8.0), FL2FXCONST_SGL( 0.96715662938132f / 8.0) }, - { FL2FXCONST_SGL( 0.44844799194357f / 8.0), FL2FXCONST_SGL( 0.99211574628306f / 8.0) }, - { FL2FXCONST_SGL( 0.66614891079092f / 8.0), FL2FXCONST_SGL( 0.96590176169121f / 8.0) }, - { FL2FXCONST_SGL( 0.74922239129237f / 8.0), FL2FXCONST_SGL(-0.89879858826087f / 8.0) }, - { FL2FXCONST_SGL(-0.99571588506485f / 8.0), FL2FXCONST_SGL( 0.52785521494349f / 8.0) }, - { FL2FXCONST_SGL( 0.97401082477563f / 8.0), FL2FXCONST_SGL(-0.16855870075190f / 8.0) }, - { FL2FXCONST_SGL( 0.72683747733879f / 8.0), FL2FXCONST_SGL(-0.48060774432251f / 8.0) }, - { FL2FXCONST_SGL( 0.95432193457128f / 8.0), FL2FXCONST_SGL( 0.68849603408441f / 8.0) }, - { FL2FXCONST_SGL(-0.72962208425191f / 8.0), FL2FXCONST_SGL(-0.76608443420917f / 8.0) }, - { FL2FXCONST_SGL(-0.85359479233537f / 8.0), FL2FXCONST_SGL( 0.88738125901579f / 8.0) }, - { FL2FXCONST_SGL(-0.81412430338535f / 8.0), FL2FXCONST_SGL(-0.97480768049637f / 8.0) }, - { FL2FXCONST_SGL(-0.87930772356786f / 8.0), FL2FXCONST_SGL( 0.74748307690436f / 8.0) }, - { FL2FXCONST_SGL(-0.71573331064977f / 8.0), FL2FXCONST_SGL(-0.98570608178923f / 8.0) }, - { FL2FXCONST_SGL( 0.83524300028228f / 8.0), FL2FXCONST_SGL( 0.83702537075163f / 8.0) }, - { FL2FXCONST_SGL(-0.48086065601423f / 8.0), FL2FXCONST_SGL(-0.98848504923531f / 8.0) }, - { FL2FXCONST_SGL( 0.97139128574778f / 8.0), FL2FXCONST_SGL( 0.80093621198236f / 8.0) }, - { FL2FXCONST_SGL( 0.51992825347895f / 8.0), FL2FXCONST_SGL( 0.80247631400510f / 8.0) }, - { FL2FXCONST_SGL(-0.00848591195325f / 8.0), FL2FXCONST_SGL(-0.76670128000486f / 8.0) }, - { FL2FXCONST_SGL(-0.70294374303036f / 8.0), FL2FXCONST_SGL( 0.55359910445577f / 8.0) }, - { FL2FXCONST_SGL(-0.95894428168140f / 8.0), FL2FXCONST_SGL(-0.43265504344783f / 8.0) }, - { FL2FXCONST_SGL( 0.97079252950321f / 8.0), FL2FXCONST_SGL( 0.09325857238682f / 8.0) }, - { FL2FXCONST_SGL(-0.92404293670797f / 8.0), FL2FXCONST_SGL( 0.85507704027855f / 8.0) }, - { FL2FXCONST_SGL(-0.69506469500450f / 8.0), FL2FXCONST_SGL( 0.98633412625459f / 8.0) }, - { FL2FXCONST_SGL( 0.26559203620024f / 8.0), FL2FXCONST_SGL( 0.73314307966524f / 8.0) }, - { FL2FXCONST_SGL( 0.28038443336943f / 8.0), FL2FXCONST_SGL( 0.14537913654427f / 8.0) }, - { FL2FXCONST_SGL(-0.74138124825523f / 8.0), FL2FXCONST_SGL( 0.99310339807762f / 8.0) }, - { FL2FXCONST_SGL(-0.01752795995444f / 8.0), FL2FXCONST_SGL(-0.82616635284178f / 8.0) }, - { FL2FXCONST_SGL(-0.55126773094930f / 8.0), FL2FXCONST_SGL(-0.98898543862153f / 8.0) }, - { FL2FXCONST_SGL( 0.97960898850996f / 8.0), FL2FXCONST_SGL(-0.94021446752851f / 8.0) }, - { FL2FXCONST_SGL(-0.99196309146936f / 8.0), FL2FXCONST_SGL( 0.67019017358456f / 8.0) }, - { FL2FXCONST_SGL(-0.67684928085260f / 8.0), FL2FXCONST_SGL( 0.12631491649378f / 8.0) }, - { FL2FXCONST_SGL( 0.09140039465500f / 8.0), FL2FXCONST_SGL(-0.20537731453108f / 8.0) }, - { FL2FXCONST_SGL(-0.71658965751996f / 8.0), FL2FXCONST_SGL(-0.97788200391224f / 8.0) }, - { FL2FXCONST_SGL( 0.81014640078925f / 8.0), FL2FXCONST_SGL( 0.53722648362443f / 8.0) }, - { FL2FXCONST_SGL( 0.40616991671205f / 8.0), FL2FXCONST_SGL(-0.26469008598449f / 8.0) }, - { FL2FXCONST_SGL(-0.67680188682972f / 8.0), FL2FXCONST_SGL( 0.94502052337695f / 8.0) }, - { FL2FXCONST_SGL( 0.86849774348749f / 8.0), FL2FXCONST_SGL(-0.18333598647899f / 8.0) }, - { FL2FXCONST_SGL(-0.99500381284851f / 8.0), FL2FXCONST_SGL(-0.02634122068550f / 8.0) }, - { FL2FXCONST_SGL( 0.84329189340667f / 8.0), FL2FXCONST_SGL( 0.10406957462213f / 8.0) }, - { FL2FXCONST_SGL(-0.09215968531446f / 8.0), FL2FXCONST_SGL( 0.69540012101253f / 8.0) }, - { FL2FXCONST_SGL( 0.99956173327206f / 8.0), FL2FXCONST_SGL(-0.12358542001404f / 8.0) }, - { FL2FXCONST_SGL(-0.79732779473535f / 8.0), FL2FXCONST_SGL(-0.91582524736159f / 8.0) }, - { FL2FXCONST_SGL( 0.96349973642406f / 8.0), FL2FXCONST_SGL( 0.96640458041000f / 8.0) }, - { FL2FXCONST_SGL(-0.79942778496547f / 8.0), FL2FXCONST_SGL( 0.64323902822857f / 8.0) }, - { FL2FXCONST_SGL(-0.11566039853896f / 8.0), FL2FXCONST_SGL( 0.28587846253726f / 8.0) }, - { FL2FXCONST_SGL(-0.39922954514662f / 8.0), FL2FXCONST_SGL( 0.94129601616966f / 8.0) }, - { FL2FXCONST_SGL( 0.99089197565987f / 8.0), FL2FXCONST_SGL(-0.92062625581587f / 8.0) }, - { FL2FXCONST_SGL( 0.28631285179909f / 8.0), FL2FXCONST_SGL(-0.91035047143603f / 8.0) }, - { FL2FXCONST_SGL(-0.83302725605608f / 8.0), FL2FXCONST_SGL(-0.67330410892084f / 8.0) }, - { FL2FXCONST_SGL( 0.95404443402072f / 8.0), FL2FXCONST_SGL( 0.49162765398743f / 8.0) }, - { FL2FXCONST_SGL(-0.06449863579434f / 8.0), FL2FXCONST_SGL( 0.03250560813135f / 8.0) }, - { FL2FXCONST_SGL(-0.99575054486311f / 8.0), FL2FXCONST_SGL( 0.42389784469507f / 8.0) }, - { FL2FXCONST_SGL(-0.65501142790847f / 8.0), FL2FXCONST_SGL( 0.82546114655624f / 8.0) }, - { FL2FXCONST_SGL(-0.81254441908887f / 8.0), FL2FXCONST_SGL(-0.51627234660629f / 8.0) }, - { FL2FXCONST_SGL(-0.99646369485481f / 8.0), FL2FXCONST_SGL( 0.84490533520752f / 8.0) }, - { FL2FXCONST_SGL( 0.00287840603348f / 8.0), FL2FXCONST_SGL( 0.64768261158166f / 8.0) }, - { FL2FXCONST_SGL( 0.70176989408455f / 8.0), FL2FXCONST_SGL(-0.20453028573322f / 8.0) }, - { FL2FXCONST_SGL( 0.96361882270190f / 8.0), FL2FXCONST_SGL( 0.40706967140989f / 8.0) }, - { FL2FXCONST_SGL(-0.68883758192426f / 8.0), FL2FXCONST_SGL( 0.91338958840772f / 8.0) }, - { FL2FXCONST_SGL(-0.34875585502238f / 8.0), FL2FXCONST_SGL( 0.71472290693300f / 8.0) }, - { FL2FXCONST_SGL( 0.91980081243087f / 8.0), FL2FXCONST_SGL( 0.66507455644919f / 8.0) }, - { FL2FXCONST_SGL(-0.99009048343881f / 8.0), FL2FXCONST_SGL( 0.85868021604848f / 8.0) }, - { FL2FXCONST_SGL( 0.68865791458395f / 8.0), FL2FXCONST_SGL( 0.55660316809678f / 8.0) }, - { FL2FXCONST_SGL(-0.99484402129368f / 8.0), FL2FXCONST_SGL(-0.20052559254934f / 8.0) }, - { FL2FXCONST_SGL( 0.94214511408023f / 8.0), FL2FXCONST_SGL(-0.99696425367461f / 8.0) }, - { FL2FXCONST_SGL(-0.67414626793544f / 8.0), FL2FXCONST_SGL( 0.49548221180078f / 8.0) }, - { FL2FXCONST_SGL(-0.47339353684664f / 8.0), FL2FXCONST_SGL(-0.85904328834047f / 8.0) }, - { FL2FXCONST_SGL( 0.14323651387360f / 8.0), FL2FXCONST_SGL(-0.94145598222488f / 8.0) }, - { FL2FXCONST_SGL(-0.29268293575672f / 8.0), FL2FXCONST_SGL( 0.05759224927952f / 8.0) }, - { FL2FXCONST_SGL( 0.43793861458754f / 8.0), FL2FXCONST_SGL(-0.78904969892724f / 8.0) }, - { FL2FXCONST_SGL(-0.36345126374441f / 8.0), FL2FXCONST_SGL( 0.64874435357162f / 8.0) }, - { FL2FXCONST_SGL(-0.08750604656825f / 8.0), FL2FXCONST_SGL( 0.97686944362527f / 8.0) }, - { FL2FXCONST_SGL(-0.96495267812511f / 8.0), FL2FXCONST_SGL(-0.53960305946511f / 8.0) }, - { FL2FXCONST_SGL( 0.55526940659947f / 8.0), FL2FXCONST_SGL( 0.78891523734774f / 8.0) }, - { FL2FXCONST_SGL( 0.73538215752630f / 8.0), FL2FXCONST_SGL( 0.96452072373404f / 8.0) }, - { FL2FXCONST_SGL(-0.30889773919437f / 8.0), FL2FXCONST_SGL(-0.80664389776860f / 8.0) }, - { FL2FXCONST_SGL( 0.03574995626194f / 8.0), FL2FXCONST_SGL(-0.97325616900959f / 8.0) }, - { FL2FXCONST_SGL( 0.98720684660488f / 8.0), FL2FXCONST_SGL( 0.48409133691962f / 8.0) }, - { FL2FXCONST_SGL(-0.81689296271203f / 8.0), FL2FXCONST_SGL(-0.90827703628298f / 8.0) }, - { FL2FXCONST_SGL( 0.67866860118215f / 8.0), FL2FXCONST_SGL( 0.81284503870856f / 8.0) }, - { FL2FXCONST_SGL(-0.15808569732583f / 8.0), FL2FXCONST_SGL( 0.85279555024382f / 8.0) }, - { FL2FXCONST_SGL( 0.80723395114371f / 8.0), FL2FXCONST_SGL(-0.24717418514605f / 8.0) }, - { FL2FXCONST_SGL( 0.47788757329038f / 8.0), FL2FXCONST_SGL(-0.46333147839295f / 8.0) }, - { FL2FXCONST_SGL( 0.96367554763201f / 8.0), FL2FXCONST_SGL( 0.38486749303242f / 8.0) }, - { FL2FXCONST_SGL(-0.99143875716818f / 8.0), FL2FXCONST_SGL(-0.24945277239809f / 8.0) }, - { FL2FXCONST_SGL( 0.83081876925833f / 8.0), FL2FXCONST_SGL(-0.94780851414763f / 8.0) }, - { FL2FXCONST_SGL(-0.58753191905341f / 8.0), FL2FXCONST_SGL( 0.01290772389163f / 8.0) }, - { FL2FXCONST_SGL( 0.95538108220960f / 8.0), FL2FXCONST_SGL(-0.85557052096538f / 8.0) }, - { FL2FXCONST_SGL(-0.96490920476211f / 8.0), FL2FXCONST_SGL(-0.64020970923102f / 8.0) }, - { FL2FXCONST_SGL(-0.97327101028521f / 8.0), FL2FXCONST_SGL( 0.12378128133110f / 8.0) }, - { FL2FXCONST_SGL( 0.91400366022124f / 8.0), FL2FXCONST_SGL( 0.57972471346930f / 8.0) }, - { FL2FXCONST_SGL(-0.99925837363824f / 8.0), FL2FXCONST_SGL( 0.71084847864067f / 8.0) }, - { FL2FXCONST_SGL(-0.86875903507313f / 8.0), FL2FXCONST_SGL(-0.20291699203564f / 8.0) }, - { FL2FXCONST_SGL(-0.26240034795124f / 8.0), FL2FXCONST_SGL(-0.68264554369108f / 8.0) }, - { FL2FXCONST_SGL(-0.24664412953388f / 8.0), FL2FXCONST_SGL(-0.87642273115183f / 8.0) }, - { FL2FXCONST_SGL( 0.02416275806869f / 8.0), FL2FXCONST_SGL( 0.27192914288905f / 8.0) }, - { FL2FXCONST_SGL( 0.82068619590515f / 8.0), FL2FXCONST_SGL(-0.85087787994476f / 8.0) }, - { FL2FXCONST_SGL( 0.88547373760759f / 8.0), FL2FXCONST_SGL(-0.89636802901469f / 8.0) }, - { FL2FXCONST_SGL(-0.18173078152226f / 8.0), FL2FXCONST_SGL(-0.26152145156800f / 8.0) }, - { FL2FXCONST_SGL( 0.09355476558534f / 8.0), FL2FXCONST_SGL( 0.54845123045604f / 8.0) }, - { FL2FXCONST_SGL(-0.54668414224090f / 8.0), FL2FXCONST_SGL( 0.95980774020221f / 8.0) }, - { FL2FXCONST_SGL( 0.37050990604091f / 8.0), FL2FXCONST_SGL(-0.59910140383171f / 8.0) }, - { FL2FXCONST_SGL(-0.70373594262891f / 8.0), FL2FXCONST_SGL( 0.91227665827081f / 8.0) }, - { FL2FXCONST_SGL(-0.34600785879594f / 8.0), FL2FXCONST_SGL(-0.99441426144200f / 8.0) }, - { FL2FXCONST_SGL(-0.68774481731008f / 8.0), FL2FXCONST_SGL(-0.30238837956299f / 8.0) }, - { FL2FXCONST_SGL(-0.26843291251234f / 8.0), FL2FXCONST_SGL( 0.83115668004362f / 8.0) }, - { FL2FXCONST_SGL( 0.49072334613242f / 8.0), FL2FXCONST_SGL(-0.45359708737775f / 8.0) }, - { FL2FXCONST_SGL( 0.38975993093975f / 8.0), FL2FXCONST_SGL( 0.95515358099121f / 8.0) }, - { FL2FXCONST_SGL(-0.97757125224150f / 8.0), FL2FXCONST_SGL( 0.05305894580606f / 8.0) }, - { FL2FXCONST_SGL(-0.17325552859616f / 8.0), FL2FXCONST_SGL(-0.92770672250494f / 8.0) }, - { FL2FXCONST_SGL( 0.99948035025744f / 8.0), FL2FXCONST_SGL( 0.58285545563426f / 8.0) }, - { FL2FXCONST_SGL(-0.64946246527458f / 8.0), FL2FXCONST_SGL( 0.68645507104960f / 8.0) }, - { FL2FXCONST_SGL(-0.12016920576437f / 8.0), FL2FXCONST_SGL(-0.57147322153312f / 8.0) }, - { FL2FXCONST_SGL(-0.58947456517751f / 8.0), FL2FXCONST_SGL(-0.34847132454388f / 8.0) }, - { FL2FXCONST_SGL(-0.41815140454465f / 8.0), FL2FXCONST_SGL( 0.16276422358861f / 8.0) }, - { FL2FXCONST_SGL( 0.99885650204884f / 8.0), FL2FXCONST_SGL( 0.11136095490444f / 8.0) }, - { FL2FXCONST_SGL(-0.56649614128386f / 8.0), FL2FXCONST_SGL(-0.90494866361587f / 8.0) }, - { FL2FXCONST_SGL( 0.94138021032330f / 8.0), FL2FXCONST_SGL( 0.35281916733018f / 8.0) }, - { FL2FXCONST_SGL(-0.75725076534641f / 8.0), FL2FXCONST_SGL( 0.53650549640587f / 8.0) }, - { FL2FXCONST_SGL( 0.20541973692630f / 8.0), FL2FXCONST_SGL(-0.94435144369918f / 8.0) }, - { FL2FXCONST_SGL( 0.99980371023351f / 8.0), FL2FXCONST_SGL( 0.79835913565599f / 8.0) }, - { FL2FXCONST_SGL( 0.29078277605775f / 8.0), FL2FXCONST_SGL( 0.35393777921520f / 8.0) }, - { FL2FXCONST_SGL(-0.62858772103030f / 8.0), FL2FXCONST_SGL( 0.38765693387102f / 8.0) }, - { FL2FXCONST_SGL( 0.43440904467688f / 8.0), FL2FXCONST_SGL(-0.98546330463232f / 8.0) }, - { FL2FXCONST_SGL(-0.98298583762390f / 8.0), FL2FXCONST_SGL( 0.21021524625209f / 8.0) }, - { FL2FXCONST_SGL( 0.19513029146934f / 8.0), FL2FXCONST_SGL(-0.94239832251867f / 8.0) }, - { FL2FXCONST_SGL(-0.95476662400101f / 8.0), FL2FXCONST_SGL( 0.98364554179143f / 8.0) }, - { FL2FXCONST_SGL( 0.93379635304810f / 8.0), FL2FXCONST_SGL(-0.70881994583682f / 8.0) }, - { FL2FXCONST_SGL(-0.85235410573336f / 8.0), FL2FXCONST_SGL(-0.08342347966410f / 8.0) }, - { FL2FXCONST_SGL(-0.86425093011245f / 8.0), FL2FXCONST_SGL(-0.45795025029466f / 8.0) }, - { FL2FXCONST_SGL( 0.38879779059045f / 8.0), FL2FXCONST_SGL( 0.97274429344593f / 8.0) }, - { FL2FXCONST_SGL( 0.92045124735495f / 8.0), FL2FXCONST_SGL(-0.62433652524220f / 8.0) }, - { FL2FXCONST_SGL( 0.89162532251878f / 8.0), FL2FXCONST_SGL( 0.54950955570563f / 8.0) }, - { FL2FXCONST_SGL(-0.36834336949252f / 8.0), FL2FXCONST_SGL( 0.96458298020975f / 8.0) }, - { FL2FXCONST_SGL( 0.93891760988045f / 8.0), FL2FXCONST_SGL(-0.89968353740388f / 8.0) }, - { FL2FXCONST_SGL( 0.99267657565094f / 8.0), FL2FXCONST_SGL(-0.03757034316958f / 8.0) }, - { FL2FXCONST_SGL(-0.94063471614176f / 8.0), FL2FXCONST_SGL( 0.41332338538963f / 8.0) }, - { FL2FXCONST_SGL( 0.99740224117019f / 8.0), FL2FXCONST_SGL(-0.16830494996370f / 8.0) }, - { FL2FXCONST_SGL(-0.35899413170555f / 8.0), FL2FXCONST_SGL(-0.46633226649613f / 8.0) }, - { FL2FXCONST_SGL( 0.05237237274947f / 8.0), FL2FXCONST_SGL(-0.25640361602661f / 8.0) }, - { FL2FXCONST_SGL( 0.36703583957424f / 8.0), FL2FXCONST_SGL(-0.38653265641875f / 8.0) }, - { FL2FXCONST_SGL( 0.91653180367913f / 8.0), FL2FXCONST_SGL(-0.30587628726597f / 8.0) }, - { FL2FXCONST_SGL( 0.69000803499316f / 8.0), FL2FXCONST_SGL( 0.90952171386132f / 8.0) }, - { FL2FXCONST_SGL(-0.38658751133527f / 8.0), FL2FXCONST_SGL( 0.99501571208985f / 8.0) }, - { FL2FXCONST_SGL(-0.29250814029851f / 8.0), FL2FXCONST_SGL( 0.37444994344615f / 8.0) }, - { FL2FXCONST_SGL(-0.60182204677608f / 8.0), FL2FXCONST_SGL( 0.86779651036123f / 8.0) }, - { FL2FXCONST_SGL(-0.97418588163217f / 8.0), FL2FXCONST_SGL( 0.96468523666475f / 8.0) }, - { FL2FXCONST_SGL( 0.88461574003963f / 8.0), FL2FXCONST_SGL( 0.57508405276414f / 8.0) }, - { FL2FXCONST_SGL( 0.05198933055162f / 8.0), FL2FXCONST_SGL( 0.21269661669964f / 8.0) }, - { FL2FXCONST_SGL(-0.53499621979720f / 8.0), FL2FXCONST_SGL( 0.97241553731237f / 8.0) }, - { FL2FXCONST_SGL(-0.49429560226497f / 8.0), FL2FXCONST_SGL( 0.98183865291903f / 8.0) }, - { FL2FXCONST_SGL(-0.98935142339139f / 8.0), FL2FXCONST_SGL(-0.40249159006933f / 8.0) }, - { FL2FXCONST_SGL(-0.98081380091130f / 8.0), FL2FXCONST_SGL(-0.72856895534041f / 8.0) }, - { FL2FXCONST_SGL(-0.27338148835532f / 8.0), FL2FXCONST_SGL( 0.99950922447209f / 8.0) }, - { FL2FXCONST_SGL( 0.06310802338302f / 8.0), FL2FXCONST_SGL(-0.54539587529618f / 8.0) }, - { FL2FXCONST_SGL(-0.20461677199539f / 8.0), FL2FXCONST_SGL(-0.14209977628489f / 8.0) }, - { FL2FXCONST_SGL( 0.66223843141647f / 8.0), FL2FXCONST_SGL( 0.72528579940326f / 8.0) }, - { FL2FXCONST_SGL(-0.84764345483665f / 8.0), FL2FXCONST_SGL( 0.02372316801261f / 8.0) }, - { FL2FXCONST_SGL(-0.89039863483811f / 8.0), FL2FXCONST_SGL( 0.88866581484602f / 8.0) }, - { FL2FXCONST_SGL( 0.95903308477986f / 8.0), FL2FXCONST_SGL( 0.76744927173873f / 8.0) }, - { FL2FXCONST_SGL( 0.73504123909879f / 8.0), FL2FXCONST_SGL(-0.03747203173192f / 8.0) }, - { FL2FXCONST_SGL(-0.31744434966056f / 8.0), FL2FXCONST_SGL(-0.36834111883652f / 8.0) }, - { FL2FXCONST_SGL(-0.34110827591623f / 8.0), FL2FXCONST_SGL( 0.40211222807691f / 8.0) }, - { FL2FXCONST_SGL( 0.47803883714199f / 8.0), FL2FXCONST_SGL(-0.39423219786288f / 8.0) }, - { FL2FXCONST_SGL( 0.98299195879514f / 8.0), FL2FXCONST_SGL( 0.01989791390047f / 8.0) }, - { FL2FXCONST_SGL(-0.30963073129751f / 8.0), FL2FXCONST_SGL(-0.18076720599336f / 8.0) }, - { FL2FXCONST_SGL( 0.99992588229018f / 8.0), FL2FXCONST_SGL(-0.26281872094289f / 8.0) }, - { FL2FXCONST_SGL(-0.93149731080767f / 8.0), FL2FXCONST_SGL(-0.98313162570490f / 8.0) }, - { FL2FXCONST_SGL( 0.99923472302773f / 8.0), FL2FXCONST_SGL(-0.80142993767554f / 8.0) }, - { FL2FXCONST_SGL(-0.26024169633417f / 8.0), FL2FXCONST_SGL(-0.75999759855752f / 8.0) }, - { FL2FXCONST_SGL(-0.35712514743563f / 8.0), FL2FXCONST_SGL( 0.19298963768574f / 8.0) }, - { FL2FXCONST_SGL(-0.99899084509530f / 8.0), FL2FXCONST_SGL( 0.74645156992493f / 8.0) }, - { FL2FXCONST_SGL( 0.86557171579452f / 8.0), FL2FXCONST_SGL( 0.55593866696299f / 8.0) }, - { FL2FXCONST_SGL( 0.33408042438752f / 8.0), FL2FXCONST_SGL( 0.86185953874709f / 8.0) }, - { FL2FXCONST_SGL( 0.99010736374716f / 8.0), FL2FXCONST_SGL( 0.04602397576623f / 8.0) }, - { FL2FXCONST_SGL(-0.66694269691195f / 8.0), FL2FXCONST_SGL(-0.91643611810148f / 8.0) }, - { FL2FXCONST_SGL( 0.64016792079480f / 8.0), FL2FXCONST_SGL( 0.15649530836856f / 8.0) }, - { FL2FXCONST_SGL( 0.99570534804836f / 8.0), FL2FXCONST_SGL( 0.45844586038111f / 8.0) }, - { FL2FXCONST_SGL(-0.63431466947340f / 8.0), FL2FXCONST_SGL( 0.21079116459234f / 8.0) }, - { FL2FXCONST_SGL(-0.07706847005931f / 8.0), FL2FXCONST_SGL(-0.89581437101329f / 8.0) }, - { FL2FXCONST_SGL( 0.98590090577724f / 8.0), FL2FXCONST_SGL( 0.88241721133981f / 8.0) }, - { FL2FXCONST_SGL( 0.80099335254678f / 8.0), FL2FXCONST_SGL(-0.36851896710853f / 8.0) }, - { FL2FXCONST_SGL( 0.78368131392666f / 8.0), FL2FXCONST_SGL( 0.45506999802597f / 8.0) }, - { FL2FXCONST_SGL( 0.08707806671691f / 8.0), FL2FXCONST_SGL( 0.80938994918745f / 8.0) }, - { FL2FXCONST_SGL(-0.86811883080712f / 8.0), FL2FXCONST_SGL( 0.39347308654705f / 8.0) }, - { FL2FXCONST_SGL(-0.39466529740375f / 8.0), FL2FXCONST_SGL(-0.66809432114456f / 8.0) }, - { FL2FXCONST_SGL( 0.97875325649683f / 8.0), FL2FXCONST_SGL(-0.72467840967746f / 8.0) }, - { FL2FXCONST_SGL(-0.95038560288864f / 8.0), FL2FXCONST_SGL( 0.89563219587625f / 8.0) }, - { FL2FXCONST_SGL( 0.17005239424212f / 8.0), FL2FXCONST_SGL( 0.54683053962658f / 8.0) }, - { FL2FXCONST_SGL(-0.76910792026848f / 8.0), FL2FXCONST_SGL(-0.96226617549298f / 8.0) }, - { FL2FXCONST_SGL( 0.99743281016846f / 8.0), FL2FXCONST_SGL( 0.42697157037567f / 8.0) }, - { FL2FXCONST_SGL( 0.95437383549973f / 8.0), FL2FXCONST_SGL( 0.97002324109952f / 8.0) }, - { FL2FXCONST_SGL( 0.99578905365569f / 8.0), FL2FXCONST_SGL(-0.54106826257356f / 8.0) }, - { FL2FXCONST_SGL( 0.28058259829990f / 8.0), FL2FXCONST_SGL(-0.85361420634036f / 8.0) }, - { FL2FXCONST_SGL( 0.85256524470573f / 8.0), FL2FXCONST_SGL(-0.64567607735589f / 8.0) }, - { FL2FXCONST_SGL(-0.50608540105128f / 8.0), FL2FXCONST_SGL(-0.65846015480300f / 8.0) }, - { FL2FXCONST_SGL(-0.97210735183243f / 8.0), FL2FXCONST_SGL(-0.23095213067791f / 8.0) }, - { FL2FXCONST_SGL( 0.95424048234441f / 8.0), FL2FXCONST_SGL(-0.99240147091219f / 8.0) }, - { FL2FXCONST_SGL(-0.96926570524023f / 8.0), FL2FXCONST_SGL( 0.73775654896574f / 8.0) }, - { FL2FXCONST_SGL( 0.30872163214726f / 8.0), FL2FXCONST_SGL( 0.41514960556126f / 8.0) }, - { FL2FXCONST_SGL(-0.24523839572639f / 8.0), FL2FXCONST_SGL( 0.63206633394807f / 8.0) }, - { FL2FXCONST_SGL(-0.33813265086024f / 8.0), FL2FXCONST_SGL(-0.38661779441897f / 8.0) }, - { FL2FXCONST_SGL(-0.05826828420146f / 8.0), FL2FXCONST_SGL(-0.06940774188029f / 8.0) }, - { FL2FXCONST_SGL(-0.22898461455054f / 8.0), FL2FXCONST_SGL( 0.97054853316316f / 8.0) }, - { FL2FXCONST_SGL(-0.18509915019881f / 8.0), FL2FXCONST_SGL( 0.47565762892084f / 8.0) }, - { FL2FXCONST_SGL(-0.10488238045009f / 8.0), FL2FXCONST_SGL(-0.87769947402394f / 8.0) }, - { FL2FXCONST_SGL(-0.71886586182037f / 8.0), FL2FXCONST_SGL( 0.78030982480538f / 8.0) }, - { FL2FXCONST_SGL( 0.99793873738654f / 8.0), FL2FXCONST_SGL( 0.90041310491497f / 8.0) }, - { FL2FXCONST_SGL( 0.57563307626120f / 8.0), FL2FXCONST_SGL(-0.91034337352097f / 8.0) }, - { FL2FXCONST_SGL( 0.28909646383717f / 8.0), FL2FXCONST_SGL( 0.96307783970534f / 8.0) }, - { FL2FXCONST_SGL( 0.42188998312520f / 8.0), FL2FXCONST_SGL( 0.48148651230437f / 8.0) }, - { FL2FXCONST_SGL( 0.93335049681047f / 8.0), FL2FXCONST_SGL(-0.43537023883588f / 8.0) }, - { FL2FXCONST_SGL(-0.97087374418267f / 8.0), FL2FXCONST_SGL( 0.86636445711364f / 8.0) }, - { FL2FXCONST_SGL( 0.36722871286923f / 8.0), FL2FXCONST_SGL( 0.65291654172961f / 8.0) }, - { FL2FXCONST_SGL(-0.81093025665696f / 8.0), FL2FXCONST_SGL( 0.08778370229363f / 8.0) }, - { FL2FXCONST_SGL(-0.26240603062237f / 8.0), FL2FXCONST_SGL(-0.92774095379098f / 8.0) }, - { FL2FXCONST_SGL( 0.83996497984604f / 8.0), FL2FXCONST_SGL( 0.55839849139647f / 8.0) }, - { FL2FXCONST_SGL(-0.99909615720225f / 8.0), FL2FXCONST_SGL(-0.96024605713970f / 8.0) }, - { FL2FXCONST_SGL( 0.74649464155061f / 8.0), FL2FXCONST_SGL( 0.12144893606462f / 8.0) }, - { FL2FXCONST_SGL(-0.74774595569805f / 8.0), FL2FXCONST_SGL(-0.26898062008959f / 8.0) }, - { FL2FXCONST_SGL( 0.95781667469567f / 8.0), FL2FXCONST_SGL(-0.79047927052628f / 8.0) }, - { FL2FXCONST_SGL( 0.95472308713099f / 8.0), FL2FXCONST_SGL(-0.08588776019550f / 8.0) }, - { FL2FXCONST_SGL( 0.48708332746299f / 8.0), FL2FXCONST_SGL( 0.99999041579432f / 8.0) }, - { FL2FXCONST_SGL( 0.46332038247497f / 8.0), FL2FXCONST_SGL( 0.10964126185063f / 8.0) }, - { FL2FXCONST_SGL(-0.76497004940162f / 8.0), FL2FXCONST_SGL( 0.89210929242238f / 8.0) }, - { FL2FXCONST_SGL( 0.57397389364339f / 8.0), FL2FXCONST_SGL( 0.35289703373760f / 8.0) }, - { FL2FXCONST_SGL( 0.75374316974495f / 8.0), FL2FXCONST_SGL( 0.96705214651335f / 8.0) }, - { FL2FXCONST_SGL(-0.59174397685714f / 8.0), FL2FXCONST_SGL(-0.89405370422752f / 8.0) }, - { FL2FXCONST_SGL( 0.75087906691890f / 8.0), FL2FXCONST_SGL(-0.29612672982396f / 8.0) }, - { FL2FXCONST_SGL(-0.98607857336230f / 8.0), FL2FXCONST_SGL( 0.25034911730023f / 8.0) }, - { FL2FXCONST_SGL(-0.40761056640505f / 8.0), FL2FXCONST_SGL(-0.90045573444695f / 8.0) }, - { FL2FXCONST_SGL( 0.66929266740477f / 8.0), FL2FXCONST_SGL( 0.98629493401748f / 8.0) }, - { FL2FXCONST_SGL(-0.97463695257310f / 8.0), FL2FXCONST_SGL(-0.00190223301301f / 8.0) }, - { FL2FXCONST_SGL( 0.90145509409859f / 8.0), FL2FXCONST_SGL( 0.99781390365446f / 8.0) }, - { FL2FXCONST_SGL(-0.87259289048043f / 8.0), FL2FXCONST_SGL( 0.99233587353666f / 8.0) }, - { FL2FXCONST_SGL(-0.91529461447692f / 8.0), FL2FXCONST_SGL(-0.15698707534206f / 8.0) }, - { FL2FXCONST_SGL(-0.03305738840705f / 8.0), FL2FXCONST_SGL(-0.37205262859764f / 8.0) }, - { FL2FXCONST_SGL( 0.07223051368337f / 8.0), FL2FXCONST_SGL(-0.88805001733626f / 8.0) }, - { FL2FXCONST_SGL( 0.99498012188353f / 8.0), FL2FXCONST_SGL( 0.97094358113387f / 8.0) }, - { FL2FXCONST_SGL(-0.74904939500519f / 8.0), FL2FXCONST_SGL( 0.99985483641521f / 8.0) }, - { FL2FXCONST_SGL( 0.04585228574211f / 8.0), FL2FXCONST_SGL( 0.99812337444082f / 8.0) }, - { FL2FXCONST_SGL(-0.89054954257993f / 8.0), FL2FXCONST_SGL(-0.31791913188064f / 8.0) }, - { FL2FXCONST_SGL(-0.83782144651251f / 8.0), FL2FXCONST_SGL( 0.97637632547466f / 8.0) }, - { FL2FXCONST_SGL( 0.33454804933804f / 8.0), FL2FXCONST_SGL(-0.86231516800408f / 8.0) }, - { FL2FXCONST_SGL(-0.99707579362824f / 8.0), FL2FXCONST_SGL( 0.93237990079441f / 8.0) }, - { FL2FXCONST_SGL(-0.22827527843994f / 8.0), FL2FXCONST_SGL( 0.18874759397997f / 8.0) }, - { FL2FXCONST_SGL( 0.67248046289143f / 8.0), FL2FXCONST_SGL(-0.03646211390569f / 8.0) }, - { FL2FXCONST_SGL(-0.05146538187944f / 8.0), FL2FXCONST_SGL(-0.92599700120679f / 8.0) }, - { FL2FXCONST_SGL( 0.99947295749905f / 8.0), FL2FXCONST_SGL( 0.93625229707912f / 8.0) }, - { FL2FXCONST_SGL( 0.66951124390363f / 8.0), FL2FXCONST_SGL( 0.98905825623893f / 8.0) }, - { FL2FXCONST_SGL(-0.99602956559179f / 8.0), FL2FXCONST_SGL(-0.44654715757688f / 8.0) }, - { FL2FXCONST_SGL( 0.82104905483590f / 8.0), FL2FXCONST_SGL( 0.99540741724928f / 8.0) }, - { FL2FXCONST_SGL( 0.99186510988782f / 8.0), FL2FXCONST_SGL( 0.72023001312947f / 8.0) }, - { FL2FXCONST_SGL(-0.65284592392918f / 8.0), FL2FXCONST_SGL( 0.52186723253637f / 8.0) }, - { FL2FXCONST_SGL( 0.93885443798188f / 8.0), FL2FXCONST_SGL(-0.74895312615259f / 8.0) }, - { FL2FXCONST_SGL( 0.96735248738388f / 8.0), FL2FXCONST_SGL( 0.90891816978629f / 8.0) }, - { FL2FXCONST_SGL(-0.22225968841114f / 8.0), FL2FXCONST_SGL( 0.57124029781228f / 8.0) }, - { FL2FXCONST_SGL(-0.44132783753414f / 8.0), FL2FXCONST_SGL(-0.92688840659280f / 8.0) }, - { FL2FXCONST_SGL(-0.85694974219574f / 8.0), FL2FXCONST_SGL( 0.88844532719844f / 8.0) }, - { FL2FXCONST_SGL( 0.91783042091762f / 8.0), FL2FXCONST_SGL(-0.46356892383970f / 8.0) }, - { FL2FXCONST_SGL( 0.72556974415690f / 8.0), FL2FXCONST_SGL(-0.99899555770747f / 8.0) }, - { FL2FXCONST_SGL(-0.99711581834508f / 8.0), FL2FXCONST_SGL( 0.58211560180426f / 8.0) }, - { FL2FXCONST_SGL( 0.77638976371966f / 8.0), FL2FXCONST_SGL( 0.94321834873819f / 8.0) }, - { FL2FXCONST_SGL( 0.07717324253925f / 8.0), FL2FXCONST_SGL( 0.58638399856595f / 8.0) }, - { FL2FXCONST_SGL(-0.56049829194163f / 8.0), FL2FXCONST_SGL( 0.82522301569036f / 8.0) }, - { FL2FXCONST_SGL( 0.98398893639988f / 8.0), FL2FXCONST_SGL( 0.39467440420569f / 8.0) }, - { FL2FXCONST_SGL( 0.47546946844938f / 8.0), FL2FXCONST_SGL( 0.68613044836811f / 8.0) }, - { FL2FXCONST_SGL( 0.65675089314631f / 8.0), FL2FXCONST_SGL( 0.18331637134880f / 8.0) }, - { FL2FXCONST_SGL( 0.03273375457980f / 8.0), FL2FXCONST_SGL(-0.74933109564108f / 8.0) }, - { FL2FXCONST_SGL(-0.38684144784738f / 8.0), FL2FXCONST_SGL( 0.51337349030406f / 8.0) }, - { FL2FXCONST_SGL(-0.97346267944545f / 8.0), FL2FXCONST_SGL(-0.96549364384098f / 8.0) }, - { FL2FXCONST_SGL(-0.53282156061942f / 8.0), FL2FXCONST_SGL(-0.91423265091354f / 8.0) }, - { FL2FXCONST_SGL( 0.99817310731176f / 8.0), FL2FXCONST_SGL( 0.61133572482148f / 8.0) }, - { FL2FXCONST_SGL(-0.50254500772635f / 8.0), FL2FXCONST_SGL(-0.88829338134294f / 8.0) }, - { FL2FXCONST_SGL( 0.01995873238855f / 8.0), FL2FXCONST_SGL( 0.85223515096765f / 8.0) }, - { FL2FXCONST_SGL( 0.99930381973804f / 8.0), FL2FXCONST_SGL( 0.94578896296649f / 8.0) }, - { FL2FXCONST_SGL( 0.82907767600783f / 8.0), FL2FXCONST_SGL(-0.06323442598128f / 8.0) }, - { FL2FXCONST_SGL(-0.58660709669728f / 8.0), FL2FXCONST_SGL( 0.96840773806582f / 8.0) }, - { FL2FXCONST_SGL(-0.17573736667267f / 8.0), FL2FXCONST_SGL(-0.48166920859485f / 8.0) }, - { FL2FXCONST_SGL( 0.83434292401346f / 8.0), FL2FXCONST_SGL(-0.13023450646997f / 8.0) }, - { FL2FXCONST_SGL( 0.05946491307025f / 8.0), FL2FXCONST_SGL( 0.20511047074866f / 8.0) }, - { FL2FXCONST_SGL( 0.81505484574602f / 8.0), FL2FXCONST_SGL(-0.94685947861369f / 8.0) }, - { FL2FXCONST_SGL(-0.44976380954860f / 8.0), FL2FXCONST_SGL( 0.40894572671545f / 8.0) }, - { FL2FXCONST_SGL(-0.89746474625671f / 8.0), FL2FXCONST_SGL( 0.99846578838537f / 8.0) }, - { FL2FXCONST_SGL( 0.39677256130792f / 8.0), FL2FXCONST_SGL(-0.74854668609359f / 8.0) }, - { FL2FXCONST_SGL(-0.07588948563079f / 8.0), FL2FXCONST_SGL( 0.74096214084170f / 8.0) }, - { FL2FXCONST_SGL( 0.76343198951445f / 8.0), FL2FXCONST_SGL( 0.41746629422634f / 8.0) }, - { FL2FXCONST_SGL(-0.74490104699626f / 8.0), FL2FXCONST_SGL( 0.94725911744610f / 8.0) }, - { FL2FXCONST_SGL( 0.64880119792759f / 8.0), FL2FXCONST_SGL( 0.41336660830571f / 8.0) }, - { FL2FXCONST_SGL( 0.62319537462542f / 8.0), FL2FXCONST_SGL(-0.93098313552599f / 8.0) }, - { FL2FXCONST_SGL( 0.42215817594807f / 8.0), FL2FXCONST_SGL(-0.07712787385208f / 8.0) }, - { FL2FXCONST_SGL( 0.02704554141885f / 8.0), FL2FXCONST_SGL(-0.05417518053666f / 8.0) }, - { FL2FXCONST_SGL( 0.80001773566818f / 8.0), FL2FXCONST_SGL( 0.91542195141039f / 8.0) }, - { FL2FXCONST_SGL(-0.79351832348816f / 8.0), FL2FXCONST_SGL(-0.36208897989136f / 8.0) }, - { FL2FXCONST_SGL( 0.63872359151636f / 8.0), FL2FXCONST_SGL( 0.08128252493444f / 8.0) }, - { FL2FXCONST_SGL( 0.52890520960295f / 8.0), FL2FXCONST_SGL( 0.60048872455592f / 8.0) }, - { FL2FXCONST_SGL( 0.74238552914587f / 8.0), FL2FXCONST_SGL( 0.04491915291044f / 8.0) }, - { FL2FXCONST_SGL( 0.99096131449250f / 8.0), FL2FXCONST_SGL(-0.19451182854402f / 8.0) }, - { FL2FXCONST_SGL(-0.80412329643109f / 8.0), FL2FXCONST_SGL(-0.88513818199457f / 8.0) }, - { FL2FXCONST_SGL(-0.64612616129736f / 8.0), FL2FXCONST_SGL( 0.72198674804544f / 8.0) }, - { FL2FXCONST_SGL( 0.11657770663191f / 8.0), FL2FXCONST_SGL(-0.83662833815041f / 8.0) }, - { FL2FXCONST_SGL(-0.95053182488101f / 8.0), FL2FXCONST_SGL(-0.96939905138082f / 8.0) }, - { FL2FXCONST_SGL(-0.62228872928622f / 8.0), FL2FXCONST_SGL( 0.82767262846661f / 8.0) }, - { FL2FXCONST_SGL( 0.03004475787316f / 8.0), FL2FXCONST_SGL(-0.99738896333384f / 8.0) }, - { FL2FXCONST_SGL(-0.97987214341034f / 8.0), FL2FXCONST_SGL( 0.36526129686425f / 8.0) }, - { FL2FXCONST_SGL(-0.99986980746200f / 8.0), FL2FXCONST_SGL(-0.36021610299715f / 8.0) }, - { FL2FXCONST_SGL( 0.89110648599879f / 8.0), FL2FXCONST_SGL(-0.97894250343044f / 8.0) }, - { FL2FXCONST_SGL( 0.10407960510582f / 8.0), FL2FXCONST_SGL( 0.77357793811619f / 8.0) }, - { FL2FXCONST_SGL( 0.95964737821728f / 8.0), FL2FXCONST_SGL(-0.35435818285502f / 8.0) }, - { FL2FXCONST_SGL( 0.50843233159162f / 8.0), FL2FXCONST_SGL( 0.96107691266205f / 8.0) }, - { FL2FXCONST_SGL( 0.17006334670615f / 8.0), FL2FXCONST_SGL(-0.76854025314829f / 8.0) }, - { FL2FXCONST_SGL( 0.25872675063360f / 8.0), FL2FXCONST_SGL( 0.99893303933816f / 8.0) }, - { FL2FXCONST_SGL(-0.01115998681937f / 8.0), FL2FXCONST_SGL( 0.98496019742444f / 8.0) }, - { FL2FXCONST_SGL(-0.79598702973261f / 8.0), FL2FXCONST_SGL( 0.97138411318894f / 8.0) }, - { FL2FXCONST_SGL(-0.99264708948101f / 8.0), FL2FXCONST_SGL(-0.99542822402536f / 8.0) }, - { FL2FXCONST_SGL(-0.99829663752818f / 8.0), FL2FXCONST_SGL( 0.01877138824311f / 8.0) }, - { FL2FXCONST_SGL(-0.70801016548184f / 8.0), FL2FXCONST_SGL( 0.33680685948117f / 8.0) }, - { FL2FXCONST_SGL(-0.70467057786826f / 8.0), FL2FXCONST_SGL( 0.93272777501857f / 8.0) }, - { FL2FXCONST_SGL( 0.99846021905254f / 8.0), FL2FXCONST_SGL(-0.98725746254433f / 8.0) }, - { FL2FXCONST_SGL(-0.63364968534650f / 8.0), FL2FXCONST_SGL(-0.16473594423746f / 8.0) }, - { FL2FXCONST_SGL(-0.16258217500792f / 8.0), FL2FXCONST_SGL(-0.95939125400802f / 8.0) }, - { FL2FXCONST_SGL(-0.43645594360633f / 8.0), FL2FXCONST_SGL(-0.94805030113284f / 8.0) }, - { FL2FXCONST_SGL(-0.99848471702976f / 8.0), FL2FXCONST_SGL( 0.96245166923809f / 8.0) }, - { FL2FXCONST_SGL(-0.16796458968998f / 8.0), FL2FXCONST_SGL(-0.98987511890470f / 8.0) }, - { FL2FXCONST_SGL(-0.87979225745213f / 8.0), FL2FXCONST_SGL(-0.71725725041680f / 8.0) }, - { FL2FXCONST_SGL( 0.44183099021786f / 8.0), FL2FXCONST_SGL(-0.93568974498761f / 8.0) }, - { FL2FXCONST_SGL( 0.93310180125532f / 8.0), FL2FXCONST_SGL(-0.99913308068246f / 8.0) }, - { FL2FXCONST_SGL(-0.93941931782002f / 8.0), FL2FXCONST_SGL(-0.56409379640356f / 8.0) }, - { FL2FXCONST_SGL(-0.88590003188677f / 8.0), FL2FXCONST_SGL( 0.47624600491382f / 8.0) }, - { FL2FXCONST_SGL( 0.99971463703691f / 8.0), FL2FXCONST_SGL(-0.83889954253462f / 8.0) }, - { FL2FXCONST_SGL(-0.75376385639978f / 8.0), FL2FXCONST_SGL( 0.00814643438625f / 8.0) }, - { FL2FXCONST_SGL( 0.93887685615875f / 8.0), FL2FXCONST_SGL(-0.11284528204636f / 8.0) }, - { FL2FXCONST_SGL( 0.85126435782309f / 8.0), FL2FXCONST_SGL( 0.52349251543547f / 8.0) }, - { FL2FXCONST_SGL( 0.39701421446381f / 8.0), FL2FXCONST_SGL( 0.81779634174316f / 8.0) }, - { FL2FXCONST_SGL(-0.37024464187437f / 8.0), FL2FXCONST_SGL(-0.87071656222959f / 8.0) }, - { FL2FXCONST_SGL(-0.36024828242896f / 8.0), FL2FXCONST_SGL( 0.34655735648287f / 8.0) }, - { FL2FXCONST_SGL(-0.93388812549209f / 8.0), FL2FXCONST_SGL(-0.84476541096429f / 8.0) }, - { FL2FXCONST_SGL(-0.65298804552119f / 8.0), FL2FXCONST_SGL(-0.18439575450921f / 8.0) }, - { FL2FXCONST_SGL( 0.11960319006843f / 8.0), FL2FXCONST_SGL( 0.99899346780168f / 8.0) }, - { FL2FXCONST_SGL( 0.94292565553160f / 8.0), FL2FXCONST_SGL( 0.83163906518293f / 8.0) }, - { FL2FXCONST_SGL( 0.75081145286948f / 8.0), FL2FXCONST_SGL(-0.35533223142265f / 8.0) }, - { FL2FXCONST_SGL( 0.56721979748394f / 8.0), FL2FXCONST_SGL(-0.24076836414499f / 8.0) }, - { FL2FXCONST_SGL( 0.46857766746029f / 8.0), FL2FXCONST_SGL(-0.30140233457198f / 8.0) }, - { FL2FXCONST_SGL( 0.97312313923635f / 8.0), FL2FXCONST_SGL(-0.99548191630031f / 8.0) }, - { FL2FXCONST_SGL(-0.38299976567017f / 8.0), FL2FXCONST_SGL( 0.98516909715427f / 8.0) }, - { FL2FXCONST_SGL( 0.41025800019463f / 8.0), FL2FXCONST_SGL( 0.02116736935734f / 8.0) }, - { FL2FXCONST_SGL( 0.09638062008048f / 8.0), FL2FXCONST_SGL( 0.04411984381457f / 8.0) }, - { FL2FXCONST_SGL(-0.85283249275397f / 8.0), FL2FXCONST_SGL( 0.91475563922421f / 8.0) }, - { FL2FXCONST_SGL( 0.88866808958124f / 8.0), FL2FXCONST_SGL(-0.99735267083226f / 8.0) }, - { FL2FXCONST_SGL(-0.48202429536989f / 8.0), FL2FXCONST_SGL(-0.96805608884164f / 8.0) }, - { FL2FXCONST_SGL( 0.27572582416567f / 8.0), FL2FXCONST_SGL( 0.58634753335832f / 8.0) }, - { FL2FXCONST_SGL(-0.65889129659168f / 8.0), FL2FXCONST_SGL( 0.58835634138583f / 8.0) }, - { FL2FXCONST_SGL( 0.98838086953732f / 8.0), FL2FXCONST_SGL( 0.99994349600236f / 8.0) }, - { FL2FXCONST_SGL(-0.20651349620689f / 8.0), FL2FXCONST_SGL( 0.54593044066355f / 8.0) }, - { FL2FXCONST_SGL(-0.62126416356920f / 8.0), FL2FXCONST_SGL(-0.59893681700392f / 8.0) }, - { FL2FXCONST_SGL( 0.20320105410437f / 8.0), FL2FXCONST_SGL(-0.86879180355289f / 8.0) }, - { FL2FXCONST_SGL(-0.97790548600584f / 8.0), FL2FXCONST_SGL( 0.96290806999242f / 8.0) }, - { FL2FXCONST_SGL( 0.11112534735126f / 8.0), FL2FXCONST_SGL( 0.21484763313301f / 8.0) }, - { FL2FXCONST_SGL(-0.41368337314182f / 8.0), FL2FXCONST_SGL( 0.28216837680365f / 8.0) }, - { FL2FXCONST_SGL( 0.24133038992960f / 8.0), FL2FXCONST_SGL( 0.51294362630238f / 8.0) }, - { FL2FXCONST_SGL(-0.66393410674885f / 8.0), FL2FXCONST_SGL(-0.08249679629081f / 8.0) }, - { FL2FXCONST_SGL(-0.53697829178752f / 8.0), FL2FXCONST_SGL(-0.97649903936228f / 8.0) }, - { FL2FXCONST_SGL(-0.97224737889348f / 8.0), FL2FXCONST_SGL( 0.22081333579837f / 8.0) }, - { FL2FXCONST_SGL( 0.87392477144549f / 8.0), FL2FXCONST_SGL(-0.12796173740361f / 8.0) }, - { FL2FXCONST_SGL( 0.19050361015753f / 8.0), FL2FXCONST_SGL( 0.01602615387195f / 8.0) }, - { FL2FXCONST_SGL(-0.46353441212724f / 8.0), FL2FXCONST_SGL(-0.95249041539006f / 8.0) }, - { FL2FXCONST_SGL(-0.07064096339021f / 8.0), FL2FXCONST_SGL(-0.94479803205886f / 8.0) }, - { FL2FXCONST_SGL(-0.92444085484466f / 8.0), FL2FXCONST_SGL(-0.10457590187436f / 8.0) }, - { FL2FXCONST_SGL(-0.83822593578728f / 8.0), FL2FXCONST_SGL(-0.01695043208885f / 8.0) }, - { FL2FXCONST_SGL( 0.75214681811150f / 8.0), FL2FXCONST_SGL(-0.99955681042665f / 8.0) }, - { FL2FXCONST_SGL(-0.42102998829339f / 8.0), FL2FXCONST_SGL( 0.99720941999394f / 8.0) }, - { FL2FXCONST_SGL(-0.72094786237696f / 8.0), FL2FXCONST_SGL(-0.35008961934255f / 8.0) }, - { FL2FXCONST_SGL( 0.78843311019251f / 8.0), FL2FXCONST_SGL( 0.52851398958271f / 8.0) }, - { FL2FXCONST_SGL( 0.97394027897442f / 8.0), FL2FXCONST_SGL(-0.26695944086561f / 8.0) }, - { FL2FXCONST_SGL( 0.99206463477946f / 8.0), FL2FXCONST_SGL(-0.57010120849429f / 8.0) }, - { FL2FXCONST_SGL( 0.76789609461795f / 8.0), FL2FXCONST_SGL(-0.76519356730966f / 8.0) }, - { FL2FXCONST_SGL(-0.82002421836409f / 8.0), FL2FXCONST_SGL(-0.73530179553767f / 8.0) }, - { FL2FXCONST_SGL( 0.81924990025724f / 8.0), FL2FXCONST_SGL( 0.99698425250579f / 8.0) }, - { FL2FXCONST_SGL(-0.26719850873357f / 8.0), FL2FXCONST_SGL( 0.68903369776193f / 8.0) }, - { FL2FXCONST_SGL(-0.43311260380975f / 8.0), FL2FXCONST_SGL( 0.85321815947490f / 8.0) }, - { FL2FXCONST_SGL( 0.99194979673836f / 8.0), FL2FXCONST_SGL( 0.91876249766422f / 8.0) }, - { FL2FXCONST_SGL(-0.80692001248487f / 8.0), FL2FXCONST_SGL(-0.32627540663214f / 8.0) }, - { FL2FXCONST_SGL( 0.43080003649976f / 8.0), FL2FXCONST_SGL(-0.21919095636638f / 8.0) }, - { FL2FXCONST_SGL( 0.67709491937357f / 8.0), FL2FXCONST_SGL(-0.95478075822906f / 8.0) }, - { FL2FXCONST_SGL( 0.56151770568316f / 8.0), FL2FXCONST_SGL(-0.70693811747778f / 8.0) }, - { FL2FXCONST_SGL( 0.10831862810749f / 8.0), FL2FXCONST_SGL(-0.08628837174592f / 8.0) }, - { FL2FXCONST_SGL( 0.91229417540436f / 8.0), FL2FXCONST_SGL(-0.65987351408410f / 8.0) }, - { FL2FXCONST_SGL(-0.48972893932274f / 8.0), FL2FXCONST_SGL( 0.56289246362686f / 8.0) }, - { FL2FXCONST_SGL(-0.89033658689697f / 8.0), FL2FXCONST_SGL(-0.71656563987082f / 8.0) }, - { FL2FXCONST_SGL( 0.65269447475094f / 8.0), FL2FXCONST_SGL( 0.65916004833932f / 8.0) }, - { FL2FXCONST_SGL( 0.67439478141121f / 8.0), FL2FXCONST_SGL(-0.81684380846796f / 8.0) }, - { FL2FXCONST_SGL(-0.47770832416973f / 8.0), FL2FXCONST_SGL(-0.16789556203025f / 8.0) }, - { FL2FXCONST_SGL(-0.99715979260878f / 8.0), FL2FXCONST_SGL(-0.93565784007648f / 8.0) }, - { FL2FXCONST_SGL(-0.90889593602546f / 8.0), FL2FXCONST_SGL( 0.62034397054380f / 8.0) }, - { FL2FXCONST_SGL(-0.06618622548177f / 8.0), FL2FXCONST_SGL(-0.23812217221359f / 8.0) }, - { FL2FXCONST_SGL( 0.99430266919728f / 8.0), FL2FXCONST_SGL( 0.18812555317553f / 8.0) }, - { FL2FXCONST_SGL( 0.97686402381843f / 8.0), FL2FXCONST_SGL(-0.28664534366620f / 8.0) }, - { FL2FXCONST_SGL( 0.94813650221268f / 8.0), FL2FXCONST_SGL(-0.97506640027128f / 8.0) }, - { FL2FXCONST_SGL(-0.95434497492853f / 8.0), FL2FXCONST_SGL(-0.79607978501983f / 8.0) }, - { FL2FXCONST_SGL(-0.49104783137150f / 8.0), FL2FXCONST_SGL( 0.32895214359663f / 8.0) }, - { FL2FXCONST_SGL( 0.99881175120751f / 8.0), FL2FXCONST_SGL( 0.88993983831354f / 8.0) }, - { FL2FXCONST_SGL( 0.50449166760303f / 8.0), FL2FXCONST_SGL(-0.85995072408434f / 8.0) }, - { FL2FXCONST_SGL( 0.47162891065108f / 8.0), FL2FXCONST_SGL(-0.18680204049569f / 8.0) }, - { FL2FXCONST_SGL(-0.62081581361840f / 8.0), FL2FXCONST_SGL( 0.75000676218956f / 8.0) }, - { FL2FXCONST_SGL(-0.43867015250812f / 8.0), FL2FXCONST_SGL( 0.99998069244322f / 8.0) }, - { FL2FXCONST_SGL( 0.98630563232075f / 8.0), FL2FXCONST_SGL(-0.53578899600662f / 8.0) }, - { FL2FXCONST_SGL(-0.61510362277374f / 8.0), FL2FXCONST_SGL(-0.89515019899997f / 8.0) }, - { FL2FXCONST_SGL(-0.03841517601843f / 8.0), FL2FXCONST_SGL(-0.69888815681179f / 8.0) }, - { FL2FXCONST_SGL(-0.30102157304644f / 8.0), FL2FXCONST_SGL(-0.07667808922205f / 8.0) }, - { FL2FXCONST_SGL( 0.41881284182683f / 8.0), FL2FXCONST_SGL( 0.02188098922282f / 8.0) }, - { FL2FXCONST_SGL(-0.86135454941237f / 8.0), FL2FXCONST_SGL( 0.98947480909359f / 8.0) }, - { FL2FXCONST_SGL( 0.67226861393788f / 8.0), FL2FXCONST_SGL(-0.13494389011014f / 8.0) }, - { FL2FXCONST_SGL(-0.70737398842068f / 8.0), FL2FXCONST_SGL(-0.76547349325992f / 8.0) }, - { FL2FXCONST_SGL( 0.94044946687963f / 8.0), FL2FXCONST_SGL( 0.09026201157416f / 8.0) }, - { FL2FXCONST_SGL(-0.82386352534327f / 8.0), FL2FXCONST_SGL( 0.08924768823676f / 8.0) }, - { FL2FXCONST_SGL(-0.32070666698656f / 8.0), FL2FXCONST_SGL( 0.50143421908753f / 8.0) }, - { FL2FXCONST_SGL( 0.57593163224487f / 8.0), FL2FXCONST_SGL(-0.98966422921509f / 8.0) }, - { FL2FXCONST_SGL(-0.36326018419965f / 8.0), FL2FXCONST_SGL( 0.07440243123228f / 8.0) }, - { FL2FXCONST_SGL( 0.99979044674350f / 8.0), FL2FXCONST_SGL(-0.14130287347405f / 8.0) }, - { FL2FXCONST_SGL(-0.92366023326932f / 8.0), FL2FXCONST_SGL(-0.97979298068180f / 8.0) }, - { FL2FXCONST_SGL(-0.44607178518598f / 8.0), FL2FXCONST_SGL(-0.54233252016394f / 8.0) }, - { FL2FXCONST_SGL( 0.44226800932956f / 8.0), FL2FXCONST_SGL( 0.71326756742752f / 8.0) }, - { FL2FXCONST_SGL( 0.03671907158312f / 8.0), FL2FXCONST_SGL( 0.63606389366675f / 8.0) }, - { FL2FXCONST_SGL( 0.52175424682195f / 8.0), FL2FXCONST_SGL(-0.85396826735705f / 8.0) }, - { FL2FXCONST_SGL(-0.94701139690956f / 8.0), FL2FXCONST_SGL(-0.01826348194255f / 8.0) }, - { FL2FXCONST_SGL(-0.98759606946049f / 8.0), FL2FXCONST_SGL( 0.82288714303073f / 8.0) }, - { FL2FXCONST_SGL( 0.87434794743625f / 8.0), FL2FXCONST_SGL( 0.89399495655433f / 8.0) }, - { FL2FXCONST_SGL(-0.93412041758744f / 8.0), FL2FXCONST_SGL( 0.41374052024363f / 8.0) }, - { FL2FXCONST_SGL( 0.96063943315511f / 8.0), FL2FXCONST_SGL( 0.93116709541280f / 8.0) }, - { FL2FXCONST_SGL( 0.97534253457837f / 8.0), FL2FXCONST_SGL( 0.86150930812689f / 8.0) }, - { FL2FXCONST_SGL( 0.99642466504163f / 8.0), FL2FXCONST_SGL( 0.70190043427512f / 8.0) }, - { FL2FXCONST_SGL(-0.94705089665984f / 8.0), FL2FXCONST_SGL(-0.29580042814306f / 8.0) }, - { FL2FXCONST_SGL( 0.91599807087376f / 8.0), FL2FXCONST_SGL(-0.98147830385781f / 8.0) } -}; -//@} - -/* -static const FIXP_SGL harmonicPhase [2][4] = { - { 1.0, 0.0, -1.0, 0.0}, - { 0.0, 1.0, 0.0, -1.0} -}; -*/ - - -/* The CLDFB-80 is not linear phase (unsymmetric), but the exact - phase difference between adjacent bands, at exact positions - (in this case exactly in the frequency band centre), can of - course be determined anyway. While the standard symmetric QMF - bank has a phase difference of 0.5*pi, the CLDFB-80 - bank has the difference 0.2337*pi. */ -const FIXP_SGL harmonicPhaseX [2][4] = { - { FL2FXCONST_SGL( 7.423735494778151e-001), FL2FXCONST_SGL(-6.699862036159475e-001), - FL2FXCONST_SGL(-7.423735494778152e-001), FL2FXCONST_SGL( 6.699862036159474e-001) }, - { FL2FXCONST_SGL( 7.423735494778151e-001), FL2FXCONST_SGL( 6.699862036159476e-001), - FL2FXCONST_SGL(-7.423735494778151e-001), FL2FXCONST_SGL(-6.699862036159476e-001) } -}; - -/* tables for SBR and AAC LD */ -/* table for 8 time slot index */ -const int FDK_sbrDecoder_envelopeTable_8 [8][5] = { -/* transientIndex nEnv, tranIdx, shortEnv, border1, border2, ... */ -/* borders from left to right side; -1 = not in use */ - /*[|T-|------]*/ { 2, 0, 0, 1, -1 }, - /*[|-T-|-----]*/ { 2, 0, 0, 2, -1 }, - /*[--|T-|----]*/ { 3, 1, 1, 2, 4 }, - /*[---|T-|---]*/ { 3, 1, 1, 3, 5 }, - /*[----|T-|--]*/ { 3, 1, 1, 4, 6 }, - /*[-----|T--|]*/ { 2, 1, 1, 5, -1 }, - /*[------|T-|]*/ { 2, 1, 1, 6, -1 }, - /*[-------|T|]*/ { 2, 1, 1, 7, -1 }, -}; - -/* table for 15 time slot index */ -const int FDK_sbrDecoder_envelopeTable_15 [15][6] = { - /* transientIndex nEnv, tranIdx, shortEnv, border1, border2, ... */ - /* length from left to right side; -1 = not in use */ - /*[|T---|------------]*/ { 2, 0, 0, 4, -1, -1}, - /*[|-T---|-----------]*/ { 2, 0, 0, 5, -1, -1}, - /*[|--|T---|---------]*/ { 3, 1, 1, 2, 6, -1}, - /*[|---|T---|--------]*/ { 3, 1, 1, 3, 7, -1}, - /*[|----|T---|-------]*/ { 3, 1, 1, 4, 8, -1}, - /*[|-----|T---|------]*/ { 3, 1, 1, 5, 9, -1}, - /*[|------|T---|-----]*/ { 3, 1, 1, 6, 10, -1}, - /*[|-------|T---|----]*/ { 3, 1, 1, 7, 11, -1}, - /*[|--------|T---|---]*/ { 3, 1, 1, 8, 12, -1}, - /*[|---------|T---|--]*/ { 3, 1, 1, 9, 13, -1}, - /*[|----------|T----|]*/ { 2, 1, 1,10, -1, -1}, - /*[|-----------|T---|]*/ { 2, 1, 1,11, -1, -1}, - /*[|------------|T--|]*/ { 2, 1, 1,12, -1, -1}, - /*[|-------------|T-|]*/ { 2, 1, 1,13, -1, -1}, - /*[|--------------|T|]*/ { 2, 1, 1,14, -1, -1}, -}; - -/* table for 16 time slot index */ -const int FDK_sbrDecoder_envelopeTable_16 [16][6] = { - /* transientIndex nEnv, tranIdx, shortEnv, border1, border2, ... */ - /* length from left to right side; -1 = not in use */ - /*[|T---|------------|]*/ { 2, 0, 0, 4, -1, -1}, - /*[|-T---|-----------|]*/ { 2, 0, 0, 5, -1, -1}, - /*[|--|T---|----------]*/ { 3, 1, 1, 2, 6, -1}, - /*[|---|T---|---------]*/ { 3, 1, 1, 3, 7, -1}, - /*[|----|T---|--------]*/ { 3, 1, 1, 4, 8, -1}, - /*[|-----|T---|-------]*/ { 3, 1, 1, 5, 9, -1}, - /*[|------|T---|------]*/ { 3, 1, 1, 6, 10, -1}, - /*[|-------|T---|-----]*/ { 3, 1, 1, 7, 11, -1}, - /*[|--------|T---|----]*/ { 3, 1, 1, 8, 12, -1}, - /*[|---------|T---|---]*/ { 3, 1, 1, 9, 13, -1}, - /*[|----------|T---|--]*/ { 3, 1, 1,10, 14, -1}, - /*[|-----------|T----|]*/ { 2, 1, 1,11, -1, -1}, - /*[|------------|T---|]*/ { 2, 1, 1,12, -1, -1}, - /*[|-------------|T--|]*/ { 2, 1, 1,13, -1, -1}, - /*[|--------------|T-|]*/ { 2, 1, 1,14, -1, -1}, - /*[|---------------|T|]*/ { 2, 1, 1,15, -1, -1}, -}; - -/*! - \name FrameInfoDefaults - - Predefined envelope positions for the FIX-FIX case (static framing) -*/ -//@{ -const FRAME_INFO FDK_sbrDecoder_sbr_frame_info1_15 = { 0, 1, {0, 15, 0, 0, 0, 0}, {1, 0, 0, 0, 0}, -1, 1, {0, 15, 0} }; -const FRAME_INFO FDK_sbrDecoder_sbr_frame_info2_15 = { 0, 2, {0, 8, 15, 0, 0, 0}, {1, 1, 0, 0, 0}, -1, 2, {0, 8, 15} }; -const FRAME_INFO FDK_sbrDecoder_sbr_frame_info4_15 = { 0, 4, {0, 4, 8, 12, 15, 0}, {1, 1, 1, 1, 0}, -1, 2, {0, 8, 15} }; -#if (MAX_ENVELOPES >= 8) -const FRAME_INFO FDK_sbrDecoder_sbr_frame_info8_15 = { 0, 8, {0, 2, 4, 6, 8, 10, 12, 14, 15}, {1, 1, 1, 1, 1, 1, 1, 1}, -1, 2, {0, 8, 15} }; -#endif - -const FRAME_INFO FDK_sbrDecoder_sbr_frame_info1_16 = { 0, 1, {0, 16, 0, 0, 0, 0}, {1, 0, 0, 0, 0}, -1, 1, {0, 16, 0} }; -const FRAME_INFO FDK_sbrDecoder_sbr_frame_info2_16 = { 0, 2, {0, 8, 16, 0, 0, 0}, {1, 1, 0, 0, 0}, -1, 2, {0, 8, 16} }; -const FRAME_INFO FDK_sbrDecoder_sbr_frame_info4_16 = { 0, 4, {0, 4, 8, 12, 16, 0}, {1, 1, 1, 1, 0}, -1, 2, {0, 8, 16} }; - -#if (MAX_ENVELOPES >= 8) -const FRAME_INFO FDK_sbrDecoder_sbr_frame_info8_16 = { 0, 8, {0, 2, 4, 6, 8, 10, 12, 14, 16}, {1, 1, 1, 1, 1, 1, 1, 1}, -1, 2, {0, 8, 16} }; -#endif - - -//@} - -/*! - \name SBR_HuffmanTables - - SBR Huffman Table Overview: \n - \n - o envelope level, 1.5 dB: \n - 1) sbr_huffBook_EnvLevel10T[120][2] \n - 2) sbr_huffBook_EnvLevel10F[120][2] \n - \n - o envelope balance, 1.5 dB: \n - 3) sbr_huffBook_EnvBalance10T[48][2] \n - 4) sbr_huffBook_EnvBalance10F[48][2] \n - \n - o envelope level, 3.0 dB: \n - 5) sbr_huffBook_EnvLevel11T[62][2] \n - 6) sbr_huffBook_EnvLevel11F[62][2] \n - \n - o envelope balance, 3.0 dB: \n - 7) sbr_huffBook_EnvBalance11T[24][2] \n - 8) sbr_huffBook_EnvBalance11F[24][2] \n - \n - o noise level, 3.0 dB: \n - 9) sbr_huffBook_NoiseLevel11T[62][2] \n - -) (sbr_huffBook_EnvLevel11F[62][2] is used for freq dir)\n - \n - o noise balance, 3.0 dB: \n - 10) sbr_huffBook_NoiseBalance11T[24][2]\n - -) (sbr_huffBook_EnvBalance11F[24][2] is used for freq dir)\n - \n - (1.5 dB is never used for noise) - -*/ -//@{ -const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvLevel10T[120][2] = { - { 1, 2 }, { -64, -65 }, { 3, 4 }, { -63, -66 }, - { 5, 6 }, { -62, -67 }, { 7, 8 }, { -61, -68 }, - { 9, 10 }, { -60, -69 }, { 11, 12 }, { -59, -70 }, - { 13, 14 }, { -58, -71 }, { 15, 16 }, { -57, -72 }, - { 17, 18 }, { -73, -56 }, { 19, 21 }, { -74, 20 }, - { -55, -75 }, { 22, 26 }, { 23, 24 }, { -54, -76 }, - { -77, 25 }, { -53, -78 }, { 27, 34 }, { 28, 29 }, - { -52, -79 }, { 30, 31 }, { -80, -51 }, { 32, 33 }, - { -83, -82 }, { -81, -50 }, { 35, 57 }, { 36, 40 }, - { 37, 38 }, { -88, -84 }, { -48, 39 }, { -90, -85 }, - { 41, 46 }, { 42, 43 }, { -49, -87 }, { 44, 45 }, - { -89, -86 }, {-124,-123 }, { 47, 50 }, { 48, 49 }, - {-122,-121 }, {-120,-119 }, { 51, 54 }, { 52, 53 }, - {-118,-117 }, {-116,-115 }, { 55, 56 }, {-114,-113 }, - {-112,-111 }, { 58, 89 }, { 59, 74 }, { 60, 67 }, - { 61, 64 }, { 62, 63 }, {-110,-109 }, {-108,-107 }, - { 65, 66 }, {-106,-105 }, {-104,-103 }, { 68, 71 }, - { 69, 70 }, {-102,-101 }, {-100, -99 }, { 72, 73 }, - { -98, -97 }, { -96, -95 }, { 75, 82 }, { 76, 79 }, - { 77, 78 }, { -94, -93 }, { -92, -91 }, { 80, 81 }, - { -47, -46 }, { -45, -44 }, { 83, 86 }, { 84, 85 }, - { -43, -42 }, { -41, -40 }, { 87, 88 }, { -39, -38 }, - { -37, -36 }, { 90, 105 }, { 91, 98 }, { 92, 95 }, - { 93, 94 }, { -35, -34 }, { -33, -32 }, { 96, 97 }, - { -31, -30 }, { -29, -28 }, { 99, 102 }, { 100, 101 }, - { -27, -26 }, { -25, -24 }, { 103, 104 }, { -23, -22 }, - { -21, -20 }, { 106, 113 }, { 107, 110 }, { 108, 109 }, - { -19, -18 }, { -17, -16 }, { 111, 112 }, { -15, -14 }, - { -13, -12 }, { 114, 117 }, { 115, 116 }, { -11, -10 }, - { -9, -8 }, { 118, 119 }, { -7, -6 }, { -5, -4 } -}; - -const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvLevel10F[120][2] = { - { 1, 2 }, { -64, -65 }, { 3, 4 }, { -63, -66 }, - { 5, 6 }, { -67, -62 }, { 7, 8 }, { -68, -61 }, - { 9, 10 }, { -69, -60 }, { 11, 13 }, { -70, 12 }, - { -59, -71 }, { 14, 16 }, { -58, 15 }, { -72, -57 }, - { 17, 19 }, { -73, 18 }, { -56, -74 }, { 20, 23 }, - { 21, 22 }, { -55, -75 }, { -54, -53 }, { 24, 27 }, - { 25, 26 }, { -76, -52 }, { -77, -51 }, { 28, 31 }, - { 29, 30 }, { -50, -78 }, { -79, -49 }, { 32, 36 }, - { 33, 34 }, { -48, -47 }, { -80, 35 }, { -81, -82 }, - { 37, 47 }, { 38, 41 }, { 39, 40 }, { -83, -46 }, - { -45, -84 }, { 42, 44 }, { -85, 43 }, { -44, -43 }, - { 45, 46 }, { -88, -87 }, { -86, -90 }, { 48, 66 }, - { 49, 56 }, { 50, 53 }, { 51, 52 }, { -92, -42 }, - { -41, -39 }, { 54, 55 }, {-105, -89 }, { -38, -37 }, - { 57, 60 }, { 58, 59 }, { -94, -91 }, { -40, -36 }, - { 61, 63 }, { -20, 62 }, {-115,-110 }, { 64, 65 }, - {-108,-107 }, {-101, -97 }, { 67, 89 }, { 68, 75 }, - { 69, 72 }, { 70, 71 }, { -95, -93 }, { -34, -27 }, - { 73, 74 }, { -22, -17 }, { -16,-124 }, { 76, 82 }, - { 77, 79 }, {-123, 78 }, {-122,-121 }, { 80, 81 }, - {-120,-119 }, {-118,-117 }, { 83, 86 }, { 84, 85 }, - {-116,-114 }, {-113,-112 }, { 87, 88 }, {-111,-109 }, - {-106,-104 }, { 90, 105 }, { 91, 98 }, { 92, 95 }, - { 93, 94 }, {-103,-102 }, {-100, -99 }, { 96, 97 }, - { -98, -96 }, { -35, -33 }, { 99, 102 }, { 100, 101 }, - { -32, -31 }, { -30, -29 }, { 103, 104 }, { -28, -26 }, - { -25, -24 }, { 106, 113 }, { 107, 110 }, { 108, 109 }, - { -23, -21 }, { -19, -18 }, { 111, 112 }, { -15, -14 }, - { -13, -12 }, { 114, 117 }, { 115, 116 }, { -11, -10 }, - { -9, -8 }, { 118, 119 }, { -7, -6 }, { -5, -4 } -}; - -const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvBalance10T[48][2] = { - { -64, 1 }, { -63, 2 }, { -65, 3 }, { -62, 4 }, - { -66, 5 }, { -61, 6 }, { -67, 7 }, { -60, 8 }, - { -68, 9 }, { 10, 11 }, { -69, -59 }, { 12, 13 }, - { -70, -58 }, { 14, 28 }, { 15, 21 }, { 16, 18 }, - { -57, 17 }, { -71, -56 }, { 19, 20 }, { -88, -87 }, - { -86, -85 }, { 22, 25 }, { 23, 24 }, { -84, -83 }, - { -82, -81 }, { 26, 27 }, { -80, -79 }, { -78, -77 }, - { 29, 36 }, { 30, 33 }, { 31, 32 }, { -76, -75 }, - { -74, -73 }, { 34, 35 }, { -72, -55 }, { -54, -53 }, - { 37, 41 }, { 38, 39 }, { -52, -51 }, { -50, 40 }, - { -49, -48 }, { 42, 45 }, { 43, 44 }, { -47, -46 }, - { -45, -44 }, { 46, 47 }, { -43, -42 }, { -41, -40 } -}; - -const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvBalance10F[48][2] = { - { -64, 1 }, { -65, 2 }, { -63, 3 }, { -66, 4 }, - { -62, 5 }, { -61, 6 }, { -67, 7 }, { -68, 8 }, - { -60, 9 }, { 10, 11 }, { -69, -59 }, { -70, 12 }, - { -58, 13 }, { 14, 17 }, { -71, 15 }, { -57, 16 }, - { -56, -73 }, { 18, 32 }, { 19, 25 }, { 20, 22 }, - { -72, 21 }, { -88, -87 }, { 23, 24 }, { -86, -85 }, - { -84, -83 }, { 26, 29 }, { 27, 28 }, { -82, -81 }, - { -80, -79 }, { 30, 31 }, { -78, -77 }, { -76, -75 }, - { 33, 40 }, { 34, 37 }, { 35, 36 }, { -74, -55 }, - { -54, -53 }, { 38, 39 }, { -52, -51 }, { -50, -49 }, - { 41, 44 }, { 42, 43 }, { -48, -47 }, { -46, -45 }, - { 45, 46 }, { -44, -43 }, { -42, 47 }, { -41, -40 } -}; - -const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvLevel11T[62][2] = { - { -64, 1 }, { -65, 2 }, { -63, 3 }, { -66, 4 }, - { -62, 5 }, { -67, 6 }, { -61, 7 }, { -68, 8 }, - { -60, 9 }, { 10, 11 }, { -69, -59 }, { 12, 14 }, - { -70, 13 }, { -71, -58 }, { 15, 18 }, { 16, 17 }, - { -72, -57 }, { -73, -74 }, { 19, 22 }, { -56, 20 }, - { -55, 21 }, { -54, -77 }, { 23, 31 }, { 24, 25 }, - { -75, -76 }, { 26, 27 }, { -78, -53 }, { 28, 29 }, - { -52, -95 }, { -94, 30 }, { -93, -92 }, { 32, 47 }, - { 33, 40 }, { 34, 37 }, { 35, 36 }, { -91, -90 }, - { -89, -88 }, { 38, 39 }, { -87, -86 }, { -85, -84 }, - { 41, 44 }, { 42, 43 }, { -83, -82 }, { -81, -80 }, - { 45, 46 }, { -79, -51 }, { -50, -49 }, { 48, 55 }, - { 49, 52 }, { 50, 51 }, { -48, -47 }, { -46, -45 }, - { 53, 54 }, { -44, -43 }, { -42, -41 }, { 56, 59 }, - { 57, 58 }, { -40, -39 }, { -38, -37 }, { 60, 61 }, - { -36, -35 }, { -34, -33 } -}; - -const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvLevel11F[62][2] = { - { -64, 1 }, { -65, 2 }, { -63, 3 }, { -66, 4 }, - { -62, 5 }, { -67, 6 }, { 7, 8 }, { -61, -68 }, - { 9, 10 }, { -60, -69 }, { 11, 12 }, { -59, -70 }, - { 13, 14 }, { -58, -71 }, { 15, 16 }, { -57, -72 }, - { 17, 19 }, { -56, 18 }, { -55, -73 }, { 20, 24 }, - { 21, 22 }, { -74, -54 }, { -53, 23 }, { -75, -76 }, - { 25, 30 }, { 26, 27 }, { -52, -51 }, { 28, 29 }, - { -77, -79 }, { -50, -49 }, { 31, 39 }, { 32, 35 }, - { 33, 34 }, { -78, -46 }, { -82, -88 }, { 36, 37 }, - { -83, -48 }, { -47, 38 }, { -86, -85 }, { 40, 47 }, - { 41, 44 }, { 42, 43 }, { -80, -44 }, { -43, -42 }, - { 45, 46 }, { -39, -87 }, { -84, -40 }, { 48, 55 }, - { 49, 52 }, { 50, 51 }, { -95, -94 }, { -93, -92 }, - { 53, 54 }, { -91, -90 }, { -89, -81 }, { 56, 59 }, - { 57, 58 }, { -45, -41 }, { -38, -37 }, { 60, 61 }, - { -36, -35 }, { -34, -33 } -}; - -const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvBalance11T[24][2] = { - { -64, 1 }, { -63, 2 }, { -65, 3 }, { -66, 4 }, - { -62, 5 }, { -61, 6 }, { -67, 7 }, { -68, 8 }, - { -60, 9 }, { 10, 16 }, { 11, 13 }, { -69, 12 }, - { -76, -75 }, { 14, 15 }, { -74, -73 }, { -72, -71 }, - { 17, 20 }, { 18, 19 }, { -70, -59 }, { -58, -57 }, - { 21, 22 }, { -56, -55 }, { -54, 23 }, { -53, -52 } -}; - -const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvBalance11F[24][2] = { - { -64, 1 }, { -65, 2 }, { -63, 3 }, { -66, 4 }, - { -62, 5 }, { -61, 6 }, { -67, 7 }, { -68, 8 }, - { -60, 9 }, { 10, 13 }, { -69, 11 }, { -59, 12 }, - { -58, -76 }, { 14, 17 }, { 15, 16 }, { -75, -74 }, - { -73, -72 }, { 18, 21 }, { 19, 20 }, { -71, -70 }, - { -57, -56 }, { 22, 23 }, { -55, -54 }, { -53, -52 } -}; - -const SCHAR FDK_sbrDecoder_sbr_huffBook_NoiseLevel11T[62][2] = { - { -64, 1 }, { -63, 2 }, { -65, 3 }, { -66, 4 }, - { -62, 5 }, { -67, 6 }, { 7, 8 }, { -61, -68 }, - { 9, 30 }, { 10, 15 }, { -60, 11 }, { -69, 12 }, - { 13, 14 }, { -59, -53 }, { -95, -94 }, { 16, 23 }, - { 17, 20 }, { 18, 19 }, { -93, -92 }, { -91, -90 }, - { 21, 22 }, { -89, -88 }, { -87, -86 }, { 24, 27 }, - { 25, 26 }, { -85, -84 }, { -83, -82 }, { 28, 29 }, - { -81, -80 }, { -79, -78 }, { 31, 46 }, { 32, 39 }, - { 33, 36 }, { 34, 35 }, { -77, -76 }, { -75, -74 }, - { 37, 38 }, { -73, -72 }, { -71, -70 }, { 40, 43 }, - { 41, 42 }, { -58, -57 }, { -56, -55 }, { 44, 45 }, - { -54, -52 }, { -51, -50 }, { 47, 54 }, { 48, 51 }, - { 49, 50 }, { -49, -48 }, { -47, -46 }, { 52, 53 }, - { -45, -44 }, { -43, -42 }, { 55, 58 }, { 56, 57 }, - { -41, -40 }, { -39, -38 }, { 59, 60 }, { -37, -36 }, - { -35, 61 }, { -34, -33 } -}; - -const SCHAR FDK_sbrDecoder_sbr_huffBook_NoiseBalance11T[24][2] = { - { -64, 1 }, { -65, 2 }, { -63, 3 }, { 4, 9 }, - { -66, 5 }, { -62, 6 }, { 7, 8 }, { -76, -75 }, - { -74, -73 }, { 10, 17 }, { 11, 14 }, { 12, 13 }, - { -72, -71 }, { -70, -69 }, { 15, 16 }, { -68, -67 }, - { -61, -60 }, { 18, 21 }, { 19, 20 }, { -59, -58 }, - { -57, -56 }, { 22, 23 }, { -55, -54 }, { -53, -52 } -}; -//@} - - - - -/*! - \name parametric stereo - \brief constants used by the parametric stereo part of the decoder - -*/ - - -/* constants used in psbitdec.cpp */ - -/* FIX_BORDER can have 0, 1, 2, 4 envelopes */ -const UCHAR FDK_sbrDecoder_aFixNoEnvDecode[4] = {0, 1, 2, 4}; - - -/* IID & ICC Huffman codebooks */ -const SCHAR aBookPsIidTimeDecode[28][2] = { - { -64, 1 }, { -65, 2 }, { -63, 3 }, { -66, 4 }, - { -62, 5 }, { -67, 6 }, { -61, 7 }, { -68, 8 }, - { -60, 9 }, { -69, 10 }, { -59, 11 }, { -70, 12 }, - { -58, 13 }, { -57, 14 }, { -71, 15 }, { 16, 17 }, - { -56, -72 }, { 18, 21 }, { 19, 20 }, { -55, -78 }, - { -77, -76 }, { 22, 25 }, { 23, 24 }, { -75, -74 }, - { -73, -54 }, { 26, 27 }, { -53, -52 }, { -51, -50 } -}; - -const SCHAR aBookPsIidFreqDecode[28][2] = { - { -64, 1 }, { 2, 3 }, { -63, -65 }, { 4, 5 }, - { -62, -66 }, { 6, 7 }, { -61, -67 }, { 8, 9 }, - { -68, -60 }, { -59, 10 }, { -69, 11 }, { -58, 12 }, - { -70, 13 }, { -71, 14 }, { -57, 15 }, { 16, 17 }, - { -56, -72 }, { 18, 19 }, { -55, -54 }, { 20, 21 }, - { -73, -53 }, { 22, 24 }, { -74, 23 }, { -75, -78 }, - { 25, 26 }, { -77, -76 }, { -52, 27 }, { -51, -50 } -}; - -const SCHAR aBookPsIccTimeDecode[14][2] = { - { -64, 1 }, { -63, 2 }, { -65, 3 }, { -62, 4 }, - { -66, 5 }, { -61, 6 }, { -67, 7 }, { -60, 8 }, - { -68, 9 }, { -59, 10 }, { -69, 11 }, { -58, 12 }, - { -70, 13 }, { -71, -57 } -}; - -const SCHAR aBookPsIccFreqDecode[14][2] = { - { -64, 1 }, { -63, 2 }, { -65, 3 }, { -62, 4 }, - { -66, 5 }, { -61, 6 }, { -67, 7 }, { -60, 8 }, - { -59, 9 }, { -68, 10 }, { -58, 11 }, { -69, 12 }, - { -57, 13 }, { -70, -71 } -}; - -/* IID-fine Huffman codebooks */ - -const SCHAR aBookPsIidFineTimeDecode[60][2] = { - { 1, -64 }, { -63, 2 }, { 3, -65 }, { 4, 59 }, - { 5, 7 }, { 6, -67 }, { -68, -60 }, { -61, 8 }, - { 9, 11 }, { -59, 10 }, { -70, -58 }, { 12, 41 }, - { 13, 20 }, { 14, -71 }, { -55, 15 }, { -53, 16 }, - { 17, -77 }, { 18, 19 }, { -85, -84 }, { -46, -45 }, - { -57, 21 }, { 22, 40 }, { 23, 29 }, { -51, 24 }, - { 25, 26 }, { -83, -82 }, { 27, 28 }, { -90, -38 }, - { -92, -91 }, { 30, 37 }, { 31, 34 }, { 32, 33 }, - { -35, -34 }, { -37, -36 }, { 35, 36 }, { -94, -93 }, - { -89, -39 }, { 38, -79 }, { 39, -81 }, { -88, -40 }, - { -74, -54 }, { 42, -69 }, { 43, 44 }, { -72, -56 }, - { 45, 52 }, { 46, 50 }, { 47, -76 }, { -49, 48 }, - { -47, 49 }, { -87, -41 }, { -52, 51 }, { -78, -50 }, - { 53, -73 }, { 54, -75 }, { 55, 57 }, { 56, -80 }, - { -86, -42 }, { -48, 58 }, { -44, -43 }, { -66, -62 } -}; - - -const SCHAR aBookPsIidFineFreqDecode[60][2] = { - { 1, -64 }, { 2, 4 }, { 3, -65 }, { -66, -62 }, - { -63, 5 }, { 6, 7 }, { -67, -61 }, { 8, 9 }, - { -68, -60 }, { 10, 11 }, { -69, -59 }, { 12, 13 }, - { -70, -58 }, { 14, 18 }, { -57, 15 }, { 16, -72 }, - { -54, 17 }, { -75, -53 }, { 19, 37 }, { -56, 20 }, - { 21, -73 }, { 22, 29 }, { 23, -76 }, { 24, -78 }, - { 25, 28 }, { 26, 27 }, { -85, -43 }, { -83, -45 }, - { -81, -47 }, { -52, 30 }, { -50, 31 }, { 32, -79 }, - { 33, 34 }, { -82, -46 }, { 35, 36 }, { -90, -89 }, - { -92, -91 }, { 38, -71 }, { -55, 39 }, { 40, -74 }, - { 41, 50 }, { 42, -77 }, { -49, 43 }, { 44, 47 }, - { 45, 46 }, { -86, -42 }, { -88, -87 }, { 48, 49 }, - { -39, -38 }, { -41, -40 }, { -51, 51 }, { 52, 59 }, - { 53, 56 }, { 54, 55 }, { -35, -34 }, { -37, -36 }, - { 57, 58 }, { -94, -93 }, { -84, -44 }, { -80, -48 } -}; - -/* constants used in psdec.cpp */ - -const FIXP_DBL decayScaleFactTable[64] = { - - FL2FXCONST_DBL(1.000000), FL2FXCONST_DBL(1.000000), FL2FXCONST_DBL(1.000000), FL2FXCONST_DBL(1.000000), - FL2FXCONST_DBL(0.950000), FL2FXCONST_DBL(0.900000), FL2FXCONST_DBL(0.850000), FL2FXCONST_DBL(0.800000), - FL2FXCONST_DBL(0.750000), FL2FXCONST_DBL(0.700000), FL2FXCONST_DBL(0.650000), FL2FXCONST_DBL(0.600000), - FL2FXCONST_DBL(0.550000), FL2FXCONST_DBL(0.500000), FL2FXCONST_DBL(0.450000), FL2FXCONST_DBL(0.400000), - FL2FXCONST_DBL(0.350000), FL2FXCONST_DBL(0.300000), FL2FXCONST_DBL(0.250000), FL2FXCONST_DBL(0.200000), - FL2FXCONST_DBL(0.150000), FL2FXCONST_DBL(0.100000), FL2FXCONST_DBL(0.050000), FL2FXCONST_DBL(0.000000), - FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), - FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), - FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), - FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), - FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), - FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), - FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), - FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), - FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), - FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000) }; - -/* the values of the following 3 tables are shiftet right by 1 ! */ -const FIXP_DBL ScaleFactors[NO_IID_LEVELS] = { - - 0x5a5ded00, 0x59cd0400, 0x58c29680, 0x564c2e80, 0x52a3d480, - 0x4c8be080, 0x46df3080, 0x40000000, 0x384ba5c0, 0x304c2980, - 0x24e9f640, 0x1b4a2940, 0x11b5c0a0, 0x0b4e2540, 0x0514ea90 -}; - -const FIXP_DBL ScaleFactorsFine[NO_IID_LEVELS_FINE] = { - - 0x5a825c00, 0x5a821c00, 0x5a815100, 0x5a7ed000, 0x5a76e600, - 0x5a5ded00, 0x5a39b880, 0x59f1fd00, 0x5964d680, 0x5852ca00, - 0x564c2e80, 0x54174480, 0x50ea7500, 0x4c8be080, 0x46df3080, - 0x40000000, 0x384ba5c0, 0x304c2980, 0x288dd240, 0x217a2900, - 0x1b4a2940, 0x13c5ece0, 0x0e2b0090, 0x0a178ef0, 0x072ab798, - 0x0514ea90, 0x02dc5944, 0x019bf87c, 0x00e7b173, 0x00824b8b, - 0x00494568 -}; -const FIXP_DBL Alphas[NO_ICC_LEVELS] = { - - 0x00000000, 0x0b6b5be0, 0x12485f80, 0x1da2fa40, - 0x2637ebc0, 0x3243f6c0, 0x466b7480, 0x6487ed80 -}; - -#if defined(ARCH_PREFER_MULT_32x16) -#define FIXP_PS FIXP_SGL -#define FXP_CAST(a) FX_DBL2FX_SGL((FIXP_DBL)a) -#define FL2FXCONST_PS FL2FXCONST_SGL -#else -#define FIXP_PS FIXP_DBL -#define FXP_CAST(x) ((FIXP_DBL)(x)) -#define FL2FXCONST_PS FL2FXCONST_DBL -#endif - -const FIXP_PS aAllpassLinkDecaySer[NO_SERIAL_ALLPASS_LINKS] = { -FXP_CAST(0x53625b00), FXP_CAST(0x4848af00), FXP_CAST(0x3ea94d00) }; - -const FIXP_PS aaFractDelayPhaseFactorReQmf[NO_QMF_CHANNELS] = { -FXP_CAST(0x68b92180), FXP_CAST(0xde396900), FXP_CAST(0x80650380), FXP_CAST(0xcb537e40), FXP_CAST(0x5beb8f00), FXP_CAST(0x72f29200), FXP_CAST(0xf1f43c50), FXP_CAST(0x83896280), -FXP_CAST(0xb9b99c00), FXP_CAST(0x4cda8f00), FXP_CAST(0x7a576e00), FXP_CAST(0x060799e0), FXP_CAST(0x89be5280), FXP_CAST(0xa9dab600), FXP_CAST(0x3be51b00), FXP_CAST(0x7eb91900), -FXP_CAST(0x19f4f540), FXP_CAST(0x92dcb380), FXP_CAST(0x9c1ad700), FXP_CAST(0x29761940), FXP_CAST(0x7ffbf500), FXP_CAST(0x2d3eb180), FXP_CAST(0x9eab0a00), FXP_CAST(0x90d0aa80), -FXP_CAST(0x1601bcc0), FXP_CAST(0x7e180e80), FXP_CAST(0x3f6b3940), FXP_CAST(0xacdeeb00), FXP_CAST(0x88435b00), FXP_CAST(0x0202a768), FXP_CAST(0x79194f80), FXP_CAST(0x5007fd00), -FXP_CAST(0xbd1ecf00), FXP_CAST(0x82a8d100), FXP_CAST(0xedf6e5e0), FXP_CAST(0x711f3500), FXP_CAST(0x5eac4480), FXP_CAST(0xcf0447c0), FXP_CAST(0x80245f80), FXP_CAST(0xda5cd4c0), -FXP_CAST(0x665c0800), FXP_CAST(0x6afbc500), FXP_CAST(0xe21e85e0), FXP_CAST(0x80c5e500), FXP_CAST(0xc7b003c0), FXP_CAST(0x59139f80), FXP_CAST(0x74a8e400), FXP_CAST(0xf5f51f40), -FXP_CAST(0x84896680), FXP_CAST(0xb6662b00), FXP_CAST(0x4999b600), FXP_CAST(0x7b76a300), FXP_CAST(0x0a0b0650), FXP_CAST(0x8b572b80), FXP_CAST(0xa6ec4580), FXP_CAST(0x384fda80), -FXP_CAST(0x7f3a1f00), FXP_CAST(0x1de19ec0), FXP_CAST(0x95045000), FXP_CAST(0x99a3e180), FXP_CAST(0x25a30740), FXP_CAST(0x7fdb9e80), FXP_CAST(0x30fbdb00), FXP_CAST(0xa153d500) }; - -const FIXP_PS aaFractDelayPhaseFactorImQmf[NO_QMF_CHANNELS] = { -FXP_CAST(0xb6663a80), FXP_CAST(0x84896200), FXP_CAST(0xf5f50c70), FXP_CAST(0x74a8dc80), FXP_CAST(0x5913ad00), FXP_CAST(0xc7b01480), FXP_CAST(0x80c5e300), FXP_CAST(0xe21e73a0), -FXP_CAST(0x6afbba80), FXP_CAST(0x665c1380), FXP_CAST(0xda5ce6c0), FXP_CAST(0x80246080), FXP_CAST(0xcf043640), FXP_CAST(0x5eac3800), FXP_CAST(0x711f3e00), FXP_CAST(0xedf6f8a0), -FXP_CAST(0x82a8d500), FXP_CAST(0xbd1ebe80), FXP_CAST(0x5007ee00), FXP_CAST(0x79195580), FXP_CAST(0x0202ba40), FXP_CAST(0x88436180), FXP_CAST(0xacdedc80), FXP_CAST(0x3f6b28c0), -FXP_CAST(0x7e181180), FXP_CAST(0x1601cf40), FXP_CAST(0x90d0b380), FXP_CAST(0x9eaafd80), FXP_CAST(0x2d3e9fc0), FXP_CAST(0x7ffbf580), FXP_CAST(0x29762b00), FXP_CAST(0x9c1ae280), -FXP_CAST(0x92dca980), FXP_CAST(0x19f4e2c0), FXP_CAST(0x7eb91680), FXP_CAST(0x3be52b80), FXP_CAST(0xa9dac400), FXP_CAST(0x89be4b80), FXP_CAST(0x06078710), FXP_CAST(0x7a576880), -FXP_CAST(0x4cda9e00), FXP_CAST(0xb9b9ac00), FXP_CAST(0x83895e00), FXP_CAST(0xf1f42990), FXP_CAST(0x72f28a00), FXP_CAST(0x5beb9c00), FXP_CAST(0xcb538f40), FXP_CAST(0x80650200), -FXP_CAST(0xde3956c0), FXP_CAST(0x68b91680), FXP_CAST(0x68b92c00), FXP_CAST(0xde397b40), FXP_CAST(0x80650500), FXP_CAST(0xcb536d00), FXP_CAST(0x5beb8180), FXP_CAST(0x72f29a80), -FXP_CAST(0xf1f44f10), FXP_CAST(0x83896700), FXP_CAST(0xb9b98c80), FXP_CAST(0x4cda8000), FXP_CAST(0x7a577380), FXP_CAST(0x0607acb8), FXP_CAST(0x89be5a00), FXP_CAST(0xa9daa800) }; - -const FIXP_PS aaFractDelayPhaseFactorReSubQmf20[NO_SUB_QMF_CHANNELS] = { -FXP_CAST(0x7e807380), FXP_CAST(0x72b9bb00), FXP_CAST(0x5c44ee80), FXP_CAST(0x3d3938c0), FXP_CAST(0x80000000), FXP_CAST(0x80000000), -FXP_CAST(0x72b9bb00), FXP_CAST(0x7e807380), FXP_CAST(0xba914700), FXP_CAST(0x050677b0), FXP_CAST(0x895cc380), FXP_CAST(0x834e4900) }; - -const FIXP_PS aaFractDelayPhaseFactorImSubQmf20[NO_SUB_QMF_CHANNELS] = { -FXP_CAST(0xec791720), FXP_CAST(0xc73ca080), FXP_CAST(0xa748ea00), FXP_CAST(0x8f976980), FXP_CAST(0x00000000), FXP_CAST(0x00000000), -FXP_CAST(0x38c35f80), FXP_CAST(0x1386e8e0), FXP_CAST(0x9477d000), FXP_CAST(0x80194380), FXP_CAST(0xcff26140), FXP_CAST(0x1ce70d40) }; - -const FIXP_PS aaFractDelayPhaseFactorSerReQmf[NO_QMF_CHANNELS][NO_SERIAL_ALLPASS_LINKS] = { -{FXP_CAST(0x63e52480), FXP_CAST(0x30fbc540), FXP_CAST(0x6d73af00)}, {FXP_CAST(0xc7b01280), FXP_CAST(0x89be5100), FXP_CAST(0xf7c31cb0)}, {FXP_CAST(0x83896200), FXP_CAST(0x7641af00), FXP_CAST(0x8aee2700)}, -{FXP_CAST(0x0202b330), FXP_CAST(0xcf043ac0), FXP_CAST(0x9bfab500)}, {FXP_CAST(0x7d572c80), FXP_CAST(0xcf043ac0), FXP_CAST(0x1893b960)}, {FXP_CAST(0x34ac7fc0), FXP_CAST(0x7641af00), FXP_CAST(0x7abf7980)}, -{FXP_CAST(0x99a3ee00), FXP_CAST(0x89be5100), FXP_CAST(0x58eead80)}, {FXP_CAST(0x9eab0580), FXP_CAST(0x30fbc540), FXP_CAST(0xd77dae40)}, {FXP_CAST(0x3be52140), FXP_CAST(0x30fbc540), FXP_CAST(0x819b8500)}, -{FXP_CAST(0x7b769d80), FXP_CAST(0x89be5100), FXP_CAST(0xb3a12280)}, {FXP_CAST(0xf9f86878), FXP_CAST(0x7641af00), FXP_CAST(0x37c519c0)}, {FXP_CAST(0x81e7ef80), FXP_CAST(0xcf043ac0), FXP_CAST(0x7ff16880)}, -{FXP_CAST(0xcf043cc0), FXP_CAST(0xcf043ac0), FXP_CAST(0x3e8b2340)}, {FXP_CAST(0x68b92280), FXP_CAST(0x7641af00), FXP_CAST(0xb9e4a900)}, {FXP_CAST(0x5eac3980), FXP_CAST(0x89be5100), FXP_CAST(0x80a05200)}, -{FXP_CAST(0xc094cd00), FXP_CAST(0x30fbc540), FXP_CAST(0xd051dc80)}, {FXP_CAST(0x85a89400), FXP_CAST(0x30fbc540), FXP_CAST(0x53483b00)}, {FXP_CAST(0x0a0af5e0), FXP_CAST(0x89be5100), FXP_CAST(0x7cb1b680)}, -{FXP_CAST(0x7eb91900), FXP_CAST(0x7641af00), FXP_CAST(0x2006e8c0)}, {FXP_CAST(0x2d3ea680), FXP_CAST(0xcf043ac0), FXP_CAST(0xa0ec1c00)}, {FXP_CAST(0x95044180), FXP_CAST(0xcf043ac0), FXP_CAST(0x880d2180)}, -{FXP_CAST(0xa4147300), FXP_CAST(0x7641af00), FXP_CAST(0xf0282870)}, {FXP_CAST(0x42e13f80), FXP_CAST(0x89be5100), FXP_CAST(0x694c4a00)}, {FXP_CAST(0x79195200), FXP_CAST(0x30fbc540), FXP_CAST(0x71374780)}, -{FXP_CAST(0xf1f43550), FXP_CAST(0x30fbc540), FXP_CAST(0xff6593ea)}, {FXP_CAST(0x80c5e280), FXP_CAST(0x89be5100), FXP_CAST(0x8e39ec00)}, {FXP_CAST(0xd689e480), FXP_CAST(0x7641af00), FXP_CAST(0x97648100)}, -{FXP_CAST(0x6d235300), FXP_CAST(0xcf043ac0), FXP_CAST(0x110a20c0)}, {FXP_CAST(0x5913a800), FXP_CAST(0xcf043ac0), FXP_CAST(0x785d4f80)}, {FXP_CAST(0xb9b99a00), FXP_CAST(0x7641af00), FXP_CAST(0x5e440880)}, -{FXP_CAST(0x88436100), FXP_CAST(0x89be5100), FXP_CAST(0xdece7000)}, {FXP_CAST(0x12091320), FXP_CAST(0x30fbc540), FXP_CAST(0x8309f800)}, {FXP_CAST(0x7f9afd00), FXP_CAST(0x30fbc540), FXP_CAST(0xada33f00)}, -{FXP_CAST(0x25a31700), FXP_CAST(0x89be5100), FXP_CAST(0x30cc3600)}, {FXP_CAST(0x90d0ab80), FXP_CAST(0x7641af00), FXP_CAST(0x7f7cbe80)}, {FXP_CAST(0xa9dabf00), FXP_CAST(0xcf043ac0), FXP_CAST(0x45182580)}, -{FXP_CAST(0x4999cb80), FXP_CAST(0xcf043ac0), FXP_CAST(0xc0681c80)}, {FXP_CAST(0x7641ac80), FXP_CAST(0x7641af00), FXP_CAST(0x80194380)}, {FXP_CAST(0xe9fe3300), FXP_CAST(0x89be5100), FXP_CAST(0xc95184c0)}, -{FXP_CAST(0x80246000), FXP_CAST(0x30fbc540), FXP_CAST(0x4d55d800)}, {FXP_CAST(0xde396fc0), FXP_CAST(0x30fbc540), FXP_CAST(0x7e324000)}, {FXP_CAST(0x711f3f00), FXP_CAST(0x89be5100), FXP_CAST(0x275ce480)}, -{FXP_CAST(0x53211700), FXP_CAST(0x7641af00), FXP_CAST(0xa6343580)}, {FXP_CAST(0xb3256780), FXP_CAST(0xcf043ac0), FXP_CAST(0x85997b80)}, {FXP_CAST(0x8b572680), FXP_CAST(0xcf043ac0), FXP_CAST(0xe89ba660)}, -{FXP_CAST(0x19f4f780), FXP_CAST(0x7641af00), FXP_CAST(0x64c4e100)}, {FXP_CAST(0x7ffbf580), FXP_CAST(0x89be5100), FXP_CAST(0x7493a380)}, {FXP_CAST(0x1de18100), FXP_CAST(0x30fbc540), FXP_CAST(0x070897f0)}, -{FXP_CAST(0x8d0d6a80), FXP_CAST(0x30fbc540), FXP_CAST(0x91ed6f00)}, {FXP_CAST(0xaff81380), FXP_CAST(0x89be5100), FXP_CAST(0x932db000)}, {FXP_CAST(0x5007fb00), FXP_CAST(0x7641af00), FXP_CAST(0x0970feb0)}, -{FXP_CAST(0x72f28d00), FXP_CAST(0xcf043ac0), FXP_CAST(0x758d6500)}, {FXP_CAST(0xe21e6cc0), FXP_CAST(0xcf043ac0), FXP_CAST(0x63436f80)}, {FXP_CAST(0x80040b00), FXP_CAST(0x7641af00), FXP_CAST(0xe63d7600)}, -{FXP_CAST(0xe60b1ae0), FXP_CAST(0x89be5100), FXP_CAST(0x84ea5c80)}, {FXP_CAST(0x74a8e100), FXP_CAST(0x30fbc540), FXP_CAST(0xa7f07500)}, {FXP_CAST(0x4cda8980), FXP_CAST(0x30fbc540), FXP_CAST(0x29a6d340)}, -{FXP_CAST(0xacdeda80), FXP_CAST(0x89be5100), FXP_CAST(0x7e93d600)}, {FXP_CAST(0x8ee0c980), FXP_CAST(0x7641af00), FXP_CAST(0x4b662680)}, {FXP_CAST(0x21c6a280), FXP_CAST(0xcf043ac0), FXP_CAST(0xc7258c80)}, -{FXP_CAST(0x7fdb9f00), FXP_CAST(0xcf043ac0), FXP_CAST(0x8006d500)}, {FXP_CAST(0x1601ba60), FXP_CAST(0x7641af00), FXP_CAST(0xc2830940)}, {FXP_CAST(0x89be4c80), FXP_CAST(0x89be5100), FXP_CAST(0x471cf100)}, -{FXP_CAST(0xb6664400), FXP_CAST(0x30fbc540), FXP_CAST(0x7f3fb800)}}; - -const FIXP_PS aaFractDelayPhaseFactorSerImQmf[NO_QMF_CHANNELS][NO_SERIAL_ALLPASS_LINKS] = { -{FXP_CAST(0xaff80c80), FXP_CAST(0x89be5100), FXP_CAST(0xbda29e00)}, {FXP_CAST(0x8d0d6f00), FXP_CAST(0x30fbc540), FXP_CAST(0x8043ee80)}, {FXP_CAST(0x1de18a20), FXP_CAST(0x30fbc540), FXP_CAST(0xcc3e7840)}, -{FXP_CAST(0x7ffbf500), FXP_CAST(0x89be5100), FXP_CAST(0x4fdfc180)}, {FXP_CAST(0x19f4ee40), FXP_CAST(0x7641af00), FXP_CAST(0x7d9e4c00)}, {FXP_CAST(0x8b572300), FXP_CAST(0xcf043ac0), FXP_CAST(0x244a2940)}, -{FXP_CAST(0xb3256f00), FXP_CAST(0xcf043ac0), FXP_CAST(0xa3f0a500)}, {FXP_CAST(0x53211e00), FXP_CAST(0x7641af00), FXP_CAST(0x86944500)}, {FXP_CAST(0x711f3a80), FXP_CAST(0x89be5100), FXP_CAST(0xebc72040)}, -{FXP_CAST(0xde3966c0), FXP_CAST(0x30fbc540), FXP_CAST(0x66b87e00)}, {FXP_CAST(0x80246080), FXP_CAST(0x30fbc540), FXP_CAST(0x73362c00)}, {FXP_CAST(0xe9fe3c40), FXP_CAST(0x89be5100), FXP_CAST(0x03d1d110)}, -{FXP_CAST(0x7641b000), FXP_CAST(0x7641af00), FXP_CAST(0x90520c80)}, {FXP_CAST(0x4999c380), FXP_CAST(0xcf043ac0), FXP_CAST(0x94e80a80)}, {FXP_CAST(0xa9dab800), FXP_CAST(0xcf043ac0), FXP_CAST(0x0ca570e0)}, -{FXP_CAST(0x90d0b000), FXP_CAST(0x7641af00), FXP_CAST(0x76c9bc80)}, {FXP_CAST(0x25a32000), FXP_CAST(0x89be5100), FXP_CAST(0x61338500)}, {FXP_CAST(0x7f9afc80), FXP_CAST(0x30fbc540), FXP_CAST(0xe318f060)}, -{FXP_CAST(0x120909c0), FXP_CAST(0x30fbc540), FXP_CAST(0x84124e00)}, {FXP_CAST(0x88435d80), FXP_CAST(0x89be5100), FXP_CAST(0xaa4d2f80)}, {FXP_CAST(0xb9b9a200), FXP_CAST(0x7641af00), FXP_CAST(0x2cae1800)}, -{FXP_CAST(0x5913ae80), FXP_CAST(0xcf043ac0), FXP_CAST(0x7f040680)}, {FXP_CAST(0x6d234e00), FXP_CAST(0xcf043ac0), FXP_CAST(0x48c6a100)}, {FXP_CAST(0xd689db80), FXP_CAST(0x7641af00), FXP_CAST(0xc44860c0)}, -{FXP_CAST(0x80c5e380), FXP_CAST(0x89be5100), FXP_CAST(0x80005d00)}, {FXP_CAST(0xf1f43eb0), FXP_CAST(0x30fbc540), FXP_CAST(0xc55a3a00)}, {FXP_CAST(0x79195500), FXP_CAST(0x30fbc540), FXP_CAST(0x49c3de00)}, -{FXP_CAST(0x42e13700), FXP_CAST(0x89be5100), FXP_CAST(0x7edc5b00)}, {FXP_CAST(0xa4146c80), FXP_CAST(0x7641af00), FXP_CAST(0x2b8c2c00)}, {FXP_CAST(0x95044680), FXP_CAST(0xcf043ac0), FXP_CAST(0xa968c100)}, -{FXP_CAST(0x2d3eaf40), FXP_CAST(0xcf043ac0), FXP_CAST(0x8460fd80)}, {FXP_CAST(0x7eb91780), FXP_CAST(0x7641af00), FXP_CAST(0xe44621e0)}, {FXP_CAST(0x0a0aec80), FXP_CAST(0x89be5100), FXP_CAST(0x61fb5c00)}, -{FXP_CAST(0x85a89100), FXP_CAST(0x30fbc540), FXP_CAST(0x76555780)}, {FXP_CAST(0xc094d500), FXP_CAST(0x30fbc540), FXP_CAST(0x0b71f790)}, {FXP_CAST(0x5eac4000), FXP_CAST(0x89be5100), FXP_CAST(0x94401a80)}, -{FXP_CAST(0x68b91d80), FXP_CAST(0x7641af00), FXP_CAST(0x90ea3980)}, {FXP_CAST(0xcf043440), FXP_CAST(0xcf043ac0), FXP_CAST(0x05067a08)}, {FXP_CAST(0x81e7f180), FXP_CAST(0xcf043ac0), FXP_CAST(0x73bb6d00)}, -{FXP_CAST(0xf9f871e0), FXP_CAST(0x7641af00), FXP_CAST(0x65ff0e00)}, {FXP_CAST(0x7b76a000), FXP_CAST(0x89be5100), FXP_CAST(0xea9664c0)}, {FXP_CAST(0x3be518c0), FXP_CAST(0x30fbc540), FXP_CAST(0x8633e880)}, -{FXP_CAST(0x9eaaff00), FXP_CAST(0x30fbc540), FXP_CAST(0xa4c84500)}, {FXP_CAST(0x99a3f400), FXP_CAST(0x89be5100), FXP_CAST(0x2571eac0)}, {FXP_CAST(0x34ac8840), FXP_CAST(0x7641af00), FXP_CAST(0x7dd82b00)}, -{FXP_CAST(0x7d572a80), FXP_CAST(0xcf043ac0), FXP_CAST(0x4eed8400)}, {FXP_CAST(0x0202a9c4), FXP_CAST(0xcf043ac0), FXP_CAST(0xcb249700)}, {FXP_CAST(0x83896000), FXP_CAST(0x7641af00), FXP_CAST(0x80318200)}, -{FXP_CAST(0xc7b01b00), FXP_CAST(0x89be5100), FXP_CAST(0xbeab7580)}, {FXP_CAST(0x63e52a80), FXP_CAST(0x30fbc540), FXP_CAST(0x4364b700)}, {FXP_CAST(0x63e51f00), FXP_CAST(0x30fbc540), FXP_CAST(0x7fa6bd00)}, -{FXP_CAST(0xc7b00a00), FXP_CAST(0x89be5100), FXP_CAST(0x32a67940)}, {FXP_CAST(0x83896400), FXP_CAST(0x7641af00), FXP_CAST(0xaf2fd200)}, {FXP_CAST(0x0202bc9c), FXP_CAST(0xcf043ac0), FXP_CAST(0x829e6e80)}, -{FXP_CAST(0x7d572e80), FXP_CAST(0xcf043ac0), FXP_CAST(0xdcde6b80)}, {FXP_CAST(0x34ac7700), FXP_CAST(0x7641af00), FXP_CAST(0x5ce4e280)}, {FXP_CAST(0x99a3e880), FXP_CAST(0x89be5100), FXP_CAST(0x79089c00)}, -{FXP_CAST(0x9eab0b80), FXP_CAST(0x30fbc540), FXP_CAST(0x1307ae80)}, {FXP_CAST(0x3be52980), FXP_CAST(0x30fbc540), FXP_CAST(0x98906880)}, {FXP_CAST(0x7b769b00), FXP_CAST(0x89be5100), FXP_CAST(0x8d51b300)}, -{FXP_CAST(0xf9f85f10), FXP_CAST(0x7641af00), FXP_CAST(0xfd62ee24)}, {FXP_CAST(0x81e7ee00), FXP_CAST(0xcf043ac0), FXP_CAST(0x70439680)}, {FXP_CAST(0xcf044580), FXP_CAST(0xcf043ac0), FXP_CAST(0x6a6d9600)}, -{FXP_CAST(0x68b92800), FXP_CAST(0x7641af00), FXP_CAST(0xf2275f80)}}; - -const FIXP_PS aaFractDelayPhaseFactorSerReSubQmf20[NO_SUB_QMF_CHANNELS][NO_SERIAL_ALLPASS_LINKS] = { -{FXP_CAST(0x7e2df000), FXP_CAST(0x7a7d0580), FXP_CAST(0x7ed03e00)}, {FXP_CAST(0x6fec9a80), FXP_CAST(0x5133cc80), FXP_CAST(0x7573df00)}, {FXP_CAST(0x55063900), FXP_CAST(0x0c8bd360), FXP_CAST(0x636c0400)}, -{FXP_CAST(0x3084ca00), FXP_CAST(0xc3a94580), FXP_CAST(0x4a0d6700)}, {FXP_CAST(0x80000000), FXP_CAST(0x80000000), FXP_CAST(0x80000000)}, {FXP_CAST(0x80000000), FXP_CAST(0x80000000), FXP_CAST(0x80000000)}, -{FXP_CAST(0x6fec9a80), FXP_CAST(0x5133cc80), FXP_CAST(0x7573df00)}, {FXP_CAST(0x7e2df000), FXP_CAST(0x7a7d0580), FXP_CAST(0x7ed03e00)}, {FXP_CAST(0xa4c84280), FXP_CAST(0xb8e31300), FXP_CAST(0xd5af0140)}, -{FXP_CAST(0xf0f488a0), FXP_CAST(0x8275a100), FXP_CAST(0x1a72e360)}, {FXP_CAST(0x80aaa680), FXP_CAST(0x471ced00), FXP_CAST(0x9d2ead80)}, {FXP_CAST(0x9477d100), FXP_CAST(0x7d8a5f00), FXP_CAST(0x8151df80)}}; - -const FIXP_PS aaFractDelayPhaseFactorSerImSubQmf20[NO_SUB_QMF_CHANNELS][NO_SERIAL_ALLPASS_LINKS] = { -{FXP_CAST(0xea7d08a0), FXP_CAST(0xdad7f3c0), FXP_CAST(0xee9c9f60)}, {FXP_CAST(0xc1e54140), FXP_CAST(0x9d0dfe80), FXP_CAST(0xcd1e7300)}, {FXP_CAST(0xa051a580), FXP_CAST(0x809dc980), FXP_CAST(0xaf61c400)}, -{FXP_CAST(0x898d4e00), FXP_CAST(0x8f1d3400), FXP_CAST(0x97988280)}, {FXP_CAST(0x00000000), FXP_CAST(0x00000000), FXP_CAST(0x00000000)}, {FXP_CAST(0x00000000), FXP_CAST(0x00000000), FXP_CAST(0x00000000)}, -{FXP_CAST(0x3e1abec0), FXP_CAST(0x62f20180), FXP_CAST(0x32e18d00)}, {FXP_CAST(0x1582f760), FXP_CAST(0x25280c40), FXP_CAST(0x116360a0)}, {FXP_CAST(0xa6343800), FXP_CAST(0x6a6d9880), FXP_CAST(0x87327a00)}, -{FXP_CAST(0x80e32200), FXP_CAST(0xe70747c0), FXP_CAST(0x82c32b00)}, {FXP_CAST(0xf2f42420), FXP_CAST(0x6a6d9880), FXP_CAST(0xaea47080)}, {FXP_CAST(0x456eba00), FXP_CAST(0xe70747c0), FXP_CAST(0xedaa8640)}}; - -const FIXP_PS p8_13_20[13] = -{ - FL2FXCONST_PS(0.00746082949812f), FL2FXCONST_PS(0.02270420949825f), FL2FXCONST_PS(0.04546865930473f), FL2FXCONST_PS(0.07266113929591f), - FL2FXCONST_PS(0.09885108575264f), FL2FXCONST_PS(0.11793710567217f), FL2FXCONST_PS(0.125f ), FL2FXCONST_PS(0.11793710567217f), - FL2FXCONST_PS(0.09885108575264f), FL2FXCONST_PS(0.07266113929591f), FL2FXCONST_PS(0.04546865930473f), FL2FXCONST_PS(0.02270420949825f), - FL2FXCONST_PS(0.00746082949812f) -}; - -const FIXP_PS p2_13_20[13] = -{ - FL2FXCONST_PS(0.0f), FL2FXCONST_PS( 0.01899487526049f), FL2FXCONST_PS(0.0f), FL2FXCONST_PS(-0.07293139167538f), - FL2FXCONST_PS(0.0f), FL2FXCONST_PS( 0.30596630545168f), FL2FXCONST_PS(0.5f), FL2FXCONST_PS( 0.30596630545168f), - FL2FXCONST_PS(0.0f), FL2FXCONST_PS(-0.07293139167538f), FL2FXCONST_PS(0.0f), FL2FXCONST_PS( 0.01899487526049f), - FL2FXCONST_PS(0.0f) -}; - - - -const UCHAR aAllpassLinkDelaySer[] = { 3, 4, 5}; - -const UCHAR delayIndexQmf[NO_QMF_CHANNELS] = { - 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, - 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, - 14, 14, 14, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, - 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1 -}; - -const UCHAR groupBorders20[NO_IID_GROUPS + 1] = -{ - 6, 7, 0, 1, 2, 3, /* 6 subqmf subbands - 0th qmf subband */ - 9, 8, /* 2 subqmf subbands - 1st qmf subband */ - 10, 11, /* 2 subqmf subbands - 2nd qmf subband */ - 3, 4, 5, 6, 7, 8, - 9, 11, 14, 18, 23, 35, 64 -}; - -const UCHAR groupBorders34[NO_IID_GROUPS_HI_RES + 1] = -{ - 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, /* 12 subqmf subbands - 0th qmf subband */ - 12, 13, 14, 15, 16, 17, 18, 19, /* 8 subqmf subbands - 1st qmf subband */ - 20, 21, 22, 23, /* 4 subqmf subbands - 2nd qmf subband */ - 24, 25, 26, 27, /* 4 subqmf subbands - 3nd qmf subband */ - 28, 29, 30, 31, /* 4 subqmf subbands - 4nd qmf subband */ - 32-27, 33-27, 34-27, 35-27, 36-27, 37-27, 38-27, - 40-27, 42-27, 44-27, 46-27, 48-27, 51-27, 54-27, - 57-27, 60-27, 64-27, 68-27, 91-27 -}; - -const UCHAR bins2groupMap20[NO_IID_GROUPS] = -{ - 1, 0, - 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19 -}; - -const UCHAR quantizedIIDs[NO_IID_STEPS] = -{ - 2, 4, 7, 10, 14, 18, 25 -}; -const UCHAR quantizedIIDsFine[NO_IID_STEPS_FINE] = -{ - 2, 4, 6, 8, 10, 13, 16, 19, 22, 25, 30, 35, 40, 45, 50 -}; - -const UCHAR FDK_sbrDecoder_aNoIidBins[3] = {NO_LOW_RES_IID_BINS, - NO_MID_RES_IID_BINS, - NO_HI_RES_IID_BINS}; - -const UCHAR FDK_sbrDecoder_aNoIccBins[3] = {NO_LOW_RES_ICC_BINS, - NO_MID_RES_ICC_BINS, - NO_HI_RES_ICC_BINS}; - - - -/************************************************************************/ -/*! - \brief Create lookup tables for some arithmetic functions - - The tables would normally be defined as const arrays, - but initialization at run time allows to specify their accuracy. -*/ -/************************************************************************/ - -/* 1/x-table: (example for INV_TABLE_BITS 8) - - The table covers an input range from 0.5 to 1.0 with a step size of 1/512, - starting at 0.5 + 1/512. - Each table entry corresponds to an input interval starting 1/1024 below the - exact value and ending 1/1024 above it. - - The table is actually a 0.5/x-table, so that the output range is again - 0.5...1.0 and the exponent of the result must be increased by 1. - - Input range Index in table result - ------------------------------------------------------------------- - 0.500000...0.500976 - 0.5 / 0.500000 = 1.000000 - 0.500976...0.502930 0 0.5 / 0.501953 = 0.996109 - 0.502930...0.500488 1 0.5 / 0.503906 = 0.992248 - ... - 0.999023...1.000000 255 0.5 / 1.000000 = 0.500000 - - for (i=0; i=0.5 - FRACT_BITS-1 zero e.g |max| = 0 -\endverbatim - - Dynamic scaling is used to achieve sufficient accuracy even when the signal - energy is low. The dynamic framing of SBR produces a variable overlap area - where samples from the previous QMF-Analysis are stored. Depending on the - start position and stop position of the current SBR envelopes, the processing - buffer consists of differently scaled regions like illustrated in the below - figure. - - \image html scales.png Scale -*/ - - -#endif diff --git a/libSBRdec/src/sbrdec_drc.cpp b/libSBRdec/src/sbrdec_drc.cpp deleted file mode 100644 index a834c0b..0000000 --- a/libSBRdec/src/sbrdec_drc.cpp +++ /dev/null @@ -1,525 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/***************************** MPEG-4 AAC Decoder ************************** - - Author(s): Christian Griebel - Description: Dynamic range control (DRC) decoder tool for SBR - -******************************************************************************/ - -#include "sbrdec_drc.h" - - -/* DRC - Offset table for QMF interpolation. */ -static const int offsetTab[2][16] = -{ - { 0, 4, 8, 12, 16, 20, 24, 28, 0, 0, 0, 0, 0, 0, 0, 0 }, /* 1024 framing */ - { 0, 4, 8, 12, 16, 19, 22, 26, 0, 0, 0, 0, 0, 0, 0, 0 } /* 960 framing */ -}; - -/*! - \brief Initialize DRC QMF factors - - \hDrcData Handle to DRC channel data. - - \return none -*/ -void sbrDecoder_drcInitChannel ( - HANDLE_SBR_DRC_CHANNEL hDrcData ) -{ - int band; - - if (hDrcData == NULL) { - return; - } - - for (band = 0; band < (64); band++) { - hDrcData->prevFact_mag[band] = FL2FXCONST_DBL(0.5f); - } - - for (band = 0; band < SBRDEC_MAX_DRC_BANDS; band++) { - hDrcData->currFact_mag[band] = FL2FXCONST_DBL(0.5f); - hDrcData->nextFact_mag[band] = FL2FXCONST_DBL(0.5f); - } - - hDrcData->prevFact_exp = 1; - hDrcData->currFact_exp = 1; - hDrcData->nextFact_exp = 1; - - hDrcData->numBandsCurr = 1; - hDrcData->numBandsNext = 1; - - hDrcData->winSequenceCurr = 0; - hDrcData->winSequenceNext = 0; - - hDrcData->drcInterpolationSchemeCurr = 0; - hDrcData->drcInterpolationSchemeNext = 0; - - hDrcData->enable = 0; -} - - -/*! - \brief Swap DRC QMF scaling factors after they have been applied. - - \hDrcData Handle to DRC channel data. - - \return none -*/ -void sbrDecoder_drcUpdateChannel ( - HANDLE_SBR_DRC_CHANNEL hDrcData ) -{ - if (hDrcData == NULL) { - return; - } - if (hDrcData->enable != 1) { - return; - } - - /* swap previous data */ - FDKmemcpy( hDrcData->currFact_mag, - hDrcData->nextFact_mag, - SBRDEC_MAX_DRC_BANDS * sizeof(FIXP_DBL) ); - - hDrcData->currFact_exp = hDrcData->nextFact_exp; - - hDrcData->numBandsCurr = hDrcData->numBandsNext; - - FDKmemcpy( hDrcData->bandTopCurr, - hDrcData->bandTopNext, - SBRDEC_MAX_DRC_BANDS * sizeof(USHORT) ); - - hDrcData->drcInterpolationSchemeCurr = hDrcData->drcInterpolationSchemeNext; - - hDrcData->winSequenceCurr = hDrcData->winSequenceNext; -} - - -/*! - \brief Apply DRC factors slot based. - - \hDrcData Handle to DRC channel data. - \qmfRealSlot Pointer to real valued QMF data of one time slot. - \qmfImagSlot Pointer to the imaginary QMF data of one time slot. - \col Number of the time slot. - \numQmfSubSamples Total number of time slots for one frame. - \scaleFactor Pointer to the out scale factor of the time slot. - - \return None. -*/ -void sbrDecoder_drcApplySlot ( - HANDLE_SBR_DRC_CHANNEL hDrcData, - FIXP_DBL *qmfRealSlot, - FIXP_DBL *qmfImagSlot, - int col, - int numQmfSubSamples, - int maxShift - ) -{ - const int *offset; - - int band, bottomMdct, topMdct, bin, useLP; - int indx = numQmfSubSamples - (numQmfSubSamples >> 1) - 10; /* l_border */ - int frameLenFlag = (numQmfSubSamples == 30) ? 1 : 0; - - const FIXP_DBL *fact_mag = NULL; - INT fact_exp = 0; - UINT numBands = 0; - USHORT *bandTop = NULL; - int shortDrc = 0; - - FIXP_DBL alphaValue = FL2FXCONST_DBL(0.0f); - - if (hDrcData == NULL) { - return; - } - if (hDrcData->enable != 1) { - return; - } - - offset = offsetTab[frameLenFlag]; - - useLP = (qmfImagSlot == NULL) ? 1 : 0; - - col += indx; - bottomMdct = 0; - bin = 0; - - /* get respective data and calc interpolation factor */ - if (col < (numQmfSubSamples>>1)) { /* first half of current frame */ - if (hDrcData->winSequenceCurr != 2) { /* long window */ - int j = col + (numQmfSubSamples>>1); - - if (hDrcData->drcInterpolationSchemeCurr == 0) { - INT k = (frameLenFlag) ? 0x4444444 : 0x4000000; - - alphaValue = (FIXP_DBL)(j * k); - } - else { - if (j >= offset[hDrcData->drcInterpolationSchemeCurr - 1]) { - alphaValue = (FIXP_DBL)MAXVAL_DBL; - } - } - } - else { /* short windows */ - shortDrc = 1; - } - - fact_mag = hDrcData->currFact_mag; - fact_exp = hDrcData->currFact_exp; - numBands = hDrcData->numBandsCurr; - bandTop = hDrcData->bandTopCurr; - } - else if (col < numQmfSubSamples) { /* second half of current frame */ - if (hDrcData->winSequenceNext != 2) { /* next: long window */ - int j = col - (numQmfSubSamples>>1); - - if (hDrcData->drcInterpolationSchemeNext == 0) { - INT k = (frameLenFlag) ? 0x4444444 : 0x4000000; - - alphaValue = (FIXP_DBL)(j * k); - } - else { - if (j >= offset[hDrcData->drcInterpolationSchemeNext - 1]) { - alphaValue = (FIXP_DBL)MAXVAL_DBL; - } - } - - fact_mag = hDrcData->nextFact_mag; - fact_exp = hDrcData->nextFact_exp; - numBands = hDrcData->numBandsNext; - bandTop = hDrcData->bandTopNext; - } - else { /* next: short windows */ - if (hDrcData->winSequenceCurr != 2) { /* current: long window */ - alphaValue = (FIXP_DBL)0; - - fact_mag = hDrcData->nextFact_mag; - fact_exp = hDrcData->nextFact_exp; - numBands = hDrcData->numBandsNext; - bandTop = hDrcData->bandTopNext; - } - else { /* current: short windows */ - shortDrc = 1; - - fact_mag = hDrcData->currFact_mag; - fact_exp = hDrcData->currFact_exp; - numBands = hDrcData->numBandsCurr; - bandTop = hDrcData->bandTopCurr; - } - } - } - else { /* first half of next frame */ - if (hDrcData->winSequenceNext != 2) { /* long window */ - int j = col - (numQmfSubSamples>>1); - - if (hDrcData->drcInterpolationSchemeNext == 0) { - INT k = (frameLenFlag) ? 0x4444444 : 0x4000000; - - alphaValue = (FIXP_DBL)(j * k); - } - else { - if (j >= offset[hDrcData->drcInterpolationSchemeNext - 1]) { - alphaValue = (FIXP_DBL)MAXVAL_DBL; - } - } - } - else { /* short windows */ - shortDrc = 1; - } - - fact_mag = hDrcData->nextFact_mag; - fact_exp = hDrcData->nextFact_exp; - numBands = hDrcData->numBandsNext; - bandTop = hDrcData->bandTopNext; - - col -= numQmfSubSamples; - } - - - /* process bands */ - for (band = 0; band < (int)numBands; band++) { - int bottomQmf, topQmf; - - FIXP_DBL drcFact_mag = (FIXP_DBL)MAXVAL_DBL; - - topMdct = (bandTop[band]+1) << 2; - - if (!shortDrc) { /* long window */ - if (frameLenFlag) { - /* 960 framing */ - bottomMdct = 30 * (bottomMdct / 30); - topMdct = 30 * (topMdct / 30); - - bottomQmf = fMultIfloor((FIXP_DBL)0x4444444, bottomMdct); - topQmf = fMultIfloor((FIXP_DBL)0x4444444, topMdct); - } - else { - /* 1024 framing */ - bottomMdct &= ~0x1f; - topMdct &= ~0x1f; - - bottomQmf = bottomMdct >> 5; - topQmf = topMdct >> 5; - } - - if (band == ((int)numBands-1)) { - topQmf = (64); - } - - for (bin = bottomQmf; bin < topQmf; bin++) { - FIXP_DBL drcFact1_mag = hDrcData->prevFact_mag[bin]; - FIXP_DBL drcFact2_mag = fact_mag[band]; - - /* normalize scale factors */ - if (hDrcData->prevFact_exp < maxShift) { - drcFact1_mag >>= maxShift - hDrcData->prevFact_exp; - } - if (fact_exp < maxShift) { - drcFact2_mag >>= maxShift - fact_exp; - } - - /* interpolate */ - if (alphaValue == (FIXP_DBL)0) { - drcFact_mag = drcFact1_mag; - } else if (alphaValue == (FIXP_DBL)MAXVAL_DBL) { - drcFact_mag = drcFact2_mag; - } else { - drcFact_mag = fMult(alphaValue, drcFact2_mag) + fMult(((FIXP_DBL)MAXVAL_DBL - alphaValue), drcFact1_mag); - } - - /* apply scaling */ - qmfRealSlot[bin] = fMult(qmfRealSlot[bin], drcFact_mag); - if (!useLP) { - qmfImagSlot[bin] = fMult(qmfImagSlot[bin], drcFact_mag); - } - - /* save previous factors */ - if (col == (numQmfSubSamples>>1)-1) { - hDrcData->prevFact_mag[bin] = fact_mag[band]; - } - } - } - else { /* short windows */ - int startSample, stopSample; - FIXP_DBL invFrameSizeDiv8 = (frameLenFlag) ? (FIXP_DBL)0x1111111 : (FIXP_DBL)0x1000000; - - if (frameLenFlag) { - /* 960 framing */ - bottomMdct = 30/8 * (bottomMdct*8/30); - topMdct = 30/8 * (topMdct*8/30); - } - else { - /* 1024 framing */ - bottomMdct &= ~0x03; - topMdct &= ~0x03; - } - - /* startSample is truncated to the nearest corresponding start subsample in - the QMF of the short window bottom is present in:*/ - startSample = ((fMultIfloor( invFrameSizeDiv8, bottomMdct ) & 0x7) * numQmfSubSamples) >> 3; - - /* stopSample is rounded upwards to the nearest corresponding stop subsample - in the QMF of the short window top is present in. */ - stopSample = ((fMultIceil( invFrameSizeDiv8, topMdct ) & 0xf) * numQmfSubSamples) >> 3; - - bottomQmf = fMultIfloor( invFrameSizeDiv8, ((bottomMdct%(numQmfSubSamples<<2)) << 5) ); - topQmf = fMultIfloor( invFrameSizeDiv8, ((topMdct%(numQmfSubSamples<<2)) << 5) ); - - /* extend last band */ - if (band == ((int)numBands-1)) { - topQmf = (64); - stopSample = numQmfSubSamples; - } - - if (topQmf == 0) { - topQmf = (64); - } - - /* save previous factors */ - if (stopSample == numQmfSubSamples) { - int tmpBottom = bottomQmf; - - if (((numQmfSubSamples-1) & ~0x03) > startSample) { - tmpBottom = 0; /* band starts in previous short window */ - } - - for (bin = tmpBottom; bin < topQmf; bin++) { - hDrcData->prevFact_mag[bin] = fact_mag[band]; - } - } - - /* apply */ - if ((col >= startSample) && (col < stopSample)) { - if ((col & ~0x03) > startSample) { - bottomQmf = 0; /* band starts in previous short window */ - } - if (col < ((stopSample-1) & ~0x03)) { - topQmf = (64); /* band ends in next short window */ - } - - drcFact_mag = fact_mag[band]; - - /* normalize scale factor */ - if (fact_exp < maxShift) { - drcFact_mag >>= maxShift - fact_exp; - } - - /* apply scaling */ - for (bin = bottomQmf; bin < topQmf; bin++) { - qmfRealSlot[bin] = fMult(qmfRealSlot[bin], drcFact_mag); - if (!useLP) { - qmfImagSlot[bin] = fMult(qmfImagSlot[bin], drcFact_mag); - } - } - } - } - - bottomMdct = topMdct; - } /* end of bands loop */ - - if (col == (numQmfSubSamples>>1)-1) { - hDrcData->prevFact_exp = fact_exp; - } -} - - -/*! - \brief Apply DRC factors frame based. - - \hDrcData Handle to DRC channel data. - \qmfRealSlot Pointer to real valued QMF data of the whole frame. - \qmfImagSlot Pointer to the imaginary QMF data of the whole frame. - \numQmfSubSamples Total number of time slots for one frame. - \scaleFactor Pointer to the out scale factor of the frame. - - \return None. -*/ -void sbrDecoder_drcApply ( - HANDLE_SBR_DRC_CHANNEL hDrcData, - FIXP_DBL **QmfBufferReal, - FIXP_DBL **QmfBufferImag, - int numQmfSubSamples, - int *scaleFactor - ) -{ - int col; - int maxShift = 0; - - if (hDrcData == NULL) { - return; - } - if (hDrcData->enable == 0) { - return; /* Avoid changing the scaleFactor even though the processing is disabled. */ - } - - /* get max scale factor */ - if (hDrcData->prevFact_exp > maxShift) { - maxShift = hDrcData->prevFact_exp; - } - if (hDrcData->currFact_exp > maxShift) { - maxShift = hDrcData->currFact_exp; - } - if (hDrcData->nextFact_exp > maxShift) { - maxShift = hDrcData->nextFact_exp; - } - - for (col = 0; col < numQmfSubSamples; col++) - { - FIXP_DBL *qmfSlotReal = QmfBufferReal[col]; - FIXP_DBL *qmfSlotImag = (QmfBufferImag == NULL) ? NULL : QmfBufferImag[col]; - - sbrDecoder_drcApplySlot ( - hDrcData, - qmfSlotReal, - qmfSlotImag, - col, - numQmfSubSamples, - maxShift - ); - } - - *scaleFactor += maxShift; -} - diff --git a/libSBRdec/src/sbrdec_drc.h b/libSBRdec/src/sbrdec_drc.h deleted file mode 100644 index 7eed53a..0000000 --- a/libSBRdec/src/sbrdec_drc.h +++ /dev/null @@ -1,151 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/***************************** MPEG-4 AAC Decoder ************************** - - Author(s): Christian Griebel - Description: Dynamic range control (DRC) decoder tool for SBR - -******************************************************************************/ - -#ifndef _SBRDEC_DRC_H_ -#define _SBRDEC_DRC_H_ - -#include "sbrdecoder.h" - - - -#define SBRDEC_MAX_DRC_CHANNELS (8) -#define SBRDEC_MAX_DRC_BANDS ( 16 ) - -typedef struct -{ - FIXP_DBL prevFact_mag[(64)]; - INT prevFact_exp; - - FIXP_DBL currFact_mag[SBRDEC_MAX_DRC_BANDS]; - FIXP_DBL nextFact_mag[SBRDEC_MAX_DRC_BANDS]; - INT currFact_exp; - INT nextFact_exp; - - UINT numBandsCurr; - UINT numBandsNext; - USHORT bandTopCurr[SBRDEC_MAX_DRC_BANDS]; - USHORT bandTopNext[SBRDEC_MAX_DRC_BANDS]; - - SHORT drcInterpolationSchemeCurr; - SHORT drcInterpolationSchemeNext; - - SHORT enable; - - UCHAR winSequenceCurr; - UCHAR winSequenceNext; - -} SBRDEC_DRC_CHANNEL; - -typedef SBRDEC_DRC_CHANNEL * HANDLE_SBR_DRC_CHANNEL; - - -void sbrDecoder_drcInitChannel ( - HANDLE_SBR_DRC_CHANNEL hDrcData ); - -void sbrDecoder_drcUpdateChannel ( - HANDLE_SBR_DRC_CHANNEL hDrcData ); - -void sbrDecoder_drcApplySlot ( - HANDLE_SBR_DRC_CHANNEL hDrcData, - FIXP_DBL *qmfRealSlot, - FIXP_DBL *qmfImagSlot, - int col, - int numQmfSubSamples, - int maxShift ); - -void sbrDecoder_drcApply ( - HANDLE_SBR_DRC_CHANNEL hDrcData, - FIXP_DBL **QmfBufferReal, - FIXP_DBL **QmfBufferImag, - int numQmfSubSamples, - int *scaleFactor ); - - -#endif /* _SBRDEC_DRC_H_ */ diff --git a/libSBRdec/src/sbrdec_freq_sca.cpp b/libSBRdec/src/sbrdec_freq_sca.cpp deleted file mode 100644 index 8adfbb1..0000000 --- a/libSBRdec/src/sbrdec_freq_sca.cpp +++ /dev/null @@ -1,812 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief Frequency scale calculation -*/ - -#include "sbrdec_freq_sca.h" - -#include "transcendent.h" -#include "sbr_rom.h" -#include "env_extr.h" - -#include "genericStds.h" /* need log() for debug-code only */ - -#define MAX_OCTAVE 29 -#define MAX_SECOND_REGION 50 - - -static int numberOfBands(FIXP_SGL bpo_div16, int start, int stop, int warpFlag); -static void CalcBands(UCHAR * diff, UCHAR start, UCHAR stop, UCHAR num_bands); -static SBR_ERROR modifyBands(UCHAR max_band, UCHAR * diff, UCHAR length); -static void cumSum(UCHAR start_value, UCHAR* diff, UCHAR length, UCHAR *start_adress); - - - -/*! - \brief Retrieve QMF-band where the SBR range starts - - Convert startFreq which was read from the bitstream into a - QMF-channel number. - - \return Number of start band -*/ -static UCHAR -getStartBand(UINT fs, /*!< Output sampling frequency */ - UCHAR startFreq, /*!< Index to table of possible start bands */ - UINT headerDataFlags) /*!< Info to SBR mode */ -{ - INT band; - UINT fsMapped; - - fsMapped = fs; - - switch (fsMapped) { - case 96000: - case 88200: - band = FDK_sbrDecoder_sbr_start_freq_88[startFreq]; - break; - case 64000: - band = FDK_sbrDecoder_sbr_start_freq_64[startFreq]; - break; - case 48000: - band = FDK_sbrDecoder_sbr_start_freq_48[startFreq]; - break; - case 44100: - band = FDK_sbrDecoder_sbr_start_freq_44[startFreq]; - break; - case 32000: - band = FDK_sbrDecoder_sbr_start_freq_32[startFreq]; - break; - case 24000: - band = FDK_sbrDecoder_sbr_start_freq_24[startFreq]; - break; - case 22050: - band = FDK_sbrDecoder_sbr_start_freq_22[startFreq]; - break; - case 16000: - band = FDK_sbrDecoder_sbr_start_freq_16[startFreq]; - break; - default: - band = 255; - } - - return band; -} - - -/*! - \brief Retrieve QMF-band where the SBR range starts - - Convert startFreq which was read from the bitstream into a - QMF-channel number. - - \return Number of start band -*/ -static UCHAR -getStopBand(UINT fs, /*!< Output sampling frequency */ - UCHAR stopFreq, /*!< Index to table of possible start bands */ - UINT headerDataFlags, /*!< Info to SBR mode */ - UCHAR k0) /*!< Start freq index */ -{ - UCHAR k2; - - if (stopFreq < 14) { - INT stopMin; - UCHAR diff_tot[MAX_OCTAVE + MAX_SECOND_REGION]; - UCHAR *diff0 = diff_tot; - UCHAR *diff1 = diff_tot+MAX_OCTAVE; - - if (fs < 32000) { - stopMin = (((2*6000*2*(64)) / fs) + 1) >> 1; - } - else { - if (fs < 64000) { - stopMin = (((2*8000*2*(64)) / fs) + 1) >> 1; - } - else { - stopMin = (((2*10000*2*(64)) / fs) + 1) >> 1; - } - } - - /* - Choose a stop band between k1 and 64 depending on stopFreq (0..13), - based on a logarithmic scale. - The vectors diff0 and diff1 are used temporarily here. - */ - CalcBands( diff0, stopMin, 64, 13); - shellsort( diff0, 13); - cumSum(stopMin, diff0, 13, diff1); - k2 = diff1[stopFreq]; - } - else if (stopFreq==14) - k2 = 2*k0; - else - k2 = 3*k0; - - /* Limit to Nyquist */ - if (k2 > (64)) - k2 = (64); - - - /* Range checks */ - /* 1 <= difference <= 48; 1 <= fs <= 96000 */ - if ( ((k2 - k0) > MAX_FREQ_COEFFS) || (k2 <= k0) ) { - return 255; - } - - if (headerDataFlags & (SBRDEC_SYNTAX_USAC|SBRDEC_SYNTAX_RSVD50)) { - /* 1 <= difference <= 35; 42000 <= fs <= 96000 */ - if ( (fs >= 42000) && ( (k2 - k0) > MAX_FREQ_COEFFS_FS44100 ) ) { - return 255; - } - /* 1 <= difference <= 32; 46009 <= fs <= 96000 */ - if ( (fs >= 46009) && ( (k2 - k0) > MAX_FREQ_COEFFS_FS48000 ) ) { - return 255; - } - } - else { - /* 1 <= difference <= 35; fs == 44100 */ - if ( (fs == 44100) && ( (k2 - k0) > MAX_FREQ_COEFFS_FS44100 ) ) { - return 255; - } - /* 1 <= difference <= 32; 48000 <= fs <= 96000 */ - if ( (fs >= 48000) && ( (k2 - k0) > MAX_FREQ_COEFFS_FS48000 ) ) { - return 255; - } - } - - return k2; -} - - -/*! - \brief Generates master frequency tables - - Frequency tables are calculated according to the selected domain - (linear/logarithmic) and granularity. - IEC 14496-3 4.6.18.3.2.1 - - \return errorCode, 0 if successful -*/ -SBR_ERROR -sbrdecUpdateFreqScale(UCHAR * v_k_master, /*!< Master table to be created */ - UCHAR *numMaster, /*!< Number of entries in master table */ - UINT fs, /*!< SBR working sampling rate */ - HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Control data from bitstream */ - UINT flags) -{ - FIXP_SGL bpo_div16; /* bands_per_octave divided by 16 */ - INT dk=0; - - /* Internal variables */ - UCHAR k0, k2, i; - UCHAR num_bands0 = 0; - UCHAR num_bands1 = 0; - UCHAR diff_tot[MAX_OCTAVE + MAX_SECOND_REGION]; - UCHAR *diff0 = diff_tot; - UCHAR *diff1 = diff_tot+MAX_OCTAVE; - INT k2_achived; - INT k2_diff; - INT incr=0; - - /* - Determine start band - */ - k0 = getStartBand(fs, hHeaderData->bs_data.startFreq, flags); - if (k0 == 255) { - return SBRDEC_UNSUPPORTED_CONFIG; - } - - /* - Determine stop band - */ - k2 = getStopBand(fs, hHeaderData->bs_data.stopFreq, flags, k0); - if (k2 == 255) { - return SBRDEC_UNSUPPORTED_CONFIG; - } - - if(hHeaderData->bs_data.freqScale>0) { /* Bark */ - INT k1; - - if(hHeaderData->bs_data.freqScale==1) { - bpo_div16 = FL2FXCONST_SGL(12.0f/16.0f); - } - else if(hHeaderData->bs_data.freqScale==2) { - bpo_div16 = FL2FXCONST_SGL(10.0f/16.0f); - } - else { - bpo_div16 = FL2FXCONST_SGL(8.0f/16.0f); - } - - - if( 1000 * k2 > 2245 * k0 ) { /* Two or more regions */ - k1 = 2*k0; - - num_bands0 = numberOfBands(bpo_div16, k0, k1, 0); - num_bands1 = numberOfBands(bpo_div16, k1, k2, hHeaderData->bs_data.alterScale ); - if ( num_bands0 < 1) { - return SBRDEC_UNSUPPORTED_CONFIG; - } - if ( num_bands1 < 1 ) { - return SBRDEC_UNSUPPORTED_CONFIG; - } - - CalcBands(diff0, k0, k1, num_bands0); - shellsort( diff0, num_bands0); - if (diff0[0] == 0) { -#ifdef DEBUG_TOOLS -#endif - return SBRDEC_UNSUPPORTED_CONFIG; - } - - cumSum(k0, diff0, num_bands0, v_k_master); - - CalcBands(diff1, k1, k2, num_bands1); - shellsort( diff1, num_bands1); - if(diff0[num_bands0-1] > diff1[0]) { - SBR_ERROR err; - - err = modifyBands(diff0[num_bands0-1],diff1, num_bands1); - if (err) - return SBRDEC_UNSUPPORTED_CONFIG; - } - - /* Add 2nd region */ - cumSum(k1, diff1, num_bands1, &v_k_master[num_bands0]); - *numMaster = num_bands0 + num_bands1; /* Output nr of bands */ - - } - else { /* Only one region */ - k1=k2; - - num_bands0 = numberOfBands(bpo_div16, k0, k1, 0); - if ( num_bands0 < 1) { - return SBRDEC_UNSUPPORTED_CONFIG; - } - CalcBands(diff0, k0, k1, num_bands0); - shellsort(diff0, num_bands0); - if (diff0[0] == 0) { -#ifdef DEBUG_TOOLS -#endif - return SBRDEC_UNSUPPORTED_CONFIG; - } - - cumSum(k0, diff0, num_bands0, v_k_master); - *numMaster = num_bands0; /* Output nr of bands */ - - } - } - else { /* Linear mode */ - if (hHeaderData->bs_data.alterScale==0) { - dk = 1; - /* FLOOR to get to few number of bands (next lower even number) */ - num_bands0 = (k2 - k0) & 254; - } else { - dk = 2; - num_bands0 = ( ((k2 - k0) >> 1) + 1 ) & 254; /* ROUND to the closest fit */ - } - - if (num_bands0 < 1) { - return SBRDEC_UNSUPPORTED_CONFIG; - /* We must return already here because 'i' can become negative below. */ - } - - k2_achived = k0 + num_bands0*dk; - k2_diff = k2 - k2_achived; - - for(i=0;i 0) { - incr = -1; - i = num_bands0-1; - } - - /* Adjust diff vector to get sepc. SBR range */ - while (k2_diff != 0) { - diff_tot[i] = diff_tot[i] - incr; - i = i + incr; - k2_diff = k2_diff + incr; - } - - cumSum(k0, diff_tot, num_bands0, v_k_master);/* cumsum */ - *numMaster = num_bands0; /* Output nr of bands */ - } - - if (*numMaster < 1) { - return SBRDEC_UNSUPPORTED_CONFIG; - } - - - /* - Print out the calculated table - */ - - return SBRDEC_OK; -} - - -/*! - \brief Calculate frequency ratio of one SBR band - - All SBR bands should span a constant frequency range in the logarithmic - domain. This function calculates the ratio of any SBR band's upper and lower - frequency. - - \return num_band-th root of k_start/k_stop -*/ -static FIXP_SGL calcFactorPerBand(int k_start, int k_stop, int num_bands) -{ -/* Scaled bandfactor and step 1 bit right to avoid overflow - * use double data type */ - FIXP_DBL bandfactor = FL2FXCONST_DBL(0.25f); /* Start value */ - FIXP_DBL step = FL2FXCONST_DBL(0.125f); /* Initial increment for factor */ - - int direction = 1; - -/* Because saturation can't be done in INT IIS, - * changed start and stop data type from FIXP_SGL to FIXP_DBL */ - FIXP_DBL start = k_start << (DFRACT_BITS-8); - FIXP_DBL stop = k_stop << (DFRACT_BITS-8); - - FIXP_DBL temp; - - int j, i=0; - - while ( step > FL2FXCONST_DBL(0.0f)) { - i++; - temp = stop; - - /* Calculate temp^num_bands: */ - for (j=0; j> 1); - direction = 1; - bandfactor = bandfactor + step; - } - else { /* Factor is too weak: make it stronger */ - if (direction == 1) - step = (FIXP_DBL)((LONG)step >> 1); - direction = 0; - bandfactor = bandfactor - step; - } - - if (i>100) { - step = FL2FXCONST_DBL(0.0f); - } - } - return FX_DBL2FX_SGL(bandfactor<<1); -} - - -/*! - \brief Calculate number of SBR bands between start and stop band - - Given the number of bands per octave, this function calculates how many - bands fit in the given frequency range. - When the warpFlag is set, the 'band density' is decreased by a factor - of 1/1.3 - - \return number of bands -*/ -static int -numberOfBands(FIXP_SGL bpo_div16, /*!< Input: number of bands per octave divided by 16 */ - int start, /*!< First QMF band of SBR frequency range */ - int stop, /*!< Last QMF band of SBR frequency range + 1 */ - int warpFlag) /*!< Stretching flag */ -{ - FIXP_SGL num_bands_div128; - int num_bands; - - num_bands_div128 = FX_DBL2FX_SGL(fMult(FDK_getNumOctavesDiv8(start,stop),bpo_div16)); - - if (warpFlag) { - /* Apply the warp factor of 1.3 to get wider bands. We use a value - of 32768/25200 instead of the exact value to avoid critical cases - of rounding. - */ - num_bands_div128 = FX_DBL2FX_SGL(fMult(num_bands_div128, FL2FXCONST_SGL(25200.0/32768.0))); - } - - /* add scaled 1 for rounding to even numbers: */ - num_bands_div128 = num_bands_div128 + FL2FXCONST_SGL( 1.0f/128.0f ); - /* scale back to right aligned integer and double the value: */ - num_bands = 2 * ((LONG)num_bands_div128 >> (FRACT_BITS - 7)); - - return(num_bands); -} - - -/*! - \brief Calculate width of SBR bands - - Given the desired number of bands within the SBR frequency range, - this function calculates the width of each SBR band in QMF channels. - The bands get wider from start to stop (bark scale). -*/ -static void -CalcBands(UCHAR * diff, /*!< Vector of widths to be calculated */ - UCHAR start, /*!< Lower end of subband range */ - UCHAR stop, /*!< Upper end of subband range */ - UCHAR num_bands) /*!< Desired number of bands */ -{ - int i; - int previous; - int current; - FIXP_SGL exact, temp; - FIXP_SGL bandfactor = calcFactorPerBand(start, stop, num_bands); - - previous = stop; /* Start with highest QMF channel */ - exact = (FIXP_SGL)(stop << (FRACT_BITS-8)); /* Shift left to gain some accuracy */ - - for(i=num_bands-1; i>=0; i--) { - /* Calculate border of next lower sbr band */ - exact = FX_DBL2FX_SGL(fMult(exact,bandfactor)); - - /* Add scaled 0.5 for rounding: - We use a value 128/256 instead of 0.5 to avoid some critical cases of rounding. */ - temp = exact + FL2FXCONST_SGL(128.0/32768.0); - - /* scale back to right alinged integer: */ - current = (LONG)temp >> (FRACT_BITS-8); - - /* Save width of band i */ - diff[i] = previous - current; - previous = current; - } -} - - -/*! - \brief Calculate cumulated sum vector from delta vector -*/ -static void -cumSum(UCHAR start_value, UCHAR* diff, UCHAR length, UCHAR *start_adress) -{ - int i; - start_adress[0]=start_value; - for(i=1; i<=length; i++) - start_adress[i] = start_adress[i-1] + diff[i-1]; -} - - -/*! - \brief Adapt width of frequency bands in the second region - - If SBR spans more than 2 octaves, the upper part of a bark-frequency-scale - is calculated separately. This function tries to avoid that the second region - starts with a band smaller than the highest band of the first region. -*/ -static SBR_ERROR -modifyBands(UCHAR max_band_previous, UCHAR * diff, UCHAR length) -{ - int change = max_band_previous - diff[0]; - - /* Limit the change so that the last band cannot get narrower than the first one */ - if ( change > (diff[length-1]-diff[0])>>1 ) - change = (diff[length-1]-diff[0])>>1; - - diff[0] += change; - diff[length-1] -= change; - shellsort(diff, length); - - return SBRDEC_OK; -} - - -/*! - \brief Update high resolution frequency band table -*/ -static void -sbrdecUpdateHiRes(UCHAR * h_hires, - UCHAR * num_hires, - UCHAR * v_k_master, - UCHAR num_bands, - UCHAR xover_band) -{ - UCHAR i; - - *num_hires = num_bands-xover_band; - - for(i=xover_band; i<=num_bands; i++) { - h_hires[i-xover_band] = v_k_master[i]; - } -} - - -/*! - \brief Build low resolution table out of high resolution table -*/ -static void -sbrdecUpdateLoRes(UCHAR * h_lores, - UCHAR * num_lores, - UCHAR * h_hires, - UCHAR num_hires) -{ - UCHAR i; - - if( (num_hires & 1) == 0) { - /* If even number of hires bands */ - *num_lores = num_hires >> 1; - /* Use every second lores=hires[0,2,4...] */ - for(i=0; i<=*num_lores; i++) - h_lores[i] = h_hires[i*2]; - } - else { - /* Odd number of hires, which means xover is odd */ - *num_lores = (num_hires+1) >> 1; - /* Use lores=hires[0,1,3,5 ...] */ - h_lores[0] = h_hires[0]; - for(i=1; i<=*num_lores; i++) { - h_lores[i] = h_hires[i*2-1]; - } - } -} - - -/*! - \brief Derive a low-resolution frequency-table from the master frequency table -*/ -void -sbrdecDownSampleLoRes(UCHAR *v_result, - UCHAR num_result, - UCHAR *freqBandTableRef, - UCHAR num_Ref) -{ - int step; - int i,j; - int org_length,result_length; - int v_index[MAX_FREQ_COEFFS>>1]; - - /* init */ - org_length = num_Ref; - result_length = num_result; - - v_index[0] = 0; /* Always use left border */ - i=0; - while(org_length > 0) { - /* Create downsample vector */ - i++; - step = org_length / result_length; - org_length = org_length - step; - result_length--; - v_index[i] = v_index[i-1] + step; - } - - for(j=0;j<=i;j++) { - /* Use downsample vector to index LoResolution vector */ - v_result[j]=freqBandTableRef[v_index[j]]; - } - -} - - -/*! - \brief Sorting routine -*/ -void shellsort(UCHAR *in, UCHAR n) -{ - - int i, j, v, w; - int inc = 1; - - do - inc = 3 * inc + 1; - while (inc <= n); - - do { - inc = inc / 3; - for (i = inc; i < n; i++) { - v = in[i]; - j = i; - while ((w=in[j-inc]) > v) { - in[j] = w; - j -= inc; - if (j < inc) - break; - } - in[j] = v; - } - } while (inc > 1); - -} - - - -/*! - \brief Reset frequency band tables - \return errorCode, 0 if successful -*/ -SBR_ERROR -resetFreqBandTables(HANDLE_SBR_HEADER_DATA hHeaderData, const UINT flags) -{ - SBR_ERROR err = SBRDEC_OK; - int k2,kx, lsb, usb; - int intTemp; - UCHAR nBandsLo, nBandsHi; - HANDLE_FREQ_BAND_DATA hFreq = &hHeaderData->freqBandData; - - /* Calculate master frequency function */ - err = sbrdecUpdateFreqScale(hFreq->v_k_master, - &hFreq->numMaster, - hHeaderData->sbrProcSmplRate, - hHeaderData, - flags); - - if ( err || (hHeaderData->bs_info.xover_band > hFreq->numMaster) ) { - return SBRDEC_UNSUPPORTED_CONFIG; - } - - /* Derive Hiresolution from master frequency function */ - sbrdecUpdateHiRes(hFreq->freqBandTable[1], &nBandsHi, hFreq->v_k_master, hFreq->numMaster, hHeaderData->bs_info.xover_band ); - /* Derive Loresolution from Hiresolution */ - sbrdecUpdateLoRes(hFreq->freqBandTable[0], &nBandsLo, hFreq->freqBandTable[1], nBandsHi); - - - hFreq->nSfb[0] = nBandsLo; - hFreq->nSfb[1] = nBandsHi; - - /* Check index to freqBandTable[0] */ - if ( !(nBandsLo > 0) || (nBandsLo > (MAX_FREQ_COEFFS>>1)) ) { - return SBRDEC_UNSUPPORTED_CONFIG; - } - - lsb = hFreq->freqBandTable[0][0]; - usb = hFreq->freqBandTable[0][nBandsLo]; - - /* Additional check for lsb */ - if ( (lsb > (32)) || (lsb >= usb) ) { - return SBRDEC_UNSUPPORTED_CONFIG; - } - - - /* Calculate number of noise bands */ - - k2 = hFreq->freqBandTable[1][nBandsHi]; - kx = hFreq->freqBandTable[1][0]; - - if (hHeaderData->bs_data.noise_bands == 0) - { - hFreq->nNfb = 1; - } - else /* Calculate no of noise bands 1,2 or 3 bands/octave */ - { - /* Fetch number of octaves divided by 32 */ - intTemp = (LONG)FDK_getNumOctavesDiv8(kx,k2) >> 2; - - /* Integer-Multiplication with number of bands: */ - intTemp = intTemp * hHeaderData->bs_data.noise_bands; - - /* Add scaled 0.5 for rounding: */ - intTemp = intTemp + (LONG)FL2FXCONST_SGL(0.5f/32.0f); - - /* Convert to right-aligned integer: */ - intTemp = intTemp >> (FRACT_BITS - 1 /*sign*/ - 5 /* rescale */); - - /* Compare with float calculation */ - FDK_ASSERT( intTemp == (int)((hHeaderData->bs_data.noise_bands * FDKlog( (float)k2/kx) / (float)(FDKlog(2.0)))+0.5) ); - - if( intTemp==0) - intTemp=1; - - hFreq->nNfb = intTemp; - } - - hFreq->nInvfBands = hFreq->nNfb; - - if( hFreq->nNfb > MAX_NOISE_COEFFS ) { - return SBRDEC_UNSUPPORTED_CONFIG; - } - - /* Get noise bands */ - sbrdecDownSampleLoRes(hFreq->freqBandTableNoise, - hFreq->nNfb, - hFreq->freqBandTable[0], - nBandsLo); - - - - - hFreq->lowSubband = lsb; - hFreq->highSubband = usb; - - return SBRDEC_OK; -} diff --git a/libSBRdec/src/sbrdec_freq_sca.h b/libSBRdec/src/sbrdec_freq_sca.h deleted file mode 100644 index cfe4f0e..0000000 --- a/libSBRdec/src/sbrdec_freq_sca.h +++ /dev/null @@ -1,107 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief Frequency scale prototypes -*/ -#ifndef __FREQ_SCA_H -#define __FREQ_SCA_H - -#include "sbrdecoder.h" -#include "env_extr.h" - -int -sbrdecUpdateFreqScale(UCHAR * v_k_master, - UCHAR *numMaster, - HANDLE_SBR_HEADER_DATA headerData); - -void sbrdecDownSampleLoRes(UCHAR *v_result, UCHAR num_result, - UCHAR *freqBandTableRef, UCHAR num_Ref); - -void shellsort(UCHAR *in, UCHAR n); - -SBR_ERROR -resetFreqBandTables(HANDLE_SBR_HEADER_DATA hHeaderData, const UINT flags); - -#endif diff --git a/libSBRdec/src/sbrdecoder.cpp b/libSBRdec/src/sbrdecoder.cpp deleted file mode 100644 index 7d9468c..0000000 --- a/libSBRdec/src/sbrdecoder.cpp +++ /dev/null @@ -1,1764 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief SBR decoder frontend - This module provides a frontend to the SBR decoder. The function openSBR() is called for - initialization. The function sbrDecoder_Apply() is called for each frame. sbr_Apply() will call the - required functions to decode the raw SBR data (provided by env_extr.cpp), to decode the envelope data and noise floor levels [decodeSbrData()], - and to finally apply SBR to the current frame [sbr_dec()]. - - \sa sbrDecoder_Apply(), \ref documentationOverview -*/ - -/*! - \page documentationOverview Overview of important information resources and source code documentation - - The primary source code documentation is based on generated and cross-referenced HTML files using - doxygen. As part of this documentation - you can find more extensive descriptions about key concepts and algorithms at the following locations: - -

Programming

- - \li Buffer management: sbrDecoder_Apply() and sbr_dec() - \li Internal scale factors to maximize SNR on fixed point processors: #QMF_SCALE_FACTOR - \li Special mantissa-exponent format: Created in requantizeEnvelopeData() and used in calculateSbrEnvelope() - -

Algorithmic details

- \li About the SBR data format: \ref SBR_HEADER_ELEMENT and \ref SBR_STANDARD_ELEMENT - \li Details about the bitstream decoder: env_extr.cpp - \li Details about the QMF filterbank and the provided polyphase implementation: qmf_dec.cpp - \li Details about the transposer: lpp_tran.cpp - \li Details about the envelope adjuster: env_calc.cpp - -*/ - -#include "sbrdecoder.h" - -#include "FDK_bitstream.h" - -#include "sbrdec_freq_sca.h" -#include "env_extr.h" -#include "sbr_dec.h" -#include "env_dec.h" -#include "sbr_crc.h" -#include "sbr_ram.h" -#include "sbr_rom.h" -#include "lpp_tran.h" -#include "transcendent.h" - -#include "FDK_crc.h" - -#include "sbrdec_drc.h" - -#include "psbitdec.h" - - -/* Decoder library info */ -#define SBRDECODER_LIB_VL0 2 -#define SBRDECODER_LIB_VL1 2 -#define SBRDECODER_LIB_VL2 12 -#define SBRDECODER_LIB_TITLE "SBR Decoder" -#ifdef __ANDROID__ -#define SBRDECODER_LIB_BUILD_DATE "" -#define SBRDECODER_LIB_BUILD_TIME "" -#else -#define SBRDECODER_LIB_BUILD_DATE __DATE__ -#define SBRDECODER_LIB_BUILD_TIME __TIME__ -#endif - - - - -static UCHAR getHeaderSlot( UCHAR currentSlot, UCHAR hdrSlotUsage[(1)+1] ) -{ - UINT occupied = 0; - int s; - UCHAR slot = hdrSlotUsage[currentSlot]; - - FDK_ASSERT((1)+1 < 32); - - for (s = 0; s < (1)+1; s++) { - if ( (hdrSlotUsage[s] == slot) - && (s != slot) ) { - occupied = 1; - break; - } - } - - if (occupied) { - occupied = 0; - - for (s = 0; s < (1)+1; s++) { - occupied |= 1 << hdrSlotUsage[s]; - } - for (s = 0; s < (1)+1; s++) { - if ( !(occupied & 0x1) ) { - slot = s; - break; - } - occupied >>= 1; - } - } - - return slot; -} - -static void copySbrHeader( HANDLE_SBR_HEADER_DATA hDst, const HANDLE_SBR_HEADER_DATA hSrc ) -{ - /* copy the whole header memory (including pointers) */ - FDKmemcpy( hDst, hSrc, sizeof(SBR_HEADER_DATA) ); - - /* update pointers */ - hDst->freqBandData.freqBandTable[0] = hDst->freqBandData.freqBandTableLo; - hDst->freqBandData.freqBandTable[1] = hDst->freqBandData.freqBandTableHi; -} - -static int compareSbrHeader( const HANDLE_SBR_HEADER_DATA hHdr1, const HANDLE_SBR_HEADER_DATA hHdr2 ) -{ - int result = 0; - - /* compare basic data */ - result |= (hHdr1->syncState != hHdr2->syncState) ? 1 : 0; - result |= (hHdr1->status != hHdr2->status) ? 1 : 0; - result |= (hHdr1->frameErrorFlag != hHdr2->frameErrorFlag) ? 1 : 0; - result |= (hHdr1->numberTimeSlots != hHdr2->numberTimeSlots) ? 1 : 0; - result |= (hHdr1->numberOfAnalysisBands != hHdr2->numberOfAnalysisBands) ? 1 : 0; - result |= (hHdr1->timeStep != hHdr2->timeStep) ? 1 : 0; - result |= (hHdr1->sbrProcSmplRate != hHdr2->sbrProcSmplRate) ? 1 : 0; - - /* compare bitstream data */ - result |= FDKmemcmp( &hHdr1->bs_data, &hHdr2->bs_data, sizeof(SBR_HEADER_DATA_BS) ); - result |= FDKmemcmp( &hHdr1->bs_info, &hHdr2->bs_info, sizeof(SBR_HEADER_DATA_BS_INFO) ); - - /* compare frequency band data */ - result |= FDKmemcmp( &hHdr1->freqBandData, &hHdr2->freqBandData, (8+MAX_NUM_LIMITERS+1)*sizeof(UCHAR) ); - result |= FDKmemcmp( hHdr1->freqBandData.freqBandTableLo, hHdr2->freqBandData.freqBandTableLo, (MAX_FREQ_COEFFS/2+1)*sizeof(UCHAR) ); - result |= FDKmemcmp( hHdr1->freqBandData.freqBandTableHi, hHdr2->freqBandData.freqBandTableHi, (MAX_FREQ_COEFFS+1)*sizeof(UCHAR) ); - result |= FDKmemcmp( hHdr1->freqBandData.freqBandTableNoise, hHdr2->freqBandData.freqBandTableNoise, (MAX_NOISE_COEFFS+1)*sizeof(UCHAR) ); - result |= FDKmemcmp( hHdr1->freqBandData.v_k_master, hHdr2->freqBandData.v_k_master, (MAX_FREQ_COEFFS+1)*sizeof(UCHAR) ); - - return result; -} - - -/*! - \brief Reset SBR decoder. - - Reset should only be called if SBR has been sucessfully detected by - an appropriate checkForPayload() function. - - \return Error code. -*/ -static -SBR_ERROR sbrDecoder_ResetElement ( - HANDLE_SBRDECODER self, - int sampleRateIn, - int sampleRateOut, - int samplesPerFrame, - const MP4_ELEMENT_ID elementID, - const int elementIndex, - const int overlap - ) -{ - SBR_ERROR sbrError = SBRDEC_OK; - HANDLE_SBR_HEADER_DATA hSbrHeader; - UINT qmfFlags = 0; - - int i, synDownsampleFac; - - /* Check in/out samplerates */ - if ( sampleRateIn < 6400 - || sampleRateIn > 48000 - ) - { - sbrError = SBRDEC_UNSUPPORTED_CONFIG; - goto bail; - } - - if ( sampleRateOut > 96000 ) - { - sbrError = SBRDEC_UNSUPPORTED_CONFIG; - goto bail; - } - - /* Set QMF mode flags */ - if (self->flags & SBRDEC_LOW_POWER) - qmfFlags |= QMF_FLAG_LP; - - if (self->coreCodec == AOT_ER_AAC_ELD) { - if (self->flags & SBRDEC_LD_MPS_QMF) { - qmfFlags |= QMF_FLAG_MPSLDFB; - } else { - qmfFlags |= QMF_FLAG_CLDFB; - } - } - - /* Set downsampling factor for synthesis filter bank */ - if (sampleRateOut == 0) - { - /* no single rate mode */ - sampleRateOut = sampleRateIn<<1; /* In case of implicit signalling, assume dual rate SBR */ - } - - if ( sampleRateIn == sampleRateOut ) { - synDownsampleFac = 2; - self->flags |= SBRDEC_DOWNSAMPLE; - } else { - synDownsampleFac = 1; - self->flags &= ~SBRDEC_DOWNSAMPLE; - } - - self->synDownsampleFac = synDownsampleFac; - self->sampleRateOut = sampleRateOut; - - { - int i; - - for (i = 0; i < (1)+1; i++) - { - hSbrHeader = &(self->sbrHeader[elementIndex][i]); - - /* init a default header such that we can at least do upsampling later */ - sbrError = initHeaderData( - hSbrHeader, - sampleRateIn, - sampleRateOut, - samplesPerFrame, - self->flags - ); - } - } - - if (sbrError != SBRDEC_OK) { - goto bail; - } - - /* Init SBR channels going to be assigned to a SBR element */ - { - int ch; - - for (ch=0; chpSbrElement[elementIndex]->nChannels; ch++) - { - /* and create sbrDec */ - sbrError = createSbrDec (self->pSbrElement[elementIndex]->pSbrChannel[ch], - hSbrHeader, - &self->pSbrElement[elementIndex]->transposerSettings, - synDownsampleFac, - qmfFlags, - self->flags, - overlap, - ch ); - - if (sbrError != SBRDEC_OK) { - goto bail; - } - } - } - - //FDKmemclear(sbr_OverlapBuffer, sizeof(sbr_OverlapBuffer)); - - if (self->numSbrElements == 1) { - switch ( self->coreCodec ) { - case AOT_AAC_LC: - case AOT_SBR: - case AOT_PS: - case AOT_ER_AAC_SCAL: - case AOT_DRM_AAC: - if (CreatePsDec ( &self->hParametricStereoDec, samplesPerFrame )) { - sbrError = SBRDEC_CREATE_ERROR; - goto bail; - } - break; - default: - break; - } - } - - /* Init frame delay slot handling */ - self->pSbrElement[elementIndex]->useFrameSlot = 0; - for (i = 0; i < ((1)+1); i++) { - self->pSbrElement[elementIndex]->useHeaderSlot[i] = i; - } - -bail: - - return sbrError; -} - - -SBR_ERROR sbrDecoder_Open ( HANDLE_SBRDECODER * pSelf ) -{ - HANDLE_SBRDECODER self = NULL; - SBR_ERROR sbrError = SBRDEC_OK; - - /* Get memory for this instance */ - self = GetRam_SbrDecoder(); - if (self == NULL) { - sbrError = SBRDEC_MEM_ALLOC_FAILED; - goto bail; - } - - self->workBuffer1 = GetRam_SbrDecWorkBuffer1(); - self->workBuffer2 = GetRam_SbrDecWorkBuffer2(); - - if ( self->workBuffer1 == NULL - || self->workBuffer2 == NULL ) - { - sbrError = SBRDEC_MEM_ALLOC_FAILED; - goto bail; - } - - /* - Already zero because of calloc - self->numSbrElements = 0; - self->numSbrChannels = 0; - self->codecFrameSize = 0; - */ - - self->numDelayFrames = (1); /* set to the max value by default */ - - *pSelf = self; - -bail: - return sbrError; -} - -/** - * \brief determine if the given core codec AOT can be processed or not. - * \param coreCodec core codec audio object type. - * \return 1 if SBR can be processed, 0 if SBR cannot be processed/applied. - */ -static -int sbrDecoder_isCoreCodecValid(AUDIO_OBJECT_TYPE coreCodec) -{ - switch (coreCodec) { - case AOT_AAC_LC: - case AOT_SBR: - case AOT_PS: - case AOT_ER_AAC_SCAL: - case AOT_ER_AAC_ELD: - case AOT_DRM_AAC: - return 1; - default: - return 0; - } -} - -static -void sbrDecoder_DestroyElement ( - HANDLE_SBRDECODER self, - const int elementIndex - ) -{ - if (self->pSbrElement[elementIndex] != NULL) { - int ch; - - for (ch=0; chpSbrElement[elementIndex]->pSbrChannel[ch] != NULL) { - deleteSbrDec( self->pSbrElement[elementIndex]->pSbrChannel[ch] ); - FreeRam_SbrDecChannel( &self->pSbrElement[elementIndex]->pSbrChannel[ch] ); - self->numSbrChannels -= 1; - } - } - FreeRam_SbrDecElement( &self->pSbrElement[elementIndex] ); - self->numSbrElements -= 1; - } -} - - -SBR_ERROR sbrDecoder_InitElement ( - HANDLE_SBRDECODER self, - const int sampleRateIn, - const int sampleRateOut, - const int samplesPerFrame, - const AUDIO_OBJECT_TYPE coreCodec, - const MP4_ELEMENT_ID elementID, - const int elementIndex - ) -{ - SBR_ERROR sbrError = SBRDEC_OK; - int chCnt=0; - int nSbrElementsStart = self->numSbrElements; - - /* Check core codec AOT */ - if (! sbrDecoder_isCoreCodecValid(coreCodec) || elementIndex >= (8)) { - sbrError = SBRDEC_UNSUPPORTED_CONFIG; - goto bail; - } - - if ( elementID != ID_SCE && elementID != ID_CPE && elementID != ID_LFE ) - { - sbrError = SBRDEC_UNSUPPORTED_CONFIG; - goto bail; - } - - if ( self->sampleRateIn == sampleRateIn - && self->codecFrameSize == samplesPerFrame - && self->coreCodec == coreCodec - && self->pSbrElement[elementIndex] != NULL - && self->pSbrElement[elementIndex]->elementID == elementID - && !(self->flags & SBRDEC_FORCE_RESET) - ) - { - /* Nothing to do */ - return SBRDEC_OK; - } - - self->sampleRateIn = sampleRateIn; - self->codecFrameSize = samplesPerFrame; - self->coreCodec = coreCodec; - - self->flags = 0; - self->flags |= (coreCodec == AOT_ER_AAC_ELD) ? SBRDEC_ELD_GRID : 0; - self->flags |= (coreCodec == AOT_ER_AAC_SCAL) ? SBRDEC_SYNTAX_SCAL : 0; - self->flags |= (coreCodec == AOT_DRM_AAC) ? SBRDEC_SYNTAX_SCAL|SBRDEC_SYNTAX_DRM : 0; - - /* Init SBR elements */ - { - int elChannels, ch; - - if (self->pSbrElement[elementIndex] == NULL) { - self->pSbrElement[elementIndex] = GetRam_SbrDecElement(elementIndex); - if (self->pSbrElement[elementIndex] == NULL) { - sbrError = SBRDEC_MEM_ALLOC_FAILED; - goto bail; - } - self->numSbrElements ++; - } else { - self->numSbrChannels -= self->pSbrElement[elementIndex]->nChannels; - } - - /* Save element ID for sanity checks and to have a fallback for concealment. */ - self->pSbrElement[elementIndex]->elementID = elementID; - - /* Determine amount of channels for this element */ - switch (elementID) { - case ID_NONE: - case ID_CPE: elChannels=2; - break; - case ID_LFE: - case ID_SCE: elChannels=1; - break; - default: elChannels=0; - break; - } - - /* Handle case of Parametric Stereo */ - if ( elementIndex == 0 && elementID == ID_SCE ) { - switch (coreCodec) { - case AOT_AAC_LC: - case AOT_SBR: - case AOT_PS: - case AOT_ER_AAC_SCAL: - case AOT_DRM_AAC: - elChannels = 2; - break; - default: - break; - } - } - - self->pSbrElement[elementIndex]->nChannels = elChannels; - - for (ch=0; chpSbrElement[elementIndex]->pSbrChannel[ch] == NULL) { - self->pSbrElement[elementIndex]->pSbrChannel[ch] = GetRam_SbrDecChannel(chCnt); - if (self->pSbrElement[elementIndex]->pSbrChannel[ch] == NULL) { - sbrError = SBRDEC_MEM_ALLOC_FAILED; - goto bail; - } - } - self->numSbrChannels ++; - - sbrDecoder_drcInitChannel( &self->pSbrElement[elementIndex]->pSbrChannel[ch]->SbrDec.sbrDrcChannel ); - - /* Add reference pointer to workbuffers. */ - self->pSbrElement[elementIndex]->pSbrChannel[ch]->SbrDec.WorkBuffer1 = self->workBuffer1; - self->pSbrElement[elementIndex]->pSbrChannel[ch]->SbrDec.WorkBuffer2 = self->workBuffer2; - chCnt++; - } - if (elChannels == 1 && self->pSbrElement[elementIndex]->pSbrChannel[ch] != NULL) { - deleteSbrDec( self->pSbrElement[elementIndex]->pSbrChannel[ch] ); - FreeRam_SbrDecChannel( &self->pSbrElement[elementIndex]->pSbrChannel[ch] ); - } - } - - /* clear error flags for all delay slots */ - FDKmemclear(self->pSbrElement[elementIndex]->frameErrorFlag, ((1)+1)*sizeof(UCHAR)); - - /* Initialize this instance */ - sbrError = sbrDecoder_ResetElement( - self, - sampleRateIn, - sampleRateOut, - samplesPerFrame, - elementID, - elementIndex, - (coreCodec == AOT_ER_AAC_ELD) ? 0 : (6) - ); - - - -bail: - if (sbrError != SBRDEC_OK) { - if (nSbrElementsStart < self->numSbrElements) { - /* Free the memory allocated for this element */ - sbrDecoder_DestroyElement( self, elementIndex ); - } else if ( (self->pSbrElement[elementIndex] != NULL) - && (elementIndex < (8))) - { /* Set error flag to trigger concealment */ - self->pSbrElement[elementIndex]->frameErrorFlag[self->pSbrElement[elementIndex]->useFrameSlot] = 1; - } - } - - return sbrError; -} - -/** - * \brief Apply decoded SBR header for one element. - * \param self SBR decoder instance handle - * \param hSbrHeader SBR header handle to be processed. - * \param hSbrChannel pointer array to the SBR element channels corresponding to the SBR header. - * \param headerStatus header status value returned from SBR header parser. - * \param numElementChannels amount of channels for the SBR element whos header is to be processed. - */ -static -SBR_ERROR sbrDecoder_HeaderUpdate( - HANDLE_SBRDECODER self, - HANDLE_SBR_HEADER_DATA hSbrHeader, - SBR_HEADER_STATUS headerStatus, - HANDLE_SBR_CHANNEL hSbrChannel[], - const int numElementChannels - ) -{ - SBR_ERROR errorStatus = SBRDEC_OK; - - /* - change of control data, reset decoder - */ - errorStatus = resetFreqBandTables(hSbrHeader, self->flags); - - if (errorStatus == SBRDEC_OK) { - if (hSbrHeader->syncState == UPSAMPLING && headerStatus != HEADER_RESET) - { - /* As the default header would limit the frequency range, - lowSubband and highSubband must be patched. */ - hSbrHeader->freqBandData.lowSubband = hSbrHeader->numberOfAnalysisBands; - hSbrHeader->freqBandData.highSubband = hSbrHeader->numberOfAnalysisBands; - } - - /* Trigger a reset before processing this slot */ - hSbrHeader->status |= SBRDEC_HDR_STAT_RESET; - } - - return errorStatus; -} - -INT sbrDecoder_Header ( - HANDLE_SBRDECODER self, - HANDLE_FDK_BITSTREAM hBs, - const INT sampleRateIn, - const INT sampleRateOut, - const INT samplesPerFrame, - const AUDIO_OBJECT_TYPE coreCodec, - const MP4_ELEMENT_ID elementID, - const INT elementIndex - ) -{ - SBR_HEADER_STATUS headerStatus; - HANDLE_SBR_HEADER_DATA hSbrHeader; - SBR_ERROR sbrError = SBRDEC_OK; - int headerIndex; - - if ( self == NULL || elementIndex > (8) ) - { - return SBRDEC_UNSUPPORTED_CONFIG; - } - - if (! sbrDecoder_isCoreCodecValid(coreCodec)) { - return SBRDEC_UNSUPPORTED_CONFIG; - } - - sbrError = sbrDecoder_InitElement( - self, - sampleRateIn, - sampleRateOut, - samplesPerFrame, - coreCodec, - elementID, - elementIndex - ); - - if (sbrError != SBRDEC_OK) { - goto bail; - } - - headerIndex = getHeaderSlot(self->pSbrElement[elementIndex]->useFrameSlot, - self->pSbrElement[elementIndex]->useHeaderSlot); - hSbrHeader = &(self->sbrHeader[elementIndex][headerIndex]); - - headerStatus = sbrGetHeaderData ( hSbrHeader, - hBs, - self->flags, - 0); - - - { - SBR_DECODER_ELEMENT *pSbrElement; - - pSbrElement = self->pSbrElement[elementIndex]; - - /* Sanity check */ - if (pSbrElement != NULL) { - if ( (elementID == ID_CPE && pSbrElement->nChannels != 2) - || (elementID != ID_CPE && pSbrElement->nChannels != 1) ) - { - return SBRDEC_UNSUPPORTED_CONFIG; - } - if ( headerStatus == HEADER_RESET ) { - - sbrError = sbrDecoder_HeaderUpdate( - self, - hSbrHeader, - headerStatus, - pSbrElement->pSbrChannel, - pSbrElement->nChannels - ); - - if (sbrError == SBRDEC_OK) { - hSbrHeader->syncState = SBR_HEADER; - hSbrHeader->status |= SBRDEC_HDR_STAT_UPDATE; - } - /* else { - Since we already have overwritten the old SBR header the only way out is UPSAMPLING! - This will be prepared in the next step. - } */ - } - } - } -bail: - return sbrError; -} - - -SBR_ERROR sbrDecoder_SetParam (HANDLE_SBRDECODER self, - const SBRDEC_PARAM param, - const INT value ) -{ - SBR_ERROR errorStatus = SBRDEC_OK; - - /* configure the subsystems */ - switch (param) - { - case SBR_SYSTEM_BITSTREAM_DELAY: - if (value < 0 || value > (1)) { - errorStatus = SBRDEC_SET_PARAM_FAIL; - break; - } - if (self == NULL) { - errorStatus = SBRDEC_NOT_INITIALIZED; - } else { - self->numDelayFrames = (UCHAR)value; - } - break; - case SBR_QMF_MODE: - if (self == NULL) { - errorStatus = SBRDEC_NOT_INITIALIZED; - } else { - if (value == 1) { - self->flags |= SBRDEC_LOW_POWER; - } else { - self->flags &= ~SBRDEC_LOW_POWER; - } - } - break; - case SBR_LD_QMF_TIME_ALIGN: - if (self == NULL) { - errorStatus = SBRDEC_NOT_INITIALIZED; - } else { - if (value == 1) { - self->flags |= SBRDEC_LD_MPS_QMF; - } else { - self->flags &= ~SBRDEC_LD_MPS_QMF; - } - } - break; - case SBR_FLUSH_DATA: - if (value != 0) { - if (self == NULL) { - errorStatus = SBRDEC_NOT_INITIALIZED; - } else { - self->flags |= SBRDEC_FLUSH; - } - } - break; - case SBR_CLEAR_HISTORY: - if (value != 0) { - if (self == NULL) { - errorStatus = SBRDEC_NOT_INITIALIZED; - } else { - self->flags |= SBRDEC_FORCE_RESET; - } - } - break; - case SBR_BS_INTERRUPTION: - { - int elementIndex; - - if (self == NULL) { - errorStatus = SBRDEC_NOT_INITIALIZED; - break; - } - - /* Loop over SBR elements */ - for (elementIndex = 0; elementIndex < self->numSbrElements; elementIndex++) { - if (self->pSbrElement[elementIndex] != NULL) - { - HANDLE_SBR_HEADER_DATA hSbrHeader; - int headerIndex = getHeaderSlot(self->pSbrElement[elementIndex]->useFrameSlot, - self->pSbrElement[elementIndex]->useHeaderSlot); - - hSbrHeader = &(self->sbrHeader[elementIndex][headerIndex]); - - /* Set sync state UPSAMPLING for the corresponding slot. - This switches off bitstream parsing until a new header arrives. */ - hSbrHeader->syncState = UPSAMPLING; - hSbrHeader->status |= SBRDEC_HDR_STAT_UPDATE; - } } - } - break; - default: - errorStatus = SBRDEC_SET_PARAM_FAIL; - break; - } /* switch(param) */ - - return (errorStatus); -} - -static -SBRDEC_DRC_CHANNEL * sbrDecoder_drcGetChannel( const HANDLE_SBRDECODER self, const INT channel ) -{ - SBRDEC_DRC_CHANNEL *pSbrDrcChannelData = NULL; - int elementIndex, elChanIdx=0, numCh=0; - - for (elementIndex = 0; (elementIndex < (8)) && (numCh <= channel); elementIndex++) - { - SBR_DECODER_ELEMENT *pSbrElement = self->pSbrElement[elementIndex]; - int c, elChannels; - - elChanIdx = 0; - if (pSbrElement == NULL) break; - - /* Determine amount of channels for this element */ - switch (pSbrElement->elementID) { - case ID_CPE: elChannels = 2; - break; - case ID_LFE: - case ID_SCE: elChannels = 1; - break; - case ID_NONE: - default: elChannels = 0; - break; - } - - /* Limit with actual allocated element channels */ - elChannels = FDKmin(elChannels, pSbrElement->nChannels); - - for (c = 0; (c < elChannels) && (numCh <= channel); c++) { - if (pSbrElement->pSbrChannel[elChanIdx] != NULL) { - numCh++; - elChanIdx++; - } - } - } - elementIndex -= 1; - elChanIdx -= 1; - - if (elChanIdx < 0 || elementIndex < 0) { - return NULL; - } - - if ( self->pSbrElement[elementIndex] != NULL ) { - if ( self->pSbrElement[elementIndex]->pSbrChannel[elChanIdx] != NULL ) - { - pSbrDrcChannelData = &self->pSbrElement[elementIndex]->pSbrChannel[elChanIdx]->SbrDec.sbrDrcChannel; - } - } - - return (pSbrDrcChannelData); -} - -SBR_ERROR sbrDecoder_drcFeedChannel ( HANDLE_SBRDECODER self, - INT ch, - UINT numBands, - FIXP_DBL *pNextFact_mag, - INT nextFact_exp, - SHORT drcInterpolationScheme, - UCHAR winSequence, - USHORT *pBandTop ) -{ - SBRDEC_DRC_CHANNEL *pSbrDrcChannelData = NULL; - int band, isValidData = 0; - - if (self == NULL) { - return SBRDEC_NOT_INITIALIZED; - } - if (ch > (8) || pNextFact_mag == NULL) { - return SBRDEC_SET_PARAM_FAIL; - } - - /* Search for gain values different to 1.0f */ - for (band = 0; band < numBands; band += 1) { - if ( !((pNextFact_mag[band] == FL2FXCONST_DBL(0.5)) && (nextFact_exp == 1)) - && !((pNextFact_mag[band] == (FIXP_DBL)MAXVAL_DBL) && (nextFact_exp == 0)) ) { - isValidData = 1; - break; - } - } - - /* Find the right SBR channel */ - pSbrDrcChannelData = sbrDecoder_drcGetChannel( self, ch ); - - if ( pSbrDrcChannelData != NULL ) { - if ( pSbrDrcChannelData->enable || isValidData ) - { /* Activate processing only with real and valid data */ - int i; - - pSbrDrcChannelData->enable = 1; - pSbrDrcChannelData->numBandsNext = numBands; - - pSbrDrcChannelData->winSequenceNext = winSequence; - pSbrDrcChannelData->drcInterpolationSchemeNext = drcInterpolationScheme; - pSbrDrcChannelData->nextFact_exp = nextFact_exp; - - for (i = 0; i < (int)numBands; i++) { - pSbrDrcChannelData->bandTopNext[i] = pBandTop[i]; - pSbrDrcChannelData->nextFact_mag[i] = pNextFact_mag[i]; - } - } - } - - return SBRDEC_OK; -} - - -void sbrDecoder_drcDisable ( HANDLE_SBRDECODER self, - INT ch ) -{ - SBRDEC_DRC_CHANNEL *pSbrDrcChannelData = NULL; - - if ( (self == NULL) - || (ch > (8)) - || (self->numSbrElements == 0) - || (self->numSbrChannels == 0) ) { - return; - } - - /* Find the right SBR channel */ - pSbrDrcChannelData = sbrDecoder_drcGetChannel( self, ch ); - - if ( pSbrDrcChannelData != NULL ) { - sbrDecoder_drcInitChannel( pSbrDrcChannelData ); - } -} - - - -SBR_ERROR sbrDecoder_Parse( - HANDLE_SBRDECODER self, - HANDLE_FDK_BITSTREAM hBs, - int *count, - int bsPayLen, - int crcFlag, - MP4_ELEMENT_ID prevElement, - int elementIndex, - int fGlobalIndependencyFlag - ) -{ - SBR_DECODER_ELEMENT *hSbrElement; - HANDLE_SBR_HEADER_DATA hSbrHeader = NULL; - HANDLE_SBR_CHANNEL *pSbrChannel; - - SBR_FRAME_DATA *hFrameDataLeft; - SBR_FRAME_DATA *hFrameDataRight; - - SBR_ERROR errorStatus = SBRDEC_OK; - SBR_HEADER_STATUS headerStatus = HEADER_NOT_PRESENT; - - INT startPos; - INT CRCLen = 0; - HANDLE_FDK_BITSTREAM hBsOriginal = hBs; - FDK_CRCINFO crcInfo; /* shall be used for all other CRCs in the future (TBD) */ - INT crcReg = 0; - USHORT drmSbrCrc = 0; - - int stereo; - int fDoDecodeSbrData = 1; - - int lastSlot, lastHdrSlot = 0, thisHdrSlot; - - /* Reverse bits of DRM SBR payload */ - if ( (self->flags & SBRDEC_SYNTAX_DRM) && *count > 0 ) - { - UCHAR *bsBufferDrm = (UCHAR*)self->workBuffer1; - HANDLE_FDK_BITSTREAM hBsBwd = (HANDLE_FDK_BITSTREAM) (bsBufferDrm + (512)); - int dataBytes, dataBits; - - dataBits = *count; - - if (dataBits > ((512)*8)) { - /* do not flip more data than needed */ - dataBits = (512)*8; - } - - dataBytes = (dataBits+7)>>3; - - int j; - - if ((j = (int)FDKgetValidBits(hBs)) != 8) { - FDKpushBiDirectional(hBs, (j-8)); - } - - j = 0; - for ( ; dataBytes > 0; dataBytes--) - { - int i; - UCHAR tmpByte; - UCHAR buffer = 0x00; - - tmpByte = (UCHAR) FDKreadBits(hBs, 8); - for (i = 0; i < 4; i++) { - int shift = 2 * i + 1; - buffer |= (tmpByte & (0x08>>i)) << shift; - buffer |= (tmpByte & (0x10<> shift; - } - bsBufferDrm[j++] = buffer; - FDKpushBack(hBs, 16); - } - - FDKinitBitStream(hBsBwd, bsBufferDrm, (512), dataBits, BS_READER); - - /* Use reversed data */ - hBs = hBsBwd; - bsPayLen = *count; - } - - /* Remember start position of SBR element */ - startPos = FDKgetValidBits(hBs); - - /* SBR sanity checks */ - if ( self == NULL || self->pSbrElement[elementIndex] == NULL ) { - errorStatus = SBRDEC_NOT_INITIALIZED; - goto bail; - } - - hSbrElement = self->pSbrElement[elementIndex]; - - lastSlot = (hSbrElement->useFrameSlot > 0) ? hSbrElement->useFrameSlot-1 : self->numDelayFrames; - lastHdrSlot = hSbrElement->useHeaderSlot[lastSlot]; - thisHdrSlot = getHeaderSlot( hSbrElement->useFrameSlot, hSbrElement->useHeaderSlot ); /* Get a free header slot not used by frames not processed yet. */ - - /* Assign the free slot to store a new header if there is one. */ - hSbrHeader = &self->sbrHeader[elementIndex][thisHdrSlot]; - - pSbrChannel = hSbrElement->pSbrChannel; - stereo = (hSbrElement->elementID == ID_CPE) ? 1 : 0; - - hFrameDataLeft = &self->pSbrElement[elementIndex]->pSbrChannel[0]->frameData[hSbrElement->useFrameSlot]; - hFrameDataRight = &self->pSbrElement[elementIndex]->pSbrChannel[1]->frameData[hSbrElement->useFrameSlot]; - - - /* reset PS flag; will be set after PS was found */ - self->flags &= ~SBRDEC_PS_DECODED; - - if (hSbrHeader->status & SBRDEC_HDR_STAT_UPDATE) { - /* Got a new header from extern (e.g. from an ASC) */ - headerStatus = HEADER_OK; - hSbrHeader->status &= ~SBRDEC_HDR_STAT_UPDATE; - } - else if (thisHdrSlot != lastHdrSlot) { - /* Copy the last header into this slot otherwise the - header compare will trigger more HEADER_RESETs than needed. */ - copySbrHeader( hSbrHeader, &self->sbrHeader[elementIndex][lastHdrSlot] ); - } - - /* - Check if bit stream data is valid and matches the element context - */ - if ( ((prevElement != ID_SCE) && (prevElement != ID_CPE)) || prevElement != hSbrElement->elementID) { - /* In case of LFE we also land here, since there is no LFE SBR element (do upsampling only) */ - fDoDecodeSbrData = 0; - } - - if (fDoDecodeSbrData) - { - if ((INT)FDKgetValidBits(hBs) <= 0) { - fDoDecodeSbrData = 0; - } - } - - /* - SBR CRC-check - */ - if (fDoDecodeSbrData) - { - if (crcFlag) { - switch (self->coreCodec) { - case AOT_ER_AAC_ELD: - FDKpushFor (hBs, 10); - /* check sbrcrc later: we don't know the payload length now */ - break; - case AOT_DRM_AAC: - drmSbrCrc = (USHORT)FDKreadBits(hBs, 8); - /* Setup CRC decoder */ - FDKcrcInit(&crcInfo, 0x001d, 0xFFFF, 8); - /* Start CRC region */ - crcReg = FDKcrcStartReg(&crcInfo, hBs, 0); - break; - default: - CRCLen = bsPayLen - 10; /* change: 0 => i */ - if (CRCLen < 0) { - fDoDecodeSbrData = 0; - } else { - fDoDecodeSbrData = SbrCrcCheck (hBs, CRCLen); - } - break; - } - } - } /* if (fDoDecodeSbrData) */ - - /* - Read in the header data and issue a reset if change occured - */ - if (fDoDecodeSbrData) - { - int sbrHeaderPresent; - - { - sbrHeaderPresent = FDKreadBit(hBs); - } - - if ( sbrHeaderPresent ) { - headerStatus = sbrGetHeaderData (hSbrHeader, - hBs, - self->flags, - 1); - } - - if (headerStatus == HEADER_RESET) - { - errorStatus = sbrDecoder_HeaderUpdate( - self, - hSbrHeader, - headerStatus, - pSbrChannel, - hSbrElement->nChannels - ); - - if (errorStatus == SBRDEC_OK) { - hSbrHeader->syncState = SBR_HEADER; - } else { - hSbrHeader->syncState = SBR_NOT_INITIALIZED; - headerStatus = HEADER_ERROR; - } - } - - if (errorStatus != SBRDEC_OK) { - fDoDecodeSbrData = 0; - } - } /* if (fDoDecodeSbrData) */ - - /* - Print debugging output only if state has changed - */ - - /* read frame data */ - if ((hSbrHeader->syncState >= SBR_HEADER) && fDoDecodeSbrData) { - int sbrFrameOk; - /* read the SBR element data */ - if (stereo) { - sbrFrameOk = sbrGetChannelPairElement(hSbrHeader, - hFrameDataLeft, - hFrameDataRight, - hBs, - self->flags, - self->pSbrElement[elementIndex]->transposerSettings.overlap); - } - else { - if (self->hParametricStereoDec != NULL) { - /* update slot index for PS bitstream parsing */ - self->hParametricStereoDec->bsLastSlot = self->hParametricStereoDec->bsReadSlot; - self->hParametricStereoDec->bsReadSlot = hSbrElement->useFrameSlot; - } - sbrFrameOk = sbrGetSingleChannelElement(hSbrHeader, - hFrameDataLeft, - hBs, - self->hParametricStereoDec, - self->flags, - self->pSbrElement[elementIndex]->transposerSettings.overlap); - } - if (!sbrFrameOk) { - fDoDecodeSbrData = 0; - } - else { - INT valBits; - - if (bsPayLen > 0) { - valBits = bsPayLen - ((INT)startPos - (INT)FDKgetValidBits(hBs)); - } else { - valBits = (INT)FDKgetValidBits(hBs); - } - - if ( crcFlag ) { - switch (self->coreCodec) { - case AOT_ER_AAC_ELD: - { - /* late crc check for eld */ - INT payloadbits = (INT)startPos - (INT)FDKgetValidBits(hBs) - startPos; - INT crcLen = payloadbits - 10; - FDKpushBack(hBs, payloadbits); - fDoDecodeSbrData = SbrCrcCheck (hBs, crcLen); - FDKpushFor(hBs, crcLen); - } - break; - case AOT_DRM_AAC: - /* End CRC region */ - FDKcrcEndReg(&crcInfo, hBs, crcReg); - /* Check CRC */ - if ((FDKcrcGetCRC(&crcInfo)^0xFF) != drmSbrCrc) { - fDoDecodeSbrData = 0; - } - break; - default: - break; - } - } - - /* sanity check of remaining bits */ - if (valBits < 0) { - fDoDecodeSbrData = 0; - } else { - switch (self->coreCodec) { - case AOT_SBR: - case AOT_PS: - case AOT_AAC_LC: - { - /* This sanity check is only meaningful with General Audio bitstreams */ - int alignBits = valBits & 0x7; - - if (valBits > alignBits) { - fDoDecodeSbrData = 0; - } - } - break; - default: - /* No sanity check available */ - break; - } - } - } - } else { - /* The returned bit count will not be the actual payload size since we did not - parse the frame data. Return an error so that the caller can react respectively. */ - errorStatus = SBRDEC_PARSE_ERROR; - } - - if (!fDoDecodeSbrData) { - /* Set error flag for this slot to trigger concealment */ - self->pSbrElement[elementIndex]->frameErrorFlag[hSbrElement->useFrameSlot] = 1; - errorStatus = SBRDEC_PARSE_ERROR; - } else { - /* Everything seems to be ok so clear the error flag */ - self->pSbrElement[elementIndex]->frameErrorFlag[hSbrElement->useFrameSlot] = 0; - } - - if (!stereo) { - /* Turn coupling off explicitely to avoid access to absent right frame data - that might occur with corrupt bitstreams. */ - hFrameDataLeft->coupling = COUPLING_OFF; - } - -bail: - - if ( self->flags & SBRDEC_SYNTAX_DRM ) - { - hBs = hBsOriginal; - } - - if ( (errorStatus == SBRDEC_OK) - || ( (errorStatus == SBRDEC_PARSE_ERROR) - && (headerStatus != HEADER_ERROR) ) ) - { - int useOldHdr = ( (headerStatus == HEADER_NOT_PRESENT) - || (headerStatus == HEADER_ERROR) ) ? 1 : 0; - - if (!useOldHdr && (thisHdrSlot != lastHdrSlot)) { - useOldHdr |= ( compareSbrHeader( hSbrHeader, - &self->sbrHeader[elementIndex][lastHdrSlot] ) == 0 ) ? 1 : 0; - } - - if (useOldHdr != 0) { - /* Use the old header for this frame */ - hSbrElement->useHeaderSlot[hSbrElement->useFrameSlot] = lastHdrSlot; - } else { - /* Use the new header for this frame */ - hSbrElement->useHeaderSlot[hSbrElement->useFrameSlot] = thisHdrSlot; - } - - /* Move frame pointer to the next slot which is up to be decoded/applied next */ - hSbrElement->useFrameSlot = (hSbrElement->useFrameSlot+1) % (self->numDelayFrames+1); - } - - *count -= startPos - FDKgetValidBits(hBs); - - return errorStatus; -} - - -/** - * \brief Render one SBR element into time domain signal. - * \param self SBR decoder handle - * \param timeData pointer to output buffer - * \param interleaved flag indicating interleaved channel output - * \param channelMapping pointer to UCHAR array where next 2 channel offsets are stored. - * \param elementIndex enumerating index of the SBR element to render. - * \param numInChannels number of channels from core coder (reading stride). - * \param numOutChannels pointer to a location to return number of output channels. - * \param psPossible flag indicating if PS is possible or not. - * \return SBRDEC_OK if successfull, else error code - */ -static SBR_ERROR -sbrDecoder_DecodeElement ( - HANDLE_SBRDECODER self, - INT_PCM *timeData, - const int interleaved, - const UCHAR *channelMapping, - const int elementIndex, - const int numInChannels, - int *numOutChannels, - const int psPossible - ) -{ - SBR_DECODER_ELEMENT *hSbrElement = self->pSbrElement[elementIndex]; - HANDLE_SBR_CHANNEL *pSbrChannel = self->pSbrElement[elementIndex]->pSbrChannel; - HANDLE_SBR_HEADER_DATA hSbrHeader = &self->sbrHeader[elementIndex][hSbrElement->useHeaderSlot[hSbrElement->useFrameSlot]]; - HANDLE_PS_DEC h_ps_d = self->hParametricStereoDec; - - /* get memory for frame data from scratch */ - SBR_FRAME_DATA *hFrameDataLeft = &hSbrElement->pSbrChannel[0]->frameData[hSbrElement->useFrameSlot]; - SBR_FRAME_DATA *hFrameDataRight = &hSbrElement->pSbrChannel[1]->frameData[hSbrElement->useFrameSlot]; - - SBR_ERROR errorStatus = SBRDEC_OK; - - - INT strideIn, strideOut, offset0, offset1; - INT codecFrameSize = self->codecFrameSize; - - int stereo = (hSbrElement->elementID == ID_CPE) ? 1 : 0; - int numElementChannels = hSbrElement->nChannels; /* Number of channels of the current SBR element */ - - if (self->flags & SBRDEC_FLUSH) { - if ( self->numFlushedFrames > self->numDelayFrames ) { - int hdrIdx; - /* No valid SBR payload available, hence switch to upsampling (in all headers) */ - for (hdrIdx = 0; hdrIdx < ((1)+1); hdrIdx += 1) { - self->sbrHeader[elementIndex][hdrIdx].syncState = UPSAMPLING; - } - } - else { - /* Move frame pointer to the next slot which is up to be decoded/applied next */ - hSbrElement->useFrameSlot = (hSbrElement->useFrameSlot+1) % (self->numDelayFrames+1); - /* Update header and frame data pointer because they have already been set */ - hSbrHeader = &self->sbrHeader[elementIndex][hSbrElement->useHeaderSlot[hSbrElement->useFrameSlot]]; - hFrameDataLeft = &hSbrElement->pSbrChannel[0]->frameData[hSbrElement->useFrameSlot]; - hFrameDataRight = &hSbrElement->pSbrChannel[1]->frameData[hSbrElement->useFrameSlot]; - } - } - - /* Update the header error flag */ - hSbrHeader->frameErrorFlag = hSbrElement->frameErrorFlag[hSbrElement->useFrameSlot]; - - /* - Prepare filterbank for upsampling if no valid bit stream data is available. - */ - if ( hSbrHeader->syncState == SBR_NOT_INITIALIZED ) - { - errorStatus = initHeaderData( - hSbrHeader, - self->sampleRateIn, - self->sampleRateOut, - codecFrameSize, - self->flags - ); - - if (errorStatus != SBRDEC_OK) { - return errorStatus; - } - - hSbrHeader->syncState = UPSAMPLING; - - errorStatus = sbrDecoder_HeaderUpdate( - self, - hSbrHeader, - HEADER_NOT_PRESENT, - pSbrChannel, - hSbrElement->nChannels - ); - - if (errorStatus != SBRDEC_OK) { - hSbrHeader->syncState = SBR_NOT_INITIALIZED; - return errorStatus; - } - } - - /* reset */ - if (hSbrHeader->status & SBRDEC_HDR_STAT_RESET) { - int ch; - for (ch = 0 ; ch < numElementChannels; ch++) { - SBR_ERROR errorStatusTmp = SBRDEC_OK; - - errorStatusTmp = resetSbrDec ( - &pSbrChannel[ch]->SbrDec, - hSbrHeader, - &pSbrChannel[ch]->prevFrameData, - self->flags & SBRDEC_LOW_POWER, - self->synDownsampleFac - ); - - if (errorStatusTmp != SBRDEC_OK) { - errorStatus = errorStatusTmp; - } - } - hSbrHeader->status &= ~SBRDEC_HDR_STAT_RESET; - } - - /* decoding */ - if ( (hSbrHeader->syncState == SBR_ACTIVE) - || ((hSbrHeader->syncState == SBR_HEADER) && (hSbrHeader->frameErrorFlag == 0)) ) - { - errorStatus = SBRDEC_OK; - - decodeSbrData (hSbrHeader, - hFrameDataLeft, - &pSbrChannel[0]->prevFrameData, - (stereo) ? hFrameDataRight : NULL, - (stereo) ? &pSbrChannel[1]->prevFrameData : NULL); - - - /* Now we have a full parameter set and can do parameter - based concealment instead of plain upsampling. */ - hSbrHeader->syncState = SBR_ACTIVE; - } - - /* decode PS data if available */ - if (h_ps_d != NULL && psPossible) { - int applyPs = 1; - - /* define which frame delay line slot to process */ - h_ps_d->processSlot = hSbrElement->useFrameSlot; - - applyPs = DecodePs(h_ps_d, hSbrHeader->frameErrorFlag); - self->flags |= (applyPs) ? SBRDEC_PS_DECODED : 0; - } - - if (channelMapping[0] == 255 || channelMapping[1] == 255) - return SBRDEC_UNSUPPORTED_CONFIG; - if (!pSbrChannel[0]->SbrDec.LppTrans.pSettings) - return SBRDEC_UNSUPPORTED_CONFIG; - if (stereo && !pSbrChannel[1]->SbrDec.LppTrans.pSettings) - return SBRDEC_UNSUPPORTED_CONFIG; - - /* Set strides for reading and writing */ - if (interleaved) { - strideIn = numInChannels; - if ( psPossible ) - strideOut = (numInChannels < 2) ? 2 : numInChannels; - else - strideOut = numInChannels; - offset0 = channelMapping[0]; - offset1 = channelMapping[1]; - } else { - strideIn = 1; - strideOut = 1; - offset0 = channelMapping[0]*2*codecFrameSize; - offset1 = channelMapping[1]*2*codecFrameSize; - } - - /* use same buffers for left and right channel and apply PS per timeslot */ - /* Process left channel */ -//FDKprintf("self->codecFrameSize %d\t%d\n",self->codecFrameSize,self->sampleRateIn); - sbr_dec (&pSbrChannel[0]->SbrDec, - timeData + offset0, - timeData + offset0, - &pSbrChannel[1]->SbrDec, - timeData + offset1, - strideIn, - strideOut, - hSbrHeader, - hFrameDataLeft, - &pSbrChannel[0]->prevFrameData, - (hSbrHeader->syncState == SBR_ACTIVE), - h_ps_d, - self->flags, - codecFrameSize - ); - - if (stereo) { - /* Process right channel */ - sbr_dec (&pSbrChannel[1]->SbrDec, - timeData + offset1, - timeData + offset1, - NULL, - NULL, - strideIn, - strideOut, - hSbrHeader, - hFrameDataRight, - &pSbrChannel[1]->prevFrameData, - (hSbrHeader->syncState == SBR_ACTIVE), - NULL, - self->flags, - codecFrameSize - ); - } - - if (h_ps_d != NULL) { - /* save PS status for next run */ - h_ps_d->psDecodedPrv = (self->flags & SBRDEC_PS_DECODED) ? 1 : 0 ; - } - - if ( psPossible - ) - { - FDK_ASSERT(strideOut > 1); - if ( !(self->flags & SBRDEC_PS_DECODED) ) { - /* A decoder which is able to decode PS has to produce a stereo output even if no PS data is availble. */ - /* So copy left channel to right channel. */ - int copyFrameSize = codecFrameSize * 2 / self->synDownsampleFac; - if (interleaved) { - INT_PCM *ptr; - INT i; - FDK_ASSERT(strideOut == 2); - - ptr = timeData; - for (i = copyFrameSize>>1; i--; ) - { - INT_PCM tmp; /* This temporal variable is required because some compilers can't do *ptr++ = *ptr++ correctly. */ - tmp = *ptr++; *ptr++ = tmp; - tmp = *ptr++; *ptr++ = tmp; - } - } else { - FDKmemcpy( timeData+copyFrameSize, timeData, copyFrameSize*sizeof(INT_PCM) ); - } - } - *numOutChannels = 2; /* Output minimum two channels when PS is enabled. */ - } - - return errorStatus; -} - - -SBR_ERROR sbrDecoder_Apply ( HANDLE_SBRDECODER self, - INT_PCM *timeData, - int *numChannels, - int *sampleRate, - const UCHAR channelMapping[(8)], - const int interleaved, - const int coreDecodedOk, - UCHAR *psDecoded ) -{ - SBR_ERROR errorStatus = SBRDEC_OK; - - int psPossible = 0; - int sbrElementNum; - int numCoreChannels = *numChannels; - int numSbrChannels = 0; - - psPossible = *psDecoded; - - if (self->numSbrElements < 1) { - /* exit immediately to avoid access violations */ - return SBRDEC_CREATE_ERROR; - } - - /* Sanity check of allocated SBR elements. */ - for (sbrElementNum=0; sbrElementNumnumSbrElements; sbrElementNum++) { - if (self->pSbrElement[sbrElementNum] == NULL) { - return SBRDEC_CREATE_ERROR; - } - } - - if (self->numSbrElements != 1 || self->pSbrElement[0]->elementID != ID_SCE) { - psPossible = 0; - } - - - /* In case of non-interleaved time domain data and upsampling, make room for bigger SBR output. */ - if (self->synDownsampleFac == 1 && interleaved == 0) { - int c, outputFrameSize; - - outputFrameSize = - self->pSbrElement[0]->pSbrChannel[0]->SbrDec.SynthesisQMF.no_channels - * self->pSbrElement[0]->pSbrChannel[0]->SbrDec.SynthesisQMF.no_col; - - for (c=numCoreChannels-1; c>0; c--) { - FDKmemmove(timeData + c*outputFrameSize, timeData + c*self->codecFrameSize , self->codecFrameSize*sizeof(INT_PCM)); - } - } - - - /* Make sure that even if no SBR data was found/parsed *psDecoded is returned 1 if psPossible was 0. */ - if (psPossible == 0) { - self->flags &= ~SBRDEC_PS_DECODED; - } - - if ( self->flags & SBRDEC_FLUSH ) { - /* flushing is signalized, hence increment the flush frame counter */ - self->numFlushedFrames++; - } - else { - /* no flushing is signalized, hence reset the flush frame counter */ - self->numFlushedFrames = 0; - } - - /* Loop over SBR elements */ - for (sbrElementNum = 0; sbrElementNumnumSbrElements; sbrElementNum++) - { - int numElementChan; - - if (psPossible && self->pSbrElement[sbrElementNum]->pSbrChannel[1] == NULL) { - /* Disable PS and try decoding SBR mono. */ - psPossible = 0; - } - - numElementChan = (self->pSbrElement[sbrElementNum]->elementID == ID_CPE) ? 2 : 1; - - /* If core signal is bad then force upsampling */ - if ( ! coreDecodedOk ) { - self->pSbrElement[sbrElementNum]->frameErrorFlag[self->pSbrElement[sbrElementNum]->useFrameSlot] = 1; - } - - errorStatus = sbrDecoder_DecodeElement ( - self, - timeData, - interleaved, - channelMapping, - sbrElementNum, - numCoreChannels, - &numElementChan, - psPossible - ); - - if (errorStatus != SBRDEC_OK) { - goto bail; - } - - numSbrChannels += numElementChan; - channelMapping += numElementChan; - - if (numSbrChannels >= numCoreChannels) { - break; - } - } - - /* Update numChannels and samplerate */ - *numChannels = numSbrChannels; - *sampleRate = self->sampleRateOut; - *psDecoded = (self->flags & SBRDEC_PS_DECODED) ? 1 : 0; - - - - /* Clear reset and flush flag because everything seems to be done successfully. */ - self->flags &= ~SBRDEC_FORCE_RESET; - self->flags &= ~SBRDEC_FLUSH; - -bail: - - return errorStatus; -} - - -SBR_ERROR sbrDecoder_Close ( HANDLE_SBRDECODER *pSelf ) -{ - HANDLE_SBRDECODER self = *pSelf; - int i; - - if (self != NULL) - { - if (self->hParametricStereoDec != NULL) { - DeletePsDec ( &self->hParametricStereoDec ); - } - - if (self->workBuffer1 != NULL) { - FreeRam_SbrDecWorkBuffer1(&self->workBuffer1); - } - if (self->workBuffer2 != NULL) { - FreeRam_SbrDecWorkBuffer2(&self->workBuffer2); - } - - for (i = 0; i < (8); i++) { - sbrDecoder_DestroyElement( self, i ); - } - - FreeRam_SbrDecoder(pSelf); - } - - return SBRDEC_OK; -} - - -INT sbrDecoder_GetLibInfo( LIB_INFO *info ) -{ - int i; - - if (info == NULL) { - return -1; - } - - /* search for next free tab */ - for (i = 0; i < FDK_MODULE_LAST; i++) { - if (info[i].module_id == FDK_NONE) - break; - } - if (i == FDK_MODULE_LAST) - return -1; - info += i; - - info->module_id = FDK_SBRDEC; - info->version = LIB_VERSION(SBRDECODER_LIB_VL0, SBRDECODER_LIB_VL1, SBRDECODER_LIB_VL2); - LIB_VERSION_STRING(info); - info->build_date = (char *)SBRDECODER_LIB_BUILD_DATE; - info->build_time = (char *)SBRDECODER_LIB_BUILD_TIME; - info->title = (char *)SBRDECODER_LIB_TITLE; - - /* Set flags */ - info->flags = 0 - | CAPF_SBR_HQ - | CAPF_SBR_LP - | CAPF_SBR_PS_MPEG - | CAPF_SBR_DRM_BS - | CAPF_SBR_CONCEALMENT - | CAPF_SBR_DRC - ; - /* End of flags */ - - return 0; -} - - -UINT sbrDecoder_GetDelay( const HANDLE_SBRDECODER self ) -{ - UINT outputDelay = 0; - - if ( self != NULL) { - UINT flags = self->flags; - - /* See chapter 1.6.7.2 of ISO/IEC 14496-3 for the GA-SBR figures below. */ - - /* Are we initialized? */ - if ( (self->numSbrChannels > 0) - && (self->numSbrElements > 0) ) - { - /* Add QMF synthesis delay */ - if ( (flags & SBRDEC_ELD_GRID) - && IS_LOWDELAY(self->coreCodec) ) { - /* Low delay SBR: */ - { - outputDelay += (flags & SBRDEC_DOWNSAMPLE) ? 32 : 64; /* QMF synthesis */ - if (flags & SBRDEC_LD_MPS_QMF) { - outputDelay += 32; - } - } - } - else if (!IS_USAC(self->coreCodec)) { - /* By the method of elimination this is the GA (AAC-LC, HE-AAC, ...) branch: */ - outputDelay += (flags & SBRDEC_DOWNSAMPLE) ? 481 : 962; - } - } - } - - return (outputDelay); -} diff --git a/libSBRdec/src/transcendent.h b/libSBRdec/src/transcendent.h deleted file mode 100644 index ad88bc9..0000000 --- a/libSBRdec/src/transcendent.h +++ /dev/null @@ -1,355 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief FDK Fixed Point Arithmetic Library Interface -*/ - -#ifndef __TRANSCENDENT_H -#define __TRANSCENDENT_H - -#include "sbrdecoder.h" -#include "sbr_rom.h" - -/************************************************************************/ -/*! - \brief Get number of octaves between frequencies a and b - - The Result is scaled with 1/8. - The valid range for a and b is 1 to LOG_DUALIS_TABLE_SIZE. - - \return ld(a/b) / 8 -*/ -/************************************************************************/ -static inline FIXP_SGL FDK_getNumOctavesDiv8(INT a, /*!< lower band */ - INT b) /*!< upper band */ -{ - return ( (SHORT)((LONG)(CalcLdInt(b) - CalcLdInt(a))>>(FRACT_BITS-3)) ); -} - - -/************************************************************************/ -/*! - \brief Add two values given by mantissa and exponent. - - Mantissas are in fract format with values between 0 and 1.
- The base for exponents is 2. Example: \f$ a = a\_m * 2^{a\_e} \f$
-*/ -/************************************************************************/ -inline void FDK_add_MantExp(FIXP_SGL a_m, /*!< Mantissa of 1st operand a */ - SCHAR a_e, /*!< Exponent of 1st operand a */ - FIXP_SGL b_m, /*!< Mantissa of 2nd operand b */ - SCHAR b_e, /*!< Exponent of 2nd operand b */ - FIXP_SGL *ptrSum_m, /*!< Mantissa of result */ - SCHAR *ptrSum_e) /*!< Exponent of result */ -{ - FIXP_DBL accu; - int shift; - int shiftAbs; - - FIXP_DBL shiftedMantissa; - FIXP_DBL otherMantissa; - - /* Equalize exponents of the summands. - For the smaller summand, the exponent is adapted and - for compensation, the mantissa is shifted right. */ - - shift = (int)(a_e - b_e); - - shiftAbs = (shift>0)? shift : -shift; - shiftAbs = (shiftAbs < DFRACT_BITS-1)? shiftAbs : DFRACT_BITS-1; - shiftedMantissa = (shift>0)? (FX_SGL2FX_DBL(b_m) >> shiftAbs) : (FX_SGL2FX_DBL(a_m) >> shiftAbs); - otherMantissa = (shift>0)? FX_SGL2FX_DBL(a_m) : FX_SGL2FX_DBL(b_m); - *ptrSum_e = (shift>0)? a_e : b_e; - - accu = (shiftedMantissa >> 1) + (otherMantissa >> 1); - /* shift by 1 bit to avoid overflow */ - - if ( (accu >= (FL2FXCONST_DBL(0.5f) - (FIXP_DBL)1)) || (accu <= FL2FXCONST_DBL(-0.5f)) ) - *ptrSum_e += 1; - else - accu = (shiftedMantissa + otherMantissa); - - *ptrSum_m = FX_DBL2FX_SGL(accu); - -} - -inline void FDK_add_MantExp(FIXP_DBL a, /*!< Mantissa of 1st operand a */ - SCHAR a_e, /*!< Exponent of 1st operand a */ - FIXP_DBL b, /*!< Mantissa of 2nd operand b */ - SCHAR b_e, /*!< Exponent of 2nd operand b */ - FIXP_DBL *ptrSum, /*!< Mantissa of result */ - SCHAR *ptrSum_e) /*!< Exponent of result */ -{ - FIXP_DBL accu; - int shift; - int shiftAbs; - - FIXP_DBL shiftedMantissa; - FIXP_DBL otherMantissa; - - /* Equalize exponents of the summands. - For the smaller summand, the exponent is adapted and - for compensation, the mantissa is shifted right. */ - - shift = (int)(a_e - b_e); - - shiftAbs = (shift>0)? shift : -shift; - shiftAbs = (shiftAbs < DFRACT_BITS-1)? shiftAbs : DFRACT_BITS-1; - shiftedMantissa = (shift>0)? (b >> shiftAbs) : (a >> shiftAbs); - otherMantissa = (shift>0)? a : b; - *ptrSum_e = (shift>0)? a_e : b_e; - - accu = (shiftedMantissa >> 1) + (otherMantissa >> 1); - /* shift by 1 bit to avoid overflow */ - - if ( (accu >= (FL2FXCONST_DBL(0.5f) - (FIXP_DBL)1)) || (accu <= FL2FXCONST_DBL(-0.5f)) ) - *ptrSum_e += 1; - else - accu = (shiftedMantissa + otherMantissa); - - *ptrSum = accu; - -} - -/************************************************************************/ -/*! - \brief Divide two values given by mantissa and exponent. - - Mantissas are in fract format with values between 0 and 1.
- The base for exponents is 2. Example: \f$ a = a\_m * 2^{a\_e} \f$
- - For performance reasons, the division is based on a table lookup - which limits accuracy. -*/ -/************************************************************************/ -static inline void FDK_divide_MantExp(FIXP_SGL a_m, /*!< Mantissa of dividend a */ - SCHAR a_e, /*!< Exponent of dividend a */ - FIXP_SGL b_m, /*!< Mantissa of divisor b */ - SCHAR b_e, /*!< Exponent of divisor b */ - FIXP_SGL *ptrResult_m, /*!< Mantissa of quotient a/b */ - SCHAR *ptrResult_e) /*!< Exponent of quotient a/b */ - -{ - int preShift, postShift, index, shift; - FIXP_DBL ratio_m; - FIXP_SGL bInv_m = FL2FXCONST_SGL(0.0f); - - preShift = CntLeadingZeros(FX_SGL2FX_DBL(b_m)); - - /* - Shift b into the range from 0..INV_TABLE_SIZE-1, - - E.g. 10 bits must be skipped for INV_TABLE_BITS 8: - - leave 8 bits as index for table - - skip sign bit, - - skip first bit of mantissa, because this is always the same (>0.5) - - We are dealing with energies, so we need not care - about negative numbers - */ - - /* - The first interval has half width so the lowest bit of the index is - needed for a doubled resolution. - */ - shift = (FRACT_BITS - 2 - INV_TABLE_BITS - preShift); - - index = (shift<0)? (LONG)b_m << (-shift) : (LONG)b_m >> shift; - - - /* The index has INV_TABLE_BITS +1 valid bits here. Clear the other bits. */ - index &= (1 << (INV_TABLE_BITS+1)) - 1; - - /* Remove offset of half an interval */ - index--; - - /* Now the lowest bit is shifted out */ - index = index >> 1; - - /* Fetch inversed mantissa from table: */ - bInv_m = (index<0)? bInv_m : FDK_sbrDecoder_invTable[index]; - - /* Multiply a with the inverse of b: */ - ratio_m = (index<0)? FX_SGL2FX_DBL(a_m >> 1) : fMultDiv2(bInv_m,a_m); - - postShift = CntLeadingZeros(ratio_m)-1; - - *ptrResult_m = FX_DBL2FX_SGL(ratio_m << postShift); - *ptrResult_e = a_e - b_e + 1 + preShift - postShift; -} - -static inline void FDK_divide_MantExp(FIXP_DBL a_m, /*!< Mantissa of dividend a */ - SCHAR a_e, /*!< Exponent of dividend a */ - FIXP_DBL b_m, /*!< Mantissa of divisor b */ - SCHAR b_e, /*!< Exponent of divisor b */ - FIXP_DBL *ptrResult_m, /*!< Mantissa of quotient a/b */ - SCHAR *ptrResult_e) /*!< Exponent of quotient a/b */ - -{ - int preShift, postShift, index, shift; - FIXP_DBL ratio_m; - FIXP_SGL bInv_m = FL2FXCONST_SGL(0.0f); - - preShift = CntLeadingZeros(b_m); - - /* - Shift b into the range from 0..INV_TABLE_SIZE-1, - - E.g. 10 bits must be skipped for INV_TABLE_BITS 8: - - leave 8 bits as index for table - - skip sign bit, - - skip first bit of mantissa, because this is always the same (>0.5) - - We are dealing with energies, so we need not care - about negative numbers - */ - - /* - The first interval has half width so the lowest bit of the index is - needed for a doubled resolution. - */ - shift = (DFRACT_BITS - 2 - INV_TABLE_BITS - preShift); - - index = (shift<0)? (LONG)b_m << (-shift) : (LONG)b_m >> shift; - - - /* The index has INV_TABLE_BITS +1 valid bits here. Clear the other bits. */ - index &= (1 << (INV_TABLE_BITS+1)) - 1; - - /* Remove offset of half an interval */ - index--; - - /* Now the lowest bit is shifted out */ - index = index >> 1; - - /* Fetch inversed mantissa from table: */ - bInv_m = (index<0)? bInv_m : FDK_sbrDecoder_invTable[index]; - - /* Multiply a with the inverse of b: */ - ratio_m = (index<0)? (a_m >> 1) : fMultDiv2(bInv_m,a_m); - - postShift = CntLeadingZeros(ratio_m)-1; - - *ptrResult_m = ratio_m << postShift; - *ptrResult_e = a_e - b_e + 1 + preShift - postShift; -} - -/*! - \brief Calculate the squareroot of a number given by mantissa and exponent - - Mantissa is in fract format with values between 0 and 1.
- The base for the exponent is 2. Example: \f$ a = a\_m * 2^{a\_e} \f$
- The operand is addressed via pointers and will be overwritten with the result. - - For performance reasons, the square root is based on a table lookup - which limits accuracy. -*/ -static inline void FDK_sqrt_MantExp(FIXP_DBL *mantissa, /*!< Pointer to mantissa */ - SCHAR *exponent, - const SCHAR *destScale) -{ - FIXP_DBL input_m = *mantissa; - int input_e = (int) *exponent; - FIXP_DBL result = FL2FXCONST_DBL(0.0f); - int result_e = -FRACT_BITS; - - /* Call lookup square root, which does internally normalization. */ - result = sqrtFixp_lookup(input_m, &input_e); - result_e = input_e; - - /* Write result */ - if (exponent==destScale) { - *mantissa = result; - *exponent = result_e; - } else { - int shift = result_e - *destScale; - *mantissa = (shift>=0) ? result << (INT)fixMin(DFRACT_BITS-1,shift) - : result >> (INT)fixMin(DFRACT_BITS-1,-shift); - *exponent = *destScale; - } -} - - -#endif diff --git a/libSBRenc/include/sbr_encoder.h b/libSBRenc/include/sbr_encoder.h deleted file mode 100644 index aec0398..0000000 --- a/libSBRenc/include/sbr_encoder.h +++ /dev/null @@ -1,430 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*************************** Fraunhofer IIS *********************** - - Author(s): - Description: SBR encoder top level processing prototype - -******************************************************************************/ - -#ifndef __SBR_ENCODER_H -#define __SBR_ENCODER_H - -#include "common_fix.h" -#include "FDK_audio.h" - -#include "FDK_bitstream.h" - -/* core coder helpers */ -#define MAX_TRANS_FAC 8 -#define MAX_CODEC_FRAME_RATIO 2 -#define MAX_PAYLOAD_SIZE 256 - -typedef enum codecType -{ - CODEC_AAC=0, - CODEC_AACLD=1, - CODEC_UNSPECIFIED=99 -} CODEC_TYPE; - - -typedef struct -{ - INT bitRate; - INT nChannels; - INT sampleFreq; - INT transFac; - INT standardBitrate; -} CODEC_PARAM; - -typedef enum -{ - SBR_MONO, - SBR_LEFT_RIGHT, - SBR_COUPLING, - SBR_SWITCH_LRC -} SBR_STEREO_MODE; - -/* bitstream syntax flags */ -enum -{ - SBR_SYNTAX_LOW_DELAY = 0x0001, - SBR_SYNTAX_SCALABLE = 0x0002, - SBR_SYNTAX_CRC = 0x0004, - SBR_SYNTAX_DRM_CRC = 0x0008 -}; - -typedef enum -{ - FREQ_RES_LOW = 0, - FREQ_RES_HIGH -} FREQ_RES; - -typedef struct -{ - CODEC_TYPE coreCoder; /*!< LC or ELD */ - UINT bitrateFrom; /*!< inclusive */ - UINT bitrateTo; /*!< exclusive */ - - UINT sampleRate; /*!< */ - UCHAR numChannels; /*!< */ - - UCHAR startFreq; /*!< bs_start_freq */ - UCHAR startFreqSpeech; /*!< bs_start_freq for speech config flag */ - UCHAR stopFreq; /*!< bs_stop_freq */ - UCHAR stopFreqSpeech; /*!< bs_stop_freq for speech config flag */ - - UCHAR numNoiseBands; /*!< */ - UCHAR noiseFloorOffset; /*!< */ - SCHAR noiseMaxLevel; /*!< */ - SBR_STEREO_MODE stereoMode; /*!< */ - UCHAR freqScale; /*!< */ -} sbrTuningTable_t; - -typedef struct sbrConfiguration -{ - /* - core coder dependent configurations - */ - CODEC_PARAM codecSettings; /*!< Core coder settings. To be set from core coder. */ - INT SendHeaderDataTime; /*!< SBR header send update frequency in ms. */ - INT useWaveCoding; /*!< Flag: usage of wavecoding tool. */ - INT crcSbr; /*!< Flag: usage of SBR-CRC. */ - INT dynBwSupported; /*!< Flag: support for dynamic bandwidth in this combination. */ - INT parametricCoding; /*!< Flag: usage of parametric coding tool. */ - INT downSampleFactor; /*!< Sampling rate relation between the SBR and the core encoder. */ - FREQ_RES freq_res_fixfix[2];/*!< Frequency resolution of envelopes in frame class FIXFIX, for non-split case and split case */ - UCHAR fResTransIsLow; /*!< Frequency resolution of envelopes in transient frames: low (0) or variable (1) */ - - /* - core coder dependent tuning parameters - */ - INT tran_thr; /*!< SBR transient detector threshold (* 100). */ - INT noiseFloorOffset; /*!< Noise floor offset. */ - UINT useSpeechConfig; /*!< Flag: adapt tuning parameters according to speech. */ - - - - /* - core coder independent configurations - */ - INT sbrFrameSize; /*!< SBR frame size in samples. Will be calculated from core coder settings. */ - INT sbr_data_extra; /*!< Flag usage of data extra. */ - INT amp_res; /*!< Amplitude resolution. */ - INT ana_max_level; /*!< Noise insertion maximum level. */ - INT tran_fc; /*!< Transient detector start frequency. */ - INT tran_det_mode; /*!< Transient detector mode. */ - INT spread; /*!< Flag: usage of SBR spread. */ - INT stat; /*!< Flag: usage of static framing. */ - INT e; /*!< Number of envelopes when static framing is chosen. */ - SBR_STEREO_MODE stereoMode; /*!< SBR stereo mode. */ - INT deltaTAcrossFrames; /*!< Flag: allow time-delta coding. */ - FIXP_DBL dF_edge_1stEnv; /*!< Extra fraction delta-F coding is allowed to be more expensive. */ - FIXP_DBL dF_edge_incr; /*!< Increment dF_edge_1stEnv this much if dT-coding was used this frame. */ - INT sbr_invf_mode; /*!< Inverse filtering mode. */ - INT sbr_xpos_mode; /*!< Transposer mode. */ - INT sbr_xpos_ctrl; /*!< Transposer control. */ - INT sbr_xpos_level; /*!< Transposer 3rd order level. */ - INT startFreq; /*!< The start frequency table index. */ - INT stopFreq; /*!< The stop frequency table index. */ - INT useSaPan; /*!< Flag: usage of SAPAN stereo. */ - INT dynBwEnabled; /*!< Flag: usage of dynamic bandwidth. */ - INT bParametricStereo; /*!< Flag: usage of parametric stereo coding tool. */ - - /* - header_extra1 configuration - */ - UCHAR freqScale; /*!< Frequency grouping. */ - INT alterScale; /*!< Scale resolution. */ - INT sbr_noise_bands; /*!< Number of noise bands. */ - - - /* - header_extra2 configuration - */ - INT sbr_limiter_bands; /*!< Number of limiter bands. */ - INT sbr_limiter_gains; /*!< Gain of limiter. */ - INT sbr_interpol_freq; /*!< Flag: use interpolation in freq. direction. */ - INT sbr_smoothing_length; /*!< Flag: choose length 4 or 0 (=on, off). */ - UCHAR init_amp_res_FF; - FIXP_DBL threshold_AmpRes_FF_m; - SCHAR threshold_AmpRes_FF_e; -} sbrConfiguration, *sbrConfigurationPtr ; - -typedef struct SBR_CONFIG_DATA -{ - UINT sbrSyntaxFlags; /**< SBR syntax flags derived from AOT. */ - INT nChannels; /**< Number of channels. */ - - INT nSfb[2]; /**< Number of SBR scalefactor bands for LO_RES and HI_RES (?) */ - INT num_Master; /**< Number of elements in v_k_master. */ - INT sampleFreq; /**< SBR sampling frequency. */ - INT frameSize; - INT xOverFreq; /**< The SBR start frequency. */ - INT dynXOverFreq; /**< Used crossover frequency when dynamic bandwidth is enabled. */ - INT noQmfBands; /**< Number of QMF frequency bands. */ - INT noQmfSlots; /**< Number of QMF slots. */ - - UCHAR *freqBandTable[2]; /**< Frequency table for low and hires, only MAX_FREQ_COEFFS/2 +1 coeffs actually needed for lowres. */ - UCHAR *v_k_master; /**< Master BandTable where freqBandTable is derived from. */ - - - SBR_STEREO_MODE stereoMode; - INT noEnvChannels; /**< Number of envelope channels. */ - - INT useWaveCoding; /**< Flag indicates whether to use wave coding at all. */ - INT useParametricCoding; /**< Flag indicates whether to use para coding at all. */ - INT xposCtrlSwitch; /**< Flag indicates whether to switch xpos ctrl on the fly. */ - INT switchTransposers; /**< Flag indicates whether to switch xpos on the fly . */ - UCHAR initAmpResFF; - FIXP_DBL thresholdAmpResFF_m; - SCHAR thresholdAmpResFF_e; -} SBR_CONFIG_DATA, *HANDLE_SBR_CONFIG_DATA; - -typedef struct { - MP4_ELEMENT_ID elType; - INT bitRate; - int instanceTag; - UCHAR fParametricStereo; - UCHAR nChannelsInEl; - UCHAR ChannelIndex[2]; -} SBR_ELEMENT_INFO; - -#ifdef __cplusplus -extern "C" { -#endif - -typedef struct SBR_ENCODER *HANDLE_SBR_ENCODER; - -/** - * \brief Get the max required input buffer size including delay balancing space - * for N audio channels. - * \param noChannels Number of audio channels. - * \return Max required input buffer size in bytes. - */ -INT sbrEncoder_GetInBufferSize(int noChannels); - -INT sbrEncoder_Open( - HANDLE_SBR_ENCODER *phSbrEncoder, - INT nElements, - INT nChannels, - INT supportPS - ); - -/** - * \brief Get closest working bitrate to specified desired - * bitrate for a single SBR element. - * \param bitRate The desired target bit rate - * \param numChannels The amount of audio channels - * \param coreSampleRate The sample rate of the core coder - * \param aot The current Audio Object Type - * \return Closest working bit rate to bitRate value - */ -UINT sbrEncoder_LimitBitRate(UINT bitRate, UINT numChannels, UINT coreSampleRate, AUDIO_OBJECT_TYPE aot); - - -/** - * \brief Check whether downsampled SBR single rate is possible - * with given audio object type. - * \param aot The Audio object type. - * \return 0 when downsampled SBR is not possible, - * 1 when downsampled SBR is possible. - */ -UINT sbrEncoder_IsSingleRatePossible(AUDIO_OBJECT_TYPE aot); - -/** - * \brief Initialize SBR Encoder instance. - * \param phSbrEncoder Pointer to a SBR Encoder instance. - * \param elInfo Structure that describes the element/channel arrangement. - * \param noElements Amount of elements described in elInfo. - * \param inputBuffer Pointer to the encoder audio buffer - * \param bandwidth Returns the core audio encoder bandwidth (output) - * \param bufferOffset Returns the offset for the audio input data in order to do delay balancing. - * \param numChannels Input: Encoder input channels. output: core encoder channels. - * \param sampleRate Input: Encoder samplerate. output core encoder samplerate. - * \param downSampleFactor Input: Relation between SBR and core coder sampling rate; - * \param frameLength Input: Encoder frameLength. output core encoder frameLength. - * \param aot Input: Desired AOT. output AOT to be used after parameter checking. - * \param delay Input: core encoder delay. Output: total delay because of SBR. - * \param transformFactor The core encoder transform factor (blockswitching). - * \param headerPeriod Repetition rate of the SBR header: - * - (-1) means intern configuration. - * - (1-10) corresponds to header repetition rate in frames. - * \return 0 on success, and non-zero if failed. - */ -INT sbrEncoder_Init( - HANDLE_SBR_ENCODER hSbrEncoder, - SBR_ELEMENT_INFO elInfo[(8)], - int noElements, - INT_PCM *inputBuffer, - INT *coreBandwidth, - INT *inputBufferOffset, - INT *numChannels, - INT *sampleRate, - UINT *downSampleFactor, - INT *frameLength, - AUDIO_OBJECT_TYPE aot, - int *delay, - int transformFactor, - const int headerPeriod, - ULONG statesInitFlag - ); - -/** - * \brief Do delay line buffers housekeeping. To be called after each encoded audio frame. - * \param hEnvEnc SBR Encoder handle. - * \param timeBuffer Pointer to the encoder audio buffer. - * \return 0 on success, and non-zero if failed. - */ -INT sbrEncoder_UpdateBuffers(HANDLE_SBR_ENCODER hEnvEnc, - INT_PCM *timeBuffer - ); - -/** - * \brief Close SBR encoder instance. - * \param phEbrEncoder Handle of SBR encoder instance to be closed. - * \return void - */ -void sbrEncoder_Close(HANDLE_SBR_ENCODER *phEbrEncoder); - -/** - * \brief Encode SBR data of one complete audio frame. - * \param hEnvEncoder Handle of SBR encoder instance. - * \param samples Time samples, always interleaved. - * \param timeInStride Channel stride factor of samples buffer. - * \param sbrDataBits Size of SBR payload in bits. - * \param sbrData SBR payload. - * \return 0 on success, and non-zero if failed. - */ -INT sbrEncoder_EncodeFrame(HANDLE_SBR_ENCODER hEnvEncoder, - INT_PCM *samples, - UINT timeInStride, - UINT sbrDataBits[(8)], - UCHAR sbrData[(8)][MAX_PAYLOAD_SIZE] - ); - -/** - * \brief Write SBR headers of one SBR element. - * \param sbrEncoder Handle of the SBR encoder instance. - * \param hBs Handle of bit stream handle to write SBR header to. - * \param element_index Index of the SBR element which header should be written. - * \param fSendHeaders Flag indicating that the SBR encoder should send more headers in the SBR payload or not. - * \return void - */ -void sbrEncoder_GetHeader(HANDLE_SBR_ENCODER sbrEncoder, - HANDLE_FDK_BITSTREAM hBs, - INT element_index, - int fSendHeaders); - -/** - * \brief SBR encoder bitrate estimation. - * \param hSbrEncoder SBR encoder handle. - * \return Estimated bitrate. - */ -INT sbrEncoder_GetEstimateBitrate(HANDLE_SBR_ENCODER hSbrEncoder); - - -/** - * \brief Delay between input data and downsampled output data. - * \param hSbrEncoder SBR encoder handle. - * \return Delay. - */ -INT sbrEncoder_GetInputDataDelay(HANDLE_SBR_ENCODER hSbrEncoder); - -/** - * \brief Get decoder library version info. - * \param info Pointer to an allocated LIB_INFO struct, where library info is written to. - * \return 0 on sucess. - */ -INT sbrEncoder_GetLibInfo(LIB_INFO *info); - -void sbrPrintRAM(void); - -void sbrPrintROM(void); - -#ifdef __cplusplus - } -#endif - -#endif /* ifndef __SBR_MAIN_H */ diff --git a/libSBRenc/src/bit_sbr.cpp b/libSBRenc/src/bit_sbr.cpp deleted file mode 100644 index 9200e01..0000000 --- a/libSBRenc/src/bit_sbr.cpp +++ /dev/null @@ -1,1057 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief SBR bit writing routines -*/ - - -#include "bit_sbr.h" - -#include "code_env.h" -#include "cmondata.h" -#include "sbr.h" - -#include "ps_main.h" - -typedef enum { - SBR_ID_SCE = 1, - SBR_ID_CPE -} SBR_ELEMENT_TYPE; - - -static INT encodeSbrData (HANDLE_SBR_ENV_DATA sbrEnvDataLeft, - HANDLE_SBR_ENV_DATA sbrEnvDataRight, - HANDLE_PARAMETRIC_STEREO hParametricStereo, - HANDLE_COMMON_DATA cmonData, - SBR_ELEMENT_TYPE sbrElem, - INT coupling, - UINT sbrSyntaxFlags); - -static INT encodeSbrHeader (HANDLE_SBR_HEADER_DATA sbrHeaderData, - HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, - HANDLE_COMMON_DATA cmonData); - - -static INT encodeSbrHeaderData (HANDLE_SBR_HEADER_DATA sbrHeaderData, - HANDLE_FDK_BITSTREAM hBitStream); - -static INT encodeSbrSingleChannelElement (HANDLE_SBR_ENV_DATA sbrEnvData, - HANDLE_FDK_BITSTREAM hBitStream - ,HANDLE_PARAMETRIC_STEREO hParametricStereo - ,UINT sbrSyntaxFlags - ); - - - -static INT encodeSbrChannelPairElement (HANDLE_SBR_ENV_DATA sbrEnvDataLeft, - HANDLE_SBR_ENV_DATA sbrEnvDataRight, - HANDLE_PARAMETRIC_STEREO hParametricStereo, - HANDLE_FDK_BITSTREAM hBitStream, - INT coupling); - - -static INT encodeSbrGrid (HANDLE_SBR_ENV_DATA sbrEnvData, - HANDLE_FDK_BITSTREAM hBitStream); - -static int encodeLowDelaySbrGrid ( HANDLE_SBR_ENV_DATA sbrEnvData, - HANDLE_FDK_BITSTREAM hBitStream, - int transmitFreqs); - -static INT encodeSbrDtdf (HANDLE_SBR_ENV_DATA sbrEnvData, - HANDLE_FDK_BITSTREAM hBitStream); - -static INT writeNoiseLevelData (HANDLE_SBR_ENV_DATA sbrEnvData, - HANDLE_FDK_BITSTREAM hBitStream, - INT coupling); - -static INT writeEnvelopeData (HANDLE_SBR_ENV_DATA sbrEnvData, - HANDLE_FDK_BITSTREAM hBitStream, - INT coupling); - -static INT writeSyntheticCodingData (HANDLE_SBR_ENV_DATA sbrEnvData, - HANDLE_FDK_BITSTREAM hBitStream); - - -static INT encodeExtendedData (HANDLE_PARAMETRIC_STEREO hParametricStereo, - HANDLE_FDK_BITSTREAM hBitStream); - - - -static INT getSbrExtendedDataSize (HANDLE_PARAMETRIC_STEREO hParametricStereo); - -/***************************************************************************** - - functionname: FDKsbrEnc_WriteEnvSingleChannelElement - description: writes pure SBR single channel data element - returns: number of bits written - input: - output: - -*****************************************************************************/ -INT -FDKsbrEnc_WriteEnvSingleChannelElement( - HANDLE_SBR_HEADER_DATA sbrHeaderData, - HANDLE_PARAMETRIC_STEREO hParametricStereo, - HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, - HANDLE_SBR_ENV_DATA sbrEnvData, - HANDLE_COMMON_DATA cmonData, - UINT sbrSyntaxFlags - ) - -{ - INT payloadBits = 0; - - cmonData->sbrHdrBits = 0; - cmonData->sbrDataBits = 0; - - /* write pure sbr data */ - if (sbrEnvData != NULL) { - - /* write header */ - payloadBits += encodeSbrHeader (sbrHeaderData, - sbrBitstreamData, - cmonData); - - - /* write data */ - payloadBits += encodeSbrData (sbrEnvData, - NULL, - hParametricStereo, - cmonData, - SBR_ID_SCE, - 0, - sbrSyntaxFlags); - - } - return payloadBits; -} - -/***************************************************************************** - - functionname: FDKsbrEnc_WriteEnvChannelPairElement - description: writes pure SBR channel pair data element - returns: number of bits written - input: - output: - -*****************************************************************************/ -INT -FDKsbrEnc_WriteEnvChannelPairElement (HANDLE_SBR_HEADER_DATA sbrHeaderData, - HANDLE_PARAMETRIC_STEREO hParametricStereo, - HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, - HANDLE_SBR_ENV_DATA sbrEnvDataLeft, - HANDLE_SBR_ENV_DATA sbrEnvDataRight, - HANDLE_COMMON_DATA cmonData, - UINT sbrSyntaxFlags) - -{ - INT payloadBits = 0; - cmonData->sbrHdrBits = 0; - cmonData->sbrDataBits = 0; - - /* write pure sbr data */ - if ((sbrEnvDataLeft != NULL) && (sbrEnvDataRight != NULL)) { - - /* write header */ - payloadBits += encodeSbrHeader (sbrHeaderData, - sbrBitstreamData, - cmonData); - - /* write data */ - payloadBits += encodeSbrData (sbrEnvDataLeft, - sbrEnvDataRight, - hParametricStereo, - cmonData, - SBR_ID_CPE, - sbrHeaderData->coupling, - sbrSyntaxFlags); - - } - return payloadBits; -} - -INT -FDKsbrEnc_CountSbrChannelPairElement (HANDLE_SBR_HEADER_DATA sbrHeaderData, - HANDLE_PARAMETRIC_STEREO hParametricStereo, - HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, - HANDLE_SBR_ENV_DATA sbrEnvDataLeft, - HANDLE_SBR_ENV_DATA sbrEnvDataRight, - HANDLE_COMMON_DATA cmonData, - UINT sbrSyntaxFlags) -{ - INT payloadBits; - INT bitPos = FDKgetValidBits(&cmonData->sbrBitbuf); - - payloadBits = FDKsbrEnc_WriteEnvChannelPairElement(sbrHeaderData, - hParametricStereo, - sbrBitstreamData, - sbrEnvDataLeft, - sbrEnvDataRight, - cmonData, - sbrSyntaxFlags); - - FDKpushBack(&cmonData->sbrBitbuf, (FDKgetValidBits(&cmonData->sbrBitbuf) - bitPos) ); - - return payloadBits; -} - - -void sbrEncoder_GetHeader(SBR_ENCODER *sbrEncoder, - HANDLE_FDK_BITSTREAM hBs, - INT element_index, - int fSendHeaders) -{ - encodeSbrHeaderData (&sbrEncoder->sbrElement[element_index]->sbrHeaderData, hBs); - - if (fSendHeaders == 0) { - /* Prevent header being embedded into the SBR payload. */ - sbrEncoder->sbrElement[element_index]->sbrBitstreamData.NrSendHeaderData = -1; - sbrEncoder->sbrElement[element_index]->sbrBitstreamData.HeaderActive = 0; - sbrEncoder->sbrElement[element_index]->sbrBitstreamData.CountSendHeaderData = -1; - } -} - - -/***************************************************************************** - - functionname: encodeSbrHeader - description: encodes SBR Header information - returns: number of bits written - input: - output: - -*****************************************************************************/ -static INT -encodeSbrHeader (HANDLE_SBR_HEADER_DATA sbrHeaderData, - HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, - HANDLE_COMMON_DATA cmonData) -{ - INT payloadBits = 0; - - if (sbrBitstreamData->HeaderActive) { - payloadBits += FDKwriteBits (&cmonData->sbrBitbuf, 1, 1); - payloadBits += encodeSbrHeaderData (sbrHeaderData, - &cmonData->sbrBitbuf); - } - else { - payloadBits += FDKwriteBits (&cmonData->sbrBitbuf, 0, 1); - } - - cmonData->sbrHdrBits = payloadBits; - - return payloadBits; -} - - - -/***************************************************************************** - - functionname: encodeSbrHeaderData - description: writes sbr_header() - bs_protocol_version through bs_header_extra_2 - returns: number of bits written - input: - output: - -*****************************************************************************/ -static INT -encodeSbrHeaderData (HANDLE_SBR_HEADER_DATA sbrHeaderData, - HANDLE_FDK_BITSTREAM hBitStream) - -{ - INT payloadBits = 0; - if (sbrHeaderData != NULL) { - payloadBits += FDKwriteBits (hBitStream, sbrHeaderData->sbr_amp_res, - SI_SBR_AMP_RES_BITS); - payloadBits += FDKwriteBits (hBitStream, sbrHeaderData->sbr_start_frequency, - SI_SBR_START_FREQ_BITS); - payloadBits += FDKwriteBits (hBitStream, sbrHeaderData->sbr_stop_frequency, - SI_SBR_STOP_FREQ_BITS); - payloadBits += FDKwriteBits (hBitStream, sbrHeaderData->sbr_xover_band, - SI_SBR_XOVER_BAND_BITS); - - payloadBits += FDKwriteBits (hBitStream, 0, - SI_SBR_RESERVED_BITS); - - payloadBits += FDKwriteBits (hBitStream, sbrHeaderData->header_extra_1, - SI_SBR_HEADER_EXTRA_1_BITS); - payloadBits += FDKwriteBits (hBitStream, sbrHeaderData->header_extra_2, - SI_SBR_HEADER_EXTRA_2_BITS); - - - if (sbrHeaderData->header_extra_1) { - payloadBits += FDKwriteBits (hBitStream, sbrHeaderData->freqScale, - SI_SBR_FREQ_SCALE_BITS); - payloadBits += FDKwriteBits (hBitStream, sbrHeaderData->alterScale, - SI_SBR_ALTER_SCALE_BITS); - payloadBits += FDKwriteBits (hBitStream, sbrHeaderData->sbr_noise_bands, - SI_SBR_NOISE_BANDS_BITS); - } /* sbrHeaderData->header_extra_1 */ - - if (sbrHeaderData->header_extra_2) { - payloadBits += FDKwriteBits (hBitStream, sbrHeaderData->sbr_limiter_bands, - SI_SBR_LIMITER_BANDS_BITS); - payloadBits += FDKwriteBits (hBitStream, sbrHeaderData->sbr_limiter_gains, - SI_SBR_LIMITER_GAINS_BITS); - payloadBits += FDKwriteBits (hBitStream, sbrHeaderData->sbr_interpol_freq, - SI_SBR_INTERPOL_FREQ_BITS); - payloadBits += FDKwriteBits (hBitStream, sbrHeaderData->sbr_smoothing_length, - SI_SBR_SMOOTHING_LENGTH_BITS); - - } /* sbrHeaderData->header_extra_2 */ - } /* sbrHeaderData != NULL */ - - return payloadBits; -} - - -/***************************************************************************** - - functionname: encodeSbrData - description: encodes sbr Data information - returns: number of bits written - input: - output: - -*****************************************************************************/ -static INT -encodeSbrData (HANDLE_SBR_ENV_DATA sbrEnvDataLeft, - HANDLE_SBR_ENV_DATA sbrEnvDataRight, - HANDLE_PARAMETRIC_STEREO hParametricStereo, - HANDLE_COMMON_DATA cmonData, - SBR_ELEMENT_TYPE sbrElem, - INT coupling, - UINT sbrSyntaxFlags) -{ - INT payloadBits = 0; - - switch (sbrElem) { - case SBR_ID_SCE: - payloadBits += encodeSbrSingleChannelElement (sbrEnvDataLeft, &cmonData->sbrBitbuf, hParametricStereo, sbrSyntaxFlags); - break; - case SBR_ID_CPE: - payloadBits += encodeSbrChannelPairElement (sbrEnvDataLeft, sbrEnvDataRight, hParametricStereo, &cmonData->sbrBitbuf, coupling); - break; - default: - /* we never should apply SBR to any other element type */ - FDK_ASSERT (0); - } - - cmonData->sbrDataBits = payloadBits; - - return payloadBits; -} - -#define MODE_FREQ_TANS 1 -#define MODE_NO_FREQ_TRAN 0 -#define LD_TRANSMISSION MODE_FREQ_TANS -static int encodeFreqs (int mode) { - return ((mode & MODE_FREQ_TANS) ? 1 : 0); -} - - -/***************************************************************************** - - functionname: encodeSbrSingleChannelElement - description: encodes sbr SCE information - returns: number of bits written - input: - output: - -*****************************************************************************/ -static INT -encodeSbrSingleChannelElement (HANDLE_SBR_ENV_DATA sbrEnvData, - HANDLE_FDK_BITSTREAM hBitStream - ,HANDLE_PARAMETRIC_STEREO hParametricStereo - ,UINT sbrSyntaxFlags - ) -{ - INT i, payloadBits = 0; - - payloadBits += FDKwriteBits (hBitStream, 0, SI_SBR_DATA_EXTRA_BITS); /* no reserved bits */ - - if (sbrEnvData->ldGrid) { - if ( sbrEnvData->hSbrBSGrid->frameClass != FIXFIXonly ) { - /* encode normal SbrGrid */ - payloadBits += encodeSbrGrid (sbrEnvData, hBitStream); - } else { - /* use FIXFIXonly frame Grid */ - payloadBits += encodeLowDelaySbrGrid ( sbrEnvData, hBitStream, encodeFreqs(LD_TRANSMISSION)); - } - } - else - { - if (sbrSyntaxFlags & SBR_SYNTAX_SCALABLE) { - payloadBits += FDKwriteBits (hBitStream, 1, SI_SBR_COUPLING_BITS); - } - payloadBits += encodeSbrGrid (sbrEnvData, hBitStream); - } - - payloadBits += encodeSbrDtdf (sbrEnvData, hBitStream); - - for (i = 0; i < sbrEnvData->noOfnoisebands; i++) { - payloadBits += FDKwriteBits (hBitStream, sbrEnvData->sbr_invf_mode_vec[i], SI_SBR_INVF_MODE_BITS); - } - - payloadBits += writeEnvelopeData (sbrEnvData, hBitStream, 0); - payloadBits += writeNoiseLevelData (sbrEnvData, hBitStream, 0); - - payloadBits += writeSyntheticCodingData (sbrEnvData,hBitStream); - - payloadBits += encodeExtendedData(hParametricStereo, hBitStream); - - return payloadBits; -} - - -/***************************************************************************** - - functionname: encodeSbrChannelPairElement - description: encodes sbr CPE information - returns: - input: - output: - -*****************************************************************************/ -static INT -encodeSbrChannelPairElement (HANDLE_SBR_ENV_DATA sbrEnvDataLeft, - HANDLE_SBR_ENV_DATA sbrEnvDataRight, - HANDLE_PARAMETRIC_STEREO hParametricStereo, - HANDLE_FDK_BITSTREAM hBitStream, - INT coupling) -{ - INT payloadBits = 0; - INT i = 0; - - payloadBits += FDKwriteBits (hBitStream, 0, SI_SBR_DATA_EXTRA_BITS); /* no reserved bits */ - - payloadBits += FDKwriteBits (hBitStream, coupling, SI_SBR_COUPLING_BITS); - - if (coupling) { - if (sbrEnvDataLeft->ldGrid) { - if ( sbrEnvDataLeft->hSbrBSGrid->frameClass != FIXFIXonly ) { - /* normal SbrGrid */ - payloadBits += encodeSbrGrid (sbrEnvDataLeft, hBitStream); - - } else { - /* FIXFIXonly frame Grid */ - payloadBits += encodeLowDelaySbrGrid ( sbrEnvDataLeft, hBitStream, encodeFreqs(LD_TRANSMISSION)); - } - } else - payloadBits += encodeSbrGrid (sbrEnvDataLeft, hBitStream); - - payloadBits += encodeSbrDtdf (sbrEnvDataLeft, hBitStream); - payloadBits += encodeSbrDtdf (sbrEnvDataRight, hBitStream); - - for (i = 0; i < sbrEnvDataLeft->noOfnoisebands; i++) { - payloadBits += FDKwriteBits (hBitStream, sbrEnvDataLeft->sbr_invf_mode_vec[i], SI_SBR_INVF_MODE_BITS); - } - - payloadBits += writeEnvelopeData (sbrEnvDataLeft, hBitStream,1); - payloadBits += writeNoiseLevelData (sbrEnvDataLeft, hBitStream,1); - payloadBits += writeEnvelopeData (sbrEnvDataRight, hBitStream,1); - payloadBits += writeNoiseLevelData (sbrEnvDataRight, hBitStream,1); - - payloadBits += writeSyntheticCodingData (sbrEnvDataLeft,hBitStream); - payloadBits += writeSyntheticCodingData (sbrEnvDataRight,hBitStream); - - } else { /* no coupling */ - FDK_ASSERT(sbrEnvDataLeft->ldGrid == sbrEnvDataRight->ldGrid); - - if (sbrEnvDataLeft->ldGrid || sbrEnvDataRight->ldGrid) { - /* sbrEnvDataLeft (left channel) */ - if ( sbrEnvDataLeft->hSbrBSGrid->frameClass != FIXFIXonly) { - /* no FIXFIXonly Frame so we dont need encodeLowDelaySbrGrid */ - /* normal SbrGrid */ - payloadBits += encodeSbrGrid (sbrEnvDataLeft, hBitStream); - - } else { - /* FIXFIXonly frame Grid */ - payloadBits += encodeLowDelaySbrGrid ( sbrEnvDataLeft, hBitStream, encodeFreqs(LD_TRANSMISSION)); - } - - /* sbrEnvDataRight (right channel) */ - if ( sbrEnvDataRight->hSbrBSGrid->frameClass != FIXFIXonly) { - /* no FIXFIXonly Frame so we dont need encodeLowDelaySbrGrid */ - /* normal SbrGrid */ - payloadBits += encodeSbrGrid (sbrEnvDataRight, hBitStream); - - } else { - /* FIXFIXonly frame Grid */ - payloadBits += encodeLowDelaySbrGrid ( sbrEnvDataRight, hBitStream, encodeFreqs(LD_TRANSMISSION)); - } - } else - { - payloadBits += encodeSbrGrid (sbrEnvDataLeft, hBitStream); - payloadBits += encodeSbrGrid (sbrEnvDataRight, hBitStream); - } - payloadBits += encodeSbrDtdf (sbrEnvDataLeft, hBitStream); - payloadBits += encodeSbrDtdf (sbrEnvDataRight, hBitStream); - - for (i = 0; i < sbrEnvDataLeft->noOfnoisebands; i++) { - payloadBits += FDKwriteBits (hBitStream, sbrEnvDataLeft->sbr_invf_mode_vec[i], - SI_SBR_INVF_MODE_BITS); - } - for (i = 0; i < sbrEnvDataRight->noOfnoisebands; i++) { - payloadBits += FDKwriteBits (hBitStream, sbrEnvDataRight->sbr_invf_mode_vec[i], - SI_SBR_INVF_MODE_BITS); - } - - payloadBits += writeEnvelopeData (sbrEnvDataLeft, hBitStream,0); - payloadBits += writeEnvelopeData (sbrEnvDataRight, hBitStream,0); - payloadBits += writeNoiseLevelData (sbrEnvDataLeft, hBitStream,0); - payloadBits += writeNoiseLevelData (sbrEnvDataRight, hBitStream,0); - - payloadBits += writeSyntheticCodingData (sbrEnvDataLeft,hBitStream); - payloadBits += writeSyntheticCodingData (sbrEnvDataRight,hBitStream); - - } /* coupling */ - - payloadBits += encodeExtendedData(hParametricStereo, hBitStream); - - return payloadBits; -} - -static INT ceil_ln2(INT x) -{ - INT tmp=-1; - while((1<<++tmp) < x); - return(tmp); -} - - -/***************************************************************************** - - functionname: encodeSbrGrid - description: if hBitStream != NULL writes bits that describes the - time/frequency grouping of a frame; else counts them only - returns: number of bits written or counted - input: - output: - -*****************************************************************************/ -static INT -encodeSbrGrid (HANDLE_SBR_ENV_DATA sbrEnvData, HANDLE_FDK_BITSTREAM hBitStream) -{ - INT payloadBits = 0; - INT i, temp; - INT bufferFrameStart = sbrEnvData->hSbrBSGrid->bufferFrameStart; - INT numberTimeSlots = sbrEnvData->hSbrBSGrid->numberTimeSlots; - - if (sbrEnvData->ldGrid) - payloadBits += FDKwriteBits (hBitStream, - sbrEnvData->hSbrBSGrid->frameClass, - SBR_CLA_BITS_LD); - else - payloadBits += FDKwriteBits (hBitStream, - sbrEnvData->hSbrBSGrid->frameClass, - SBR_CLA_BITS); - - switch (sbrEnvData->hSbrBSGrid->frameClass) { - case FIXFIXonly: - FDK_ASSERT(0 /* Fatal error in encodeSbrGrid! */); - break; - case FIXFIX: - temp = ceil_ln2(sbrEnvData->hSbrBSGrid->bs_num_env); - payloadBits += FDKwriteBits (hBitStream, temp, SBR_ENV_BITS); - if ((sbrEnvData->ldGrid) && (sbrEnvData->hSbrBSGrid->bs_num_env==1)) - payloadBits += FDKwriteBits(hBitStream, sbrEnvData->currentAmpResFF, SI_SBR_AMP_RES_BITS); - payloadBits += FDKwriteBits (hBitStream, sbrEnvData->hSbrBSGrid->v_f[0], SBR_RES_BITS); - - break; - - case FIXVAR: - case VARFIX: - if (sbrEnvData->hSbrBSGrid->frameClass == FIXVAR) - temp = sbrEnvData->hSbrBSGrid->bs_abs_bord - (bufferFrameStart + numberTimeSlots); - else - temp = sbrEnvData->hSbrBSGrid->bs_abs_bord - bufferFrameStart; - - payloadBits += FDKwriteBits (hBitStream, temp, SBR_ABS_BITS); - payloadBits += FDKwriteBits (hBitStream, sbrEnvData->hSbrBSGrid->n, SBR_NUM_BITS); - - for (i = 0; i < sbrEnvData->hSbrBSGrid->n; i++) { - temp = (sbrEnvData->hSbrBSGrid->bs_rel_bord[i] - 2) >> 1; - payloadBits += FDKwriteBits (hBitStream, temp, SBR_REL_BITS); - } - - temp = ceil_ln2(sbrEnvData->hSbrBSGrid->n + 2); - payloadBits += FDKwriteBits (hBitStream, sbrEnvData->hSbrBSGrid->p, temp); - - for (i = 0; i < sbrEnvData->hSbrBSGrid->n + 1; i++) { - payloadBits += FDKwriteBits (hBitStream, sbrEnvData->hSbrBSGrid->v_f[i], - SBR_RES_BITS); - } - break; - - case VARVAR: - temp = sbrEnvData->hSbrBSGrid->bs_abs_bord_0 - bufferFrameStart; - payloadBits += FDKwriteBits (hBitStream, temp, SBR_ABS_BITS); - temp = sbrEnvData->hSbrBSGrid->bs_abs_bord_1 - (bufferFrameStart + numberTimeSlots); - payloadBits += FDKwriteBits (hBitStream, temp, SBR_ABS_BITS); - - payloadBits += FDKwriteBits (hBitStream, sbrEnvData->hSbrBSGrid->bs_num_rel_0, SBR_NUM_BITS); - payloadBits += FDKwriteBits (hBitStream, sbrEnvData->hSbrBSGrid->bs_num_rel_1, SBR_NUM_BITS); - - for (i = 0; i < sbrEnvData->hSbrBSGrid->bs_num_rel_0; i++) { - temp = (sbrEnvData->hSbrBSGrid->bs_rel_bord_0[i] - 2) >> 1; - payloadBits += FDKwriteBits (hBitStream, temp, SBR_REL_BITS); - } - - for (i = 0; i < sbrEnvData->hSbrBSGrid->bs_num_rel_1; i++) { - temp = (sbrEnvData->hSbrBSGrid->bs_rel_bord_1[i] - 2) >> 1; - payloadBits += FDKwriteBits (hBitStream, temp, SBR_REL_BITS); - } - - temp = ceil_ln2(sbrEnvData->hSbrBSGrid->bs_num_rel_0 + - sbrEnvData->hSbrBSGrid->bs_num_rel_1 + 2); - payloadBits += FDKwriteBits (hBitStream, sbrEnvData->hSbrBSGrid->p, temp); - - temp = sbrEnvData->hSbrBSGrid->bs_num_rel_0 + - sbrEnvData->hSbrBSGrid->bs_num_rel_1 + 1; - - for (i = 0; i < temp; i++) { - payloadBits += FDKwriteBits (hBitStream, sbrEnvData->hSbrBSGrid->v_fLR[i], - SBR_RES_BITS); - } - break; - } - - return payloadBits; -} - -#define SBR_CLA_BITS_LD 1 -/***************************************************************************** - - functionname: encodeLowDelaySbrGrid - description: if hBitStream != NULL writes bits that describes the - time/frequency grouping of a frame; - else counts them only - (this function only write the FIXFIXonly Bitstream data) - returns: number of bits written or counted - input: - output: - -*****************************************************************************/ -static int -encodeLowDelaySbrGrid ( HANDLE_SBR_ENV_DATA sbrEnvData, - HANDLE_FDK_BITSTREAM hBitStream, - int transmitFreqs - ) -{ - int payloadBits = 0; - int i; - - /* write FIXFIXonly Grid */ - /* write frameClass [1 bit] for FIXFIXonly Grid */ - payloadBits += FDKwriteBits(hBitStream, 1, SBR_CLA_BITS_LD); - - /* absolute Borders are fix: 0,X,X,X,nTimeSlots; so we dont have to transmit them */ - /* only transmit the transient position! */ - /* with this info (b1) we can reconstruct the Frame on Decoder side : */ - /* border[0] = 0; border[1] = b1; border[2]=b1+2; border[3] = nrTimeSlots */ - - /* use 3 or 4bits for transient border (border) */ - if (sbrEnvData->hSbrBSGrid->numberTimeSlots == 8) - payloadBits += FDKwriteBits ( hBitStream, sbrEnvData->hSbrBSGrid->bs_abs_bord, 3); - else - payloadBits += FDKwriteBits ( hBitStream, sbrEnvData->hSbrBSGrid->bs_abs_bord, 4); - - if (transmitFreqs) { - /* write FreqRes grid */ - for (i = 0; i < sbrEnvData->hSbrBSGrid->bs_num_env; i++) { - payloadBits += FDKwriteBits (hBitStream, sbrEnvData->hSbrBSGrid->v_f[i], SBR_RES_BITS); - } - } - - return payloadBits; -} - -/***************************************************************************** - - functionname: encodeSbrDtdf - description: writes bits that describes the direction of the envelopes of a frame - returns: number of bits written - input: - output: - -*****************************************************************************/ -static INT -encodeSbrDtdf (HANDLE_SBR_ENV_DATA sbrEnvData, HANDLE_FDK_BITSTREAM hBitStream) -{ - INT i, payloadBits = 0, noOfNoiseEnvelopes; - - noOfNoiseEnvelopes = sbrEnvData->noOfEnvelopes > 1 ? 2 : 1; - - for (i = 0; i < sbrEnvData->noOfEnvelopes; ++i) { - payloadBits += FDKwriteBits (hBitStream, sbrEnvData->domain_vec[i], SBR_DIR_BITS); - } - for (i = 0; i < noOfNoiseEnvelopes; ++i) { - payloadBits += FDKwriteBits (hBitStream, sbrEnvData->domain_vec_noise[i], SBR_DIR_BITS); - } - - return payloadBits; -} - - -/***************************************************************************** - - functionname: writeNoiseLevelData - description: writes bits corresponding to the noise-floor-level - returns: number of bits written - input: - output: - -*****************************************************************************/ -static INT -writeNoiseLevelData (HANDLE_SBR_ENV_DATA sbrEnvData, HANDLE_FDK_BITSTREAM hBitStream, INT coupling) -{ - INT j, i, payloadBits = 0; - INT nNoiseEnvelopes = sbrEnvData->noOfEnvelopes > 1 ? 2 : 1; - - for (i = 0; i < nNoiseEnvelopes; i++) { - switch (sbrEnvData->domain_vec_noise[i]) { - case FREQ: - if (coupling && sbrEnvData->balance) { - payloadBits += FDKwriteBits (hBitStream, - sbrEnvData->sbr_noise_levels[i * sbrEnvData->noOfnoisebands], - sbrEnvData->si_sbr_start_noise_bits_balance); - } else { - payloadBits += FDKwriteBits (hBitStream, - sbrEnvData->sbr_noise_levels[i * sbrEnvData->noOfnoisebands], - sbrEnvData->si_sbr_start_noise_bits); - } - - for (j = 1 + i * sbrEnvData->noOfnoisebands; j < (sbrEnvData->noOfnoisebands * (1 + i)); j++) { - if (coupling) { - if (sbrEnvData->balance) { - /* coupling && balance */ - payloadBits += FDKwriteBits (hBitStream, - sbrEnvData->hufftableNoiseBalanceFreqC[sbrEnvData->sbr_noise_levels[j] + - CODE_BOOK_SCF_LAV_BALANCE11], - sbrEnvData->hufftableNoiseBalanceFreqL[sbrEnvData->sbr_noise_levels[j] + - CODE_BOOK_SCF_LAV_BALANCE11]); - } else { - /* coupling && !balance */ - payloadBits += FDKwriteBits (hBitStream, - sbrEnvData->hufftableNoiseLevelFreqC[sbrEnvData->sbr_noise_levels[j] + - CODE_BOOK_SCF_LAV11], - sbrEnvData->hufftableNoiseLevelFreqL[sbrEnvData->sbr_noise_levels[j] + - CODE_BOOK_SCF_LAV11]); - } - } else { - /* !coupling */ - payloadBits += FDKwriteBits (hBitStream, - sbrEnvData->hufftableNoiseFreqC[sbrEnvData->sbr_noise_levels[j] + - CODE_BOOK_SCF_LAV11], - sbrEnvData->hufftableNoiseFreqL[sbrEnvData->sbr_noise_levels[j] + - CODE_BOOK_SCF_LAV11]); - } - } - break; - - case TIME: - for (j = i * sbrEnvData->noOfnoisebands; j < (sbrEnvData->noOfnoisebands * (1 + i)); j++) { - if (coupling) { - if (sbrEnvData->balance) { - /* coupling && balance */ - payloadBits += FDKwriteBits (hBitStream, - sbrEnvData->hufftableNoiseBalanceTimeC[sbrEnvData->sbr_noise_levels[j] + - CODE_BOOK_SCF_LAV_BALANCE11], - sbrEnvData->hufftableNoiseBalanceTimeL[sbrEnvData->sbr_noise_levels[j] + - CODE_BOOK_SCF_LAV_BALANCE11]); - } else { - /* coupling && !balance */ - payloadBits += FDKwriteBits (hBitStream, - sbrEnvData->hufftableNoiseLevelTimeC[sbrEnvData->sbr_noise_levels[j] + - CODE_BOOK_SCF_LAV11], - sbrEnvData->hufftableNoiseLevelTimeL[sbrEnvData->sbr_noise_levels[j] + - CODE_BOOK_SCF_LAV11]); - } - } else { - /* !coupling */ - payloadBits += FDKwriteBits (hBitStream, - sbrEnvData->hufftableNoiseLevelTimeC[sbrEnvData->sbr_noise_levels[j] + - CODE_BOOK_SCF_LAV11], - sbrEnvData->hufftableNoiseLevelTimeL[sbrEnvData->sbr_noise_levels[j] + - CODE_BOOK_SCF_LAV11]); - } - } - break; - } - } - return payloadBits; -} - - -/***************************************************************************** - - functionname: writeEnvelopeData - description: writes bits corresponding to the envelope - returns: number of bits written - input: - output: - -*****************************************************************************/ -static INT -writeEnvelopeData (HANDLE_SBR_ENV_DATA sbrEnvData, HANDLE_FDK_BITSTREAM hBitStream, INT coupling) -{ - INT payloadBits = 0, j, i, delta; - - for (j = 0; j < sbrEnvData->noOfEnvelopes; j++) { /* loop over all envelopes */ - if (sbrEnvData->domain_vec[j] == FREQ) { - if (coupling && sbrEnvData->balance) { - payloadBits += FDKwriteBits (hBitStream, sbrEnvData->ienvelope[j][0], sbrEnvData->si_sbr_start_env_bits_balance); - } else { - payloadBits += FDKwriteBits (hBitStream, sbrEnvData->ienvelope[j][0], sbrEnvData->si_sbr_start_env_bits); - } - } - - for (i = 1 - sbrEnvData->domain_vec[j]; i < sbrEnvData->noScfBands[j]; i++) { - delta = sbrEnvData->ienvelope[j][i]; - if (coupling && sbrEnvData->balance) { - FDK_ASSERT (fixp_abs (delta) <= sbrEnvData->codeBookScfLavBalance); - } else { - FDK_ASSERT (fixp_abs (delta) <= sbrEnvData->codeBookScfLav); - } - if (coupling) { - if (sbrEnvData->balance) { - if (sbrEnvData->domain_vec[j]) { - /* coupling && balance && TIME */ - payloadBits += FDKwriteBits (hBitStream, - sbrEnvData->hufftableBalanceTimeC[delta + sbrEnvData->codeBookScfLavBalance], - sbrEnvData->hufftableBalanceTimeL[delta + sbrEnvData->codeBookScfLavBalance]); - } else { - /* coupling && balance && FREQ */ - payloadBits += FDKwriteBits (hBitStream, - sbrEnvData->hufftableBalanceFreqC[delta + sbrEnvData->codeBookScfLavBalance], - sbrEnvData->hufftableBalanceFreqL[delta + sbrEnvData->codeBookScfLavBalance]); - } - } else { - if (sbrEnvData->domain_vec[j]) { - /* coupling && !balance && TIME */ - payloadBits += FDKwriteBits (hBitStream, - sbrEnvData->hufftableLevelTimeC[delta + sbrEnvData->codeBookScfLav], - sbrEnvData->hufftableLevelTimeL[delta + sbrEnvData->codeBookScfLav]); - } else { - /* coupling && !balance && FREQ */ - payloadBits += FDKwriteBits (hBitStream, - sbrEnvData->hufftableLevelFreqC[delta + sbrEnvData->codeBookScfLav], - sbrEnvData->hufftableLevelFreqL[delta + sbrEnvData->codeBookScfLav]); - } - } - } else { - if (sbrEnvData->domain_vec[j]) { - /* !coupling && TIME */ - payloadBits += FDKwriteBits (hBitStream, - sbrEnvData->hufftableTimeC[delta + sbrEnvData->codeBookScfLav], - sbrEnvData->hufftableTimeL[delta + sbrEnvData->codeBookScfLav]); - } else { - /* !coupling && FREQ */ - payloadBits += FDKwriteBits (hBitStream, - sbrEnvData->hufftableFreqC[delta + sbrEnvData->codeBookScfLav], - sbrEnvData->hufftableFreqL[delta + sbrEnvData->codeBookScfLav]); - } - } - } - } - return payloadBits; -} - - -/***************************************************************************** - - functionname: encodeExtendedData - description: writes bits corresponding to the extended data - returns: number of bits written - input: - output: - -*****************************************************************************/ -static INT encodeExtendedData (HANDLE_PARAMETRIC_STEREO hParametricStereo, - HANDLE_FDK_BITSTREAM hBitStream) -{ - INT extDataSize; - INT payloadBits = 0; - - extDataSize = getSbrExtendedDataSize(hParametricStereo); - - - if (extDataSize != 0) { - INT maxExtSize = (1<addHarmonicFlag, 1); - - if (sbrEnvData->addHarmonicFlag) { - for (i = 0; i < sbrEnvData->noHarmonics; i++) { - payloadBits += FDKwriteBits (hBitStream, sbrEnvData->addHarmonic[i], 1); - } - } - - return payloadBits; -} - -/***************************************************************************** - - functionname: getSbrExtendedDataSize - description: counts the number of bits needed for encoding the - extended data (including extension id) - - returns: number of bits needed for the extended data - input: - output: - -*****************************************************************************/ -static INT -getSbrExtendedDataSize (HANDLE_PARAMETRIC_STEREO hParametricStereo) -{ - INT extDataBits = 0; - - /* add your new extended data counting methods here */ - - /* - no extended data - */ - - if(hParametricStereo){ - /* PS extended data */ - extDataBits += SI_SBR_EXTENSION_ID_BITS; - extDataBits += FDKsbrEnc_PSEnc_WritePSData(hParametricStereo, NULL); - } - - return (extDataBits+7) >> 3; -} - - - - - diff --git a/libSBRenc/src/bit_sbr.h b/libSBRenc/src/bit_sbr.h deleted file mode 100644 index de4ac89..0000000 --- a/libSBRenc/src/bit_sbr.h +++ /dev/null @@ -1,258 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief SBR bit writing -*/ -#ifndef __BIT_SBR_H -#define __BIT_SBR_H - -#include "sbr_def.h" -#include "cmondata.h" -#include "fram_gen.h" - -struct SBR_ENV_DATA; - -struct SBR_BITSTREAM_DATA -{ - INT TotalBits; - INT PayloadBits; - INT FillBits; - INT HeaderActive; - INT NrSendHeaderData; /**< input from commandline */ - INT CountSendHeaderData; /**< modulo count. If < 0 then no counting is done (no SBR headers) */ -}; - -typedef struct SBR_BITSTREAM_DATA *HANDLE_SBR_BITSTREAM_DATA; - -struct SBR_HEADER_DATA -{ - AMP_RES sbr_amp_res; - INT sbr_start_frequency; - INT sbr_stop_frequency; - INT sbr_xover_band; - INT sbr_noise_bands; - INT sbr_data_extra; - INT header_extra_1; - INT header_extra_2; - INT sbr_lc_stereo_mode; - INT sbr_limiter_bands; - INT sbr_limiter_gains; - INT sbr_interpol_freq; - INT sbr_smoothing_length; - INT alterScale; - INT freqScale; - - /* - element of channelpairelement - */ - INT coupling; - INT prev_coupling; - - /* - element of singlechannelelement - */ - -}; -typedef struct SBR_HEADER_DATA *HANDLE_SBR_HEADER_DATA; - -struct SBR_ENV_DATA -{ - - INT sbr_xpos_ctrl; - FREQ_RES freq_res_fixfix[2]; - UCHAR fResTransIsLow; - - INVF_MODE sbr_invf_mode; - INVF_MODE sbr_invf_mode_vec[MAX_NUM_NOISE_VALUES]; - - XPOS_MODE sbr_xpos_mode; - - INT ienvelope[MAX_ENVELOPES][MAX_FREQ_COEFFS]; - - INT codeBookScfLavBalance; - INT codeBookScfLav; - const INT *hufftableTimeC; - const INT *hufftableFreqC; - const UCHAR *hufftableTimeL; - const UCHAR *hufftableFreqL; - - const INT *hufftableLevelTimeC; - const INT *hufftableBalanceTimeC; - const INT *hufftableLevelFreqC; - const INT *hufftableBalanceFreqC; - const UCHAR *hufftableLevelTimeL; - const UCHAR *hufftableBalanceTimeL; - const UCHAR *hufftableLevelFreqL; - const UCHAR *hufftableBalanceFreqL; - - - const UCHAR *hufftableNoiseTimeL; - const INT *hufftableNoiseTimeC; - const UCHAR *hufftableNoiseFreqL; - const INT *hufftableNoiseFreqC; - - const UCHAR *hufftableNoiseLevelTimeL; - const INT *hufftableNoiseLevelTimeC; - const UCHAR *hufftableNoiseBalanceTimeL; - const INT *hufftableNoiseBalanceTimeC; - const UCHAR *hufftableNoiseLevelFreqL; - const INT *hufftableNoiseLevelFreqC; - const UCHAR *hufftableNoiseBalanceFreqL; - const INT *hufftableNoiseBalanceFreqC; - - HANDLE_SBR_GRID hSbrBSGrid; - - INT noHarmonics; - INT addHarmonicFlag; - UCHAR addHarmonic[MAX_FREQ_COEFFS]; - - - /* calculated helper vars */ - INT si_sbr_start_env_bits_balance; - INT si_sbr_start_env_bits; - INT si_sbr_start_noise_bits_balance; - INT si_sbr_start_noise_bits; - - INT noOfEnvelopes; - INT noScfBands[MAX_ENVELOPES]; - INT domain_vec[MAX_ENVELOPES]; - INT domain_vec_noise[MAX_ENVELOPES]; - SCHAR sbr_noise_levels[MAX_FREQ_COEFFS]; - INT noOfnoisebands; - - INT balance; - AMP_RES init_sbr_amp_res; - AMP_RES currentAmpResFF; - FIXP_DBL ton_HF[SBR_GLOBAL_TONALITY_VALUES]; /* tonality is scaled by 2^19/0.524288f (fract part of RELAXATION) */ - FIXP_DBL global_tonality; - - /* extended data */ - INT extended_data; - INT extension_size; - INT extension_id; - UCHAR extended_data_buffer[SBR_EXTENDED_DATA_MAX_CNT]; - - UCHAR ldGrid; -}; -typedef struct SBR_ENV_DATA *HANDLE_SBR_ENV_DATA; - - - -INT FDKsbrEnc_WriteEnvSingleChannelElement(struct SBR_HEADER_DATA *sbrHeaderData, - struct T_PARAMETRIC_STEREO *hParametricStereo, - struct SBR_BITSTREAM_DATA *sbrBitstreamData, - struct SBR_ENV_DATA *sbrEnvData, - struct COMMON_DATA *cmonData, - UINT sbrSyntaxFlags); - - -INT FDKsbrEnc_WriteEnvChannelPairElement(struct SBR_HEADER_DATA *sbrHeaderData, - struct T_PARAMETRIC_STEREO *hParametricStereo, - struct SBR_BITSTREAM_DATA *sbrBitstreamData, - struct SBR_ENV_DATA *sbrEnvDataLeft, - struct SBR_ENV_DATA *sbrEnvDataRight, - struct COMMON_DATA *cmonData, - UINT sbrSyntaxFlags); - - - -INT FDKsbrEnc_CountSbrChannelPairElement (struct SBR_HEADER_DATA *sbrHeaderData, - struct T_PARAMETRIC_STEREO *hParametricStereo, - struct SBR_BITSTREAM_DATA *sbrBitstreamData, - struct SBR_ENV_DATA *sbrEnvDataLeft, - struct SBR_ENV_DATA *sbrEnvDataRight, - struct COMMON_DATA *cmonData, - UINT sbrSyntaxFlags); - - - -/* debugging and tuning functions */ - -/*#define SBR_ENV_STATISTICS */ - - -/*#define SBR_PAYLOAD_MONITOR*/ - -#endif diff --git a/libSBRenc/src/cmondata.h b/libSBRenc/src/cmondata.h deleted file mode 100644 index 32e6993..0000000 --- a/libSBRenc/src/cmondata.h +++ /dev/null @@ -1,110 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief Core Coder's and SBR's shared data structure definition -*/ -#ifndef __SBR_CMONDATA_H -#define __SBR_CMONDATA_H - -#include "FDK_bitstream.h" - - -struct COMMON_DATA { - INT sbrHdrBits; /**< number of SBR header bits */ - INT sbrDataBits; /**< number of SBR data bits */ - INT sbrFillBits; /**< number of SBR fill bits */ - FDK_BITSTREAM sbrBitbuf; /**< the SBR data bitbuffer */ - FDK_BITSTREAM tmpWriteBitbuf; /**< helper var for writing header*/ - INT xOverFreq; /**< the SBR crossover frequency */ - INT dynBwEnabled; /**< indicates if dynamic bandwidth is enabled */ - INT sbrNumChannels; /**< number of channels (meaning mono or stereo) */ - INT dynXOverFreqEnc; /**< encoder dynamic crossover frequency */ -}; - -typedef struct COMMON_DATA *HANDLE_COMMON_DATA; - - - -#endif diff --git a/libSBRenc/src/code_env.cpp b/libSBRenc/src/code_env.cpp deleted file mode 100644 index e1a28d5..0000000 --- a/libSBRenc/src/code_env.cpp +++ /dev/null @@ -1,641 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -#include "code_env.h" -#include "sbr_rom.h" - -/***************************************************************************** - - functionname: FDKsbrEnc_InitSbrHuffmanTables - description: initializes Huffman Tables dependent on chosen amp_res - returns: error handle - input: - output: - -*****************************************************************************/ -INT -FDKsbrEnc_InitSbrHuffmanTables (HANDLE_SBR_ENV_DATA sbrEnvData, - HANDLE_SBR_CODE_ENVELOPE henv, - HANDLE_SBR_CODE_ENVELOPE hnoise, - AMP_RES amp_res) -{ - if ( (!henv) || (!hnoise) || (!sbrEnvData) ) - return (1); /* not init. */ - - sbrEnvData->init_sbr_amp_res = amp_res; - - switch (amp_res) { - case SBR_AMP_RES_3_0: - /*envelope data*/ - - /*Level/Pan - coding */ - sbrEnvData->hufftableLevelTimeC = v_Huff_envelopeLevelC11T; - sbrEnvData->hufftableLevelTimeL = v_Huff_envelopeLevelL11T; - sbrEnvData->hufftableBalanceTimeC = bookSbrEnvBalanceC11T; - sbrEnvData->hufftableBalanceTimeL = bookSbrEnvBalanceL11T; - - sbrEnvData->hufftableLevelFreqC = v_Huff_envelopeLevelC11F; - sbrEnvData->hufftableLevelFreqL = v_Huff_envelopeLevelL11F; - sbrEnvData->hufftableBalanceFreqC = bookSbrEnvBalanceC11F; - sbrEnvData->hufftableBalanceFreqL = bookSbrEnvBalanceL11F; - - /*Right/Left - coding */ - sbrEnvData->hufftableTimeC = v_Huff_envelopeLevelC11T; - sbrEnvData->hufftableTimeL = v_Huff_envelopeLevelL11T; - sbrEnvData->hufftableFreqC = v_Huff_envelopeLevelC11F; - sbrEnvData->hufftableFreqL = v_Huff_envelopeLevelL11F; - - sbrEnvData->codeBookScfLavBalance = CODE_BOOK_SCF_LAV_BALANCE11; - sbrEnvData->codeBookScfLav = CODE_BOOK_SCF_LAV11; - - sbrEnvData->si_sbr_start_env_bits = SI_SBR_START_ENV_BITS_AMP_RES_3_0; - sbrEnvData->si_sbr_start_env_bits_balance = SI_SBR_START_ENV_BITS_BALANCE_AMP_RES_3_0; - break; - - case SBR_AMP_RES_1_5: - /*envelope data*/ - - /*Level/Pan - coding */ - sbrEnvData->hufftableLevelTimeC = v_Huff_envelopeLevelC10T; - sbrEnvData->hufftableLevelTimeL = v_Huff_envelopeLevelL10T; - sbrEnvData->hufftableBalanceTimeC = bookSbrEnvBalanceC10T; - sbrEnvData->hufftableBalanceTimeL = bookSbrEnvBalanceL10T; - - sbrEnvData->hufftableLevelFreqC = v_Huff_envelopeLevelC10F; - sbrEnvData->hufftableLevelFreqL = v_Huff_envelopeLevelL10F; - sbrEnvData->hufftableBalanceFreqC = bookSbrEnvBalanceC10F; - sbrEnvData->hufftableBalanceFreqL = bookSbrEnvBalanceL10F; - - /*Right/Left - coding */ - sbrEnvData->hufftableTimeC = v_Huff_envelopeLevelC10T; - sbrEnvData->hufftableTimeL = v_Huff_envelopeLevelL10T; - sbrEnvData->hufftableFreqC = v_Huff_envelopeLevelC10F; - sbrEnvData->hufftableFreqL = v_Huff_envelopeLevelL10F; - - sbrEnvData->codeBookScfLavBalance = CODE_BOOK_SCF_LAV_BALANCE10; - sbrEnvData->codeBookScfLav = CODE_BOOK_SCF_LAV10; - - sbrEnvData->si_sbr_start_env_bits = SI_SBR_START_ENV_BITS_AMP_RES_1_5; - sbrEnvData->si_sbr_start_env_bits_balance = SI_SBR_START_ENV_BITS_BALANCE_AMP_RES_1_5; - break; - - default: - return (1); /* undefined amp_res mode */ - } - - /* these are common to both amp_res values */ - /*Noise data*/ - - /*Level/Pan - coding */ - sbrEnvData->hufftableNoiseLevelTimeC = v_Huff_NoiseLevelC11T; - sbrEnvData->hufftableNoiseLevelTimeL = v_Huff_NoiseLevelL11T; - sbrEnvData->hufftableNoiseBalanceTimeC = bookSbrNoiseBalanceC11T; - sbrEnvData->hufftableNoiseBalanceTimeL = bookSbrNoiseBalanceL11T; - - sbrEnvData->hufftableNoiseLevelFreqC = v_Huff_envelopeLevelC11F; - sbrEnvData->hufftableNoiseLevelFreqL = v_Huff_envelopeLevelL11F; - sbrEnvData->hufftableNoiseBalanceFreqC = bookSbrEnvBalanceC11F; - sbrEnvData->hufftableNoiseBalanceFreqL = bookSbrEnvBalanceL11F; - - - /*Right/Left - coding */ - sbrEnvData->hufftableNoiseTimeC = v_Huff_NoiseLevelC11T; - sbrEnvData->hufftableNoiseTimeL = v_Huff_NoiseLevelL11T; - sbrEnvData->hufftableNoiseFreqC = v_Huff_envelopeLevelC11F; - sbrEnvData->hufftableNoiseFreqL = v_Huff_envelopeLevelL11F; - - sbrEnvData->si_sbr_start_noise_bits = SI_SBR_START_NOISE_BITS_AMP_RES_3_0; - sbrEnvData->si_sbr_start_noise_bits_balance = SI_SBR_START_NOISE_BITS_BALANCE_AMP_RES_3_0; - - - /* init envelope tables and codebooks */ - henv->codeBookScfLavBalanceTime = sbrEnvData->codeBookScfLavBalance; - henv->codeBookScfLavBalanceFreq = sbrEnvData->codeBookScfLavBalance; - henv->codeBookScfLavLevelTime = sbrEnvData->codeBookScfLav; - henv->codeBookScfLavLevelFreq = sbrEnvData->codeBookScfLav; - henv->codeBookScfLavTime = sbrEnvData->codeBookScfLav; - henv->codeBookScfLavFreq = sbrEnvData->codeBookScfLav; - - henv->hufftableLevelTimeL = sbrEnvData->hufftableLevelTimeL; - henv->hufftableBalanceTimeL = sbrEnvData->hufftableBalanceTimeL; - henv->hufftableTimeL = sbrEnvData->hufftableTimeL; - henv->hufftableLevelFreqL = sbrEnvData->hufftableLevelFreqL; - henv->hufftableBalanceFreqL = sbrEnvData->hufftableBalanceFreqL; - henv->hufftableFreqL = sbrEnvData->hufftableFreqL; - - henv->codeBookScfLavFreq = sbrEnvData->codeBookScfLav; - henv->codeBookScfLavTime = sbrEnvData->codeBookScfLav; - - henv->start_bits = sbrEnvData->si_sbr_start_env_bits; - henv->start_bits_balance = sbrEnvData->si_sbr_start_env_bits_balance; - - - /* init noise tables and codebooks */ - - hnoise->codeBookScfLavBalanceTime = CODE_BOOK_SCF_LAV_BALANCE11; - hnoise->codeBookScfLavBalanceFreq = CODE_BOOK_SCF_LAV_BALANCE11; - hnoise->codeBookScfLavLevelTime = CODE_BOOK_SCF_LAV11; - hnoise->codeBookScfLavLevelFreq = CODE_BOOK_SCF_LAV11; - hnoise->codeBookScfLavTime = CODE_BOOK_SCF_LAV11; - hnoise->codeBookScfLavFreq = CODE_BOOK_SCF_LAV11; - - hnoise->hufftableLevelTimeL = sbrEnvData->hufftableNoiseLevelTimeL; - hnoise->hufftableBalanceTimeL = sbrEnvData->hufftableNoiseBalanceTimeL; - hnoise->hufftableTimeL = sbrEnvData->hufftableNoiseTimeL; - hnoise->hufftableLevelFreqL = sbrEnvData->hufftableNoiseLevelFreqL; - hnoise->hufftableBalanceFreqL = sbrEnvData->hufftableNoiseBalanceFreqL; - hnoise->hufftableFreqL = sbrEnvData->hufftableNoiseFreqL; - - - hnoise->start_bits = sbrEnvData->si_sbr_start_noise_bits; - hnoise->start_bits_balance = sbrEnvData->si_sbr_start_noise_bits_balance; - - /* No delta coding in time from the previous frame due to 1.5dB FIx-FIX rule */ - henv->upDate = 0; - hnoise->upDate = 0; - return (0); -} - -/******************************************************************************* - Functionname: indexLow2High - ******************************************************************************* - - Description: Nice small patch-functions in order to cope with non-factor-2 - ratios between high-res and low-res - - Arguments: INT offset, INT index, FREQ_RES res - - Return: INT - -*******************************************************************************/ -static INT indexLow2High(INT offset, INT index, FREQ_RES res) -{ - - if(res == FREQ_RES_LOW) - { - if (offset >= 0) - { - if (index < offset) - return(index); - else - return(2*index - offset); - } - else - { - offset = -offset; - if (index < offset) - return(2*index+index); - else - return(2*index + offset); - } - } - else - return(index); -} - - - -/******************************************************************************* - Functionname: mapLowResEnergyVal - ******************************************************************************* - - Description: - - Arguments: INT currVal,INT* prevData, INT offset, INT index, FREQ_RES res - - Return: none - -*******************************************************************************/ -static void mapLowResEnergyVal(SCHAR currVal, SCHAR* prevData, INT offset, INT index, FREQ_RES res) -{ - - if(res == FREQ_RES_LOW) - { - if (offset >= 0) - { - if(index < offset) - prevData[index] = currVal; - else - { - prevData[2*index - offset] = currVal; - prevData[2*index+1 - offset] = currVal; - } - } - else - { - offset = -offset; - if (index < offset) - { - prevData[3*index] = currVal; - prevData[3*index+1] = currVal; - prevData[3*index+2] = currVal; - } - else - { - prevData[2*index + offset] = currVal; - prevData[2*index + 1 + offset] = currVal; - } - } - } - else - prevData[index] = currVal; -} - - - -/******************************************************************************* - Functionname: computeBits - ******************************************************************************* - - Description: - - Arguments: INT delta, - INT codeBookScfLavLevel, - INT codeBookScfLavBalance, - const UCHAR * hufftableLevel, - const UCHAR * hufftableBalance, INT coupling, INT channel) - - Return: INT - -*******************************************************************************/ -static INT -computeBits (SCHAR *delta, - INT codeBookScfLavLevel, - INT codeBookScfLavBalance, - const UCHAR * hufftableLevel, - const UCHAR * hufftableBalance, INT coupling, INT channel) -{ - INT index; - INT delta_bits = 0; - - if (coupling) { - if (channel == 1) - { - if (*delta < 0) - index = fixMax(*delta, -codeBookScfLavBalance); - else - index = fixMin(*delta, codeBookScfLavBalance); - - if (index != *delta) { - *delta = index; - return (10000); - } - - delta_bits = hufftableBalance[index + codeBookScfLavBalance]; - } - else { - if (*delta < 0) - index = fixMax(*delta, -codeBookScfLavLevel); - else - index = fixMin(*delta, codeBookScfLavLevel); - - if (index != *delta) { - *delta = index; - return (10000); - } - delta_bits = hufftableLevel[index + codeBookScfLavLevel]; - } - } - else { - if (*delta < 0) - index = fixMax(*delta, -codeBookScfLavLevel); - else - index = fixMin(*delta, codeBookScfLavLevel); - - if (index != *delta) { - *delta = index; - return (10000); - } - delta_bits = hufftableLevel[index + codeBookScfLavLevel]; - } - - return (delta_bits); -} - - - - -/******************************************************************************* - Functionname: FDKsbrEnc_codeEnvelope - ******************************************************************************* - - Description: - - Arguments: INT *sfb_nrg, - const FREQ_RES *freq_res, - SBR_CODE_ENVELOPE * h_sbrCodeEnvelope, - INT *directionVec, INT scalable, INT nEnvelopes, INT channel, - INT headerActive) - - Return: none - h_sbrCodeEnvelope->sfb_nrg_prev is modified ! - sfb_nrg is modified - h_sbrCodeEnvelope->update is modfied ! - *directionVec is modified - -*******************************************************************************/ -void -FDKsbrEnc_codeEnvelope(SCHAR *sfb_nrg, - const FREQ_RES *freq_res, - SBR_CODE_ENVELOPE *h_sbrCodeEnvelope, - INT *directionVec, - INT coupling, - INT nEnvelopes, - INT channel, - INT headerActive) -{ - INT i, no_of_bands, band; - FIXP_DBL tmp1,tmp2,tmp3,dF_edge_1stEnv; - SCHAR *ptr_nrg; - - INT codeBookScfLavLevelTime; - INT codeBookScfLavLevelFreq; - INT codeBookScfLavBalanceTime; - INT codeBookScfLavBalanceFreq; - const UCHAR *hufftableLevelTimeL; - const UCHAR *hufftableBalanceTimeL; - const UCHAR *hufftableLevelFreqL; - const UCHAR *hufftableBalanceFreqL; - - INT offset = h_sbrCodeEnvelope->offset; - INT envDataTableCompFactor; - - INT delta_F_bits = 0, delta_T_bits = 0; - INT use_dT; - - SCHAR delta_F[MAX_FREQ_COEFFS]; - SCHAR delta_T[MAX_FREQ_COEFFS]; - SCHAR last_nrg, curr_nrg; - - tmp1 = FL2FXCONST_DBL(0.5f) >> (DFRACT_BITS-16-1); - tmp2 = h_sbrCodeEnvelope->dF_edge_1stEnv >> (DFRACT_BITS-16); - tmp3 = (FIXP_DBL)(((INT)(LONG)h_sbrCodeEnvelope->dF_edge_incr*h_sbrCodeEnvelope->dF_edge_incr_fac) >> (DFRACT_BITS-16)); - - dF_edge_1stEnv = tmp1 + tmp2 + tmp3; - - if (coupling) { - codeBookScfLavLevelTime = h_sbrCodeEnvelope->codeBookScfLavLevelTime; - codeBookScfLavLevelFreq = h_sbrCodeEnvelope->codeBookScfLavLevelFreq; - codeBookScfLavBalanceTime = h_sbrCodeEnvelope->codeBookScfLavBalanceTime; - codeBookScfLavBalanceFreq = h_sbrCodeEnvelope->codeBookScfLavBalanceFreq; - hufftableLevelTimeL = h_sbrCodeEnvelope->hufftableLevelTimeL; - hufftableBalanceTimeL = h_sbrCodeEnvelope->hufftableBalanceTimeL; - hufftableLevelFreqL = h_sbrCodeEnvelope->hufftableLevelFreqL; - hufftableBalanceFreqL = h_sbrCodeEnvelope->hufftableBalanceFreqL; - } - else { - codeBookScfLavLevelTime = h_sbrCodeEnvelope->codeBookScfLavTime; - codeBookScfLavLevelFreq = h_sbrCodeEnvelope->codeBookScfLavFreq; - codeBookScfLavBalanceTime = h_sbrCodeEnvelope->codeBookScfLavTime; - codeBookScfLavBalanceFreq = h_sbrCodeEnvelope->codeBookScfLavFreq; - hufftableLevelTimeL = h_sbrCodeEnvelope->hufftableTimeL; - hufftableBalanceTimeL = h_sbrCodeEnvelope->hufftableTimeL; - hufftableLevelFreqL = h_sbrCodeEnvelope->hufftableFreqL; - hufftableBalanceFreqL = h_sbrCodeEnvelope->hufftableFreqL; - } - - if(coupling == 1 && channel == 1) - envDataTableCompFactor = 1; /*should be one when the new huffman-tables are ready*/ - else - envDataTableCompFactor = 0; - - - if (h_sbrCodeEnvelope->deltaTAcrossFrames == 0) - h_sbrCodeEnvelope->upDate = 0; - - /* no delta coding in time in case of a header */ - if (headerActive) - h_sbrCodeEnvelope->upDate = 0; - - - for (i = 0; i < nEnvelopes; i++) - { - if (freq_res[i] == FREQ_RES_HIGH) - no_of_bands = h_sbrCodeEnvelope->nSfb[FREQ_RES_HIGH]; - else - no_of_bands = h_sbrCodeEnvelope->nSfb[FREQ_RES_LOW]; - - - ptr_nrg = sfb_nrg; - curr_nrg = *ptr_nrg; - - delta_F[0] = curr_nrg >> envDataTableCompFactor; - - if (coupling && channel == 1) - delta_F_bits = h_sbrCodeEnvelope->start_bits_balance; - else - delta_F_bits = h_sbrCodeEnvelope->start_bits; - - - if(h_sbrCodeEnvelope->upDate != 0) - { - delta_T[0] = (curr_nrg - h_sbrCodeEnvelope->sfb_nrg_prev[0]) >> envDataTableCompFactor; - - delta_T_bits = computeBits (&delta_T[0], - codeBookScfLavLevelTime, - codeBookScfLavBalanceTime, - hufftableLevelTimeL, - hufftableBalanceTimeL, coupling, channel); - } - - - mapLowResEnergyVal(curr_nrg, h_sbrCodeEnvelope->sfb_nrg_prev, offset, 0, freq_res[i]); - - /* ensure that nrg difference is not higher than codeBookScfLavXXXFreq */ - if ( coupling && channel == 1 ) { - for (band = no_of_bands - 1; band > 0; band--) { - if ( ptr_nrg[band] - ptr_nrg[band-1] > codeBookScfLavBalanceFreq ) { - ptr_nrg[band-1] = ptr_nrg[band] - codeBookScfLavBalanceFreq; - } - } - for (band = 1; band < no_of_bands; band++) { - if ( ptr_nrg[band-1] - ptr_nrg[band] > codeBookScfLavBalanceFreq ) { - ptr_nrg[band] = ptr_nrg[band-1] - codeBookScfLavBalanceFreq; - } - } - } - else { - for (band = no_of_bands - 1; band > 0; band--) { - if ( ptr_nrg[band] - ptr_nrg[band-1] > codeBookScfLavLevelFreq ) { - ptr_nrg[band-1] = ptr_nrg[band] - codeBookScfLavLevelFreq; - } - } - for (band = 1; band < no_of_bands; band++) { - if ( ptr_nrg[band-1] - ptr_nrg[band] > codeBookScfLavLevelFreq ) { - ptr_nrg[band] = ptr_nrg[band-1] - codeBookScfLavLevelFreq; - } - } - } - - - /* Coding loop*/ - for (band = 1; band < no_of_bands; band++) - { - last_nrg = (*ptr_nrg); - ptr_nrg++; - curr_nrg = (*ptr_nrg); - - delta_F[band] = (curr_nrg - last_nrg) >> envDataTableCompFactor; - - delta_F_bits += computeBits (&delta_F[band], - codeBookScfLavLevelFreq, - codeBookScfLavBalanceFreq, - hufftableLevelFreqL, - hufftableBalanceFreqL, coupling, channel); - - if(h_sbrCodeEnvelope->upDate != 0) - { - delta_T[band] = curr_nrg - h_sbrCodeEnvelope->sfb_nrg_prev[indexLow2High(offset, band, freq_res[i])]; - delta_T[band] = delta_T[band] >> envDataTableCompFactor; - } - - mapLowResEnergyVal(curr_nrg, h_sbrCodeEnvelope->sfb_nrg_prev, offset, band, freq_res[i]); - - if(h_sbrCodeEnvelope->upDate != 0) - { - delta_T_bits += computeBits (&delta_T[band], - codeBookScfLavLevelTime, - codeBookScfLavBalanceTime, - hufftableLevelTimeL, - hufftableBalanceTimeL, coupling, channel); - } - } - - /* Replace sfb_nrg with deltacoded samples and set flag */ - if (i == 0) { - INT tmp_bits; - tmp_bits = (((delta_T_bits * dF_edge_1stEnv) >> (DFRACT_BITS-18)) + (FIXP_DBL)1) >> 1; - use_dT = (h_sbrCodeEnvelope->upDate != 0 && (delta_F_bits > tmp_bits)); - } - else - use_dT = (delta_T_bits < delta_F_bits && h_sbrCodeEnvelope->upDate != 0); - - if (use_dT) - { - directionVec[i] = TIME; - FDKmemcpy (sfb_nrg, delta_T, no_of_bands * sizeof (SCHAR)); - } - else { - h_sbrCodeEnvelope->upDate = 0; - directionVec[i] = FREQ; - FDKmemcpy (sfb_nrg, delta_F, no_of_bands * sizeof (SCHAR)); - } - sfb_nrg += no_of_bands; - h_sbrCodeEnvelope->upDate = 1; - } - -} - - -/******************************************************************************* - Functionname: FDKsbrEnc_InitSbrCodeEnvelope - ******************************************************************************* - - Description: - - Arguments: - - Return: - -*******************************************************************************/ -INT -FDKsbrEnc_InitSbrCodeEnvelope (HANDLE_SBR_CODE_ENVELOPE h_sbrCodeEnvelope, - INT *nSfb, - INT deltaTAcrossFrames, - FIXP_DBL dF_edge_1stEnv, - FIXP_DBL dF_edge_incr) -{ - - FDKmemclear(h_sbrCodeEnvelope,sizeof(SBR_CODE_ENVELOPE)); - - h_sbrCodeEnvelope->deltaTAcrossFrames = deltaTAcrossFrames; - h_sbrCodeEnvelope->dF_edge_1stEnv = dF_edge_1stEnv; - h_sbrCodeEnvelope->dF_edge_incr = dF_edge_incr; - h_sbrCodeEnvelope->dF_edge_incr_fac = 0; - h_sbrCodeEnvelope->upDate = 0; - h_sbrCodeEnvelope->nSfb[FREQ_RES_LOW] = nSfb[FREQ_RES_LOW]; - h_sbrCodeEnvelope->nSfb[FREQ_RES_HIGH] = nSfb[FREQ_RES_HIGH]; - h_sbrCodeEnvelope->offset = 2*h_sbrCodeEnvelope->nSfb[FREQ_RES_LOW] - h_sbrCodeEnvelope->nSfb[FREQ_RES_HIGH]; - - return (0); -} diff --git a/libSBRenc/src/code_env.h b/libSBRenc/src/code_env.h deleted file mode 100644 index 50a365e..0000000 --- a/libSBRenc/src/code_env.h +++ /dev/null @@ -1,153 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief DPCM Envelope coding -*/ - -#ifndef __CODE_ENV_H -#define __CODE_ENV_H - -#include "sbr_def.h" -#include "bit_sbr.h" -#include "fram_gen.h" - -typedef struct -{ - INT offset; - INT upDate; - INT nSfb[2]; - SCHAR sfb_nrg_prev[MAX_FREQ_COEFFS]; - INT deltaTAcrossFrames; - FIXP_DBL dF_edge_1stEnv; - FIXP_DBL dF_edge_incr; - INT dF_edge_incr_fac; - - - INT codeBookScfLavTime; - INT codeBookScfLavFreq; - - INT codeBookScfLavLevelTime; - INT codeBookScfLavLevelFreq; - INT codeBookScfLavBalanceTime; - INT codeBookScfLavBalanceFreq; - - INT start_bits; - INT start_bits_balance; - - - const UCHAR *hufftableTimeL; - const UCHAR *hufftableFreqL; - - const UCHAR *hufftableLevelTimeL; - const UCHAR *hufftableBalanceTimeL; - const UCHAR *hufftableLevelFreqL; - const UCHAR *hufftableBalanceFreqL; -} -SBR_CODE_ENVELOPE; -typedef SBR_CODE_ENVELOPE *HANDLE_SBR_CODE_ENVELOPE; - - - -void -FDKsbrEnc_codeEnvelope (SCHAR *sfb_nrg, - const FREQ_RES *freq_res, - SBR_CODE_ENVELOPE * h_sbrCodeEnvelope, - INT *directionVec, INT coupling, INT nEnvelopes, INT channel, - INT headerActive); - -INT -FDKsbrEnc_InitSbrCodeEnvelope (HANDLE_SBR_CODE_ENVELOPE h_sbrCodeEnvelope, - INT *nSfb, - INT deltaTAcrossFrames, - FIXP_DBL dF_edge_1stEnv, - FIXP_DBL dF_edge_incr); - -INT -FDKsbrEnc_InitSbrHuffmanTables (struct SBR_ENV_DATA* sbrEnvData, - HANDLE_SBR_CODE_ENVELOPE henv, - HANDLE_SBR_CODE_ENVELOPE hnoise, - AMP_RES amp_res); - -#endif diff --git a/libSBRenc/src/env_bit.cpp b/libSBRenc/src/env_bit.cpp deleted file mode 100644 index ea31183..0000000 --- a/libSBRenc/src/env_bit.cpp +++ /dev/null @@ -1,250 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief Remaining SBR Bit Writing Routines -*/ - -#include "env_bit.h" -#include "cmondata.h" - - -#ifndef min -#define min(a,b) ( a < b ? a:b) -#endif - -#ifndef max -#define max(a,b) ( a > b ? a:b) -#endif - -/* ***************************** crcAdvance **********************************/ -/** - * @fn - * @brief updates crc data register - * @return none - * - * This function updates the crc register - * - */ -static void crcAdvance(USHORT crcPoly, - USHORT crcMask, - USHORT *crc, - ULONG bValue, - INT bBits - ) -{ - INT i; - USHORT flag; - - for (i=bBits-1; i>=0; i--) { - flag = ((*crc) & crcMask) ? (1) : (0) ; - flag ^= (bValue & (1<sbrBitbuf, BS_WRITER); - - FDKinitBitStream(&hCmonData->tmpWriteBitbuf, memoryBase, - memorySize, 0, BS_WRITER); - - if (sbrSyntaxFlags & SBR_SYNTAX_CRC) { - if (sbrSyntaxFlags & SBR_SYNTAX_DRM_CRC) - { /* Init and start CRC region */ - FDKwriteBits (&hCmonData->sbrBitbuf, 0x0, SI_SBR_DRM_CRC_BITS); - FDKcrcInit( hCrcInfo, 0x001d, 0xFFFF, SI_SBR_DRM_CRC_BITS ); - crcRegion = FDKcrcStartReg( hCrcInfo, &hCmonData->sbrBitbuf, 0 ); - } else { - FDKwriteBits (&hCmonData->sbrBitbuf, 0x0, SI_SBR_CRC_BITS); - } - } - - return (crcRegion); -} - - -/* ************************** FDKsbrEnc_AssembleSbrBitstream *******************************/ -/** - * @fn - * @brief Formats the SBR payload - * @return nothing - * - * Also the CRC will be calculated here. - * - */ - -void -FDKsbrEnc_AssembleSbrBitstream( HANDLE_COMMON_DATA hCmonData, - HANDLE_FDK_CRCINFO hCrcInfo, - INT crcRegion, - UINT sbrSyntaxFlags) -{ - USHORT crcReg = SBR_CRCINIT; - INT numCrcBits,i; - - /* check if SBR is present */ - if ( hCmonData==NULL ) - return; - - hCmonData->sbrFillBits = 0; /* Fill bits are written only for GA streams */ - - if ( sbrSyntaxFlags & SBR_SYNTAX_DRM_CRC ) - { - /* - * Calculate and write DRM CRC - */ - FDKcrcEndReg( hCrcInfo, &hCmonData->sbrBitbuf, crcRegion ); - FDKwriteBits( &hCmonData->tmpWriteBitbuf, FDKcrcGetCRC(hCrcInfo)^0xFF, SI_SBR_DRM_CRC_BITS ); - } - else - { - if ( !(sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) ) - { - /* Do alignment here, because its defined as part of the sbr_extension_data */ - int sbrLoad = hCmonData->sbrHdrBits + hCmonData->sbrDataBits; - - if ( sbrSyntaxFlags & SBR_SYNTAX_CRC ) { - sbrLoad += SI_SBR_CRC_BITS; - } - - sbrLoad += 4; /* Do byte Align with 4 bit offset. ISO/IEC 14496-3:2005(E) page 39. */ - - hCmonData->sbrFillBits = (8 - (sbrLoad % 8)) % 8; - - /* - append fill bits - */ - FDKwriteBits(&hCmonData->sbrBitbuf, 0, hCmonData->sbrFillBits ); - - FDK_ASSERT(FDKgetValidBits(&hCmonData->sbrBitbuf) % 8 == 4); - } - - /* - calculate crc - */ - if ( sbrSyntaxFlags & SBR_SYNTAX_CRC ) { - FDK_BITSTREAM tmpCRCBuf = hCmonData->sbrBitbuf; - FDKresetBitbuffer( &tmpCRCBuf, BS_READER ); - - numCrcBits = hCmonData->sbrHdrBits + hCmonData->sbrDataBits + hCmonData->sbrFillBits; - - for(i=0;itmpWriteBitbuf, crcReg, SI_SBR_CRC_BITS); - } - } - - FDKsyncCache(&hCmonData->tmpWriteBitbuf); -} - diff --git a/libSBRenc/src/env_bit.h b/libSBRenc/src/env_bit.h deleted file mode 100644 index 038a32a..0000000 --- a/libSBRenc/src/env_bit.h +++ /dev/null @@ -1,126 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief Remaining SBR Bit Writing Routines -*/ - -#ifndef BIT_ENV_H -#define BIT_ENV_H - -#include "sbr_encoder.h" -#include "FDK_crc.h" - -/* G(x) = x^10 + x^9 + x^5 + x^4 + x + 1 */ -#define SBR_CRC_POLY (0x0233) -#define SBR_CRC_MASK (0x0200) -#define SBR_CRC_RANGE (0x03FF) -#define SBR_CRC_MAXREGS 1 -#define SBR_CRCINIT (0x0) - - -#define SI_SBR_CRC_ENABLE_BITS 0 -#define SI_SBR_CRC_BITS 10 -#define SI_SBR_DRM_CRC_BITS 8 - - -struct COMMON_DATA; - -INT FDKsbrEnc_InitSbrBitstream(struct COMMON_DATA *hCmonData, - UCHAR *memoryBase, - INT memorySize, - HANDLE_FDK_CRCINFO hCrcInfo, - UINT sbrSyntaxFlags); - -void -FDKsbrEnc_AssembleSbrBitstream (struct COMMON_DATA *hCmonData, - HANDLE_FDK_CRCINFO hCrcInfo, - INT crcReg, - UINT sbrSyntaxFlags); - - - - - -#endif /* #ifndef BIT_ENV_H */ diff --git a/libSBRenc/src/env_est.cpp b/libSBRenc/src/env_est.cpp deleted file mode 100644 index 4fcda51..0000000 --- a/libSBRenc/src/env_est.cpp +++ /dev/null @@ -1,2030 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -#include "env_est.h" -#include "tran_det.h" - -#include "qmf.h" - -#include "fram_gen.h" -#include "bit_sbr.h" -#include "cmondata.h" -#include "sbr_ram.h" - - -#include "genericStds.h" - -#define QUANT_ERROR_THRES 200 -#define Y_NRG_SCALE 5 /* noCols = 32 -> shift(5) */ - - -static const UCHAR panTable[2][10] = { { 0, 2, 4, 6, 8,12,16,20,24}, - { 0, 2, 4, 8,12, 0, 0, 0, 0 } }; -static const UCHAR maxIndex[2] = {9, 5}; - - -/****************************************************************************** - Functionname: FDKsbrEnc_GetTonality -******************************************************************************/ -/***************************************************************************/ -/*! - - \brief Calculates complete energy per band from the energy values - of the QMF subsamples. - - \brief quotaMatrix - calculated in FDKsbrEnc_CalculateTonalityQuotas() - \brief noEstPerFrame - number of estimations per frame - \brief startIndex - start index for the quota matrix - \brief Energies - energy matrix - \brief startBand - start band - \brief stopBand - number of QMF bands - \brief numberCols - number of QMF subsamples - - \return mean tonality of the 5 bands with the highest energy - scaled by 2^(RELAXATION_SHIFT+2)*RELAXATION_FRACT - -****************************************************************************/ -static FIXP_DBL FDKsbrEnc_GetTonality( - const FIXP_DBL *const *quotaMatrix, - const INT noEstPerFrame, - const INT startIndex, - const FIXP_DBL *const *Energies, - const UCHAR startBand, - const INT stopBand, - const INT numberCols - ) -{ - UCHAR b, e, k; - INT no_enMaxBand[SBR_MAX_ENERGY_VALUES] = { -1, -1, -1, -1, -1 }; - FIXP_DBL energyMax[SBR_MAX_ENERGY_VALUES] = { FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f) }; - FIXP_DBL energyMaxMin = MAXVAL_DBL; /* min. energy in energyMax array */ - UCHAR posEnergyMaxMin = 0; /* min. energy in energyMax array position */ - FIXP_DBL tonalityBand[SBR_MAX_ENERGY_VALUES] = { FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f) }; - FIXP_DBL globalTonality = FL2FXCONST_DBL(0.0f); - FIXP_DBL energyBand[QMF_CHANNELS]; - INT maxNEnergyValues; /* max. number of max. energy values */ - - /*** Sum up energies for each band ***/ - FDK_ASSERT(numberCols==15||numberCols==16); - /* numberCols is always 15 or 16 for ELD. In case of 16 bands, the - energyBands are initialized with the [15]th column. - The rest of the column energies are added in the next step. */ - if (numberCols==15) { - for (b=startBand; b>4; - } - } - - for (k=0; k<15; k++) { - for (b=startBand; b>4; - } - } - - /*** Determine 5 highest band-energies ***/ - maxNEnergyValues = fMin(SBR_MAX_ENERGY_VALUES, stopBand-startBand); - - /* Get min. value in energyMax array */ - energyMaxMin = energyMax[0] = energyBand[startBand]; - no_enMaxBand[0] = startBand; - posEnergyMaxMin = 0; - for (k=1; k energyMax[k]) { - energyMaxMin = energyMax[k]; - posEnergyMaxMin = k; - } - } - - for (b=startBand+maxNEnergyValues; b energyMaxMin) { - energyMax[posEnergyMaxMin] = energyBand[b]; - no_enMaxBand[posEnergyMaxMin] = b; - - /* Again, get min. value in energyMax array */ - energyMaxMin = energyMax[0]; - posEnergyMaxMin = 0; - for (k=1; k energyMax[k]) { - energyMaxMin = energyMax[k]; - posEnergyMaxMin = k; - } - } - } - } - /*** End determine 5 highest band-energies ***/ - - /* Get tonality values for 5 highest energies */ - for (e=0; e> 1; - } - globalTonality += tonalityBand[e] >> 2; /* headroom of 2+1 (max. 5 additions) */ - } - - return globalTonality; -} - -/***************************************************************************/ -/*! - - \brief Calculates energy form real and imaginary part of - the QMF subsamples - - \return none - -****************************************************************************/ -LNK_SECTION_CODE_L1 -static void -FDKsbrEnc_getEnergyFromCplxQmfData(FIXP_DBL **RESTRICT energyValues,/*!< the result of the operation */ - FIXP_DBL **RESTRICT realValues, /*!< the real part of the QMF subsamples */ - FIXP_DBL **RESTRICT imagValues, /*!< the imaginary part of the QMF subsamples */ - INT numberBands, /*!< number of QMF bands */ - INT numberCols, /*!< number of QMF subsamples */ - INT *qmfScale, /*!< sclefactor of QMF subsamples */ - INT *energyScale) /*!< scalefactor of energies */ -{ - int j, k; - int scale; - FIXP_DBL max_val = FL2FXCONST_DBL(0.0f); - - /* Get Scratch buffer */ - C_ALLOC_SCRATCH_START(tmpNrg, FIXP_DBL, QMF_CHANNELS*QMF_MAX_TIME_SLOTS/2); - - /* Get max possible scaling of QMF data */ - scale = DFRACT_BITS; - for (k=0; k= DFRACT_BITS-1) { - scale = (FRACT_BITS-1-*qmfScale); - } - /* prevent scaling of QFM values to -1.f */ - scale = fixMax(0,scale-1); - - /* Update QMF scale */ - *qmfScale += scale; - - /* - Calculate energy of each time slot pair, max energy - and shift QMF values as far as possible to the left. - */ - { - FIXP_DBL *nrgValues = tmpNrg; - for (k=0; k> 1; - - tr1 <<= scale; - ti1 <<= scale; - energy += fPow2AddDiv2(fPow2Div2(tr1), ti1) >> 1; - - /* Write timeslot pair energy to scratch */ - *nrgValues++ = energy; - max_val = fixMax(max_val, energy); - - /* Write back scaled QMF values */ - r0[j] = tr0; r1[j] = tr1; i0[j] = ti0; i1[j] = ti1; - } - } - } - /* energyScale: scalefactor energies of current frame */ - *energyScale = 2*(*qmfScale)-1; /* if qmfScale > 0: nr of right shifts otherwise nr of left shifts */ - - /* Scale timeslot pair energies and write to output buffer */ - scale = CountLeadingBits(max_val); - { - FIXP_DBL *nrgValues = tmpNrg; - for (k=0; k>1; k++) { - scaleValues(energyValues[k], nrgValues, numberBands, scale); - nrgValues += numberBands; - } - *energyScale += scale; - } - - /* Free Scratch buffer */ - C_ALLOC_SCRATCH_END(tmpNrg, FIXP_DBL, QMF_CHANNELS*QMF_MAX_TIME_SLOTS/2); -} - -LNK_SECTION_CODE_L1 -static void -FDKsbrEnc_getEnergyFromCplxQmfDataFull(FIXP_DBL **RESTRICT energyValues,/*!< the result of the operation */ - FIXP_DBL **RESTRICT realValues, /*!< the real part of the QMF subsamples */ - FIXP_DBL **RESTRICT imagValues, /*!< the imaginary part of the QMF subsamples */ - int numberBands, /*!< number of QMF bands */ - int numberCols, /*!< number of QMF subsamples */ - int *qmfScale, /*!< sclefactor of QMF subsamples */ - int *energyScale) /*!< scalefactor of energies */ -{ - int j, k; - int scale; - FIXP_DBL max_val = FL2FXCONST_DBL(0.0f); - - /* Get Scratch buffer */ - C_ALLOC_SCRATCH_START(tmpNrg, FIXP_DBL, QMF_MAX_TIME_SLOTS*QMF_CHANNELS/2); - - FDK_ASSERT(numberBands <= QMF_CHANNELS); - FDK_ASSERT(numberCols <= QMF_MAX_TIME_SLOTS/2); - - /* Get max possible scaling of QMF data */ - scale = DFRACT_BITS; - for (k=0; k= DFRACT_BITS-1) { - scale = (FRACT_BITS-1-*qmfScale); - } - /* prevent scaling of QFM values to -1.f */ - scale = fixMax(0,scale-1); - - /* Update QMF scale */ - *qmfScale += scale; - - /* - Calculate energy of each time slot pair, max energy - and shift QMF values as far as possible to the left. - */ - { - FIXP_DBL *nrgValues = tmpNrg; - for (k=0; k 0: nr of right shifts otherwise nr of left shifts */ - - /* Scale timeslot pair energies and write to output buffer */ - scale = CountLeadingBits(max_val); - { - FIXP_DBL *nrgValues = tmpNrg; - for (k=0; k 0 ? 1 : -1; - - nrgVal *= sign; - - min_val = FDK_INT_MAX; - panIndex = 0; - for (i = 0; i < maxIndex[ampRes]; i++) { - val = fixp_abs ((nrgVal - (INT)panTable[ampRes][i])); - - if (val < min_val) { - min_val = val; - panIndex = i; - } - } - - *quantError=min_val; - - return panTable[ampRes][maxIndex[ampRes]-1] + sign * panTable[ampRes][panIndex]; -} - - -/***************************************************************************/ -/*! - - \brief Quantisation of the noise floor levels - - \return void - -****************************************************************************/ -static void -sbrNoiseFloorLevelsQuantisation(SCHAR *RESTRICT iNoiseLevels, /*! quantized noise levels */ - FIXP_DBL *RESTRICT NoiseLevels, /*! the noise levels */ - INT coupling /*! the coupling flag */ - ) -{ - INT i; - INT tmp, dummy; - - /* Quantisation, similar to sfb quant... */ - for (i = 0; i < MAX_NUM_NOISE_VALUES; i++) { - /* tmp = NoiseLevels[i] > (PFLOAT)30.0f ? 30: (INT) (NoiseLevels[i] + (PFLOAT)0.5); */ - /* 30>>6 = 0.46875 */ - if ((FIXP_DBL)NoiseLevels[i] > FL2FXCONST_DBL(0.46875f)) { - tmp = 30; - } - else { - /* tmp = (INT)((FIXP_DBL)NoiseLevels[i] + (FL2FXCONST_DBL(0.5f)>>(*/ /* FRACT_BITS+ */ /* 6-1)));*/ - /* tmp = tmp >> (DFRACT_BITS-1-6); */ /* conversion to integer happens here */ - /* rounding is done by shifting one bit less than necessary to the right, adding '1' and then shifting the final bit */ - tmp = ((((INT)NoiseLevels[i])>>(DFRACT_BITS-1-LD_DATA_SHIFT)) ); /* conversion to integer */ - if (tmp != 0) - tmp += 1; - } - - if (coupling) { - tmp = tmp < -30 ? -30 : tmp; - tmp = mapPanorama (tmp,1,&dummy); - } - iNoiseLevels[i] = tmp; - } -} - -/***************************************************************************/ -/*! - - \brief Calculation of noise floor for coupling - - \return void - -****************************************************************************/ -static void -coupleNoiseFloor(FIXP_DBL *RESTRICT noise_level_left, /*! noise level left (modified)*/ - FIXP_DBL *RESTRICT noise_level_right /*! noise level right (modified)*/ - ) -{ - FIXP_DBL cmpValLeft,cmpValRight; - INT i; - FIXP_DBL temp1,temp2; - - for (i = 0; i < MAX_NUM_NOISE_VALUES; i++) { - - /* Calculation of the power function using ld64: - z = x^y; - z' = CalcLd64(z) = y*CalcLd64(x)/64; - z = CalcInvLd64(z'); - */ - cmpValLeft = NOISE_FLOOR_OFFSET_64 - noise_level_left[i]; - cmpValRight = NOISE_FLOOR_OFFSET_64 - noise_level_right[i]; - - if (cmpValRight < FL2FXCONST_DBL(0.0f)) { - temp1 = CalcInvLdData(NOISE_FLOOR_OFFSET_64 - noise_level_right[i]); - } - else { - temp1 = CalcInvLdData(NOISE_FLOOR_OFFSET_64 - noise_level_right[i]); - temp1 = temp1 << (DFRACT_BITS-1-LD_DATA_SHIFT-1); /* INT to fract conversion of result, if input of CalcInvLdData is positiv */ - } - - if (cmpValLeft < FL2FXCONST_DBL(0.0f)) { - temp2 = CalcInvLdData(NOISE_FLOOR_OFFSET_64 - noise_level_left[i]); - } - else { - temp2 = CalcInvLdData(NOISE_FLOOR_OFFSET_64 - noise_level_left[i]); - temp2 = temp2 << (DFRACT_BITS-1-LD_DATA_SHIFT-1); /* INT to fract conversion of result, if input of CalcInvLdData is positiv */ - } - - - if ((cmpValLeft < FL2FXCONST_DBL(0.0f)) && (cmpValRight < FL2FXCONST_DBL(0.0f))) { - noise_level_left[i] = NOISE_FLOOR_OFFSET_64 - (CalcLdData(((temp1>>1) + (temp2>>1)))); /* no scaling needed! both values are dfract */ - noise_level_right[i] = CalcLdData(temp2) - CalcLdData(temp1); - } - - if ((cmpValLeft >= FL2FXCONST_DBL(0.0f)) && (cmpValRight >= FL2FXCONST_DBL(0.0f))) { - noise_level_left[i] = NOISE_FLOOR_OFFSET_64 - (CalcLdData(((temp1>>1) + (temp2>>1))) + FL2FXCONST_DBL(0.109375f)); /* scaled with 7/64 */ - noise_level_right[i] = CalcLdData(temp2) - CalcLdData(temp1); - } - - if ((cmpValLeft >= FL2FXCONST_DBL(0.0f)) && (cmpValRight < FL2FXCONST_DBL(0.0f))) { - noise_level_left[i] = NOISE_FLOOR_OFFSET_64 - (CalcLdData(((temp1>>(7+1)) + (temp2>>1))) + FL2FXCONST_DBL(0.109375f)); /* scaled with 7/64 */ - noise_level_right[i] = (CalcLdData(temp2) + FL2FXCONST_DBL(0.109375f)) - CalcLdData(temp1); - } - - if ((cmpValLeft < FL2FXCONST_DBL(0.0f)) && (cmpValRight >= FL2FXCONST_DBL(0.0f))) { - noise_level_left[i] = NOISE_FLOOR_OFFSET_64 - (CalcLdData(((temp1>>1) + (temp2>>(7+1)))) + FL2FXCONST_DBL(0.109375f)); /* scaled with 7/64 */ - noise_level_right[i] = CalcLdData(temp2) - (CalcLdData(temp1) + FL2FXCONST_DBL(0.109375f)); /* scaled with 7/64 */ - } - } -} - -/***************************************************************************/ -/*! - - \brief Calculation of energy starting in lower band (li) up to upper band (ui) - over slots (start_pos) to (stop_pos) - - \return void - -****************************************************************************/ -static FIXP_DBL -getEnvSfbEnergy(INT li, /*! lower band */ - INT ui, /*! upper band */ - INT start_pos, /*! start slot */ - INT stop_pos, /*! stop slot */ - INT border_pos, /*! slots scaling border */ - FIXP_DBL **YBuffer, /*! sfb energy buffer */ - INT YBufferSzShift, /*! Energy buffer index scale */ - INT scaleNrg0, /*! scaling of lower slots */ - INT scaleNrg1) /*! scaling of upper slots */ -{ - /* use dynamic scaling for outer energy loop; - energies are critical and every bit is important */ - int sc0, sc1, k, l; - - FIXP_DBL nrgSum, nrg1, nrg2, accu1, accu2; - INT dynScale, dynScale1, dynScale2; - if(ui-li==0) dynScale = DFRACT_BITS-1; - else - dynScale = CalcLdInt(ui-li)>>(DFRACT_BITS-1-LD_DATA_SHIFT); - - sc0 = fixMin(scaleNrg0,Y_NRG_SCALE); sc1 = fixMin(scaleNrg1,Y_NRG_SCALE); - /* dynScale{1,2} is set such that the right shift below is positive */ - dynScale1 = fixMin((scaleNrg0-sc0),dynScale); - dynScale2 = fixMin((scaleNrg1-sc1),dynScale); - nrgSum = accu1 = accu2 = (FIXP_DBL)0; - - for (k = li; k < ui; k++) { - nrg1 = nrg2 = (FIXP_DBL)0; - for (l = start_pos; l < border_pos; l++) { - nrg1 += YBuffer[l>>YBufferSzShift][k] >> sc0; - } - for (; l < stop_pos; l++) { - nrg2 += YBuffer[l>>YBufferSzShift][k] >> sc1; - } - accu1 += (nrg1>>dynScale1); - accu2 += (nrg2>>dynScale2); - } - /* This shift factor is always positive. See comment above. */ - nrgSum += ( accu1 >> fixMin((scaleNrg0-sc0-dynScale1),(DFRACT_BITS-1)) ) - + ( accu2 >> fixMin((scaleNrg1-sc1-dynScale2),(DFRACT_BITS-1)) ); - - return nrgSum; -} - -/***************************************************************************/ -/*! - - \brief Energy compensation in missing harmonic mode - - \return void - -****************************************************************************/ -static FIXP_DBL -mhLoweringEnergy(FIXP_DBL nrg, INT M) -{ - /* - Compensating for the fact that we in the decoder map the "average energy to every QMF - band, and use this when we calculate the boost-factor. Since the mapped energy isn't - the average energy but the maximum energy in case of missing harmonic creation, we will - in the boost function calculate that too much limiting has been applied and hence we will - boost the signal although it isn't called for. Hence we need to compensate for this by - lowering the transmitted energy values for the sines so they will get the correct level - after the boost is applied. - */ - if(M > 2){ - INT tmpScale; - tmpScale = CountLeadingBits(nrg); - nrg <<= tmpScale; - nrg = fMult(nrg, FL2FXCONST_DBL(0.398107267f)); /* The maximum boost is 1.584893, so the maximum attenuation should be square(1/1.584893) = 0.398107267 */ - nrg >>= tmpScale; - } - else{ - if(M > 1){ - nrg >>= 1; - } - } - - return nrg; -} - -/***************************************************************************/ -/*! - - \brief Energy compensation in none missing harmonic mode - - \return void - -****************************************************************************/ -static FIXP_DBL nmhLoweringEnergy( - FIXP_DBL nrg, - const FIXP_DBL nrgSum, - const INT nrgSum_scale, - const INT M - ) -{ - if (nrg>FL2FXCONST_DBL(0)) { - int sc=0; - /* gain = nrgSum / (nrg*(M+1)) */ - FIXP_DBL gain = fMult(fDivNorm(nrgSum, nrg, &sc), GetInvInt(M+1)); - sc += nrgSum_scale; - - /* reduce nrg if gain smaller 1.f */ - if ( !((sc>=0) && ( gain > ((FIXP_DBL)MAXVAL_DBL>>sc) )) ) { - nrg = fMult(scaleValue(gain,sc), nrg); - } - } - return nrg; -} - -/***************************************************************************/ -/*! - - \brief calculates the envelope values from the energies, depending on - framing and stereo mode - - \return void - -****************************************************************************/ -static void -calculateSbrEnvelope (FIXP_DBL **RESTRICT YBufferLeft, /*! energy buffer left */ - FIXP_DBL **RESTRICT YBufferRight, /*! energy buffer right */ - int *RESTRICT YBufferScaleLeft, /*! scale energy buffer left */ - int *RESTRICT YBufferScaleRight, /*! scale energy buffer right */ - const SBR_FRAME_INFO *frame_info, /*! frame info vector */ - SCHAR *RESTRICT sfb_nrgLeft, /*! sfb energy buffer left */ - SCHAR *RESTRICT sfb_nrgRight, /*! sfb energy buffer right */ - HANDLE_SBR_CONFIG_DATA h_con, /*! handle to config data */ - HANDLE_ENV_CHANNEL h_sbr, /*! envelope channel handle */ - SBR_STEREO_MODE stereoMode, /*! stereo coding mode */ - INT* maxQuantError, /*! maximum quantization error, for panorama. */ - int YBufferSzShift) /*! Energy buffer index scale */ - -{ - int i, j, m = 0; - INT no_of_bands, start_pos, stop_pos, li, ui; - FREQ_RES freq_res; - - INT ca = 2 - h_sbr->encEnvData.init_sbr_amp_res; - INT oneBitLess = 0; - if (ca == 2) - oneBitLess = 1; /* LD_DATA_SHIFT => ld64 scaling; one bit less for rounding */ - - INT quantError; - INT nEnvelopes = frame_info->nEnvelopes; - INT short_env = frame_info->shortEnv - 1; - INT timeStep = h_sbr->sbrExtractEnvelope.time_step; - INT commonScale,scaleLeft0,scaleLeft1; - INT scaleRight0=0,scaleRight1=0; - - commonScale = fixMin(YBufferScaleLeft[0],YBufferScaleLeft[1]); - - if (stereoMode == SBR_COUPLING) { - commonScale = fixMin(commonScale,YBufferScaleRight[0]); - commonScale = fixMin(commonScale,YBufferScaleRight[1]); - } - - commonScale = commonScale - 7; - - scaleLeft0 = YBufferScaleLeft[0] - commonScale; - scaleLeft1 = YBufferScaleLeft[1] - commonScale ; - FDK_ASSERT ((scaleLeft0 >= 0) && (scaleLeft1 >= 0)); - - if (stereoMode == SBR_COUPLING) { - scaleRight0 = YBufferScaleRight[0] - commonScale; - scaleRight1 = YBufferScaleRight[1] - commonScale; - FDK_ASSERT ((scaleRight0 >= 0) && (scaleRight1 >= 0)); - *maxQuantError = 0; - } - - for (i = 0; i < nEnvelopes; i++) { - - FIXP_DBL pNrgLeft[QMF_MAX_TIME_SLOTS]; - FIXP_DBL pNrgRight[QMF_MAX_TIME_SLOTS]; - int envNrg_scale; - FIXP_DBL envNrgLeft = FL2FXCONST_DBL(0.0f); - FIXP_DBL envNrgRight = FL2FXCONST_DBL(0.0f); - int missingHarmonic[QMF_MAX_TIME_SLOTS]; - int count[QMF_MAX_TIME_SLOTS]; - - start_pos = timeStep * frame_info->borders[i]; - stop_pos = timeStep * frame_info->borders[i + 1]; - freq_res = frame_info->freqRes[i]; - no_of_bands = h_con->nSfb[freq_res]; - envNrg_scale = DFRACT_BITS-fNormz((FIXP_DBL)no_of_bands); - - if (i == short_env) { - stop_pos -= fixMax(2, timeStep); /* consider at least 2 QMF slots less for short envelopes (envelopes just before transients) */ - } - - for (j = 0; j < no_of_bands; j++) { - FIXP_DBL nrgLeft = FL2FXCONST_DBL(0.0f); - FIXP_DBL nrgRight = FL2FXCONST_DBL(0.0f); - - li = h_con->freqBandTable[freq_res][j]; - ui = h_con->freqBandTable[freq_res][j + 1]; - - if(freq_res == FREQ_RES_HIGH){ - if(j == 0 && ui-li > 1){ - li++; - } - } - else{ - if(j == 0 && ui-li > 2){ - li++; - } - } - - /* - Find out whether a sine will be missing in the scale-factor - band that we're currently processing. - */ - missingHarmonic[j] = 0; - - if(h_sbr->encEnvData.addHarmonicFlag){ - - if(freq_res == FREQ_RES_HIGH){ - if(h_sbr->encEnvData.addHarmonic[j]){ /*A missing sine in the current band*/ - missingHarmonic[j] = 1; - } - } - else{ - INT i; - INT startBandHigh = 0; - INT stopBandHigh = 0; - - while(h_con->freqBandTable[FREQ_RES_HIGH][startBandHigh] < h_con->freqBandTable[FREQ_RES_LOW][j]) - startBandHigh++; - while(h_con->freqBandTable[FREQ_RES_HIGH][stopBandHigh] < h_con->freqBandTable[FREQ_RES_LOW][j + 1]) - stopBandHigh++; - - for(i = startBandHigh; iencEnvData.addHarmonic[i]){ - missingHarmonic[j] = 1; - } - } - } - } - - /* - If a sine is missing in a scalefactorband, with more than one qmf channel - use the nrg from the channel with the largest nrg rather than the mean. - Compensate for the boost calculation in the decdoder. - */ - int border_pos = fixMin(stop_pos, h_sbr->sbrExtractEnvelope.YBufferWriteOffset<>envNrg_scale); - envNrgRight += (nrgRight>>envNrg_scale); - } /* j */ - - for (j = 0; j < no_of_bands; j++) { - - FIXP_DBL nrgLeft2 = FL2FXCONST_DBL(0.0f); - FIXP_DBL nrgLeft = pNrgLeft[j]; - FIXP_DBL nrgRight = pNrgRight[j]; - - /* None missing harmonic Energy lowering compensation */ - if(!missingHarmonic[j] && h_sbr->fLevelProtect) { - /* in case of missing energy in base band, - reduce reference energy to prevent overflows in decoder output */ - nrgLeft = nmhLoweringEnergy(nrgLeft, envNrgLeft, envNrg_scale, no_of_bands); - if (stereoMode == SBR_COUPLING) { - nrgRight = nmhLoweringEnergy(nrgRight, envNrgRight, envNrg_scale, no_of_bands); - } - } - - if (stereoMode == SBR_COUPLING) { - /* calc operation later with log */ - nrgLeft2 = nrgLeft; - nrgLeft = (nrgRight + nrgLeft) >> 1; - } - - /* nrgLeft = f20_log2(nrgLeft / (PFLOAT)(count * h_sbr->sbrQmf.no_channels))+(PFLOAT)44; */ - /* If nrgLeft == 0 then the Log calculations below do fail. */ - if (nrgLeft > FL2FXCONST_DBL(0.0f)) - { - FIXP_DBL tmp0,tmp1,tmp2,tmp3; - INT tmpScale; - - tmpScale = CountLeadingBits(nrgLeft); - nrgLeft = nrgLeft << tmpScale; - - tmp0 = CalcLdData(nrgLeft); /* scaled by 1/64 */ - tmp1 = ((FIXP_DBL) (commonScale+tmpScale)) << (DFRACT_BITS-1-LD_DATA_SHIFT-1); /* scaled by 1/64 */ - tmp2 = ((FIXP_DBL)(count[j]*h_con->noQmfBands)) << (DFRACT_BITS-1-14-1); - tmp2 = CalcLdData(tmp2); /* scaled by 1/64 */ - tmp3 = FL2FXCONST_DBL(0.6875f-0.21875f-0.015625f)>>1; /* scaled by 1/64 */ - - nrgLeft = ((tmp0-tmp2)>>1) + (tmp3 - tmp1); - } else { - nrgLeft = FL2FXCONST_DBL(-1.0f); - } - - /* ld64 to integer conversion */ - nrgLeft = fixMin(fixMax(nrgLeft,FL2FXCONST_DBL(0.0f)),(FL2FXCONST_DBL(0.5f)>>oneBitLess)); - nrgLeft = (FIXP_DBL)(LONG)nrgLeft >> (DFRACT_BITS-1-LD_DATA_SHIFT-1-oneBitLess-1); - sfb_nrgLeft[m] = ((INT)nrgLeft+1)>>1; /* rounding */ - - if (stereoMode == SBR_COUPLING) { - FIXP_DBL scaleFract; - int sc0, sc1; - - nrgLeft2 = fixMax((FIXP_DBL)0x1, nrgLeft2); - nrgRight = fixMax((FIXP_DBL)0x1, nrgRight); - - sc0 = CountLeadingBits(nrgLeft2); - sc1 = CountLeadingBits(nrgRight); - - scaleFract = ((FIXP_DBL)(sc0-sc1)) << (DFRACT_BITS-1-LD_DATA_SHIFT); /* scale value in ld64 representation */ - nrgRight = CalcLdData(nrgLeft2<> (DFRACT_BITS-1-LD_DATA_SHIFT-1-oneBitLess); - nrgRight = (nrgRight+(FIXP_DBL)1)>>1; /* rounding */ - - sfb_nrgRight[m] = mapPanorama (nrgRight,h_sbr->encEnvData.init_sbr_amp_res,&quantError); - - *maxQuantError = fixMax(quantError, *maxQuantError); - } - - m++; - } /* j */ - - /* Do energy compensation for sines that are present in two - QMF-bands in the original, but will only occur in one band in - the decoder due to the synthetic sine coding.*/ - if (h_con->useParametricCoding) { - m-=no_of_bands; - for (j = 0; j < no_of_bands; j++) { - if (freq_res==FREQ_RES_HIGH && h_sbr->sbrExtractEnvelope.envelopeCompensation[j]){ - sfb_nrgLeft[m] -= (ca * fixp_abs((INT)h_sbr->sbrExtractEnvelope.envelopeCompensation[j])); - } - sfb_nrgLeft[m] = fixMax(0, sfb_nrgLeft[m]); - m++; - } - } /* useParametricCoding */ - - } /* i*/ -} - -/***************************************************************************/ -/*! - - \brief calculates the noise floor and the envelope values from the - energies, depending on framing and stereo mode - - FDKsbrEnc_extractSbrEnvelope is the main function for encoding and writing the - envelope and the noise floor. The function includes the following processes: - - -Analysis subband filtering. - -Encoding SA and pan parameters (if enabled). - -Transient detection. - -****************************************************************************/ - -LNK_SECTION_CODE_L1 -void -FDKsbrEnc_extractSbrEnvelope1 ( - HANDLE_SBR_CONFIG_DATA h_con, /*! handle to config data */ - HANDLE_SBR_HEADER_DATA sbrHeaderData, - HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, - HANDLE_ENV_CHANNEL hEnvChan, - HANDLE_COMMON_DATA hCmonData, - SBR_ENV_TEMP_DATA *eData, - SBR_FRAME_TEMP_DATA *fData - ) -{ - - HANDLE_SBR_EXTRACT_ENVELOPE sbrExtrEnv = &hEnvChan->sbrExtractEnvelope; - - if (sbrExtrEnv->YBufferSzShift == 0) - FDKsbrEnc_getEnergyFromCplxQmfDataFull(&sbrExtrEnv->YBuffer[sbrExtrEnv->YBufferWriteOffset], - sbrExtrEnv->rBuffer + sbrExtrEnv->rBufferReadOffset, - sbrExtrEnv->iBuffer + sbrExtrEnv->rBufferReadOffset, - h_con->noQmfBands, - sbrExtrEnv->no_cols, - &hEnvChan->qmfScale, - &sbrExtrEnv->YBufferScale[1]); - else - FDKsbrEnc_getEnergyFromCplxQmfData(&sbrExtrEnv->YBuffer[sbrExtrEnv->YBufferWriteOffset], - sbrExtrEnv->rBuffer + sbrExtrEnv->rBufferReadOffset, - sbrExtrEnv->iBuffer + sbrExtrEnv->rBufferReadOffset, - h_con->noQmfBands, - sbrExtrEnv->no_cols, - &hEnvChan->qmfScale, - &sbrExtrEnv->YBufferScale[1]); - - - - /* - Precalculation of Tonality Quotas COEFF Transform OK - */ - FDKsbrEnc_CalculateTonalityQuotas(&hEnvChan->TonCorr, - sbrExtrEnv->rBuffer, - sbrExtrEnv->iBuffer, - h_con->freqBandTable[HI][h_con->nSfb[HI]], - hEnvChan->qmfScale); - - - if(h_con->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) { - FIXP_DBL tonality = FDKsbrEnc_GetTonality ( - hEnvChan->TonCorr.quotaMatrix, - hEnvChan->TonCorr.numberOfEstimatesPerFrame, - hEnvChan->TonCorr.startIndexMatrix, - sbrExtrEnv->YBuffer + sbrExtrEnv->YBufferWriteOffset, - h_con->freqBandTable[HI][0]+1, - h_con->noQmfBands, - sbrExtrEnv->no_cols - ); - - hEnvChan->encEnvData.ton_HF[1] = hEnvChan->encEnvData.ton_HF[0]; - hEnvChan->encEnvData.ton_HF[0] = tonality; - - /* tonality is scaled by 2^19/0.524288f (fract part of RELAXATION) */ - hEnvChan->encEnvData.global_tonality = (hEnvChan->encEnvData.ton_HF[0]>>1) + (hEnvChan->encEnvData.ton_HF[1]>>1); - } - - - - /* - Transient detection COEFF Transform OK - */ - if(h_con->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) - { - FDKsbrEnc_fastTransientDetect( - &hEnvChan->sbrFastTransientDetector, - sbrExtrEnv->YBuffer, - sbrExtrEnv->YBufferScale, - sbrExtrEnv->YBufferWriteOffset, - eData->transient_info - ); - - } - else - { - FDKsbrEnc_transientDetect(&hEnvChan->sbrTransientDetector, - sbrExtrEnv->YBuffer, - sbrExtrEnv->YBufferScale, - eData->transient_info, - sbrExtrEnv->YBufferWriteOffset, - sbrExtrEnv->YBufferSzShift, - sbrExtrEnv->time_step, - hEnvChan->SbrEnvFrame.frameMiddleSlot); - } - - - - /* - Generate flags for 2 env in a FIXFIX-frame. - Remove this function to get always 1 env per FIXFIX-frame. - */ - - /* - frame Splitter COEFF Transform OK - */ - FDKsbrEnc_frameSplitter(sbrExtrEnv->YBuffer, - sbrExtrEnv->YBufferScale, - &hEnvChan->sbrTransientDetector, - h_con->freqBandTable[1], - eData->transient_info, - sbrExtrEnv->YBufferWriteOffset, - sbrExtrEnv->YBufferSzShift, - h_con->nSfb[1], - sbrExtrEnv->time_step, - sbrExtrEnv->no_cols, - &hEnvChan->encEnvData.global_tonality); - - -} - -/***************************************************************************/ -/*! - - \brief calculates the noise floor and the envelope values from the - energies, depending on framing and stereo mode - - FDKsbrEnc_extractSbrEnvelope is the main function for encoding and writing the - envelope and the noise floor. The function includes the following processes: - - -Determine time/frequency division of current granule. - -Sending transient info to bitstream. - -Set amp_res to 1.5 dB if the current frame contains only one envelope. - -Lock dynamic bandwidth frequency change if the next envelope not starts on a - frame boundary. - -MDCT transposer (needed to detect where harmonics will be missing). - -Spectrum Estimation (used for pulse train and missing harmonics detection). - -Pulse train detection. - -Inverse Filtering detection. - -Waveform Coding. - -Missing Harmonics detection. - -Extract envelope of current frame. - -Noise floor estimation. - -Noise floor quantisation and coding. - -Encode envelope of current frame. - -Send the encoded data to the bitstream. - -Write to bitstream. - -****************************************************************************/ - -LNK_SECTION_CODE_L1 -void -FDKsbrEnc_extractSbrEnvelope2 ( - HANDLE_SBR_CONFIG_DATA h_con, /*! handle to config data */ - HANDLE_SBR_HEADER_DATA sbrHeaderData, - HANDLE_PARAMETRIC_STEREO hParametricStereo, - HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, - HANDLE_ENV_CHANNEL h_envChan0, - HANDLE_ENV_CHANNEL h_envChan1, - HANDLE_COMMON_DATA hCmonData, - SBR_ENV_TEMP_DATA *eData, - SBR_FRAME_TEMP_DATA *fData, - int clearOutput - ) -{ - HANDLE_ENV_CHANNEL h_envChan[MAX_NUM_CHANNELS] = {h_envChan0, h_envChan1}; - int ch, i, j, c, YSzShift = h_envChan[0]->sbrExtractEnvelope.YBufferSzShift; - - SBR_STEREO_MODE stereoMode = h_con->stereoMode; - int nChannels = h_con->nChannels; - const int *v_tuning; - static const int v_tuningHEAAC[6] = { 0, 2, 4, 0, 0, 0 }; - - static const int v_tuningELD[6] = { 0, 2, 3, 0, 0, 0 }; - - if (h_con->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) - v_tuning = v_tuningELD; - else - v_tuning = v_tuningHEAAC; - - - /* - Select stereo mode. - */ - if (stereoMode == SBR_COUPLING) { - if (eData[0].transient_info[1] && eData[1].transient_info[1]) { - eData[0].transient_info[0] = fixMin(eData[1].transient_info[0], eData[0].transient_info[0]); - eData[1].transient_info[0] = eData[0].transient_info[0]; - } - else { - if (eData[0].transient_info[1] && !eData[1].transient_info[1]) { - eData[1].transient_info[0] = eData[0].transient_info[0]; - } - else { - if (!eData[0].transient_info[1] && eData[1].transient_info[1]) - eData[0].transient_info[0] = eData[1].transient_info[0]; - else { - eData[0].transient_info[0] = fixMax(eData[1].transient_info[0], eData[0].transient_info[0]); - eData[1].transient_info[0] = eData[0].transient_info[0]; - } - } - } - } - - /* - Determine time/frequency division of current granule - */ - eData[0].frame_info = FDKsbrEnc_frameInfoGenerator(&h_envChan[0]->SbrEnvFrame, - eData[0].transient_info, - h_envChan[0]->sbrExtractEnvelope.pre_transient_info, - h_envChan[0]->encEnvData.ldGrid, - v_tuning); - - h_envChan[0]->encEnvData.hSbrBSGrid = &h_envChan[0]->SbrEnvFrame.SbrGrid; - - /* AAC LD patch for transient prediction */ - if (h_envChan[0]->encEnvData.ldGrid && eData[0].transient_info[2]) { - /* if next frame will start with transient, set shortEnv to numEnvelopes(shortend Envelope = shortEnv-1)*/ - h_envChan[0]->SbrEnvFrame.SbrFrameInfo.shortEnv = h_envChan[0]->SbrEnvFrame.SbrFrameInfo.nEnvelopes; - } - - - switch (stereoMode) { - case SBR_LEFT_RIGHT: - case SBR_SWITCH_LRC: - eData[1].frame_info = FDKsbrEnc_frameInfoGenerator(&h_envChan[1]->SbrEnvFrame, - eData[1].transient_info, - h_envChan[1]->sbrExtractEnvelope.pre_transient_info, - h_envChan[1]->encEnvData.ldGrid, - v_tuning); - - h_envChan[1]->encEnvData.hSbrBSGrid = &h_envChan[1]->SbrEnvFrame.SbrGrid; - - if (h_envChan[1]->encEnvData.ldGrid && eData[1].transient_info[2]) { - /* if next frame will start with transient, set shortEnv to numEnvelopes(shortend Envelope = shortEnv-1)*/ - h_envChan[1]->SbrEnvFrame.SbrFrameInfo.shortEnv = h_envChan[1]->SbrEnvFrame.SbrFrameInfo.nEnvelopes; - } - - /* compare left and right frame_infos */ - if (eData[0].frame_info->nEnvelopes != eData[1].frame_info->nEnvelopes) { - stereoMode = SBR_LEFT_RIGHT; - } else { - for (i = 0; i < eData[0].frame_info->nEnvelopes + 1; i++) { - if (eData[0].frame_info->borders[i] != eData[1].frame_info->borders[i]) { - stereoMode = SBR_LEFT_RIGHT; - break; - } - } - for (i = 0; i < eData[0].frame_info->nEnvelopes; i++) { - if (eData[0].frame_info->freqRes[i] != eData[1].frame_info->freqRes[i]) { - stereoMode = SBR_LEFT_RIGHT; - break; - } - } - if (eData[0].frame_info->shortEnv != eData[1].frame_info->shortEnv) { - stereoMode = SBR_LEFT_RIGHT; - } - } - break; - case SBR_COUPLING: - eData[1].frame_info = eData[0].frame_info; - h_envChan[1]->encEnvData.hSbrBSGrid = &h_envChan[0]->SbrEnvFrame.SbrGrid; - break; - case SBR_MONO: - /* nothing to do */ - break; - default: - FDK_ASSERT (0); - } - - - for (ch = 0; ch < nChannels;ch++) - { - HANDLE_ENV_CHANNEL hEnvChan = h_envChan[ch]; - HANDLE_SBR_EXTRACT_ENVELOPE sbrExtrEnv = &hEnvChan->sbrExtractEnvelope; - SBR_ENV_TEMP_DATA *ed = &eData[ch]; - - - /* - Send transient info to bitstream and store for next call - */ - sbrExtrEnv->pre_transient_info[0] = ed->transient_info[0];/* tran_pos */ - sbrExtrEnv->pre_transient_info[1] = ed->transient_info[1];/* tran_flag */ - hEnvChan->encEnvData.noOfEnvelopes = ed->nEnvelopes = ed->frame_info->nEnvelopes; /* number of envelopes of current frame */ - - /* - Check if the current frame is divided into one envelope only. If so, set the amplitude - resolution to 1.5 dB, otherwise may set back to chosen value - */ - if( ( hEnvChan->encEnvData.hSbrBSGrid->frameClass == FIXFIX ) - && ( ed->nEnvelopes == 1 ) ) - { - - if (h_con->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) - { - /* Note: global_tonaliy_float_value == ((float)hEnvChan->encEnvData.global_tonality/((INT64)(1)<<(31-(19+2)))/0.524288*(2.0/3.0))); - threshold_float_value == ((float)h_con->thresholdAmpResFF_m/((INT64)(1)<<(31-(h_con->thresholdAmpResFF_e)))/0.524288*(2.0/3.0))); */ - /* decision of SBR_AMP_RES */ - if (fIsLessThan( /* global_tonality > threshold ? */ - h_con->thresholdAmpResFF_m, h_con->thresholdAmpResFF_e, - hEnvChan->encEnvData.global_tonality, RELAXATION_SHIFT+2 ) - ) - { - hEnvChan->encEnvData.currentAmpResFF = SBR_AMP_RES_1_5; - } - else { - hEnvChan->encEnvData.currentAmpResFF = SBR_AMP_RES_3_0; - } - } else { - hEnvChan->encEnvData.currentAmpResFF = SBR_AMP_RES_1_5; - } - - if ( hEnvChan->encEnvData.currentAmpResFF != hEnvChan->encEnvData.init_sbr_amp_res) { - - FDKsbrEnc_InitSbrHuffmanTables(&hEnvChan->encEnvData, - &hEnvChan->sbrCodeEnvelope, - &hEnvChan->sbrCodeNoiseFloor, - hEnvChan->encEnvData.currentAmpResFF); - } - } - else { - if(sbrHeaderData->sbr_amp_res != hEnvChan->encEnvData.init_sbr_amp_res ) { - - FDKsbrEnc_InitSbrHuffmanTables(&hEnvChan->encEnvData, - &hEnvChan->sbrCodeEnvelope, - &hEnvChan->sbrCodeNoiseFloor, - sbrHeaderData->sbr_amp_res); - } - } - - if (!clearOutput) { - - /* - Tonality correction parameter extraction (inverse filtering level, noise floor additional sines). - */ - FDKsbrEnc_TonCorrParamExtr(&hEnvChan->TonCorr, - hEnvChan->encEnvData.sbr_invf_mode_vec, - ed->noiseFloor, - &hEnvChan->encEnvData.addHarmonicFlag, - hEnvChan->encEnvData.addHarmonic, - sbrExtrEnv->envelopeCompensation, - ed->frame_info, - ed->transient_info, - h_con->freqBandTable[HI], - h_con->nSfb[HI], - hEnvChan->encEnvData.sbr_xpos_mode, - h_con->sbrSyntaxFlags); - - } - - /* Low energy in low band fix */ - if ( hEnvChan->sbrTransientDetector.prevLowBandEnergy < hEnvChan->sbrTransientDetector.prevHighBandEnergy - && hEnvChan->sbrTransientDetector.prevHighBandEnergy > FL2FX_DBL(0.03) - /* The fix needs the non-fast transient detector running. - It sets prevLowBandEnergy and prevHighBandEnergy. */ - && !(h_con->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) - ) - { - int i; - - hEnvChan->fLevelProtect = 1; - - for (i=0; iencEnvData.sbr_invf_mode_vec[i] = INVF_HIGH_LEVEL; - } else { - hEnvChan->fLevelProtect = 0; - } - - hEnvChan->encEnvData.sbr_invf_mode = hEnvChan->encEnvData.sbr_invf_mode_vec[0]; - - hEnvChan->encEnvData.noOfnoisebands = hEnvChan->TonCorr.sbrNoiseFloorEstimate.noNoiseBands; - - - } /* ch */ - - - - /* - Save number of scf bands per envelope - */ - for (ch = 0; ch < nChannels;ch++) { - for (i = 0; i < eData[ch].nEnvelopes; i++){ - h_envChan[ch]->encEnvData.noScfBands[i] = - (eData[ch].frame_info->freqRes[i] == FREQ_RES_HIGH ? h_con->nSfb[FREQ_RES_HIGH] : h_con->nSfb[FREQ_RES_LOW]); - } - } - - /* - Extract envelope of current frame. - */ - switch (stereoMode) { - case SBR_MONO: - calculateSbrEnvelope (h_envChan[0]->sbrExtractEnvelope.YBuffer, NULL, - h_envChan[0]->sbrExtractEnvelope.YBufferScale, NULL, - eData[0].frame_info, eData[0].sfb_nrg, NULL, - h_con, h_envChan[0], SBR_MONO, NULL, YSzShift); - break; - case SBR_LEFT_RIGHT: - calculateSbrEnvelope (h_envChan[0]->sbrExtractEnvelope.YBuffer, NULL, - h_envChan[0]->sbrExtractEnvelope.YBufferScale, NULL, - eData[0].frame_info, eData[0].sfb_nrg, NULL, - h_con, h_envChan[0], SBR_MONO, NULL, YSzShift); - calculateSbrEnvelope (h_envChan[1]->sbrExtractEnvelope.YBuffer, NULL, - h_envChan[1]->sbrExtractEnvelope.YBufferScale, NULL, - eData[1].frame_info,eData[1].sfb_nrg, NULL, - h_con, h_envChan[1], SBR_MONO, NULL, YSzShift); - break; - case SBR_COUPLING: - calculateSbrEnvelope (h_envChan[0]->sbrExtractEnvelope.YBuffer, h_envChan[1]->sbrExtractEnvelope.YBuffer, - h_envChan[0]->sbrExtractEnvelope.YBufferScale, h_envChan[1]->sbrExtractEnvelope.YBufferScale, - eData[0].frame_info, eData[0].sfb_nrg, eData[1].sfb_nrg, - h_con, h_envChan[0], SBR_COUPLING, &fData->maxQuantError, YSzShift); - break; - case SBR_SWITCH_LRC: - calculateSbrEnvelope (h_envChan[0]->sbrExtractEnvelope.YBuffer, NULL, - h_envChan[0]->sbrExtractEnvelope.YBufferScale, NULL, - eData[0].frame_info, eData[0].sfb_nrg, NULL, - h_con, h_envChan[0], SBR_MONO, NULL, YSzShift); - calculateSbrEnvelope (h_envChan[1]->sbrExtractEnvelope.YBuffer, NULL, - h_envChan[1]->sbrExtractEnvelope.YBufferScale, NULL, - eData[1].frame_info, eData[1].sfb_nrg, NULL, - h_con, h_envChan[1], SBR_MONO,NULL, YSzShift); - calculateSbrEnvelope (h_envChan[0]->sbrExtractEnvelope.YBuffer, h_envChan[1]->sbrExtractEnvelope.YBuffer, - h_envChan[0]->sbrExtractEnvelope.YBufferScale, h_envChan[1]->sbrExtractEnvelope.YBufferScale, - eData[0].frame_info, eData[0].sfb_nrg_coupling, eData[1].sfb_nrg_coupling, - h_con, h_envChan[0], SBR_COUPLING, &fData->maxQuantError, YSzShift); - break; - } - - - - /* - Noise floor quantisation and coding. - */ - - switch (stereoMode) { - case SBR_MONO: - sbrNoiseFloorLevelsQuantisation(eData[0].noise_level, eData[0].noiseFloor, 0); - - FDKsbrEnc_codeEnvelope(eData[0].noise_level, fData->res, - &h_envChan[0]->sbrCodeNoiseFloor, - h_envChan[0]->encEnvData.domain_vec_noise, 0, - (eData[0].frame_info->nEnvelopes > 1 ? 2 : 1), 0, - sbrBitstreamData->HeaderActive); - - break; - case SBR_LEFT_RIGHT: - sbrNoiseFloorLevelsQuantisation(eData[0].noise_level,eData[0].noiseFloor, 0); - - FDKsbrEnc_codeEnvelope (eData[0].noise_level, fData->res, - &h_envChan[0]->sbrCodeNoiseFloor, - h_envChan[0]->encEnvData.domain_vec_noise, 0, - (eData[0].frame_info->nEnvelopes > 1 ? 2 : 1), 0, - sbrBitstreamData->HeaderActive); - - sbrNoiseFloorLevelsQuantisation(eData[1].noise_level,eData[1].noiseFloor, 0); - - FDKsbrEnc_codeEnvelope (eData[1].noise_level, fData->res, - &h_envChan[1]->sbrCodeNoiseFloor, - h_envChan[1]->encEnvData.domain_vec_noise, 0, - (eData[1].frame_info->nEnvelopes > 1 ? 2 : 1), 0, - sbrBitstreamData->HeaderActive); - - break; - - case SBR_COUPLING: - coupleNoiseFloor(eData[0].noiseFloor,eData[1].noiseFloor); - - sbrNoiseFloorLevelsQuantisation(eData[0].noise_level,eData[0].noiseFloor, 0); - - FDKsbrEnc_codeEnvelope (eData[0].noise_level, fData->res, - &h_envChan[0]->sbrCodeNoiseFloor, - h_envChan[0]->encEnvData.domain_vec_noise, 1, - (eData[0].frame_info->nEnvelopes > 1 ? 2 : 1), 0, - sbrBitstreamData->HeaderActive); - - sbrNoiseFloorLevelsQuantisation(eData[1].noise_level,eData[1].noiseFloor, 1); - - FDKsbrEnc_codeEnvelope (eData[1].noise_level, fData->res, - &h_envChan[1]->sbrCodeNoiseFloor, - h_envChan[1]->encEnvData.domain_vec_noise, 1, - (eData[1].frame_info->nEnvelopes > 1 ? 2 : 1), 1, - sbrBitstreamData->HeaderActive); - - break; - case SBR_SWITCH_LRC: - sbrNoiseFloorLevelsQuantisation(eData[0].noise_level,eData[0].noiseFloor, 0); - sbrNoiseFloorLevelsQuantisation(eData[1].noise_level,eData[1].noiseFloor, 0); - coupleNoiseFloor(eData[0].noiseFloor,eData[1].noiseFloor); - sbrNoiseFloorLevelsQuantisation(eData[0].noise_level_coupling,eData[0].noiseFloor, 0); - sbrNoiseFloorLevelsQuantisation(eData[1].noise_level_coupling,eData[1].noiseFloor, 1); - break; - } - - - - /* - Encode envelope of current frame. - */ - switch (stereoMode) { - case SBR_MONO: - sbrHeaderData->coupling = 0; - h_envChan[0]->encEnvData.balance = 0; - FDKsbrEnc_codeEnvelope (eData[0].sfb_nrg, eData[0].frame_info->freqRes, - &h_envChan[0]->sbrCodeEnvelope, - h_envChan[0]->encEnvData.domain_vec, - sbrHeaderData->coupling, - eData[0].frame_info->nEnvelopes, 0, - sbrBitstreamData->HeaderActive); - break; - case SBR_LEFT_RIGHT: - sbrHeaderData->coupling = 0; - - h_envChan[0]->encEnvData.balance = 0; - h_envChan[1]->encEnvData.balance = 0; - - - FDKsbrEnc_codeEnvelope (eData[0].sfb_nrg, eData[0].frame_info->freqRes, - &h_envChan[0]->sbrCodeEnvelope, - h_envChan[0]->encEnvData.domain_vec, - sbrHeaderData->coupling, - eData[0].frame_info->nEnvelopes, 0, - sbrBitstreamData->HeaderActive); - FDKsbrEnc_codeEnvelope (eData[1].sfb_nrg, eData[1].frame_info->freqRes, - &h_envChan[1]->sbrCodeEnvelope, - h_envChan[1]->encEnvData.domain_vec, - sbrHeaderData->coupling, - eData[1].frame_info->nEnvelopes, 0, - sbrBitstreamData->HeaderActive); - break; - case SBR_COUPLING: - sbrHeaderData->coupling = 1; - h_envChan[0]->encEnvData.balance = 0; - h_envChan[1]->encEnvData.balance = 1; - - FDKsbrEnc_codeEnvelope (eData[0].sfb_nrg, eData[0].frame_info->freqRes, - &h_envChan[0]->sbrCodeEnvelope, - h_envChan[0]->encEnvData.domain_vec, - sbrHeaderData->coupling, - eData[0].frame_info->nEnvelopes, 0, - sbrBitstreamData->HeaderActive); - FDKsbrEnc_codeEnvelope (eData[1].sfb_nrg, eData[1].frame_info->freqRes, - &h_envChan[1]->sbrCodeEnvelope, - h_envChan[1]->encEnvData.domain_vec, - sbrHeaderData->coupling, - eData[1].frame_info->nEnvelopes, 1, - sbrBitstreamData->HeaderActive); - break; - case SBR_SWITCH_LRC: - { - INT payloadbitsLR; - INT payloadbitsCOUPLING; - - SCHAR sfbNrgPrevTemp[MAX_NUM_CHANNELS][MAX_FREQ_COEFFS]; - SCHAR noisePrevTemp[MAX_NUM_CHANNELS][MAX_NUM_NOISE_COEFFS]; - INT upDateNrgTemp[MAX_NUM_CHANNELS]; - INT upDateNoiseTemp[MAX_NUM_CHANNELS]; - INT domainVecTemp[MAX_NUM_CHANNELS][MAX_ENVELOPES]; - INT domainVecNoiseTemp[MAX_NUM_CHANNELS][MAX_ENVELOPES]; - - INT tempFlagRight = 0; - INT tempFlagLeft = 0; - - /* - Store previous values, in order to be able to "undo" what is being done. - */ - - for(ch = 0; ch < nChannels;ch++){ - FDKmemcpy (sfbNrgPrevTemp[ch], h_envChan[ch]->sbrCodeEnvelope.sfb_nrg_prev, - MAX_FREQ_COEFFS * sizeof (SCHAR)); - - FDKmemcpy (noisePrevTemp[ch], h_envChan[ch]->sbrCodeNoiseFloor.sfb_nrg_prev, - MAX_NUM_NOISE_COEFFS * sizeof (SCHAR)); - - upDateNrgTemp[ch] = h_envChan[ch]->sbrCodeEnvelope.upDate; - upDateNoiseTemp[ch] = h_envChan[ch]->sbrCodeNoiseFloor.upDate; - - /* - forbid time coding in the first envelope in case of a different - previous stereomode - */ - if(sbrHeaderData->prev_coupling){ - h_envChan[ch]->sbrCodeEnvelope.upDate = 0; - h_envChan[ch]->sbrCodeNoiseFloor.upDate = 0; - } - } /* ch */ - - - /* - Code ordinary Left/Right stereo - */ - FDKsbrEnc_codeEnvelope (eData[0].sfb_nrg, eData[0].frame_info->freqRes, - &h_envChan[0]->sbrCodeEnvelope, - h_envChan[0]->encEnvData.domain_vec, 0, - eData[0].frame_info->nEnvelopes, 0, - sbrBitstreamData->HeaderActive); - FDKsbrEnc_codeEnvelope (eData[1].sfb_nrg, eData[1].frame_info->freqRes, - &h_envChan[1]->sbrCodeEnvelope, - h_envChan[1]->encEnvData.domain_vec, 0, - eData[1].frame_info->nEnvelopes, 0, - sbrBitstreamData->HeaderActive); - - c = 0; - for (i = 0; i < eData[0].nEnvelopes; i++) { - for (j = 0; j < h_envChan[0]->encEnvData.noScfBands[i]; j++) - { - h_envChan[0]->encEnvData.ienvelope[i][j] = eData[0].sfb_nrg[c]; - h_envChan[1]->encEnvData.ienvelope[i][j] = eData[1].sfb_nrg[c]; - c++; - } - } - - - - FDKsbrEnc_codeEnvelope (eData[0].noise_level, fData->res, - &h_envChan[0]->sbrCodeNoiseFloor, - h_envChan[0]->encEnvData.domain_vec_noise, 0, - (eData[0].frame_info->nEnvelopes > 1 ? 2 : 1), 0, - sbrBitstreamData->HeaderActive); - - - for (i = 0; i < MAX_NUM_NOISE_VALUES; i++) - h_envChan[0]->encEnvData.sbr_noise_levels[i] = eData[0].noise_level[i]; - - - FDKsbrEnc_codeEnvelope (eData[1].noise_level, fData->res, - &h_envChan[1]->sbrCodeNoiseFloor, - h_envChan[1]->encEnvData.domain_vec_noise, 0, - (eData[1].frame_info->nEnvelopes > 1 ? 2 : 1), 0, - sbrBitstreamData->HeaderActive); - - for (i = 0; i < MAX_NUM_NOISE_VALUES; i++) - h_envChan[1]->encEnvData.sbr_noise_levels[i] = eData[1].noise_level[i]; - - - sbrHeaderData->coupling = 0; - h_envChan[0]->encEnvData.balance = 0; - h_envChan[1]->encEnvData.balance = 0; - - payloadbitsLR = FDKsbrEnc_CountSbrChannelPairElement (sbrHeaderData, - hParametricStereo, - sbrBitstreamData, - &h_envChan[0]->encEnvData, - &h_envChan[1]->encEnvData, - hCmonData, - h_con->sbrSyntaxFlags); - - /* - swap saved stored with current values - */ - for(ch = 0; ch < nChannels;ch++){ - INT itmp; - for(i=0;isbrCodeEnvelope.sfb_nrg_prev[i]; - h_envChan[ch]->sbrCodeEnvelope.sfb_nrg_prev[i]=sfbNrgPrevTemp[ch][i]; - sfbNrgPrevTemp[ch][i]=itmp; - } - for(i=0;isbrCodeNoiseFloor.sfb_nrg_prev[i]; - h_envChan[ch]->sbrCodeNoiseFloor.sfb_nrg_prev[i]=noisePrevTemp[ch][i]; - noisePrevTemp[ch][i]=itmp; - } - /* swap update flags */ - itmp = h_envChan[ch]->sbrCodeEnvelope.upDate; - h_envChan[ch]->sbrCodeEnvelope.upDate=upDateNrgTemp[ch]; - upDateNrgTemp[ch] = itmp; - - itmp = h_envChan[ch]->sbrCodeNoiseFloor.upDate; - h_envChan[ch]->sbrCodeNoiseFloor.upDate=upDateNoiseTemp[ch]; - upDateNoiseTemp[ch]=itmp; - - /* - save domain vecs - */ - FDKmemcpy(domainVecTemp[ch],h_envChan[ch]->encEnvData.domain_vec,sizeof(INT)*MAX_ENVELOPES); - FDKmemcpy(domainVecNoiseTemp[ch],h_envChan[ch]->encEnvData.domain_vec_noise,sizeof(INT)*MAX_ENVELOPES); - - /* - forbid time coding in the first envelope in case of a different - previous stereomode - */ - - if(!sbrHeaderData->prev_coupling){ - h_envChan[ch]->sbrCodeEnvelope.upDate = 0; - h_envChan[ch]->sbrCodeNoiseFloor.upDate = 0; - } - } /* ch */ - - - /* - Coupling - */ - - FDKsbrEnc_codeEnvelope (eData[0].sfb_nrg_coupling, eData[0].frame_info->freqRes, - &h_envChan[0]->sbrCodeEnvelope, - h_envChan[0]->encEnvData.domain_vec, 1, - eData[0].frame_info->nEnvelopes, 0, - sbrBitstreamData->HeaderActive); - - FDKsbrEnc_codeEnvelope (eData[1].sfb_nrg_coupling, eData[1].frame_info->freqRes, - &h_envChan[1]->sbrCodeEnvelope, - h_envChan[1]->encEnvData.domain_vec, 1, - eData[1].frame_info->nEnvelopes, 1, - sbrBitstreamData->HeaderActive); - - - c = 0; - for (i = 0; i < eData[0].nEnvelopes; i++) { - for (j = 0; j < h_envChan[0]->encEnvData.noScfBands[i]; j++) { - h_envChan[0]->encEnvData.ienvelope[i][j] = eData[0].sfb_nrg_coupling[c]; - h_envChan[1]->encEnvData.ienvelope[i][j] = eData[1].sfb_nrg_coupling[c]; - c++; - } - } - - FDKsbrEnc_codeEnvelope (eData[0].noise_level_coupling, fData->res, - &h_envChan[0]->sbrCodeNoiseFloor, - h_envChan[0]->encEnvData.domain_vec_noise, 1, - (eData[0].frame_info->nEnvelopes > 1 ? 2 : 1), 0, - sbrBitstreamData->HeaderActive); - - for (i = 0; i < MAX_NUM_NOISE_VALUES; i++) - h_envChan[0]->encEnvData.sbr_noise_levels[i] = eData[0].noise_level_coupling[i]; - - - FDKsbrEnc_codeEnvelope (eData[1].noise_level_coupling, fData->res, - &h_envChan[1]->sbrCodeNoiseFloor, - h_envChan[1]->encEnvData.domain_vec_noise, 1, - (eData[1].frame_info->nEnvelopes > 1 ? 2 : 1), 1, - sbrBitstreamData->HeaderActive); - - for (i = 0; i < MAX_NUM_NOISE_VALUES; i++) - h_envChan[1]->encEnvData.sbr_noise_levels[i] = eData[1].noise_level_coupling[i]; - - sbrHeaderData->coupling = 1; - - h_envChan[0]->encEnvData.balance = 0; - h_envChan[1]->encEnvData.balance = 1; - - tempFlagLeft = h_envChan[0]->encEnvData.addHarmonicFlag; - tempFlagRight = h_envChan[1]->encEnvData.addHarmonicFlag; - - payloadbitsCOUPLING = - FDKsbrEnc_CountSbrChannelPairElement (sbrHeaderData, - hParametricStereo, - sbrBitstreamData, - &h_envChan[0]->encEnvData, - &h_envChan[1]->encEnvData, - hCmonData, - h_con->sbrSyntaxFlags); - - - h_envChan[0]->encEnvData.addHarmonicFlag = tempFlagLeft; - h_envChan[1]->encEnvData.addHarmonicFlag = tempFlagRight; - - if (payloadbitsCOUPLING < payloadbitsLR) { - - /* - copy coded coupling envelope and noise data to l/r - */ - for(ch = 0; ch < nChannels;ch++){ - SBR_ENV_TEMP_DATA *ed = &eData[ch]; - FDKmemcpy (ed->sfb_nrg, ed->sfb_nrg_coupling, - MAX_NUM_ENVELOPE_VALUES * sizeof (SCHAR)); - FDKmemcpy (ed->noise_level, ed->noise_level_coupling, - MAX_NUM_NOISE_VALUES * sizeof (SCHAR)); - } - - sbrHeaderData->coupling = 1; - h_envChan[0]->encEnvData.balance = 0; - h_envChan[1]->encEnvData.balance = 1; - } - else{ - /* - restore saved l/r items - */ - for(ch = 0; ch < nChannels;ch++){ - - FDKmemcpy (h_envChan[ch]->sbrCodeEnvelope.sfb_nrg_prev, - sfbNrgPrevTemp[ch], MAX_FREQ_COEFFS * sizeof (SCHAR)); - - h_envChan[ch]->sbrCodeEnvelope.upDate = upDateNrgTemp[ch]; - - FDKmemcpy (h_envChan[ch]->sbrCodeNoiseFloor.sfb_nrg_prev, - noisePrevTemp[ch], MAX_NUM_NOISE_COEFFS * sizeof (SCHAR)); - - FDKmemcpy (h_envChan[ch]->encEnvData.domain_vec,domainVecTemp[ch],sizeof(INT)*MAX_ENVELOPES); - FDKmemcpy (h_envChan[ch]->encEnvData.domain_vec_noise,domainVecNoiseTemp[ch],sizeof(INT)*MAX_ENVELOPES); - - h_envChan[ch]->sbrCodeNoiseFloor.upDate = upDateNoiseTemp[ch]; - } - - sbrHeaderData->coupling = 0; - h_envChan[0]->encEnvData.balance = 0; - h_envChan[1]->encEnvData.balance = 0; - } - } - break; - } /* switch */ - - - /* tell the envelope encoders how long it has been, since we last sent - a frame starting with a dF-coded envelope */ - if (stereoMode == SBR_MONO ) { - if (h_envChan[0]->encEnvData.domain_vec[0] == TIME) - h_envChan[0]->sbrCodeEnvelope.dF_edge_incr_fac++; - else - h_envChan[0]->sbrCodeEnvelope.dF_edge_incr_fac = 0; - } - else { - if (h_envChan[0]->encEnvData.domain_vec[0] == TIME || - h_envChan[1]->encEnvData.domain_vec[0] == TIME) { - h_envChan[0]->sbrCodeEnvelope.dF_edge_incr_fac++; - h_envChan[1]->sbrCodeEnvelope.dF_edge_incr_fac++; - } - else { - h_envChan[0]->sbrCodeEnvelope.dF_edge_incr_fac = 0; - h_envChan[1]->sbrCodeEnvelope.dF_edge_incr_fac = 0; - } - } - - /* - Send the encoded data to the bitstream - */ - for(ch = 0; ch < nChannels;ch++){ - SBR_ENV_TEMP_DATA *ed = &eData[ch]; - c = 0; - for (i = 0; i < ed->nEnvelopes; i++) { - for (j = 0; j < h_envChan[ch]->encEnvData.noScfBands[i]; j++) { - h_envChan[ch]->encEnvData.ienvelope[i][j] = ed->sfb_nrg[c]; - - c++; - } - } - for (i = 0; i < MAX_NUM_NOISE_VALUES; i++){ - h_envChan[ch]->encEnvData.sbr_noise_levels[i] = ed->noise_level[i]; - } - }/* ch */ - - - /* - Write bitstream - */ - if (nChannels == 2) { - FDKsbrEnc_WriteEnvChannelPairElement(sbrHeaderData, - hParametricStereo, - sbrBitstreamData, - &h_envChan[0]->encEnvData, - &h_envChan[1]->encEnvData, - hCmonData, - h_con->sbrSyntaxFlags); - } - else { - FDKsbrEnc_WriteEnvSingleChannelElement(sbrHeaderData, - hParametricStereo, - sbrBitstreamData, - &h_envChan[0]->encEnvData, - hCmonData, - h_con->sbrSyntaxFlags); - } - - /* - * Update buffers. - */ - for (ch=0; chsbrExtractEnvelope.no_cols >> h_envChan[ch]->sbrExtractEnvelope.YBufferSzShift; - for (i = 0; i < h_envChan[ch]->sbrExtractEnvelope.YBufferWriteOffset; i++) { - FDKmemcpy(h_envChan[ch]->sbrExtractEnvelope.YBuffer[i], - h_envChan[ch]->sbrExtractEnvelope.YBuffer[i + YBufferLength], - sizeof(FIXP_DBL)*QMF_CHANNELS); - } - h_envChan[ch]->sbrExtractEnvelope.YBufferScale[0] = h_envChan[ch]->sbrExtractEnvelope.YBufferScale[1]; - } - - sbrHeaderData->prev_coupling = sbrHeaderData->coupling; -} - -/***************************************************************************/ -/*! - - \brief creates an envelope extractor handle - - \return error status - -****************************************************************************/ -INT -FDKsbrEnc_CreateExtractSbrEnvelope (HANDLE_SBR_EXTRACT_ENVELOPE hSbrCut, - INT channel - ,INT chInEl - ,UCHAR* dynamic_RAM - ) -{ - INT i; - FIXP_DBL* YBuffer = GetRam_Sbr_envYBuffer(channel); - - FDKmemclear(hSbrCut,sizeof(SBR_EXTRACT_ENVELOPE)); - hSbrCut->p_YBuffer = YBuffer; - - - for (i = 0; i < (QMF_MAX_TIME_SLOTS>>1); i++) { - hSbrCut->YBuffer[i] = YBuffer + (i*QMF_CHANNELS); - } - FIXP_DBL *YBufferDyn = GetRam_Sbr_envYBuffer(chInEl, dynamic_RAM); - INT n=0; - for (; i < QMF_MAX_TIME_SLOTS; i++,n++) { - hSbrCut->YBuffer[i] = YBufferDyn + (n*QMF_CHANNELS); - } - - FIXP_DBL* rBuffer = GetRam_Sbr_envRBuffer(0, dynamic_RAM); - FIXP_DBL* iBuffer = GetRam_Sbr_envIBuffer(0, dynamic_RAM); - - for (i = 0; i < QMF_MAX_TIME_SLOTS; i++) { - hSbrCut->rBuffer[i] = rBuffer + (i*QMF_CHANNELS); - hSbrCut->iBuffer[i] = iBuffer + (i*QMF_CHANNELS); - } - - return 0; -} - - -/***************************************************************************/ -/*! - - \brief Initialize an envelope extractor instance. - - \return error status - -****************************************************************************/ -INT -FDKsbrEnc_InitExtractSbrEnvelope (HANDLE_SBR_EXTRACT_ENVELOPE hSbrCut, - int no_cols, - int no_rows, - int start_index, - int time_slots, - int time_step, - int tran_off, - ULONG statesInitFlag - ,int chInEl - ,UCHAR* dynamic_RAM - ,UINT sbrSyntaxFlags - ) -{ - int YBufferLength, rBufferLength; - int i; - - if (sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) { - int off = TRANSIENT_OFFSET_LD; -#ifndef FULL_DELAY - hSbrCut->YBufferWriteOffset = (no_cols>>1)+off*time_step; -#else - hSbrCut->YBufferWriteOffset = no_cols+off*time_step; -#endif - } else - { - hSbrCut->YBufferWriteOffset = tran_off*time_step; - } - hSbrCut->rBufferReadOffset = 0; - - - YBufferLength = hSbrCut->YBufferWriteOffset + no_cols; - rBufferLength = no_cols; - - hSbrCut->pre_transient_info[0] = 0; - hSbrCut->pre_transient_info[1] = 0; - - - hSbrCut->no_cols = no_cols; - hSbrCut->no_rows = no_rows; - hSbrCut->start_index = start_index; - - hSbrCut->time_slots = time_slots; - hSbrCut->time_step = time_step; - - FDK_ASSERT(no_rows <= QMF_CHANNELS); - - /* Use half the Energy values if time step is 2 or greater */ - if (time_step >= 2) - hSbrCut->YBufferSzShift = 1; - else - hSbrCut->YBufferSzShift = 0; - - YBufferLength >>= hSbrCut->YBufferSzShift; - hSbrCut->YBufferWriteOffset >>= hSbrCut->YBufferSzShift; - - FDK_ASSERT(YBufferLength<=QMF_MAX_TIME_SLOTS); - - FIXP_DBL *YBufferDyn = GetRam_Sbr_envYBuffer(chInEl, dynamic_RAM); - INT n=0; - for (i=(QMF_MAX_TIME_SLOTS>>1); i < QMF_MAX_TIME_SLOTS; i++,n++) { - hSbrCut->YBuffer[i] = YBufferDyn + (n*QMF_CHANNELS); - } - - if(statesInitFlag) { - for (i=0; iYBuffer[i],QMF_CHANNELS*sizeof(FIXP_DBL)); - } - } - - for (i = 0; i < rBufferLength; i++) { - FDKmemclear( hSbrCut->rBuffer[i],QMF_CHANNELS*sizeof(FIXP_DBL)); - FDKmemclear( hSbrCut->iBuffer[i],QMF_CHANNELS*sizeof(FIXP_DBL)); - } - - FDKmemclear (hSbrCut->envelopeCompensation,sizeof(UCHAR)*MAX_FREQ_COEFFS); - - if(statesInitFlag) { - hSbrCut->YBufferScale[0] = hSbrCut->YBufferScale[1] = FRACT_BITS-1; - } - - return (0); -} - - - - -/***************************************************************************/ -/*! - - \brief deinitializes an envelope extractor handle - - \return void - -****************************************************************************/ - -void -FDKsbrEnc_deleteExtractSbrEnvelope (HANDLE_SBR_EXTRACT_ENVELOPE hSbrCut) -{ - - if (hSbrCut) { - FreeRam_Sbr_envYBuffer(&hSbrCut->p_YBuffer); - } -} - -INT -FDKsbrEnc_GetEnvEstDelay(HANDLE_SBR_EXTRACT_ENVELOPE hSbr) -{ - return hSbr->no_rows*((hSbr->YBufferWriteOffset)*2 /* mult 2 because nrg's are grouped half */ - - hSbr->rBufferReadOffset ); /* in reference hold half spec and calc nrg's on overlapped spec */ - -} - - - - diff --git a/libSBRenc/src/env_est.h b/libSBRenc/src/env_est.h deleted file mode 100644 index e17a974..0000000 --- a/libSBRenc/src/env_est.h +++ /dev/null @@ -1,225 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief Envelope estimation structs and prototypes -*/ -#ifndef __ENV_EST_H -#define __ENV_EST_H - -#include "sbr_def.h" -#include "sbr_encoder.h" /* SBR econfig structs */ -#include "ps_main.h" -#include "bit_sbr.h" -#include "fram_gen.h" -#include "tran_det.h" -#include "code_env.h" -#include "ton_corr.h" - -typedef struct -{ - FIXP_DBL *rBuffer[QMF_MAX_TIME_SLOTS]; - FIXP_DBL *iBuffer[QMF_MAX_TIME_SLOTS]; - - FIXP_DBL *p_YBuffer; - - FIXP_DBL *YBuffer[QMF_MAX_TIME_SLOTS]; - int YBufferScale[2]; - - UCHAR envelopeCompensation[MAX_FREQ_COEFFS]; - UCHAR pre_transient_info[2]; - - - int YBufferWriteOffset; - int YBufferSzShift; - int rBufferReadOffset; - - int no_cols; - int no_rows; - int start_index; - - int time_slots; - int time_step; -} -SBR_EXTRACT_ENVELOPE; -typedef SBR_EXTRACT_ENVELOPE *HANDLE_SBR_EXTRACT_ENVELOPE; - -struct ENV_CHANNEL -{ - FAST_TRAN_DETECTOR sbrFastTransientDetector; - SBR_TRANSIENT_DETECTOR sbrTransientDetector; - SBR_CODE_ENVELOPE sbrCodeEnvelope; - SBR_CODE_ENVELOPE sbrCodeNoiseFloor; - SBR_EXTRACT_ENVELOPE sbrExtractEnvelope; - - - SBR_ENVELOPE_FRAME SbrEnvFrame; - SBR_TON_CORR_EST TonCorr; - - struct SBR_ENV_DATA encEnvData; - - int qmfScale; - UCHAR fLevelProtect; -}; -typedef struct ENV_CHANNEL *HANDLE_ENV_CHANNEL; - -/************ Function Declarations ***************/ - -INT -FDKsbrEnc_CreateExtractSbrEnvelope (HANDLE_SBR_EXTRACT_ENVELOPE hSbrCut, - INT channel - ,INT chInEl - ,UCHAR* dynamic_RAM - ); - - -INT -FDKsbrEnc_InitExtractSbrEnvelope ( - HANDLE_SBR_EXTRACT_ENVELOPE hSbr, - int no_cols, - int no_rows, - int start_index, - int time_slots, int time_step, int tran_off, - ULONG statesInitFlag - ,int chInEl - ,UCHAR* dynamic_RAM - ,UINT sbrSyntaxFlags - ); - -void FDKsbrEnc_deleteExtractSbrEnvelope (HANDLE_SBR_EXTRACT_ENVELOPE hSbrCut); - -typedef struct { - FREQ_RES res[MAX_NUM_NOISE_VALUES]; - int maxQuantError; - -} SBR_FRAME_TEMP_DATA; - -typedef struct { - const SBR_FRAME_INFO *frame_info; - FIXP_DBL noiseFloor[MAX_NUM_NOISE_VALUES]; - SCHAR sfb_nrg_coupling[MAX_NUM_ENVELOPE_VALUES]; /* only used if stereomode = SWITCH_L_R_C */ - SCHAR sfb_nrg[MAX_NUM_ENVELOPE_VALUES]; - SCHAR noise_level_coupling[MAX_NUM_NOISE_VALUES]; /* only used if stereomode = SWITCH_L_R_C */ - SCHAR noise_level[MAX_NUM_NOISE_VALUES]; - UCHAR transient_info[3]; - UCHAR nEnvelopes; -} SBR_ENV_TEMP_DATA; - -/* - * Extract features from QMF data. Afterwards, the QMF data is not required anymore. - */ -void -FDKsbrEnc_extractSbrEnvelope1( - HANDLE_SBR_CONFIG_DATA h_con, - HANDLE_SBR_HEADER_DATA sbrHeaderData, - HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, - HANDLE_ENV_CHANNEL h_envChan, - HANDLE_COMMON_DATA cmonData, - SBR_ENV_TEMP_DATA *eData, - SBR_FRAME_TEMP_DATA *fData - ); - - -/* - * Process the previously features extracted by FDKsbrEnc_extractSbrEnvelope1 - * and create/encode SBR envelopes. - */ -void -FDKsbrEnc_extractSbrEnvelope2( - HANDLE_SBR_CONFIG_DATA h_con, - HANDLE_SBR_HEADER_DATA sbrHeaderData, - HANDLE_PARAMETRIC_STEREO hParametricStereo, - HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, - HANDLE_ENV_CHANNEL sbrEnvChannel0, - HANDLE_ENV_CHANNEL sbrEnvChannel1, - HANDLE_COMMON_DATA cmonData, - SBR_ENV_TEMP_DATA *eData, - SBR_FRAME_TEMP_DATA *fData, - int clearOutput - ); - -INT -FDKsbrEnc_GetEnvEstDelay(HANDLE_SBR_EXTRACT_ENVELOPE hSbr); - -#endif diff --git a/libSBRenc/src/fram_gen.cpp b/libSBRenc/src/fram_gen.cpp deleted file mode 100644 index 9a35111..0000000 --- a/libSBRenc/src/fram_gen.cpp +++ /dev/null @@ -1,2065 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -#include "fram_gen.h" -#include "sbr_misc.h" - -#include "genericStds.h" - -static const SBR_FRAME_INFO frameInfo1_2048 = { - 1, - { 0, 16}, - {FREQ_RES_HIGH}, - 0, - 1, - {0, 16} }; - -static const SBR_FRAME_INFO frameInfo2_2048 = { - 2, - { 0, 8, 16}, - {FREQ_RES_HIGH, FREQ_RES_HIGH}, - 0, - 2, - { 0, 8, 16} }; - -static const SBR_FRAME_INFO frameInfo4_2048 = { - 4, - { 0, 4, 8, 12, 16}, - {FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH}, - 0, - 2, - { 0, 8, 16} }; - -static const SBR_FRAME_INFO frameInfo1_2304 = { - 1, - { 0, 18}, - {FREQ_RES_HIGH}, - 0, - 1, - { 0, 18} }; - -static const SBR_FRAME_INFO frameInfo2_2304 = { - 2, - { 0, 9, 18}, - {FREQ_RES_HIGH, FREQ_RES_HIGH}, - 0, - 2, - { 0, 9, 18} }; - -static const SBR_FRAME_INFO frameInfo4_2304 = { - 4, - { 0, 5, 9, 14, 18}, - {FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH}, - 0, - 2, - { 0, 9, 18} }; - -static const SBR_FRAME_INFO frameInfo1_1920 = { - 1, - { 0, 15}, - {FREQ_RES_HIGH}, - 0, - 1, - { 0, 15} }; - -static const SBR_FRAME_INFO frameInfo2_1920 = { - 2, - { 0, 8, 15}, - {FREQ_RES_HIGH, FREQ_RES_HIGH}, - 0, - 2, - { 0, 8, 15} }; - -static const SBR_FRAME_INFO frameInfo4_1920 = { - 4, - { 0, 4, 8, 12, 15}, - {FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH}, - 0, - 2, - { 0, 8, 15} }; - -static const SBR_FRAME_INFO frameInfo1_1152 = { - 1, - { 0, 9}, - {FREQ_RES_HIGH}, - 0, - 1, - { 0, 9} }; - -static const SBR_FRAME_INFO frameInfo2_1152 = { - 2, - { 0, 5, 9}, - {FREQ_RES_HIGH, FREQ_RES_HIGH}, - 0, - 2, - { 0, 5, 9} }; - -static const SBR_FRAME_INFO frameInfo4_1152 = { - 4, - { 0, 2, 5, - 7, 9}, - {FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH}, - 0, - 2, - { 0, 5, 9} }; - - -/* AACLD frame info */ -static const SBR_FRAME_INFO frameInfo1_512LD = { - 1, - {0, 8}, - {FREQ_RES_HIGH}, - 0, - 1, - {0, 8}}; - -static const SBR_FRAME_INFO frameInfo2_512LD = { - 2, - {0, 4, 8}, - {FREQ_RES_HIGH, FREQ_RES_HIGH}, - 0, - 2, - {0, 4, 8}}; - -static const SBR_FRAME_INFO frameInfo4_512LD = { - 4, - {0, 2, 4, 6, 8}, - {FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH}, - 0, - 2, - {0, 4, 8}}; - -static int -calcFillLengthMax (int tranPos, /*!< input : transient position (ref: tran det) */ - int numberTimeSlots /*!< input : number of timeslots */ - ); - -static void -fillFrameTran (const int *v_tuningSegm, /*!< tuning: desired segment lengths */ - const int *v_tuningFreq, /*!< tuning: desired frequency resolutions */ - int tran, /*!< input : position of transient */ - int *v_bord, /*!< memNew: borders */ - int *length_v_bord, /*!< memNew: # borders */ - int *v_freq, /*!< memNew: frequency resolutions */ - int *length_v_freq, /*!< memNew: # frequency resolutions */ - int *bmin, /*!< hlpNew: first mandatory border */ - int *bmax /*!< hlpNew: last mandatory border */ - ); - -static void fillFramePre (INT dmax, INT *v_bord, INT *length_v_bord, - INT *v_freq, INT *length_v_freq, INT bmin, - INT rest); - -static void fillFramePost (INT *parts, INT *d, INT dmax, INT *v_bord, - INT *length_v_bord, INT *v_freq, - INT *length_v_freq, INT bmax, - INT bufferFrameStart, INT numberTimeSlots, INT fmax); - -static void fillFrameInter (INT *nL, const int *v_tuningSegm, INT *v_bord, - INT *length_v_bord, INT bmin, INT *v_freq, - INT *length_v_freq, INT *v_bordFollow, - INT *length_v_bordFollow, INT *v_freqFollow, - INT *length_v_freqFollow, INT i_fillFollow, - INT dmin, INT dmax, INT numberTimeSlots); - -static void calcFrameClass (FRAME_CLASS *frameClass, FRAME_CLASS *frameClassOld, INT tranFlag, - INT *spreadFlag); - -static void specialCase (INT *spreadFlag, INT allowSpread, INT *v_bord, - INT *length_v_bord, INT *v_freq, INT *length_v_freq, - INT *parts, INT d); - -static void calcCmonBorder (INT *i_cmon, INT *i_tran, INT *v_bord, - INT *length_v_bord, INT tran, - INT bufferFrameStart, INT numberTimeSlots); - -static void keepForFollowUp (INT *v_bordFollow, INT *length_v_bordFollow, - INT *v_freqFollow, INT *length_v_freqFollow, - INT *i_tranFollow, INT *i_fillFollow, - INT *v_bord, INT *length_v_bord, INT *v_freq, - INT i_cmon, INT i_tran, INT parts, INT numberTimeSlots); - -static void calcCtrlSignal (HANDLE_SBR_GRID hSbrGrid, FRAME_CLASS frameClass, - INT *v_bord, INT length_v_bord, INT *v_freq, - INT length_v_freq, INT i_cmon, INT i_tran, - INT spreadFlag, INT nL); - -static void ctrlSignal2FrameInfo (HANDLE_SBR_GRID hSbrGrid, - HANDLE_SBR_FRAME_INFO hFrameInfo, - FREQ_RES *freq_res_fixfix); - - -/* table for 8 time slot index */ -static const int envelopeTable_8 [8][5] = { -/* transientIndex nEnv, tranIdx, shortEnv, border1, border2, ... */ -/* borders from left to right side; -1 = not in use */ - /*[|T-|------]*/ { 2, 0, 0, 1, -1 }, - /*[|-T-|-----]*/ { 2, 0, 0, 2, -1 }, - /*[--|T-|----]*/ { 3, 1, 1, 2, 4 }, - /*[---|T-|---]*/ { 3, 1, 1, 3, 5 }, - /*[----|T-|--]*/ { 3, 1, 1, 4, 6 }, - /*[-----|T--|]*/ { 2, 1, 1, 5, -1 }, - /*[------|T-|]*/ { 2, 1, 1, 6, -1 }, - /*[-------|T|]*/ { 2, 1, 1, 7, -1 }, -}; - -/* table for 16 time slot index */ -static const int envelopeTable_16 [16][6] = { - /* transientIndex nEnv, tranIdx, shortEnv, border1, border2, ... */ - /* length from left to right side; -1 = not in use */ - /*[|T---|------------|]*/ { 2, 0, 0, 4, -1, -1}, - /*[|-T---|-----------|]*/ { 2, 0, 0, 5, -1, -1}, - /*[|--|T---|----------]*/ { 3, 1, 1, 2, 6, -1}, - /*[|---|T---|---------]*/ { 3, 1, 1, 3, 7, -1}, - /*[|----|T---|--------]*/ { 3, 1, 1, 4, 8, -1}, - /*[|-----|T---|-------]*/ { 3, 1, 1, 5, 9, -1}, - /*[|------|T---|------]*/ { 3, 1, 1, 6, 10, -1}, - /*[|-------|T---|-----]*/ { 3, 1, 1, 7, 11, -1}, - /*[|--------|T---|----]*/ { 3, 1, 1, 8, 12, -1}, - /*[|---------|T---|---]*/ { 3, 1, 1, 9, 13, -1}, - /*[|----------|T---|--]*/ { 3, 1, 1,10, 14, -1}, - /*[|-----------|T----|]*/ { 2, 1, 1,11, -1, -1}, - /*[|------------|T---|]*/ { 2, 1, 1,12, -1, -1}, - /*[|-------------|T--|]*/ { 2, 1, 1,13, -1, -1}, - /*[|--------------|T-|]*/ { 2, 1, 1,14, -1, -1}, - /*[|---------------|T|]*/ { 2, 1, 1,15, -1, -1}, -}; - -/* table for 15 time slot index */ -static const int envelopeTable_15 [15][6] = { - /* transientIndex nEnv, tranIdx, shortEnv, border1, border2, ... */ - /* length from left to right side; -1 = not in use */ - /*[|T---|------------]*/ { 2, 0, 0, 4, -1, -1}, - /*[|-T---|-----------]*/ { 2, 0, 0, 5, -1, -1}, - /*[|--|T---|---------]*/ { 3, 1, 1, 2, 6, -1}, - /*[|---|T---|--------]*/ { 3, 1, 1, 3, 7, -1}, - /*[|----|T---|-------]*/ { 3, 1, 1, 4, 8, -1}, - /*[|-----|T---|------]*/ { 3, 1, 1, 5, 9, -1}, - /*[|------|T---|-----]*/ { 3, 1, 1, 6, 10, -1}, - /*[|-------|T---|----]*/ { 3, 1, 1, 7, 11, -1}, - /*[|--------|T---|---]*/ { 3, 1, 1, 8, 12, -1}, - /*[|---------|T---|--]*/ { 3, 1, 1, 9, 13, -1}, - /*[|----------|T----|]*/ { 2, 1, 1,10, -1, -1}, - /*[|-----------|T---|]*/ { 2, 1, 1,11, -1, -1}, - /*[|------------|T--|]*/ { 2, 1, 1,12, -1, -1}, - /*[|-------------|T-|]*/ { 2, 1, 1,13, -1, -1}, - /*[|--------------|T|]*/ { 2, 1, 1,14, -1, -1}, -}; - -static const int minFrameTranDistance = 4; - -static const FREQ_RES freqRes_table_8[] = {FREQ_RES_LOW, FREQ_RES_LOW, FREQ_RES_LOW, FREQ_RES_LOW, FREQ_RES_LOW, - FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH}; - -static const FREQ_RES freqRes_table_16[16] = { - /* size of envelope */ -/* 0-4 */ FREQ_RES_LOW, FREQ_RES_LOW, FREQ_RES_LOW, FREQ_RES_LOW, FREQ_RES_LOW, -/* 5-9 */ FREQ_RES_LOW, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, -/* 10-16 */ FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, - FREQ_RES_HIGH }; - -static void generateFixFixOnly ( HANDLE_SBR_FRAME_INFO hSbrFrameInfo, - HANDLE_SBR_GRID hSbrGrid, - int tranPosInternal, - int numberTimeSlots, - UCHAR fResTransIsLow - ); - - -/*! - Functionname: FDKsbrEnc_frameInfoGenerator - - Description: produces the FRAME_INFO struct for the current frame - - Arguments: hSbrEnvFrame - pointer to sbr envelope handle - v_pre_transient_info - pointer to transient info vector - v_transient_info - pointer to previous transient info vector - v_tuning - pointer to tuning vector - - Return: frame_info - pointer to SBR_FRAME_INFO struct - -*******************************************************************************/ -HANDLE_SBR_FRAME_INFO -FDKsbrEnc_frameInfoGenerator (HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame, - UCHAR *v_transient_info, - UCHAR *v_transient_info_pre, - int ldGrid, - const int *v_tuning) -{ - INT numEnv, tranPosInternal=0, bmin=0, bmax=0, parts, d, i_cmon=0, i_tran=0, nL; - INT fmax = 0; - - INT *v_bord = hSbrEnvFrame->v_bord; - INT *v_freq = hSbrEnvFrame->v_freq; - INT *v_bordFollow = hSbrEnvFrame->v_bordFollow; - INT *v_freqFollow = hSbrEnvFrame->v_freqFollow; - - - INT *length_v_bordFollow = &hSbrEnvFrame->length_v_bordFollow; - INT *length_v_freqFollow = &hSbrEnvFrame->length_v_freqFollow; - INT *length_v_bord = &hSbrEnvFrame->length_v_bord; - INT *length_v_freq = &hSbrEnvFrame->length_v_freq; - INT *spreadFlag = &hSbrEnvFrame->spreadFlag; - INT *i_tranFollow = &hSbrEnvFrame->i_tranFollow; - INT *i_fillFollow = &hSbrEnvFrame->i_fillFollow; - FRAME_CLASS *frameClassOld = &hSbrEnvFrame->frameClassOld; - FRAME_CLASS frameClass = FIXFIX; - - - INT allowSpread = hSbrEnvFrame->allowSpread; - INT numEnvStatic = hSbrEnvFrame->numEnvStatic; - INT staticFraming = hSbrEnvFrame->staticFraming; - INT dmin = hSbrEnvFrame->dmin; - INT dmax = hSbrEnvFrame->dmax; - - INT bufferFrameStart = hSbrEnvFrame->SbrGrid.bufferFrameStart; - INT numberTimeSlots = hSbrEnvFrame->SbrGrid.numberTimeSlots; - INT frameMiddleSlot = hSbrEnvFrame->frameMiddleSlot; - - INT tranPos = v_transient_info[0]; - INT tranFlag = v_transient_info[1]; - - const int *v_tuningSegm = v_tuning; - const int *v_tuningFreq = v_tuning + 3; - - hSbrEnvFrame->v_tuningSegm = v_tuningSegm; - - if (ldGrid) { - /* in case there was a transient at the very end of the previous frame, start with a transient envelope */ - if ( !tranFlag && v_transient_info_pre[1] && (numberTimeSlots - v_transient_info_pre[0] < minFrameTranDistance) ){ - tranFlag = 1; - tranPos = 0; - } - } - - /* - * Synopsis: - * - * The frame generator creates the time-/frequency-grid for one SBR frame. - * Input signals are provided by the transient detector and the frame - * splitter (transientDetectNew() & FrameSplitter() in tran_det.c). The - * framing is controlled by adjusting tuning parameters stored in - * FRAME_GEN_TUNING. The parameter values are dependent on frame lengths - * and bitrates, and may in the future be signal dependent. - * - * The envelope borders are stored for frame generator internal use in - * aBorders. The contents of aBorders represent positions along the time - * axis given in the figures in fram_gen.h (the "frame-generator" rows). - * The unit is "time slot". The figures in fram_gen.h also define the - * detection ranges for the transient detector. For every border in - * aBorders, there is a corresponding entry in aFreqRes, which defines the - * frequency resolution of the envelope following (delimited by) the - * border. - * - * When no transients are present, FIXFIX class frames are used. The - * frame splitter decides whether to use one or two envelopes in the - * FIXFIX frame. "Sparse transients" (separated by a few frames without - * transients) are handeled by [FIXVAR, VARFIX] pairs or (depending on - * tuning and transient position relative the nominal frame boundaries) - * by [FIXVAR, VARVAR, VARFIX] triples. "Tight transients" (in - * consecutive frames) are handeled by [..., VARVAR, VARVAR, ...] - * sequences. - * - * The generator assumes that transients are "sparse", and designs - * borders for [FIXVAR, VARFIX] pairs right away, where the first frame - * corresponds to the present frame. At the next call of the generator - * it is known whether the transient actually is "sparse" or not. If - * 'yes', the already calculated VARFIX borders are used. If 'no', new - * borders, meeting the requirements of the "tight" transient, are - * calculated. - * - * The generator produces two outputs: A "clear-text bitstream" stored in - * SBR_GRID, and a straight-forward representation of the grid stored in - * SBR_FRAME_INFO. The former is subsequently converted to the actual - * bitstream sbr_grid() (encodeSbrGrid() in bit_sbr.c). The latter is - * used by other encoder functions, such as the envelope estimator - * (calculateSbrEnvelope() in env_est.c) and the noise floor and missing - * harmonics detector (TonCorrParamExtr() in nf_est.c). - */ - - if (staticFraming) { - /*-------------------------------------------------------------------------- - Ignore transient detector - ---------------------------------------------------------------------------*/ - - frameClass = FIXFIX; - numEnv = numEnvStatic; /* {1,2,4,8} */ - *frameClassOld = FIXFIX; /* for change to dyn */ - hSbrEnvFrame->SbrGrid.bs_num_env = numEnv; - hSbrEnvFrame->SbrGrid.frameClass = frameClass; - } - else { - /*-------------------------------------------------------------------------- - Calculate frame class to use - ---------------------------------------------------------------------------*/ - calcFrameClass (&frameClass, frameClassOld, tranFlag, spreadFlag); - - /* patch for new frame class FIXFIXonly for AAC LD */ - if (tranFlag && ldGrid) { - frameClass = FIXFIXonly; - *frameClassOld = FIXFIX; - } - - /* - * every transient is processed below by inserting - * - * - one border at the onset of the transient - * - one or more "decay borders" (after the onset of the transient) - * - optionally one "attack border" (before the onset of the transient) - * - * those borders are referred to as "mandatory borders" and are - * defined by the 'segmentLength' array in FRAME_GEN_TUNING - * - * the frequency resolutions of the corresponding envelopes are - * defined by the 'segmentRes' array in FRAME_GEN_TUNING - */ - - /*-------------------------------------------------------------------------- - Design frame (or follow-up old design) - ---------------------------------------------------------------------------*/ - if (tranFlag) { /* Always for FixVar, often but not always for VarVar */ - /*-------------------------------------------------------------------------- - Design part of T/F-grid around the new transient - ---------------------------------------------------------------------------*/ - - tranPosInternal = frameMiddleSlot + tranPos + bufferFrameStart ; /* FH 00-06-26 */ - /* - add mandatory borders around transient - */ - - fillFrameTran ( v_tuningSegm, - v_tuningFreq, - tranPosInternal, - v_bord, - length_v_bord, - v_freq, - length_v_freq, - &bmin, - &bmax ); - - /* make sure we stay within the maximum SBR frame overlap */ - fmax = calcFillLengthMax(tranPos, numberTimeSlots); - } - - switch (frameClass) { - - case FIXFIXonly: - FDK_ASSERT(ldGrid); - tranPosInternal = tranPos; - generateFixFixOnly ( &(hSbrEnvFrame->SbrFrameInfo), - &(hSbrEnvFrame->SbrGrid), - tranPosInternal, - numberTimeSlots, - hSbrEnvFrame->fResTransIsLow - ); - - return &(hSbrEnvFrame->SbrFrameInfo); - - case FIXVAR: - - /*-------------------------------------------------------------------------- - Design remaining parts of T/F-grid (assuming next frame is VarFix) - ---------------------------------------------------------------------------*/ - - /*-------------------------------------------------------------------------- - Fill region before new transient: - ---------------------------------------------------------------------------*/ - fillFramePre (dmax, v_bord, length_v_bord, v_freq, length_v_freq, - bmin, bmin - bufferFrameStart); /* FH 00-06-26 */ - - /*-------------------------------------------------------------------------- - Fill region after new transient: - ---------------------------------------------------------------------------*/ - fillFramePost (&parts, &d, dmax, v_bord, length_v_bord, v_freq, - length_v_freq, bmax, bufferFrameStart, numberTimeSlots, fmax); - - /*-------------------------------------------------------------------------- - Take care of special case: - ---------------------------------------------------------------------------*/ - if (parts == 1 && d < dmin) /* no fill, short last envelope */ - specialCase (spreadFlag, allowSpread, v_bord, length_v_bord, - v_freq, length_v_freq, &parts, d); - - /*-------------------------------------------------------------------------- - Calculate common border (split-point) - ---------------------------------------------------------------------------*/ - calcCmonBorder (&i_cmon, &i_tran, v_bord, length_v_bord, tranPosInternal, - bufferFrameStart, numberTimeSlots); /* FH 00-06-26 */ - - /*-------------------------------------------------------------------------- - Extract data for proper follow-up in next frame - ---------------------------------------------------------------------------*/ - keepForFollowUp (v_bordFollow, length_v_bordFollow, v_freqFollow, - length_v_freqFollow, i_tranFollow, i_fillFollow, - v_bord, length_v_bord, v_freq, i_cmon, i_tran, parts, numberTimeSlots); /* FH 00-06-26 */ - - /*-------------------------------------------------------------------------- - Calculate control signal - ---------------------------------------------------------------------------*/ - calcCtrlSignal (&hSbrEnvFrame->SbrGrid, frameClass, - v_bord, *length_v_bord, v_freq, *length_v_freq, - i_cmon, i_tran, *spreadFlag, DC); - break; - case VARFIX: - /*-------------------------------------------------------------------------- - Follow-up old transient - calculate control signal - ---------------------------------------------------------------------------*/ - calcCtrlSignal (&hSbrEnvFrame->SbrGrid, frameClass, - v_bordFollow, *length_v_bordFollow, v_freqFollow, - *length_v_freqFollow, DC, *i_tranFollow, - *spreadFlag, DC); - break; - case VARVAR: - if (*spreadFlag) { /* spread across three frames */ - /*-------------------------------------------------------------------------- - Follow-up old transient - calculate control signal - ---------------------------------------------------------------------------*/ - calcCtrlSignal (&hSbrEnvFrame->SbrGrid, - frameClass, v_bordFollow, *length_v_bordFollow, - v_freqFollow, *length_v_freqFollow, DC, - *i_tranFollow, *spreadFlag, DC); - - *spreadFlag = 0; - - /*-------------------------------------------------------------------------- - Extract data for proper follow-up in next frame - ---------------------------------------------------------------------------*/ - v_bordFollow[0] = hSbrEnvFrame->SbrGrid.bs_abs_bord_1 - numberTimeSlots; /* FH 00-06-26 */ - v_freqFollow[0] = 1; - *length_v_bordFollow = 1; - *length_v_freqFollow = 1; - - *i_tranFollow = -DC; - *i_fillFollow = -DC; - } - else { - /*-------------------------------------------------------------------------- - Design remaining parts of T/F-grid (assuming next frame is VarFix) - adapt or fill region before new transient: - ---------------------------------------------------------------------------*/ - fillFrameInter (&nL, v_tuningSegm, v_bord, length_v_bord, bmin, - v_freq, length_v_freq, v_bordFollow, - length_v_bordFollow, v_freqFollow, - length_v_freqFollow, *i_fillFollow, dmin, dmax, - numberTimeSlots); - - /*-------------------------------------------------------------------------- - Fill after transient: - ---------------------------------------------------------------------------*/ - fillFramePost (&parts, &d, dmax, v_bord, length_v_bord, v_freq, - length_v_freq, bmax, bufferFrameStart, numberTimeSlots, fmax); - - /*-------------------------------------------------------------------------- - Take care of special case: - ---------------------------------------------------------------------------*/ - if (parts == 1 && d < dmin) /*% no fill, short last envelope */ - specialCase (spreadFlag, allowSpread, v_bord, length_v_bord, - v_freq, length_v_freq, &parts, d); - - /*-------------------------------------------------------------------------- - Calculate common border (split-point) - ---------------------------------------------------------------------------*/ - calcCmonBorder (&i_cmon, &i_tran, v_bord, length_v_bord, tranPosInternal, - bufferFrameStart, numberTimeSlots); - - /*-------------------------------------------------------------------------- - Extract data for proper follow-up in next frame - ---------------------------------------------------------------------------*/ - keepForFollowUp (v_bordFollow, length_v_bordFollow, - v_freqFollow, length_v_freqFollow, - i_tranFollow, i_fillFollow, v_bord, - length_v_bord, v_freq, i_cmon, i_tran, parts, numberTimeSlots); - - /*-------------------------------------------------------------------------- - Calculate control signal - ---------------------------------------------------------------------------*/ - calcCtrlSignal (&hSbrEnvFrame->SbrGrid, - frameClass, v_bord, *length_v_bord, v_freq, - *length_v_freq, i_cmon, i_tran, 0, nL); - } - break; - case FIXFIX: - if (tranPos == 0) - numEnv = 1; - else - numEnv = 2; - - hSbrEnvFrame->SbrGrid.bs_num_env = numEnv; - hSbrEnvFrame->SbrGrid.frameClass = frameClass; - - break; - default: - FDK_ASSERT(0); - } - } - - /*------------------------------------------------------------------------- - Convert control signal to frame info struct - ---------------------------------------------------------------------------*/ - ctrlSignal2FrameInfo (&hSbrEnvFrame->SbrGrid, - &hSbrEnvFrame->SbrFrameInfo, - hSbrEnvFrame->freq_res_fixfix); - - return &hSbrEnvFrame->SbrFrameInfo; -} - - -/***************************************************************************/ -/*! - \brief Gnerates frame info for FIXFIXonly frame class used for low delay version - - \return nothing - ****************************************************************************/ -static void generateFixFixOnly ( HANDLE_SBR_FRAME_INFO hSbrFrameInfo, - HANDLE_SBR_GRID hSbrGrid, - int tranPosInternal, - int numberTimeSlots, - UCHAR fResTransIsLow - ) -{ - int nEnv, i, k=0, tranIdx; - const int *pTable = NULL; - const FREQ_RES *freqResTable = NULL; - - switch (numberTimeSlots) { - case 8: - pTable = envelopeTable_8[tranPosInternal]; - freqResTable = freqRes_table_8; - break; - case 15: - pTable = envelopeTable_15[tranPosInternal]; - freqResTable = freqRes_table_16; - break; - case 16: - pTable = envelopeTable_16[tranPosInternal]; - freqResTable = freqRes_table_16; - break; - } - - /* look number of envolpes in table */ - nEnv = pTable[0]; - /* look up envolpe distribution in table */ - for (i=1; iborders[i] = pTable[i+2]; - - /* open and close frame border */ - hSbrFrameInfo->borders[0] = 0; - hSbrFrameInfo->borders[nEnv] = numberTimeSlots; - - /* adjust segment-frequency-resolution according to the segment-length */ - for (i=0; iborders[i+1] - hSbrFrameInfo->borders[i]; - if (!fResTransIsLow) - hSbrFrameInfo->freqRes[i] = freqResTable[k]; - else - hSbrFrameInfo->freqRes[i] = FREQ_RES_LOW; - - hSbrGrid->v_f[i] = hSbrFrameInfo->freqRes[i]; - } - - hSbrFrameInfo->nEnvelopes = nEnv; - hSbrFrameInfo->shortEnv = pTable[2]; - /* transient idx */ - tranIdx = pTable[1]; - - /* add noise floors */ - hSbrFrameInfo->bordersNoise[0] = 0; - hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[tranIdx?tranIdx:1]; - hSbrFrameInfo->bordersNoise[2] = numberTimeSlots; - hSbrFrameInfo->nNoiseEnvelopes = 2; - - hSbrGrid->frameClass = FIXFIXonly; - hSbrGrid->bs_abs_bord = tranPosInternal; - hSbrGrid->bs_num_env = nEnv; - -} - - - -/******************************************************************************* - Functionname: FDKsbrEnc_initFrameInfoGenerator - ******************************************************************************* - - Description: - - Arguments: hSbrEnvFrame - pointer to sbr envelope handle - allowSpread - commandline parameter - numEnvStatic - commandline parameter - staticFraming - commandline parameter - - Return: none - -*******************************************************************************/ -void -FDKsbrEnc_initFrameInfoGenerator ( - HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame, - INT allowSpread, - INT numEnvStatic, - INT staticFraming, - INT timeSlots, - const FREQ_RES* freq_res_fixfix - ,UCHAR fResTransIsLow, - INT ldGrid - ) -{ /* FH 00-06-26 */ - - FDKmemclear(hSbrEnvFrame,sizeof(SBR_ENVELOPE_FRAME )); - - - /* Initialisation */ - hSbrEnvFrame->frameClassOld = FIXFIX; - hSbrEnvFrame->spreadFlag = 0; - - hSbrEnvFrame->allowSpread = allowSpread; - hSbrEnvFrame->numEnvStatic = numEnvStatic; - hSbrEnvFrame->staticFraming = staticFraming; - hSbrEnvFrame->freq_res_fixfix[0] = freq_res_fixfix[0]; - hSbrEnvFrame->freq_res_fixfix[1] = freq_res_fixfix[1]; - hSbrEnvFrame->fResTransIsLow = fResTransIsLow; - - hSbrEnvFrame->length_v_bord = 0; - hSbrEnvFrame->length_v_bordFollow = 0; - - hSbrEnvFrame->length_v_freq = 0; - hSbrEnvFrame->length_v_freqFollow = 0; - - hSbrEnvFrame->i_tranFollow = 0; - hSbrEnvFrame->i_fillFollow = 0; - - hSbrEnvFrame->SbrGrid.numberTimeSlots = timeSlots; - - if (ldGrid) { - /*case CODEC_AACLD:*/ - hSbrEnvFrame->dmin = 2; - hSbrEnvFrame->dmax = 16; - hSbrEnvFrame->frameMiddleSlot = FRAME_MIDDLE_SLOT_512LD; - hSbrEnvFrame->SbrGrid.bufferFrameStart = 0; - } else - switch(timeSlots){ - case NUMBER_TIME_SLOTS_1920: - hSbrEnvFrame->dmin = 4; - hSbrEnvFrame->dmax = 12; - hSbrEnvFrame->SbrGrid.bufferFrameStart = 0; - hSbrEnvFrame->frameMiddleSlot = FRAME_MIDDLE_SLOT_1920; - break; - case NUMBER_TIME_SLOTS_2048: - hSbrEnvFrame->dmin = 4; - hSbrEnvFrame->dmax = 12; - hSbrEnvFrame->SbrGrid.bufferFrameStart = 0; - hSbrEnvFrame->frameMiddleSlot = FRAME_MIDDLE_SLOT_2048; - break; - case NUMBER_TIME_SLOTS_1152: - hSbrEnvFrame->dmin = 2; - hSbrEnvFrame->dmax = 8; - hSbrEnvFrame->SbrGrid.bufferFrameStart = 0; - hSbrEnvFrame->frameMiddleSlot = FRAME_MIDDLE_SLOT_1152; - break; - case NUMBER_TIME_SLOTS_2304: - hSbrEnvFrame->dmin = 4; - hSbrEnvFrame->dmax = 15; - hSbrEnvFrame->SbrGrid.bufferFrameStart = 0; - hSbrEnvFrame->frameMiddleSlot = FRAME_MIDDLE_SLOT_2304; - break; - default: - FDK_ASSERT(0); - } - -} - - -/******************************************************************************* - Functionname: fillFrameTran - ******************************************************************************* - - Description: Add mandatory borders, as described by the tuning vector - and the current transient position - - Arguments: - modified: - v_bord - int pointer to v_bord vector - length_v_bord - length of v_bord vector - v_freq - int pointer to v_freq vector - length_v_freq - length of v_freq vector - bmin - int pointer to bmin (call by reference) - bmax - int pointer to bmax (call by reference) - not modified: - tran - position of transient - v_tuningSegm - int pointer to v_tuningSegm vector - v_tuningFreq - int pointer to v_tuningFreq vector - - Return: none - -*******************************************************************************/ -static void -fillFrameTran (const int *v_tuningSegm, /*!< tuning: desired segment lengths */ - const int *v_tuningFreq, /*!< tuning: desired frequency resolutions */ - int tran, /*!< input : position of transient */ - int *v_bord, /*!< memNew: borders */ - int *length_v_bord, /*!< memNew: # borders */ - int *v_freq, /*!< memNew: frequency resolutions */ - int *length_v_freq, /*!< memNew: # frequency resolutions */ - int *bmin, /*!< hlpNew: first mandatory border */ - int *bmax /*!< hlpNew: last mandatory border */ - ) -{ - int bord, i; - - *length_v_bord = 0; - *length_v_freq = 0; - - /* add attack env leading border (optional) */ - if (v_tuningSegm[0]) { - /* v_bord = [(Ba)] start of attack env */ - FDKsbrEnc_AddRight (v_bord, length_v_bord, (tran - v_tuningSegm[0])); - - /* v_freq = [(Fa)] res of attack env */ - FDKsbrEnc_AddRight (v_freq, length_v_freq, v_tuningFreq[0]); - } - - /* add attack env trailing border/first decay env leading border */ - bord = tran; - FDKsbrEnc_AddRight (v_bord, length_v_bord, tran); /* v_bord = [(Ba),Bd1] */ - - /* add first decay env trailing border/2:nd decay env leading border */ - if (v_tuningSegm[1]) { - bord += v_tuningSegm[1]; - - /* v_bord = [(Ba),Bd1,Bd2] */ - FDKsbrEnc_AddRight (v_bord, length_v_bord, bord); - - /* v_freq = [(Fa),Fd1] */ - FDKsbrEnc_AddRight (v_freq, length_v_freq, v_tuningFreq[1]); - } - - /* add 2:nd decay env trailing border (optional) */ - if (v_tuningSegm[2] != 0) { - bord += v_tuningSegm[2]; - - /* v_bord = [(Ba),Bd1, Bd2,(Bd3)] */ - FDKsbrEnc_AddRight (v_bord, length_v_bord, bord); - - /* v_freq = [(Fa),Fd1,(Fd2)] */ - FDKsbrEnc_AddRight (v_freq, length_v_freq, v_tuningFreq[2]); - } - - /* v_freq = [(Fa),Fd1,(Fd2),1] */ - FDKsbrEnc_AddRight (v_freq, length_v_freq, 1); - - - /* calc min and max values of mandatory borders */ - *bmin = v_bord[0]; - for (i = 0; i < *length_v_bord; i++) - if (v_bord[i] < *bmin) - *bmin = v_bord[i]; - - *bmax = v_bord[0]; - for (i = 0; i < *length_v_bord; i++) - if (v_bord[i] > *bmax) - *bmax = v_bord[i]; - -} - - - -/******************************************************************************* - Functionname: fillFramePre - ******************************************************************************* - - Description: Add borders before mandatory borders, if needed - - Arguments: - modified: - v_bord - int pointer to v_bord vector - length_v_bord - length of v_bord vector - v_freq - int pointer to v_freq vector - length_v_freq - length of v_freq vector - not modified: - dmax - int value - bmin - int value - rest - int value - - Return: none - -*******************************************************************************/ -static void -fillFramePre (INT dmax, - INT *v_bord, INT *length_v_bord, - INT *v_freq, INT *length_v_freq, - INT bmin, INT rest) -{ - /* - input state: - v_bord = [(Ba),Bd1, Bd2 ,(Bd3)] - v_freq = [(Fa),Fd1,(Fd2),1 ] - */ - - INT parts, d, j, S, s = 0, segm, bord; - - /* - start with one envelope - */ - - parts = 1; - d = rest; - - /* - calc # of additional envelopes and corresponding lengths - */ - - while (d > dmax) { - parts++; - - segm = rest / parts; - S = (segm - 2)>>1; - s = fixMin (8, 2 * S + 2); - d = rest - (parts - 1) * s; - } - - /* - add borders before mandatory borders - */ - - bord = bmin; - - for (j = 0; j <= parts - 2; j++) { - bord = bord - s; - - /* v_bord = [...,(Bf),(Ba),Bd1, Bd2 ,(Bd3)] */ - FDKsbrEnc_AddLeft (v_bord, length_v_bord, bord); - - /* v_freq = [...,(1 ),(Fa),Fd1,(Fd2), 1 ] */ - FDKsbrEnc_AddLeft (v_freq, length_v_freq, 1); - } -} - -/***************************************************************************/ -/*! - \brief Overlap control - - Calculate max length of trailing fill segments, such that we always get a - border within the frame overlap region - - \return void - -****************************************************************************/ -static int -calcFillLengthMax (int tranPos, /*!< input : transient position (ref: tran det) */ - int numberTimeSlots /*!< input : number of timeslots */ - ) -{ - int fmax; - - /* - calculate transient position within envelope buffer - */ - switch (numberTimeSlots) - { - case NUMBER_TIME_SLOTS_2048: - if (tranPos < 4) - fmax = 6; - else if (tranPos == 4 || tranPos == 5) - fmax = 4; - else - fmax = 8; - break; - - case NUMBER_TIME_SLOTS_1920: - if (tranPos < 4) - fmax = 5; - else if (tranPos == 4 || tranPos == 5) - fmax = 3; - else - fmax = 7; - break; - - default: - fmax = 8; - break; - } - - return fmax; -} - -/******************************************************************************* - Functionname: fillFramePost - ******************************************************************************* - - Description: -Add borders after mandatory borders, if needed - Make a preliminary design of next frame, - assuming no transient is present there - - Arguments: - modified: - parts - int pointer to parts (call by reference) - d - int pointer to d (call by reference) - v_bord - int pointer to v_bord vector - length_v_bord - length of v_bord vector - v_freq - int pointer to v_freq vector - length_v_freq - length of v_freq vector - not modified: - bmax - int value - dmax - int value - - Return: none - -*******************************************************************************/ -static void -fillFramePost (INT *parts, INT *d, INT dmax, INT *v_bord, INT *length_v_bord, - INT *v_freq, INT *length_v_freq, INT bmax, - INT bufferFrameStart, INT numberTimeSlots, INT fmax) -{ - INT j, rest, segm, S, s = 0, bord; - - /* - input state: - v_bord = [...,(Bf),(Ba),Bd1, Bd2 ,(Bd3)] - v_freq = [...,(1 ),(Fa),Fd1,(Fd2),1 ] - */ - - rest = bufferFrameStart + 2 * numberTimeSlots - bmax; - *d = rest; - - if (*d > 0) { - *parts = 1; /* start with one envelope */ - - /* calc # of additional envelopes and corresponding lengths */ - - while (*d > dmax) { - *parts = *parts + 1; - - segm = rest / (*parts); - S = (segm - 2)>>1; - s = fixMin (fmax, 2 * S + 2); - *d = rest - (*parts - 1) * s; - } - - /* add borders after mandatory borders */ - - bord = bmax; - for (j = 0; j <= *parts - 2; j++) { - bord += s; - - /* v_bord = [...,(Bf),(Ba),Bd1, Bd2 ,(Bd3),(Bf)] */ - FDKsbrEnc_AddRight (v_bord, length_v_bord, bord); - - /* v_freq = [...,(1 ),(Fa),Fd1,(Fd2), 1 , 1! ,1] */ - FDKsbrEnc_AddRight (v_freq, length_v_freq, 1); - } - } - else { - *parts = 1; - - /* remove last element from v_bord and v_freq */ - - *length_v_bord = *length_v_bord - 1; - *length_v_freq = *length_v_freq - 1; - - } -} - - - -/******************************************************************************* - Functionname: fillFrameInter - ******************************************************************************* - - Description: - - Arguments: nL - - v_tuningSegm - - v_bord - - length_v_bord - - bmin - - v_freq - - length_v_freq - - v_bordFollow - - length_v_bordFollow - - v_freqFollow - - length_v_freqFollow - - i_fillFollow - - dmin - - dmax - - - Return: none - -*******************************************************************************/ -static void -fillFrameInter (INT *nL, const int *v_tuningSegm, INT *v_bord, INT *length_v_bord, - INT bmin, INT *v_freq, INT *length_v_freq, INT *v_bordFollow, - INT *length_v_bordFollow, INT *v_freqFollow, - INT *length_v_freqFollow, INT i_fillFollow, INT dmin, - INT dmax, INT numberTimeSlots) -{ - INT middle, b_new, numBordFollow, bordMaxFollow, i; - - if (numberTimeSlots != NUMBER_TIME_SLOTS_1152) { - - /* % remove fill borders: */ - if (i_fillFollow >= 1) { - *length_v_bordFollow = i_fillFollow; - *length_v_freqFollow = i_fillFollow; - } - - numBordFollow = *length_v_bordFollow; - bordMaxFollow = v_bordFollow[numBordFollow - 1]; - - /* remove even more borders if needed */ - middle = bmin - bordMaxFollow; - while (middle < 0) { - numBordFollow--; - bordMaxFollow = v_bordFollow[numBordFollow - 1]; - middle = bmin - bordMaxFollow; - } - - *length_v_bordFollow = numBordFollow; - *length_v_freqFollow = numBordFollow; - *nL = numBordFollow - 1; - - b_new = *length_v_bord; - - - if (middle <= dmax) { - if (middle >= dmin) { /* concatenate */ - FDKsbrEnc_AddVecLeft (v_bord, length_v_bord, v_bordFollow, *length_v_bordFollow); - FDKsbrEnc_AddVecLeft (v_freq, length_v_freq, v_freqFollow, *length_v_freqFollow); - } - - else { - if (v_tuningSegm[0] != 0) { /* remove one new border and concatenate */ - *length_v_bord = b_new - 1; - FDKsbrEnc_AddVecLeft (v_bord, length_v_bord, v_bordFollow, - *length_v_bordFollow); - - *length_v_freq = b_new - 1; - FDKsbrEnc_AddVecLeft (v_freq + 1, length_v_freq, v_freqFollow, - *length_v_freqFollow); - } - else { - if (*length_v_bordFollow > 1) { /* remove one old border and concatenate */ - FDKsbrEnc_AddVecLeft (v_bord, length_v_bord, v_bordFollow, - *length_v_bordFollow - 1); - FDKsbrEnc_AddVecLeft (v_freq, length_v_freq, v_freqFollow, - *length_v_bordFollow - 1); - - *nL = *nL - 1; - } - else { /* remove new "transient" border and concatenate */ - - for (i = 0; i < *length_v_bord - 1; i++) - v_bord[i] = v_bord[i + 1]; - - for (i = 0; i < *length_v_freq - 1; i++) - v_freq[i] = v_freq[i + 1]; - - *length_v_bord = b_new - 1; - *length_v_freq = b_new - 1; - - FDKsbrEnc_AddVecLeft (v_bord, length_v_bord, v_bordFollow, - *length_v_bordFollow); - FDKsbrEnc_AddVecLeft (v_freq, length_v_freq, v_freqFollow, - *length_v_freqFollow); - } - } - } - } - else { /* middle > dmax */ - - fillFramePre (dmax, v_bord, length_v_bord, v_freq, length_v_freq, bmin, - middle); - FDKsbrEnc_AddVecLeft (v_bord, length_v_bord, v_bordFollow, *length_v_bordFollow); - FDKsbrEnc_AddVecLeft (v_freq, length_v_freq, v_freqFollow, *length_v_freqFollow); - } - - - } - else { /* numberTimeSlots==NUMBER_TIME_SLOTS_1152 */ - - INT l,m; - - - /*------------------------------------------------------------------------ - remove fill borders - ------------------------------------------------------------------------*/ - if (i_fillFollow >= 1) { - *length_v_bordFollow = i_fillFollow; - *length_v_freqFollow = i_fillFollow; - } - - numBordFollow = *length_v_bordFollow; - bordMaxFollow = v_bordFollow[numBordFollow - 1]; - - /*------------------------------------------------------------------------ - remove more borders if necessary to eliminate overlap - ------------------------------------------------------------------------*/ - - /* check for overlap */ - middle = bmin - bordMaxFollow; - - /* intervals: - i) middle < 0 : overlap, must remove borders - ii) 0 <= middle < dmin : no overlap but too tight, must remove borders - iii) dmin <= middle <= dmax : ok, just concatenate - iv) dmax <= middle : too wide, must add borders - */ - - /* first remove old non-fill-borders... */ - while (middle < 0) { - - /* ...but don't remove all of them */ - if (numBordFollow == 1) - break; - - numBordFollow--; - bordMaxFollow = v_bordFollow[numBordFollow - 1]; - middle = bmin - bordMaxFollow; - } - - /* if this isn't enough, remove new non-fill borders */ - if (middle < 0) - { - for (l = 0, m = 0 ; l < *length_v_bord ; l++) - { - if(v_bord[l]> bordMaxFollow) - { - v_bord[m] = v_bord[l]; - v_freq[m] = v_freq[l]; - m++; - } - } - - *length_v_bord = l; - *length_v_freq = l; - - bmin = v_bord[0]; - - } - - /*------------------------------------------------------------------------ - update modified follow-up data - ------------------------------------------------------------------------*/ - - *length_v_bordFollow = numBordFollow; - *length_v_freqFollow = numBordFollow; - - /* left relative borders correspond to follow-up */ - *nL = numBordFollow - 1; - - /*------------------------------------------------------------------------ - take care of intervals ii through iv - ------------------------------------------------------------------------*/ - - /* now middle should be >= 0 */ - middle = bmin - bordMaxFollow; - - if (middle <= dmin) /* (ii) */ - { - b_new = *length_v_bord; - - if (v_tuningSegm[0] != 0) - { - /* remove new "luxury" border and concatenate */ - *length_v_bord = b_new - 1; - FDKsbrEnc_AddVecLeft (v_bord, length_v_bord, v_bordFollow, - *length_v_bordFollow); - - *length_v_freq = b_new - 1; - FDKsbrEnc_AddVecLeft (v_freq + 1, length_v_freq, v_freqFollow, - *length_v_freqFollow); - - } - else if (*length_v_bordFollow > 1) - { - /* remove old border and concatenate */ - FDKsbrEnc_AddVecLeft (v_bord, length_v_bord, v_bordFollow, - *length_v_bordFollow - 1); - FDKsbrEnc_AddVecLeft (v_freq, length_v_freq, v_freqFollow, - *length_v_bordFollow - 1); - - *nL = *nL - 1; - } - else - { - /* remove new border and concatenate */ - for (i = 0; i < *length_v_bord - 1; i++) - v_bord[i] = v_bord[i + 1]; - - for (i = 0; i < *length_v_freq - 1; i++) - v_freq[i] = v_freq[i + 1]; - - *length_v_bord = b_new - 1; - *length_v_freq = b_new - 1; - - FDKsbrEnc_AddVecLeft (v_bord, length_v_bord, v_bordFollow, - *length_v_bordFollow); - FDKsbrEnc_AddVecLeft (v_freq, length_v_freq, v_freqFollow, - *length_v_freqFollow); - } - } - else if ((middle >= dmin) && (middle <= dmax)) /* (iii) */ - { - /* concatenate */ - FDKsbrEnc_AddVecLeft (v_bord, length_v_bord, v_bordFollow, *length_v_bordFollow); - FDKsbrEnc_AddVecLeft (v_freq, length_v_freq, v_freqFollow, *length_v_freqFollow); - - } - else /* (iv) */ - { - fillFramePre (dmax, v_bord, length_v_bord, v_freq, length_v_freq, bmin, - middle); - FDKsbrEnc_AddVecLeft (v_bord, length_v_bord, v_bordFollow, *length_v_bordFollow); - FDKsbrEnc_AddVecLeft (v_freq, length_v_freq, v_freqFollow, *length_v_freqFollow); - } - } -} - - - -/******************************************************************************* - Functionname: calcFrameClass - ******************************************************************************* - - Description: - - Arguments: INT* frameClass, INT* frameClassOld, INT tranFlag, INT* spreadFlag) - - Return: none - -*******************************************************************************/ -static void -calcFrameClass (FRAME_CLASS *frameClass, FRAME_CLASS *frameClassOld, INT tranFlag, - INT *spreadFlag) -{ - - switch (*frameClassOld) { - case FIXFIXonly: - case FIXFIX: - if (tranFlag) *frameClass = FIXVAR; - else *frameClass = FIXFIX; - break; - case FIXVAR: - if (tranFlag) { *frameClass = VARVAR; *spreadFlag = 0; } - else { - if (*spreadFlag) *frameClass = VARVAR; - else *frameClass = VARFIX; - } - break; - case VARFIX: - if (tranFlag) *frameClass = FIXVAR; - else *frameClass = FIXFIX; - break; - case VARVAR: - if (tranFlag) { *frameClass = VARVAR; *spreadFlag = 0; } - else { - if (*spreadFlag) *frameClass = VARVAR; - else *frameClass = VARFIX; - } - break; - }; - - *frameClassOld = *frameClass; -} - - - -/******************************************************************************* - Functionname: specialCase - ******************************************************************************* - - Description: - - Arguments: spreadFlag - allowSpread - v_bord - length_v_bord - v_freq - length_v_freq - parts - d - - Return: none - -*******************************************************************************/ -static void -specialCase (INT *spreadFlag, INT allowSpread, INT *v_bord, - INT *length_v_bord, INT *v_freq, INT *length_v_freq, INT *parts, - INT d) -{ - INT L; - - L = *length_v_bord; - - if (allowSpread) { /* add one "step 8" */ - *spreadFlag = 1; - FDKsbrEnc_AddRight (v_bord, length_v_bord, v_bord[L - 1] + 8); - FDKsbrEnc_AddRight (v_freq, length_v_freq, 1); - (*parts)++; - } - else { - if (d == 1) { /* stretch one slot */ - *length_v_bord = L - 1; - *length_v_freq = L - 1; - } - else { - if ((v_bord[L - 1] - v_bord[L - 2]) > 2) { /* compress one quant step */ - v_bord[L - 1] = v_bord[L - 1] - 2; - v_freq[*length_v_freq - 1] = 0; /* use low res for short segment */ - } - } - } -} - - - -/******************************************************************************* - Functionname: calcCmonBorder - ******************************************************************************* - - Description: - - Arguments: i_cmon - i_tran - v_bord - length_v_bord - tran - - Return: none - -*******************************************************************************/ -static void -calcCmonBorder (INT *i_cmon, INT *i_tran, INT *v_bord, INT *length_v_bord, - INT tran, INT bufferFrameStart, INT numberTimeSlots) -{ /* FH 00-06-26 */ - INT i; - - for (i = 0; i < *length_v_bord; i++) - if (v_bord[i] >= bufferFrameStart + numberTimeSlots) { /* FH 00-06-26 */ - *i_cmon = i; - break; - } - - /* keep track of transient: */ - for (i = 0; i < *length_v_bord; i++) - if (v_bord[i] >= tran) { - *i_tran = i; - break; - } - else - *i_tran = EMPTY; -} - -/******************************************************************************* - Functionname: keepForFollowUp - ******************************************************************************* - - Description: - - Arguments: v_bordFollow - length_v_bordFollow - v_freqFollow - length_v_freqFollow - i_tranFollow - i_fillFollow - v_bord - length_v_bord - v_freq - i_cmon - i_tran - parts) - - Return: none - -*******************************************************************************/ -static void -keepForFollowUp (INT *v_bordFollow, INT *length_v_bordFollow, - INT *v_freqFollow, INT *length_v_freqFollow, - INT *i_tranFollow, INT *i_fillFollow, INT *v_bord, - INT *length_v_bord, INT *v_freq, INT i_cmon, INT i_tran, - INT parts, INT numberTimeSlots) -{ /* FH 00-06-26 */ - INT L, i, j; - - L = *length_v_bord; - - (*length_v_bordFollow) = 0; - (*length_v_freqFollow) = 0; - - for (j = 0, i = i_cmon; i < L; i++, j++) { - v_bordFollow[j] = v_bord[i] - numberTimeSlots; /* FH 00-06-26 */ - v_freqFollow[j] = v_freq[i]; - (*length_v_bordFollow)++; - (*length_v_freqFollow)++; - } - if (i_tran != EMPTY) - *i_tranFollow = i_tran - i_cmon; - else - *i_tranFollow = EMPTY; - *i_fillFollow = L - (parts - 1) - i_cmon; - -} - -/******************************************************************************* - Functionname: calcCtrlSignal - ******************************************************************************* - - Description: - - Arguments: hSbrGrid - frameClass - v_bord - length_v_bord - v_freq - length_v_freq - i_cmon - i_tran - spreadFlag - nL - - Return: none - -*******************************************************************************/ -static void -calcCtrlSignal (HANDLE_SBR_GRID hSbrGrid, - FRAME_CLASS frameClass, INT *v_bord, INT length_v_bord, INT *v_freq, - INT length_v_freq, INT i_cmon, INT i_tran, INT spreadFlag, - INT nL) -{ - - - INT i, r, a, n, p, b, aL, aR, ntot, nmax, nR; - - INT *v_f = hSbrGrid->v_f; - INT *v_fLR = hSbrGrid->v_fLR; - INT *v_r = hSbrGrid->bs_rel_bord; - INT *v_rL = hSbrGrid->bs_rel_bord_0; - INT *v_rR = hSbrGrid->bs_rel_bord_1; - - INT length_v_r = 0; - INT length_v_rR = 0; - INT length_v_rL = 0; - - switch (frameClass) { - case FIXVAR: - /* absolute border: */ - - a = v_bord[i_cmon]; - - /* relative borders: */ - length_v_r = 0; - i = i_cmon; - - while (i >= 1) { - r = v_bord[i] - v_bord[i - 1]; - FDKsbrEnc_AddRight (v_r, &length_v_r, r); - i--; - } - - - /* number of relative borders: */ - n = length_v_r; - - - /* freq res: */ - for (i = 0; i < i_cmon; i++) - v_f[i] = v_freq[i_cmon - 1 - i]; - v_f[i_cmon] = 1; - - /* pointer: */ - p = (i_cmon >= i_tran && i_tran != EMPTY) ? (i_cmon - i_tran + 1) : (0) ; - - hSbrGrid->frameClass = frameClass; - hSbrGrid->bs_abs_bord = a; - hSbrGrid->n = n; - hSbrGrid->p = p; - - break; - case VARFIX: - /* absolute border: */ - a = v_bord[0]; - - /* relative borders: */ - length_v_r = 0; - - for (i = 1; i < length_v_bord; i++) { - r = v_bord[i] - v_bord[i - 1]; - FDKsbrEnc_AddRight (v_r, &length_v_r, r); - } - - /* number of relative borders: */ - n = length_v_r; - - /* freq res: */ - FDKmemcpy (v_f, v_freq, length_v_freq * sizeof (INT)); - - - /* pointer: */ - p = (i_tran >= 0 && i_tran != EMPTY) ? (i_tran + 1) : (0) ; - - hSbrGrid->frameClass = frameClass; - hSbrGrid->bs_abs_bord = a; - hSbrGrid->n = n; - hSbrGrid->p = p; - - break; - case VARVAR: - if (spreadFlag) { - /* absolute borders: */ - b = length_v_bord; - - aL = v_bord[0]; - aR = v_bord[b - 1]; - - - /* number of relative borders: */ - ntot = b - 2; - - nmax = 2; /* n: {0,1,2} */ - if (ntot > nmax) { - nL = nmax; - nR = ntot - nmax; - } - else { - nL = ntot; - nR = 0; - } - - /* relative borders: */ - length_v_rL = 0; - for (i = 1; i <= nL; i++) { - r = v_bord[i] - v_bord[i - 1]; - FDKsbrEnc_AddRight (v_rL, &length_v_rL, r); - } - - length_v_rR = 0; - i = b - 1; - while (i >= b - nR) { - r = v_bord[i] - v_bord[i - 1]; - FDKsbrEnc_AddRight (v_rR, &length_v_rR, r); - i--; - } - - /* pointer (only one due to constraint in frame info): */ - p = (i_tran > 0 && i_tran != EMPTY) ? (b - i_tran) : (0) ; - - /* freq res: */ - - for (i = 0; i < b - 1; i++) - v_fLR[i] = v_freq[i]; - } - else { - - length_v_bord = i_cmon + 1; - length_v_freq = i_cmon + 1; - - - /* absolute borders: */ - b = length_v_bord; - - aL = v_bord[0]; - aR = v_bord[b - 1]; - - /* number of relative borders: */ - ntot = b - 2; - nR = ntot - nL; - - /* relative borders: */ - length_v_rL = 0; - for (i = 1; i <= nL; i++) { - r = v_bord[i] - v_bord[i - 1]; - FDKsbrEnc_AddRight (v_rL, &length_v_rL, r); - } - - length_v_rR = 0; - i = b - 1; - while (i >= b - nR) { - r = v_bord[i] - v_bord[i - 1]; - FDKsbrEnc_AddRight (v_rR, &length_v_rR, r); - i--; - } - - /* pointer (only one due to constraint in frame info): */ - p = (i_cmon >= i_tran && i_tran != EMPTY) ? (i_cmon - i_tran + 1) : (0) ; - - /* freq res: */ - for (i = 0; i < b - 1; i++) - v_fLR[i] = v_freq[i]; - } - - hSbrGrid->frameClass = frameClass; - hSbrGrid->bs_abs_bord_0 = aL; - hSbrGrid->bs_abs_bord_1 = aR; - hSbrGrid->bs_num_rel_0 = nL; - hSbrGrid->bs_num_rel_1 = nR; - hSbrGrid->p = p; - - break; - - default: - /* do nothing */ - break; - } -} - -/******************************************************************************* - Functionname: createDefFrameInfo - ******************************************************************************* - - Description: Copies the default (static) frameInfo structs to the frameInfo - passed by reference; only used for FIXFIX frames - - Arguments: hFrameInfo - HANLDE_SBR_FRAME_INFO - nEnv - INT - nTimeSlots - INT - - Return: none; hSbrFrameInfo contains a copy of the default frameInfo - - Written: Andreas Schneider - Revised: -*******************************************************************************/ -static void -createDefFrameInfo(HANDLE_SBR_FRAME_INFO hSbrFrameInfo, INT nEnv, INT nTimeSlots) -{ - switch (nEnv) { - case 1: - switch (nTimeSlots) { - case NUMBER_TIME_SLOTS_1920: - FDKmemcpy (hSbrFrameInfo, &frameInfo1_1920, sizeof (SBR_FRAME_INFO)); - break; - case NUMBER_TIME_SLOTS_2048: - FDKmemcpy (hSbrFrameInfo, &frameInfo1_2048, sizeof (SBR_FRAME_INFO)); - break; - case NUMBER_TIME_SLOTS_1152: - FDKmemcpy (hSbrFrameInfo, &frameInfo1_1152, sizeof (SBR_FRAME_INFO)); - break; - case NUMBER_TIME_SLOTS_2304: - FDKmemcpy (hSbrFrameInfo, &frameInfo1_2304, sizeof (SBR_FRAME_INFO)); - break; - case NUMBER_TIME_SLOTS_512LD: - FDKmemcpy (hSbrFrameInfo, &frameInfo1_512LD, sizeof (SBR_FRAME_INFO)); - break; - default: - FDK_ASSERT(0); - } - break; - case 2: - switch (nTimeSlots) { - case NUMBER_TIME_SLOTS_1920: - FDKmemcpy (hSbrFrameInfo, &frameInfo2_1920, sizeof (SBR_FRAME_INFO)); - break; - case NUMBER_TIME_SLOTS_2048: - FDKmemcpy (hSbrFrameInfo, &frameInfo2_2048, sizeof (SBR_FRAME_INFO)); - break; - case NUMBER_TIME_SLOTS_1152: - FDKmemcpy (hSbrFrameInfo, &frameInfo2_1152, sizeof (SBR_FRAME_INFO)); - break; - case NUMBER_TIME_SLOTS_2304: - FDKmemcpy (hSbrFrameInfo, &frameInfo2_2304, sizeof (SBR_FRAME_INFO)); - break; - case NUMBER_TIME_SLOTS_512LD: - FDKmemcpy (hSbrFrameInfo, &frameInfo2_512LD, sizeof (SBR_FRAME_INFO)); - break; - default: - FDK_ASSERT(0); - } - break; - case 4: - switch (nTimeSlots) { - case NUMBER_TIME_SLOTS_1920: - FDKmemcpy (hSbrFrameInfo, &frameInfo4_1920, sizeof (SBR_FRAME_INFO)); - break; - case NUMBER_TIME_SLOTS_2048: - FDKmemcpy (hSbrFrameInfo, &frameInfo4_2048, sizeof (SBR_FRAME_INFO)); - break; - case NUMBER_TIME_SLOTS_1152: - FDKmemcpy (hSbrFrameInfo, &frameInfo4_1152, sizeof (SBR_FRAME_INFO)); - break; - case NUMBER_TIME_SLOTS_2304: - FDKmemcpy (hSbrFrameInfo, &frameInfo4_2304, sizeof (SBR_FRAME_INFO)); - break; - case NUMBER_TIME_SLOTS_512LD: - FDKmemcpy (hSbrFrameInfo, &frameInfo4_512LD, sizeof (SBR_FRAME_INFO)); - break; - default: - FDK_ASSERT(0); - } - break; - default: - FDK_ASSERT(0); - } -} - - -/******************************************************************************* - Functionname: ctrlSignal2FrameInfo - ******************************************************************************* - - Description: Convert "clear-text" sbr_grid() to "frame info" used by the - envelope and noise floor estimators. - This is basically (except for "low level" calculations) the - bitstream decoder defined in the MPEG-4 standard, sub clause - 4.6.18.3.3, Time / Frequency Grid. See inline comments for - explanation of the shorten and noise border algorithms. - - Arguments: hSbrGrid - source - hSbrFrameInfo - destination - freq_res_fixfix - frequency resolution for FIXFIX frames - - Return: void; hSbrFrameInfo contains the updated FRAME_INFO struct - -*******************************************************************************/ -static void -ctrlSignal2FrameInfo ( - HANDLE_SBR_GRID hSbrGrid, /* input : the grid handle */ - HANDLE_SBR_FRAME_INFO hSbrFrameInfo, /* output: the frame info handle */ - FREQ_RES *freq_res_fixfix /* in/out: frequency resolution for FIXFIX frames */ - ) -{ - INT frameSplit = 0; - INT nEnv = 0, border = 0, i, k, p /*?*/; - INT *v_r = hSbrGrid->bs_rel_bord; - INT *v_f = hSbrGrid->v_f; - - FRAME_CLASS frameClass = hSbrGrid->frameClass; - INT bufferFrameStart = hSbrGrid->bufferFrameStart; - INT numberTimeSlots = hSbrGrid->numberTimeSlots; - - switch (frameClass) { - case FIXFIX: - createDefFrameInfo(hSbrFrameInfo, hSbrGrid->bs_num_env, numberTimeSlots); - - frameSplit = (hSbrFrameInfo->nEnvelopes > 1); - for (i = 0; i < hSbrFrameInfo->nEnvelopes; i++) { - hSbrGrid->v_f[i] = hSbrFrameInfo->freqRes[i] = freq_res_fixfix[frameSplit]; - } - break; - - case FIXVAR: - case VARFIX: - nEnv = hSbrGrid->n + 1; /* read n [SBR_NUM_BITS bits] */ /*? snd*/ - FDK_ASSERT(nEnv <= MAX_ENVELOPES_FIXVAR_VARFIX); - - hSbrFrameInfo->nEnvelopes = nEnv; - - border = hSbrGrid->bs_abs_bord; /* read the absolute border */ - - if (nEnv == 1) - hSbrFrameInfo->nNoiseEnvelopes = 1; - else - hSbrFrameInfo->nNoiseEnvelopes = 2; - - break; - - default: - /* do nothing */ - break; - } - - switch (frameClass) { - case FIXVAR: - hSbrFrameInfo->borders[0] = bufferFrameStart; /* start-position of 1st envelope */ - - hSbrFrameInfo->borders[nEnv] = border; - - for (k = 0, i = nEnv - 1; k < nEnv - 1; k++, i--) { - border -= v_r[k]; - hSbrFrameInfo->borders[i] = border; - } - - /* make either envelope nr. nEnv + 1 - p short; or don't shorten if p == 0 */ - p = hSbrGrid->p; - if (p == 0) { - hSbrFrameInfo->shortEnv = 0; - } else { - hSbrFrameInfo->shortEnv = nEnv + 1 - p; - } - - for (k = 0, i = nEnv - 1; k < nEnv; k++, i--) { - hSbrFrameInfo->freqRes[i] = (FREQ_RES)v_f[k]; - } - - /* if either there is no short envelope or the last envelope is short... */ - if (p == 0 || p == 1) { - hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[nEnv - 1]; - } else { - hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[hSbrFrameInfo->shortEnv]; - } - - break; - - case VARFIX: - /* in this case 'border' indicates the start of the 1st envelope */ - hSbrFrameInfo->borders[0] = border; - - for (k = 0; k < nEnv - 1; k++) { - border += v_r[k]; - hSbrFrameInfo->borders[k + 1] = border; - } - - hSbrFrameInfo->borders[nEnv] = bufferFrameStart + numberTimeSlots; - - p = hSbrGrid->p; - if (p == 0 || p == 1) { - hSbrFrameInfo->shortEnv = 0; - } else { - hSbrFrameInfo->shortEnv = p - 1; - } - - for (k = 0; k < nEnv; k++) { - hSbrFrameInfo->freqRes[k] = (FREQ_RES)v_f[k]; - } - - switch (p) { - case 0: - hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[1]; - break; - case 1: - hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[nEnv - 1]; - break; - default: - hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[hSbrFrameInfo->shortEnv]; - break; - } - break; - - case VARVAR: - nEnv = hSbrGrid->bs_num_rel_0 + hSbrGrid->bs_num_rel_1 + 1; - FDK_ASSERT(nEnv <= MAX_ENVELOPES_VARVAR); /* just to be sure */ - hSbrFrameInfo->nEnvelopes = nEnv; - - hSbrFrameInfo->borders[0] = border = hSbrGrid->bs_abs_bord_0; - - for (k = 0, i = 1; k < hSbrGrid->bs_num_rel_0; k++, i++) { - border += hSbrGrid->bs_rel_bord_0[k]; - hSbrFrameInfo->borders[i] = border; - } - - border = hSbrGrid->bs_abs_bord_1; - hSbrFrameInfo->borders[nEnv] = border; - - for (k = 0, i = nEnv - 1; k < hSbrGrid->bs_num_rel_1; k++, i--) { - border -= hSbrGrid->bs_rel_bord_1[k]; - hSbrFrameInfo->borders[i] = border; - } - - p = hSbrGrid->p; - if (p == 0) { - hSbrFrameInfo->shortEnv = 0; - } else { - hSbrFrameInfo->shortEnv = nEnv + 1 - p; - } - - for (k = 0; k < nEnv; k++) { - hSbrFrameInfo->freqRes[k] = (FREQ_RES)hSbrGrid->v_fLR[k]; - } - - if (nEnv == 1) { - hSbrFrameInfo->nNoiseEnvelopes = 1; - hSbrFrameInfo->bordersNoise[0] = hSbrGrid->bs_abs_bord_0; - hSbrFrameInfo->bordersNoise[1] = hSbrGrid->bs_abs_bord_1; - } else { - hSbrFrameInfo->nNoiseEnvelopes = 2; - hSbrFrameInfo->bordersNoise[0] = hSbrGrid->bs_abs_bord_0; - - if (p == 0 || p == 1) { - hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[nEnv - 1]; - } else { - hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[hSbrFrameInfo->shortEnv]; - } - hSbrFrameInfo->bordersNoise[2] = hSbrGrid->bs_abs_bord_1; - } - break; - - default: - /* do nothing */ - break; - } - - if (frameClass == VARFIX || frameClass == FIXVAR) { - hSbrFrameInfo->bordersNoise[0] = hSbrFrameInfo->borders[0]; - if (nEnv == 1) { - hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[nEnv]; - } else { - hSbrFrameInfo->bordersNoise[2] = hSbrFrameInfo->borders[nEnv]; - } - } -} - diff --git a/libSBRenc/src/fram_gen.h b/libSBRenc/src/fram_gen.h deleted file mode 100644 index 00473d4..0000000 --- a/libSBRenc/src/fram_gen.h +++ /dev/null @@ -1,309 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief Framing generator prototypes and structs -*/ -#ifndef _FRAM_GEN_H -#define _FRAM_GEN_H - -#include "sbr_def.h" /* for MAX_ENVELOPES and MAX_NOISE_ENVELOPES in struct FRAME_INFO and CODEC_TYPE */ -#include "sbr_encoder.h" /* for FREQ_RES */ - -#define MAX_ENVELOPES_VARVAR MAX_ENVELOPES /*!< worst case number of envelopes in a VARVAR frame */ -#define MAX_ENVELOPES_FIXVAR_VARFIX 4 /*!< worst case number of envelopes in VARFIX and FIXVAR frames */ -#define MAX_NUM_REL 3 /*!< maximum number of relative borders in any VAR frame */ - -/* SBR frame class definitions */ -typedef enum { - FIXFIX = 0, /*!< bs_frame_class: leading and trailing frame borders are fixed */ - FIXVAR, /*!< bs_frame_class: leading frame border is fixed, trailing frame border is variable */ - VARFIX, /*!< bs_frame_class: leading frame border is variable, trailing frame border is fixed */ - VARVAR /*!< bs_frame_class: leading and trailing frame borders are variable */ - ,FIXFIXonly /*!< bs_frame_class: leading border fixed (0), trailing border fixed (nrTimeSlots) and encased borders are dynamically derived from the tranPos */ -}FRAME_CLASS; - - -/* helper constants */ -#define DC 4711 /*!< helper constant: don't care */ -#define EMPTY (-99) /*!< helper constant: empty */ - - -/* system constants: AAC+SBR, DRM Frame-Length */ -#define FRAME_MIDDLE_SLOT_1920 4 -#define NUMBER_TIME_SLOTS_1920 15 - -#define LD_PRETRAN_OFF 3 -#define FRAME_MIDDLE_SLOT_512LD 4 -#define NUMBER_TIME_SLOTS_512LD 8 -#define TRANSIENT_OFFSET_LD 0 - - - -/* -system constants: AAC+SBR or aacPRO (hybrid format), Standard Frame-Length, Multi-Rate ---------------------------------------------------------------------------- -Number of slots (numberTimeSlots): 16 (NUMBER_TIME_SLOTS_2048) -Detector-offset (frameMiddleSlot): 4 -Overlap : 3 -Buffer-offset : 8 (BUFFER_FRAME_START_2048 = 0) - - - |<------------tranPos---------->| - |c|d|e|f|0|1|2|3|4|5|6|7|8|9|a|b|c|d|e|f| - FixFix | | - FixVar | :<- ->: - VarFix :<- ->: | - VarVar :<- ->: :<- ->: - 0 1 2 3 4 5 6 7 8 9 a b c d e f 0 1 2 3 -................................................................................ - -|-|-|-|-|-|-|-|-B-|-|-|-|-|-|-|-|-|-|-|-|-|-|-|-B-|-|-|-|-|-|-|-|-|-|-|-|-|-|-|-| - -frame-generator:0 16 24 32 -analysis-buffer:8 24 32 40 -*/ -#define FRAME_MIDDLE_SLOT_2048 4 -#define NUMBER_TIME_SLOTS_2048 16 - - -/* -system constants: mp3PRO, Multi-Rate & Single-Rate --------------------------------------------------- -Number of slots (numberTimeSlots): 9 (NUMBER_TIME_SLOTS_1152) -Detector-offset (frameMiddleSlot): 4 (FRAME_MIDDLE_SLOT_1152) -Overlap : 3 -Buffer-offset : 4.5 (BUFFER_FRAME_START_1152 = 0) - - - |<----tranPos---->| - |5|6|7|8|0|1|2|3|4|5|6|7|8| - FixFix | | - FixVar | :<- ->: - VarFix :<- ->: | - VarVar :<- ->: :<- ->: - 0 1 2 3 4 5 6 7 8 0 1 2 3 - ............................................. - - -|-|-|-|-B-|-|-|-|-|-|-|-|-B-|-|-|-|-|-|-|-|-| - -frame-generator: 0 9 13 18 -analysis-buffer: 4.5 13.5 22.5 -*/ -#define FRAME_MIDDLE_SLOT_1152 4 -#define NUMBER_TIME_SLOTS_1152 9 - - -/* system constants: Layer2+SBR */ -#define FRAME_MIDDLE_SLOT_2304 8 -#define NUMBER_TIME_SLOTS_2304 18 - - -/*! - \struct SBR_GRID - \brief sbr_grid() signals to be converted to bitstream elements - - The variables hold the signals (e.g. lengths and numbers) in "clear text" -*/ - -typedef struct -{ - /* system constants */ - INT bufferFrameStart; /*!< frame generator vs analysis buffer time alignment (currently set to 0, offset added elsewhere) */ - INT numberTimeSlots; /*!< number of SBR timeslots per frame */ - - /* will be adjusted for every frame */ - FRAME_CLASS frameClass; /*!< SBR frame class */ - INT bs_num_env; /*!< bs_num_env, number of envelopes for FIXFIX */ - INT bs_abs_bord; /*!< bs_abs_bord, absolute border for VARFIX and FIXVAR */ - INT n; /*!< number of relative borders for VARFIX and FIXVAR */ - INT p; /*!< pointer-to-transient-border */ - INT bs_rel_bord[MAX_NUM_REL]; /*!< bs_rel_bord, relative borders for all VAR */ - INT v_f[MAX_ENVELOPES_FIXVAR_VARFIX]; /*!< envelope frequency resolutions for FIXVAR and VARFIX */ - - INT bs_abs_bord_0; /*!< bs_abs_bord_0, leading absolute border for VARVAR */ - INT bs_abs_bord_1; /*!< bs_abs_bord_1, trailing absolute border for VARVAR */ - INT bs_num_rel_0; /*!< bs_num_rel_0, number of relative borders associated with leading absolute border for VARVAR */ - INT bs_num_rel_1; /*!< bs_num_rel_1, number of relative borders associated with trailing absolute border for VARVAR */ - INT bs_rel_bord_0[MAX_NUM_REL]; /*!< bs_rel_bord_0, relative borders associated with leading absolute border for VARVAR */ - INT bs_rel_bord_1[MAX_NUM_REL]; /*!< bs_rel_bord_1, relative borders associated with trailing absolute border for VARVAR */ - INT v_fLR[MAX_ENVELOPES_VARVAR]; /*!< envelope frequency resolutions for VARVAR */ - -} -SBR_GRID; -typedef SBR_GRID *HANDLE_SBR_GRID; - - - -/*! - \struct SBR_FRAME_INFO - \brief time/frequency grid description for one frame -*/ -typedef struct -{ - INT nEnvelopes; /*!< number of envelopes */ - INT borders[MAX_ENVELOPES+1]; /*!< envelope borders in SBR timeslots */ - FREQ_RES freqRes[MAX_ENVELOPES]; /*!< frequency resolution of each envelope */ - INT shortEnv; /*!< number of an envelope to be shortened (starting at 1) or 0 for no shortened envelope */ - INT nNoiseEnvelopes; /*!< number of noise floors */ - INT bordersNoise[MAX_NOISE_ENVELOPES+1];/*!< noise floor borders in SBR timeslots */ -} -SBR_FRAME_INFO; -/* WARNING: When rearranging the elements of this struct keep in mind that the static - * initializations in the corresponding C-file have to be rearranged as well! - * snd 2002/01/23 - */ -typedef SBR_FRAME_INFO *HANDLE_SBR_FRAME_INFO; - - -/*! - \struct SBR_ENVELOPE_FRAME - \brief frame generator main struct - - Contains tuning parameters, time/frequency grid description, sbr_grid() bitstream elements, and generator internal signals -*/ -typedef struct -{ - /* system constants */ - INT frameMiddleSlot; /*!< transient detector offset in SBR timeslots */ - - /* basic tuning parameters */ - INT staticFraming; /*!< 1: run static framing in time, i.e. exclusive use of bs_frame_class = FIXFIX */ - INT numEnvStatic; /*!< number of envelopes per frame for static framing */ - FREQ_RES freq_res_fixfix[2]; /*!< envelope frequency resolution to use for bs_frame_class = FIXFIX; single env and split */ - UCHAR fResTransIsLow; /*!< frequency resolution for transient frames - always low (0) or according to table (1) */ - - /* expert tuning parameters */ - const int *v_tuningSegm; /*!< segment lengths to use around transient */ - const int *v_tuningFreq; /*!< frequency resolutions to use around transient */ - INT dmin; /*!< minimum length of dependent segments */ - INT dmax; /*!< maximum length of dependent segments */ - INT allowSpread; /*!< 1: allow isolated transient to influence grid of 3 consecutive frames */ - - /* internally used signals */ - FRAME_CLASS frameClassOld; /*!< frame class used for previous frame */ - INT spreadFlag; /*!< 1: use VARVAR instead of VARFIX to follow up old transient */ - - INT v_bord[2 * MAX_ENVELOPES_VARVAR + 1]; /*!< borders for current frame and preliminary borders for next frame (fixed borders excluded) */ - INT length_v_bord; /*!< helper variable: length of v_bord */ - INT v_freq[2 * MAX_ENVELOPES_VARVAR + 1]; /*!< frequency resolutions for current frame and preliminary resolutions for next frame */ - INT length_v_freq; /*!< helper variable: length of v_freq */ - - INT v_bordFollow[MAX_ENVELOPES_VARVAR]; /*!< preliminary borders for current frame (calculated during previous frame) */ - INT length_v_bordFollow; /*!< helper variable: length of v_bordFollow */ - INT i_tranFollow; /*!< points to transient border in v_bordFollow (may be negative, see keepForFollowUp()) */ - INT i_fillFollow; /*!< points to first fill border in v_bordFollow */ - INT v_freqFollow[MAX_ENVELOPES_VARVAR]; /*!< preliminary frequency resolutions for current frame (calculated during previous frame) */ - INT length_v_freqFollow; /*!< helper variable: length of v_freqFollow */ - - - /* externally needed signals */ - SBR_GRID SbrGrid; /*!< sbr_grid() signals to be converted to bitstream elements */ - SBR_FRAME_INFO SbrFrameInfo; /*!< time/frequency grid description for one frame */ -} -SBR_ENVELOPE_FRAME; -typedef SBR_ENVELOPE_FRAME *HANDLE_SBR_ENVELOPE_FRAME; - - - -void -FDKsbrEnc_initFrameInfoGenerator ( - HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame, - INT allowSpread, - INT numEnvStatic, - INT staticFraming, - INT timeSlots, - const FREQ_RES* freq_res_fixfix - ,UCHAR fResTransIsLow, - INT ldGrid - ); - -HANDLE_SBR_FRAME_INFO -FDKsbrEnc_frameInfoGenerator (HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame, - UCHAR *v_transient_info, - UCHAR *v_transient_info_pre, - int ldGrid, - const int *v_tuning); - -#endif diff --git a/libSBRenc/src/invf_est.cpp b/libSBRenc/src/invf_est.cpp deleted file mode 100644 index 32df6d9..0000000 --- a/libSBRenc/src/invf_est.cpp +++ /dev/null @@ -1,529 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -#include "invf_est.h" -#include "sbr_misc.h" - -#include "genericStds.h" - -#define MAX_NUM_REGIONS 10 -#define SCALE_FAC_QUO 512.0f -#define SCALE_FAC_NRG 256.0f - -#ifndef min -#define min(a,b) ( a < b ? a:b) -#endif - -#ifndef max -#define max(a,b) ( a > b ? a:b) -#endif - -static const FIXP_DBL quantStepsSbr[4] = { 0x00400000, 0x02800000, 0x03800000, 0x04c00000 } ; /* table scaled with SCALE_FAC_QUO */ -static const FIXP_DBL quantStepsOrig[4] = { 0x00000000, 0x00c00000, 0x01c00000, 0x02800000 } ; /* table scaled with SCALE_FAC_QUO */ -static const FIXP_DBL nrgBorders[4] = { 0x0c800000, 0x0f000000, 0x11800000, 0x14000000 } ; /* table scaled with SCALE_FAC_NRG */ - -static const DETECTOR_PARAMETERS detectorParamsAAC = { - quantStepsSbr, - quantStepsOrig, - nrgBorders, - 4, /* Number of borders SBR. */ - 4, /* Number of borders orig. */ - 4, /* Number of borders Nrg. */ - { /* Region space. */ - {INVF_MID_LEVEL, INVF_LOW_LEVEL, INVF_OFF, INVF_OFF, INVF_OFF}, /* | */ - {INVF_MID_LEVEL, INVF_LOW_LEVEL, INVF_OFF, INVF_OFF, INVF_OFF}, /* | */ - {INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_LOW_LEVEL, INVF_OFF, INVF_OFF}, /* regionSbr */ - {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF, INVF_OFF}, /* | */ - {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF, INVF_OFF} /* | */ - },/*------------------------ regionOrig ---------------------------------*/ - { /* Region space transient. */ - {INVF_LOW_LEVEL, INVF_LOW_LEVEL, INVF_LOW_LEVEL, INVF_OFF, INVF_OFF}, /* | */ - {INVF_LOW_LEVEL, INVF_LOW_LEVEL, INVF_LOW_LEVEL, INVF_OFF, INVF_OFF}, /* | */ - {INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_MID_LEVEL, INVF_OFF, INVF_OFF}, /* regionSbr */ - {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF, INVF_OFF}, /* | */ - {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF, INVF_OFF} /* | */ - },/*------------------------ regionOrig ---------------------------------*/ - {-4, -3, -2, -1, 0} /* Reduction factor of the inverse filtering for low energies.*/ -}; - -static const FIXP_DBL hysteresis = 0x00400000 ; /* Delta value for hysteresis. scaled with SCALE_FAC_QUO */ - -/* - * AAC+SBR PARAMETERS for Speech - *********************************/ -static const DETECTOR_PARAMETERS detectorParamsAACSpeech = { - quantStepsSbr, - quantStepsOrig, - nrgBorders, - 4, /* Number of borders SBR. */ - 4, /* Number of borders orig. */ - 4, /* Number of borders Nrg. */ - { /* Region space. */ - {INVF_MID_LEVEL, INVF_MID_LEVEL, INVF_LOW_LEVEL, INVF_OFF, INVF_OFF}, /* | */ - {INVF_MID_LEVEL, INVF_MID_LEVEL, INVF_LOW_LEVEL, INVF_OFF, INVF_OFF}, /* | */ - {INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_MID_LEVEL, INVF_OFF, INVF_OFF}, /* regionSbr */ - {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF, INVF_OFF}, /* | */ - {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF, INVF_OFF} /* | */ - },/*------------------------ regionOrig ---------------------------------*/ - { /* Region space transient. */ - {INVF_MID_LEVEL, INVF_MID_LEVEL, INVF_LOW_LEVEL, INVF_OFF, INVF_OFF}, /* | */ - {INVF_MID_LEVEL, INVF_MID_LEVEL, INVF_LOW_LEVEL, INVF_OFF, INVF_OFF}, /* | */ - {INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_MID_LEVEL, INVF_OFF, INVF_OFF}, /* regionSbr */ - {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF, INVF_OFF}, /* | */ - {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF, INVF_OFF} /* | */ - },/*------------------------ regionOrig ---------------------------------*/ - {-4, -3, -2, -1, 0} /* Reduction factor of the inverse filtering for low energies.*/ -}; - -/* - * Smoothing filters. - ************************/ -typedef const FIXP_DBL FIR_FILTER[5]; - -static const FIR_FILTER fir_0 = { 0x7fffffff, 0x00000000, 0x00000000, 0x00000000, 0x00000000 } ; -static const FIR_FILTER fir_1 = { 0x2aaaaa80, 0x555554ff, 0x00000000, 0x00000000, 0x00000000 } ; -static const FIR_FILTER fir_2 = { 0x10000000, 0x30000000, 0x40000000, 0x00000000, 0x00000000 } ; -static const FIR_FILTER fir_3 = { 0x077f80e8, 0x199999a0, 0x2bb3b240, 0x33333340, 0x00000000 } ; -static const FIR_FILTER fir_4 = { 0x04130598, 0x0ebdb000, 0x1becfa60, 0x2697a4c0, 0x2aaaaa80 } ; - - -static const FIR_FILTER *const fir_table[5] = { - &fir_0, - &fir_1, - &fir_2, - &fir_3, - &fir_4 -}; - -/**************************************************************************/ -/*! - \brief Calculates the values used for the detector. - - - \return none - -*/ -/**************************************************************************/ -static void -calculateDetectorValues(FIXP_DBL **quotaMatrixOrig, /*!< Matrix holding the tonality values of the original. */ - SCHAR *indexVector, /*!< Index vector to obtain the patched data. */ - FIXP_DBL *nrgVector, /*!< Energy vector. */ - DETECTOR_VALUES *detectorValues, /*!< pointer to DETECTOR_VALUES struct. */ - INT startChannel, /*!< Start channel. */ - INT stopChannel, /*!< Stop channel. */ - INT startIndex, /*!< Start index. */ - INT stopIndex, /*!< Stop index. */ - INT numberOfStrongest /*!< The number of sorted tonal components to be considered. */ - ) -{ - INT i,temp, j; - - const FIXP_DBL* filter = *fir_table[INVF_SMOOTHING_LENGTH]; - FIXP_DBL origQuotaMeanStrongest, sbrQuotaMeanStrongest; - FIXP_DBL origQuota, sbrQuota; - FIXP_DBL invIndex, invChannel, invTemp; - FIXP_DBL quotaVecOrig[64], quotaVecSbr[64]; - - FDKmemclear(quotaVecOrig,64*sizeof(FIXP_DBL)); - FDKmemclear(quotaVecSbr,64*sizeof(FIXP_DBL)); - - invIndex = GetInvInt(stopIndex-startIndex); - invChannel = GetInvInt(stopChannel-startChannel); - - /* - Calculate the mean value, over the current time segment, for the original, the HFR - and the difference, over all channels in the current frequency range. - NOTE: the averaging is done on the values quota/(1 - quota + RELAXATION). - */ - - /* The original, the sbr signal and the total energy */ - detectorValues->avgNrg = FL2FXCONST_DBL(0.0f); - for(j=startIndex; javgNrg += fMult(nrgVector[j], invIndex); - } - - /* - Calculate the mean value, over the current frequency range, for the original, the HFR - and the difference. Also calculate the same mean values for the three vectors, but only - includeing the x strongest copmponents. - */ - - origQuota = FL2FXCONST_DBL(0.0f); - sbrQuota = FL2FXCONST_DBL(0.0f); - for(i=startChannel; iorigQuotaMax = quotaVecOrig[stopChannel - 1]; - detectorValues->sbrQuotaMax = quotaVecSbr[stopChannel - 1]; - - /* - Buffer values - */ - FDKmemmove(detectorValues->origQuotaMean, detectorValues->origQuotaMean + 1, INVF_SMOOTHING_LENGTH*sizeof(FIXP_DBL)); - FDKmemmove(detectorValues->sbrQuotaMean, detectorValues->sbrQuotaMean + 1, INVF_SMOOTHING_LENGTH*sizeof(FIXP_DBL)); - FDKmemmove(detectorValues->origQuotaMeanStrongest, detectorValues->origQuotaMeanStrongest + 1, INVF_SMOOTHING_LENGTH*sizeof(FIXP_DBL)); - FDKmemmove(detectorValues->sbrQuotaMeanStrongest, detectorValues->sbrQuotaMeanStrongest + 1, INVF_SMOOTHING_LENGTH*sizeof(FIXP_DBL)); - - detectorValues->origQuotaMean[INVF_SMOOTHING_LENGTH] = origQuota<<1; - detectorValues->sbrQuotaMean[INVF_SMOOTHING_LENGTH] = sbrQuota<<1; - detectorValues->origQuotaMeanStrongest[INVF_SMOOTHING_LENGTH] = origQuotaMeanStrongest<<1; - detectorValues->sbrQuotaMeanStrongest[INVF_SMOOTHING_LENGTH] = sbrQuotaMeanStrongest<<1; - - /* - Filter values - */ - detectorValues->origQuotaMeanFilt = FL2FXCONST_DBL(0.0f); - detectorValues->sbrQuotaMeanFilt = FL2FXCONST_DBL(0.0f); - detectorValues->origQuotaMeanStrongestFilt = FL2FXCONST_DBL(0.0f); - detectorValues->sbrQuotaMeanStrongestFilt = FL2FXCONST_DBL(0.0f); - - for(i=0;iorigQuotaMeanFilt += fMult(detectorValues->origQuotaMean[i], filter[i]); - detectorValues->sbrQuotaMeanFilt += fMult(detectorValues->sbrQuotaMean[i], filter[i]); - detectorValues->origQuotaMeanStrongestFilt += fMult(detectorValues->origQuotaMeanStrongest[i], filter[i]); - detectorValues->sbrQuotaMeanStrongestFilt += fMult(detectorValues->sbrQuotaMeanStrongest[i], filter[i]); - } -} - -/**************************************************************************/ -/*! - \brief Returns the region in which the input value belongs. - - - - \return region. - -*/ -/**************************************************************************/ -static INT -findRegion(FIXP_DBL currVal, /*!< The current value. */ - const FIXP_DBL *borders, /*!< The border of the regions. */ - const INT numBorders /*!< The number of borders. */ - ) -{ - INT i; - - if(currVal < borders[0]){ - return 0; - } - - for(i = 1; i < numBorders; i++){ - if( currVal >= borders[i-1] && currVal < borders[i]){ - return i; - } - } - - if(currVal >= borders[numBorders-1]){ - return numBorders; - } - - return 0; /* We never get here, it's just to avoid compiler warnings.*/ -} - -/**************************************************************************/ -/*! - \brief Makes a clever decision based on the quota vector. - - - \return decision on which invf mode to use - -*/ -/**************************************************************************/ -static INVF_MODE -decisionAlgorithm(const DETECTOR_PARAMETERS *detectorParams, /*!< Struct with the detector parameters. */ - DETECTOR_VALUES *detectorValues, /*!< Struct with the detector values. */ - INT transientFlag, /*!< Flag indicating if there is a transient present.*/ - INT* prevRegionSbr, /*!< The previous region in which the Sbr value was. */ - INT* prevRegionOrig /*!< The previous region in which the Orig value was. */ - ) -{ - INT invFiltLevel, regionSbr, regionOrig, regionNrg; - - /* - Current thresholds. - */ - const FIXP_DBL *quantStepsSbr = detectorParams->quantStepsSbr; - const FIXP_DBL *quantStepsOrig = detectorParams->quantStepsOrig; - const FIXP_DBL *nrgBorders = detectorParams->nrgBorders; - const INT numRegionsSbr = detectorParams->numRegionsSbr; - const INT numRegionsOrig = detectorParams->numRegionsOrig; - const INT numRegionsNrg = detectorParams->numRegionsNrg; - - FIXP_DBL quantStepsSbrTmp[MAX_NUM_REGIONS]; - FIXP_DBL quantStepsOrigTmp[MAX_NUM_REGIONS]; - - /* - Current detector values. - */ - FIXP_DBL origQuotaMeanFilt; - FIXP_DBL sbrQuotaMeanFilt; - FIXP_DBL nrg; - - /* 0.375 = 3.0 / 8.0; 0.31143075889 = log2(RELAXATION)/64.0; 0.625 = log(16)/64.0; 0.6875 = 44/64.0 */ - origQuotaMeanFilt = (fMultDiv2(FL2FXCONST_DBL(2.f*0.375f), (FIXP_DBL)(CalcLdData(max(detectorValues->origQuotaMeanFilt,(FIXP_DBL)1)) + FL2FXCONST_DBL(0.31143075889f)))) << 0; /* scaled by 1/2^9 */ - sbrQuotaMeanFilt = (fMultDiv2(FL2FXCONST_DBL(2.f*0.375f), (FIXP_DBL)(CalcLdData(max(detectorValues->sbrQuotaMeanFilt,(FIXP_DBL)1)) + FL2FXCONST_DBL(0.31143075889f)))) << 0; /* scaled by 1/2^9 */ - /* If energy is zero then we will get different results for different word lengths. */ - nrg = (fMultDiv2(FL2FXCONST_DBL(2.f*0.375f), (FIXP_DBL)(CalcLdData(detectorValues->avgNrg+(FIXP_DBL)1) + FL2FXCONST_DBL(0.0625f) + FL2FXCONST_DBL(0.6875f)))) << 0; /* scaled by 1/2^8; 2^44 -> qmf energy scale */ - - FDKmemcpy(quantStepsSbrTmp,quantStepsSbr,numRegionsSbr*sizeof(FIXP_DBL)); - FDKmemcpy(quantStepsOrigTmp,quantStepsOrig,numRegionsOrig*sizeof(FIXP_DBL)); - - if(*prevRegionSbr < numRegionsSbr) - quantStepsSbrTmp[*prevRegionSbr] = quantStepsSbr[*prevRegionSbr] + hysteresis; - if(*prevRegionSbr > 0) - quantStepsSbrTmp[*prevRegionSbr - 1] = quantStepsSbr[*prevRegionSbr - 1] - hysteresis; - - if(*prevRegionOrig < numRegionsOrig) - quantStepsOrigTmp[*prevRegionOrig] = quantStepsOrig[*prevRegionOrig] + hysteresis; - if(*prevRegionOrig > 0) - quantStepsOrigTmp[*prevRegionOrig - 1] = quantStepsOrig[*prevRegionOrig - 1] - hysteresis; - - regionSbr = findRegion(sbrQuotaMeanFilt, quantStepsSbrTmp, numRegionsSbr); - regionOrig = findRegion(origQuotaMeanFilt, quantStepsOrigTmp, numRegionsOrig); - regionNrg = findRegion(nrg,nrgBorders,numRegionsNrg); - - *prevRegionSbr = regionSbr; - *prevRegionOrig = regionOrig; - - /* Use different settings if a transient is present*/ - invFiltLevel = (transientFlag == 1) ? detectorParams->regionSpaceTransient[regionSbr][regionOrig] - : detectorParams->regionSpace[regionSbr][regionOrig]; - - /* Compensate for low energy.*/ - invFiltLevel = max(invFiltLevel + detectorParams->EnergyCompFactor[regionNrg],0); - - return (INVF_MODE) (invFiltLevel); -} - -/**************************************************************************/ -/*! - \brief Estiamtion of the inverse filtering level required - in the decoder. - - A second order LPC is calculated for every filterbank channel, using - the covariance method. THe ratio between the energy of the predicted - signal and the energy of the non-predictable signal is calcualted. - - \return none. - -*/ -/**************************************************************************/ -void -FDKsbrEnc_qmfInverseFilteringDetector(HANDLE_SBR_INV_FILT_EST hInvFilt, /*!< Handle to the SBR_INV_FILT_EST struct. */ - FIXP_DBL **quotaMatrix, /*!< The matrix holding the tonality values of the original. */ - FIXP_DBL *nrgVector, /*!< The energy vector. */ - SCHAR *indexVector, /*!< Index vector to obtain the patched data. */ - INT startIndex, /*!< Start index. */ - INT stopIndex, /*!< Stop index. */ - INT transientFlag, /*!< Flag indicating if a transient is present or not.*/ - INVF_MODE* infVec /*!< Vector holding the inverse filtering levels. */ - ) -{ - INT band; - - /* - * Do the inverse filtering level estimation. - *****************************************************/ - for(band = 0 ; band < hInvFilt->noDetectorBands; band++){ - INT startChannel = hInvFilt->freqBandTableInvFilt[band]; - INT stopChannel = hInvFilt->freqBandTableInvFilt[band+1]; - - - calculateDetectorValues( quotaMatrix, - indexVector, - nrgVector, - &hInvFilt->detectorValues[band], - startChannel, - stopChannel, - startIndex, - stopIndex, - hInvFilt->numberOfStrongest); - - infVec[band]= decisionAlgorithm( hInvFilt->detectorParams, - &hInvFilt->detectorValues[band], - transientFlag, - &hInvFilt->prevRegionSbr[band], - &hInvFilt->prevRegionOrig[band]); - } - -} - - -/**************************************************************************/ -/*! - \brief Initialize an instance of the inverse filtering level estimator. - - - \return errorCode, noError if successful. - -*/ -/**************************************************************************/ -INT -FDKsbrEnc_initInvFiltDetector (HANDLE_SBR_INV_FILT_EST hInvFilt, /*!< Pointer to a handle to the SBR_INV_FILT_EST struct. */ - INT* freqBandTableDetector, /*!< Frequency band table for the inverse filtering. */ - INT numDetectorBands, /*!< Number of inverse filtering bands. */ - UINT useSpeechConfig /*!< Flag: adapt tuning parameters according to speech*/ - ) -{ - INT i; - - FDKmemclear( hInvFilt,sizeof(SBR_INV_FILT_EST)); - - hInvFilt->detectorParams = (useSpeechConfig) ? &detectorParamsAACSpeech - : &detectorParamsAAC ; - - hInvFilt->noDetectorBandsMax = numDetectorBands; - - /* - Memory initialisation - */ - for(i=0;inoDetectorBandsMax;i++){ - FDKmemclear(&hInvFilt->detectorValues[i], sizeof(DETECTOR_VALUES)); - hInvFilt->prevInvfMode[i] = INVF_OFF; - hInvFilt->prevRegionOrig[i] = 0; - hInvFilt->prevRegionSbr[i] = 0; - } - - /* - Reset the inverse fltering detector. - */ - FDKsbrEnc_resetInvFiltDetector(hInvFilt, - freqBandTableDetector, - hInvFilt->noDetectorBandsMax); - - return (0); -} - - -/**************************************************************************/ -/*! - \brief resets sbr inverse filtering structure. - - - - \return errorCode, noError if successful. - -*/ -/**************************************************************************/ -INT -FDKsbrEnc_resetInvFiltDetector(HANDLE_SBR_INV_FILT_EST hInvFilt, /*!< Handle to the SBR_INV_FILT_EST struct. */ - INT* freqBandTableDetector, /*!< Frequency band table for the inverse filtering. */ - INT numDetectorBands) /*!< Number of inverse filtering bands. */ -{ - - hInvFilt->numberOfStrongest = 1; - FDKmemcpy(hInvFilt->freqBandTableInvFilt,freqBandTableDetector,(numDetectorBands+1)*sizeof(INT)); - hInvFilt->noDetectorBands = numDetectorBands; - - return (0); -} - - diff --git a/libSBRenc/src/invf_est.h b/libSBRenc/src/invf_est.h deleted file mode 100644 index 2bd2a78..0000000 --- a/libSBRenc/src/invf_est.h +++ /dev/null @@ -1,175 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief Inverse Filtering detection prototypes -*/ -#ifndef _INV_FILT_DET_H -#define _INV_FILT_DET_H - -#include "sbr_encoder.h" -#include "sbr_def.h" - -#define INVF_SMOOTHING_LENGTH 2 - -typedef struct -{ - const FIXP_DBL *quantStepsSbr; - const FIXP_DBL *quantStepsOrig; - const FIXP_DBL *nrgBorders; - INT numRegionsSbr; - INT numRegionsOrig; - INT numRegionsNrg; - INVF_MODE regionSpace[5][5]; - INVF_MODE regionSpaceTransient[5][5]; - INT EnergyCompFactor[5]; - -}DETECTOR_PARAMETERS; - -typedef struct -{ - FIXP_DBL origQuotaMean[INVF_SMOOTHING_LENGTH+1]; - FIXP_DBL sbrQuotaMean[INVF_SMOOTHING_LENGTH+1]; - FIXP_DBL origQuotaMeanStrongest[INVF_SMOOTHING_LENGTH+1]; - FIXP_DBL sbrQuotaMeanStrongest[INVF_SMOOTHING_LENGTH+1]; - - FIXP_DBL origQuotaMeanFilt; - FIXP_DBL sbrQuotaMeanFilt; - FIXP_DBL origQuotaMeanStrongestFilt; - FIXP_DBL sbrQuotaMeanStrongestFilt; - - FIXP_DBL origQuotaMax; - FIXP_DBL sbrQuotaMax; - - FIXP_DBL avgNrg; -}DETECTOR_VALUES; - - - -typedef struct -{ - INT numberOfStrongest; - - INT prevRegionSbr[MAX_NUM_NOISE_VALUES]; - INT prevRegionOrig[MAX_NUM_NOISE_VALUES]; - - INT freqBandTableInvFilt[MAX_NUM_NOISE_VALUES]; - INT noDetectorBands; - INT noDetectorBandsMax; - - const DETECTOR_PARAMETERS *detectorParams; - - INVF_MODE prevInvfMode[MAX_NUM_NOISE_VALUES]; - DETECTOR_VALUES detectorValues[MAX_NUM_NOISE_VALUES]; - - FIXP_DBL nrgAvg; - FIXP_DBL wmQmf[MAX_NUM_NOISE_VALUES]; -} -SBR_INV_FILT_EST; - -typedef SBR_INV_FILT_EST *HANDLE_SBR_INV_FILT_EST; - -void -FDKsbrEnc_qmfInverseFilteringDetector(HANDLE_SBR_INV_FILT_EST hInvFilt, - FIXP_DBL ** quotaMatrix, - FIXP_DBL *nrgVector, - SCHAR *indexVector, - INT startIndex, - INT stopIndex, - INT transientFlag, - INVF_MODE* infVec); - -INT -FDKsbrEnc_initInvFiltDetector (HANDLE_SBR_INV_FILT_EST hInvFilt, - INT* freqBandTableDetector, - INT numDetectorBands, - UINT useSpeechConfig); - -INT -FDKsbrEnc_resetInvFiltDetector(HANDLE_SBR_INV_FILT_EST hInvFilt, - INT* freqBandTableDetector, - INT numDetectorBands); - -#endif /* _QMF_INV_FILT_H */ - diff --git a/libSBRenc/src/mh_det.cpp b/libSBRenc/src/mh_det.cpp deleted file mode 100644 index bc80a15..0000000 --- a/libSBRenc/src/mh_det.cpp +++ /dev/null @@ -1,1471 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -#include "mh_det.h" - -#include "sbr_ram.h" -#include "sbr_misc.h" - - -#include "genericStds.h" - -#define SFM_SHIFT 2 /* Attention: SFM_SCALE depends on SFM_SHIFT */ -#define SFM_SCALE (MAXVAL_DBL >> SFM_SHIFT) /* 1.0 >> SFM_SHIFT */ - - -/*!< Detector Parameters for AAC core codec. */ -static const DETECTOR_PARAMETERS_MH paramsAac = { -9, /*!< deltaTime */ -{ -FL2FXCONST_DBL(20.0f*RELAXATION_FLOAT), /*!< thresHoldDiff */ -FL2FXCONST_DBL(1.26f*RELAXATION_FLOAT), /*!< thresHoldDiffGuide */ -FL2FXCONST_DBL(15.0f*RELAXATION_FLOAT), /*!< thresHoldTone */ -FL2FXCONST_DBL((1.0f/15.0f)*RELAXATION_FLOAT), /*!< invThresHoldTone */ -FL2FXCONST_DBL(1.26f*RELAXATION_FLOAT), /*!< thresHoldToneGuide */ -FL2FXCONST_DBL(0.3f)>>SFM_SHIFT, /*!< sfmThresSbr */ -FL2FXCONST_DBL(0.1f)>>SFM_SHIFT, /*!< sfmThresOrig */ -FL2FXCONST_DBL(0.3f), /*!< decayGuideOrig */ -FL2FXCONST_DBL(0.5f), /*!< decayGuideDiff */ -FL2FXCONST_DBL(-0.000112993269), /* LD64(FL2FXCONST_DBL(0.995f)) */ /*!< derivThresMaxLD64 */ -FL2FXCONST_DBL(-0.000112993269), /* LD64(FL2FXCONST_DBL(0.995f)) */ /*!< derivThresBelowLD64 */ -FL2FXCONST_DBL(-0.005030126483f) /* LD64(FL2FXCONST_DBL(0.8f)) */ /*!< derivThresAboveLD64 */ -}, -50 /*!< maxComp */ -}; - -/*!< Detector Parameters for AAC LD core codec. */ -static const DETECTOR_PARAMETERS_MH paramsAacLd = { -16, /*!< Delta time. */ -{ -FL2FXCONST_DBL(25.0f*RELAXATION_FLOAT), /*!< thresHoldDiff */ -FL2FXCONST_DBL(1.26f*RELAXATION_FLOAT), /*!< tresHoldDiffGuide */ -FL2FXCONST_DBL(15.0f*RELAXATION_FLOAT), /*!< thresHoldTone */ -FL2FXCONST_DBL((1.0f/15.0f)*RELAXATION_FLOAT), /*!< invThresHoldTone */ -FL2FXCONST_DBL(1.26f*RELAXATION_FLOAT), /*!< thresHoldToneGuide */ -FL2FXCONST_DBL(0.3f)>>SFM_SHIFT, /*!< sfmThresSbr */ -FL2FXCONST_DBL(0.1f)>>SFM_SHIFT, /*!< sfmThresOrig */ -FL2FXCONST_DBL(0.3f), /*!< decayGuideOrig */ -FL2FXCONST_DBL(0.2f), /*!< decayGuideDiff */ -FL2FXCONST_DBL(-0.000112993269), /* LD64(FL2FXCONST_DBL(0.995f)) */ /*!< derivThresMaxLD64 */ -FL2FXCONST_DBL(-0.000112993269), /* LD64(FL2FXCONST_DBL(0.995f)) */ /*!< derivThresBelowLD64 */ -FL2FXCONST_DBL(-0.005030126483f) /* LD64(FL2FXCONST_DBL(0.8f)) */ /*!< derivThresAboveLD64 */ -}, -50 /*!< maxComp */ -}; - - -/**************************************************************************/ -/*! - \brief Calculates the difference in tonality between original and SBR - for a given time and frequency region. - - The values for pDiffMapped2Scfb are scaled by RELAXATION - - \return none. - -*/ -/**************************************************************************/ -static void diff(FIXP_DBL *RESTRICT pTonalityOrig, - FIXP_DBL *pDiffMapped2Scfb, - const UCHAR *RESTRICT pFreqBandTable, - INT nScfb, - SCHAR *indexVector) -{ - UCHAR i, ll, lu, k; - FIXP_DBL maxValOrig, maxValSbr, tmp; - INT scale; - - for(i=0; i < nScfb; i++){ - ll = pFreqBandTable[i]; - lu = pFreqBandTable[i+1]; - - maxValOrig = FL2FXCONST_DBL(0.0f); - maxValSbr = FL2FXCONST_DBL(0.0f); - - for(k=ll;k= RELAXATION)) { - tmp = fDivNorm(maxValOrig, maxValSbr, &scale); - pDiffMapped2Scfb[i] = scaleValue(fMult(tmp,RELAXATION_FRACT), fixMax(-(DFRACT_BITS-1),(scale-RELAXATION_SHIFT))); - } - else { - pDiffMapped2Scfb[i] = maxValOrig; - } - } -} - - -/**************************************************************************/ -/*! - \brief Calculates a flatness measure of the tonality measures. - - Calculation of the power function and using scalefactor for basis: - Using log2: - z = (2^k * x)^y; - z' = CalcLd(z) = y*CalcLd(x) + y*k; - z = CalcInvLd(z'); - - Using ld64: - z = (2^k * x)^y; - z' = CalcLd64(z) = y*CalcLd64(x)/64 + y*k/64; - z = CalcInvLd64(z'); - - The values pSfmOrigVec and pSfmSbrVec are scaled by the factor 1/4.0 - - \return none. - -*/ -/**************************************************************************/ -static void calculateFlatnessMeasure(FIXP_DBL *pQuotaBuffer, - SCHAR *indexVector, - FIXP_DBL *pSfmOrigVec, - FIXP_DBL *pSfmSbrVec, - const UCHAR *pFreqBandTable, - INT nSfb) -{ - INT i,j; - FIXP_DBL invBands,tmp1,tmp2; - INT shiftFac0,shiftFacSum0; - INT shiftFac1,shiftFacSum1; - FIXP_DBL accu; - - for(i=0;i>2); - pSfmSbrVec[i] = (FIXP_DBL)(MAXVAL_DBL>>2); - - if(lu - ll > 1){ - FIXP_DBL amOrig,amTransp,gmOrig,gmTransp,sfmOrig,sfmTransp; - invBands = GetInvInt(lu-ll); - shiftFacSum0 = 0; - shiftFacSum1 = 0; - amOrig = amTransp = FL2FXCONST_DBL(0.0f); - gmOrig = gmTransp = (FIXP_DBL)MAXVAL_DBL; - - for(j= ll; j FL2FXCONST_DBL(0.0f)) { - - tmp1 = CalcLdData(gmOrig); /* CalcLd64(x)/64 */ - tmp1 = fMult(invBands, tmp1); /* y*CalcLd64(x)/64 */ - - /* y*k/64 */ - accu = (FIXP_DBL)-shiftFacSum0 << (DFRACT_BITS-1-8); - tmp2 = fMultDiv2(invBands, accu) << (2+1); - - tmp2 = tmp1 + tmp2; /* y*CalcLd64(x)/64 + y*k/64 */ - gmOrig = CalcInvLdData(tmp2); /* CalcInvLd64(z'); */ - } - else { - gmOrig = FL2FXCONST_DBL(0.0f); - } - - if (gmTransp > FL2FXCONST_DBL(0.0f)) { - - tmp1 = CalcLdData(gmTransp); /* CalcLd64(x)/64 */ - tmp1 = fMult(invBands, tmp1); /* y*CalcLd64(x)/64 */ - - /* y*k/64 */ - accu = (FIXP_DBL)-shiftFacSum1 << (DFRACT_BITS-1-8); - tmp2 = fMultDiv2(invBands, accu) << (2+1); - - tmp2 = tmp1 + tmp2; /* y*CalcLd64(x)/64 + y*k/64 */ - gmTransp = CalcInvLdData(tmp2); /* CalcInvLd64(z'); */ - } - else { - gmTransp = FL2FXCONST_DBL(0.0f); - } - if ( amOrig != FL2FXCONST_DBL(0.0f) ) - pSfmOrigVec[i] = FDKsbrEnc_LSI_divide_scale_fract(gmOrig,amOrig,SFM_SCALE); - - if ( amTransp != FL2FXCONST_DBL(0.0f) ) - pSfmSbrVec[i] = FDKsbrEnc_LSI_divide_scale_fract(gmTransp,amTransp,SFM_SCALE); - } - } -} - -/**************************************************************************/ -/*! - \brief Calculates the input to the missing harmonics detection. - - - \return none. - -*/ -/**************************************************************************/ -static void calculateDetectorInput(FIXP_DBL **RESTRICT pQuotaBuffer, /*!< Pointer to tonality matrix. */ - SCHAR *RESTRICT indexVector, - FIXP_DBL **RESTRICT tonalityDiff, - FIXP_DBL **RESTRICT pSfmOrig, - FIXP_DBL **RESTRICT pSfmSbr, - const UCHAR *freqBandTable, - INT nSfb, - INT noEstPerFrame, - INT move) -{ - INT est; - - /* - New estimate. - */ - for (est=0; est < noEstPerFrame; est++) { - - diff(pQuotaBuffer[est+move], - tonalityDiff[est+move], - freqBandTable, - nSfb, - indexVector); - - calculateFlatnessMeasure(pQuotaBuffer[est+ move], - indexVector, - pSfmOrig[est + move], - pSfmSbr[est + move], - freqBandTable, - nSfb); - } -} - - -/**************************************************************************/ -/*! - \brief Checks that the detection is not due to a LP filter - - This function determines if a newly detected missing harmonics is not - in fact just a low-pass filtere input signal. If so, the detection is - removed. - - \return none. - -*/ -/**************************************************************************/ -static void removeLowPassDetection(UCHAR *RESTRICT pAddHarmSfb, - UCHAR **RESTRICT pDetectionVectors, - INT start, - INT stop, - INT nSfb, - const UCHAR *RESTRICT pFreqBandTable, - FIXP_DBL *RESTRICT pNrgVector, - THRES_HOLDS mhThresh) - -{ - INT i,est; - INT maxDerivPos = pFreqBandTable[nSfb]; - INT numBands = pFreqBandTable[nSfb]; - FIXP_DBL nrgLow,nrgHigh; - FIXP_DBL nrgLD64,nrgLowLD64,nrgHighLD64,nrgDiffLD64; - FIXP_DBL valLD64,maxValLD64,maxValAboveLD64; - INT bLPsignal = 0; - - maxValLD64 = FL2FXCONST_DBL(-1.0f); - for(i = numBands - 1 - 2; i > pFreqBandTable[0];i--){ - nrgLow = pNrgVector[i]; - nrgHigh = pNrgVector[i + 2]; - - if(nrgLow != FL2FXCONST_DBL(0.0f) && nrgLow > nrgHigh){ - nrgLowLD64 = CalcLdData(nrgLow>>1); - nrgDiffLD64 = CalcLdData((nrgLow>>1)-(nrgHigh>>1)); - valLD64 = nrgDiffLD64-nrgLowLD64; - if(valLD64 > maxValLD64){ - maxDerivPos = i; - maxValLD64 = valLD64; - } - if(maxValLD64 > mhThresh.derivThresMaxLD64) { - break; - } - } - } - - /* Find the largest "gradient" above. (should be relatively flat, hence we expect a low value - if the signal is LP.*/ - maxValAboveLD64 = FL2FXCONST_DBL(-1.0f); - for(i = numBands - 1 - 2; i > maxDerivPos + 2;i--){ - nrgLow = pNrgVector[i]; - nrgHigh = pNrgVector[i + 2]; - - if(nrgLow != FL2FXCONST_DBL(0.0f) && nrgLow > nrgHigh){ - nrgLowLD64 = CalcLdData(nrgLow>>1); - nrgDiffLD64 = CalcLdData((nrgLow>>1)-(nrgHigh>>1)); - valLD64 = nrgDiffLD64-nrgLowLD64; - if(valLD64 > maxValAboveLD64){ - maxValAboveLD64 = valLD64; - } - } - else { - if(nrgHigh != FL2FXCONST_DBL(0.0f) && nrgHigh > nrgLow){ - nrgHighLD64 = CalcLdData(nrgHigh>>1); - nrgDiffLD64 = CalcLdData((nrgHigh>>1)-(nrgLow>>1)); - valLD64 = nrgDiffLD64-nrgHighLD64; - if(valLD64 > maxValAboveLD64){ - maxValAboveLD64 = valLD64; - } - } - } - } - - if(maxValLD64 > mhThresh.derivThresMaxLD64 && maxValAboveLD64 < mhThresh.derivThresAboveLD64){ - bLPsignal = 1; - - for(i = maxDerivPos - 1; i > maxDerivPos - 5 && i >= 0 ; i--){ - if(pNrgVector[i] != FL2FXCONST_DBL(0.0f) && pNrgVector[i] > pNrgVector[maxDerivPos + 2]){ - nrgDiffLD64 = CalcLdData((pNrgVector[i]>>1)-(pNrgVector[maxDerivPos + 2]>>1)); - nrgLD64 = CalcLdData(pNrgVector[i]>>1); - valLD64 = nrgDiffLD64-nrgLD64; - if(valLD64 < mhThresh.derivThresBelowLD64) { - bLPsignal = 0; - break; - } - } - else{ - bLPsignal = 0; - break; - } - } - } - - if(bLPsignal){ - for(i=0;i= pFreqBandTable[i] && maxDerivPos < pFreqBandTable[i+1]) - break; - } - - if(pAddHarmSfb[i]){ - pAddHarmSfb[i] = 0; - for(est = start; est < stop ; est++){ - pDetectionVectors[est][i] = 0; - } - } - } -} - -/**************************************************************************/ -/*! - \brief Checks if it is allowed to detect a missing tone, that wasn't - detected previously. - - - \return newDetectionAllowed flag. - -*/ -/**************************************************************************/ -static INT isDetectionOfNewToneAllowed(const SBR_FRAME_INFO *pFrameInfo, - INT *pDetectionStartPos, - INT noEstPerFrame, - INT prevTransientFrame, - INT prevTransientPos, - INT prevTransientFlag, - INT transientPosOffset, - INT transientFlag, - INT transientPos, - INT deltaTime, - HANDLE_SBR_MISSING_HARMONICS_DETECTOR h_sbrMissingHarmonicsDetector) -{ - INT transientFrame, newDetectionAllowed; - - - /* Determine if this is a frame where a transient starts... - * If the transient flag was set the previous frame but not the - * transient frame flag, the transient frame flag is set in the current frame. - *****************************************************************************/ - transientFrame = 0; - if(transientFlag){ - if(transientPos + transientPosOffset < pFrameInfo->borders[pFrameInfo->nEnvelopes]) - transientFrame = 1; - if(noEstPerFrame > 1){ - if(transientPos + transientPosOffset > h_sbrMissingHarmonicsDetector->timeSlots >> 1){ - *pDetectionStartPos = noEstPerFrame; - } - else{ - *pDetectionStartPos = noEstPerFrame >> 1; - } - - } - else{ - *pDetectionStartPos = noEstPerFrame; - } - } - else{ - if(prevTransientFlag && !prevTransientFrame){ - transientFrame = 1; - *pDetectionStartPos = 0; - } - } - - /* - * Determine if detection of new missing harmonics are allowed. - * If the frame contains a transient it's ok. If the previous - * frame contained a transient it needs to be sufficiently close - * to the start of the current frame. - ****************************************************************/ - newDetectionAllowed = 0; - if(transientFrame){ - newDetectionAllowed = 1; - } - else { - if(prevTransientFrame && - fixp_abs(pFrameInfo->borders[0] - (prevTransientPos + transientPosOffset - - h_sbrMissingHarmonicsDetector->timeSlots)) < deltaTime) - newDetectionAllowed = 1; - *pDetectionStartPos = 0; - } - - h_sbrMissingHarmonicsDetector->previousTransientFlag = transientFlag; - h_sbrMissingHarmonicsDetector->previousTransientFrame = transientFrame; - h_sbrMissingHarmonicsDetector->previousTransientPos = transientPos; - - return (newDetectionAllowed); -} - - -/**************************************************************************/ -/*! - \brief Cleans up the detection after a transient. - - - \return none. - -*/ -/**************************************************************************/ -static void transientCleanUp(FIXP_DBL **quotaBuffer, - INT nSfb, - UCHAR **detectionVectors, - UCHAR *pAddHarmSfb, - UCHAR *pPrevAddHarmSfb, - INT ** signBuffer, - const UCHAR *pFreqBandTable, - INT start, - INT stop, - INT newDetectionAllowed, - FIXP_DBL *pNrgVector, - THRES_HOLDS mhThresh) -{ - INT i,j,li, ui,est; - - for(est=start; est < stop; est++) { - for(i=0; i maxVal1){ - maxVal1 = quotaBuffer[est][j]; - maxPos1 = j; - maxPosTime1 = est; - } - } - } - - li = pFreqBandTable[i+1]; - ui = pFreqBandTable[i+2]; - - /* Find maximum tonality in the the two scf bands.*/ - maxPosTime2 = start; - maxPos2 = li; - maxVal2 = quotaBuffer[start][li]; - for(est = start; est < stop; est++){ - for(j = li; j maxVal2){ - maxVal2 = quotaBuffer[est][j]; - maxPos2 = j; - maxPosTime2 = est; - } - } - } - - /* If the maximum values are in adjacent QMF-channels, we need to remove - the lowest of the two.*/ - if(maxPos2-maxPos1 < 2){ - - if(pPrevAddHarmSfb[i] == 1 && pPrevAddHarmSfb[i+1] == 0){ - /* Keep the lower, remove the upper.*/ - pAddHarmSfb[i+1] = 0; - for(est=start; est maxVal2){ - if(signBuffer[maxPosTime1][maxPos2] < 0 && signBuffer[maxPosTime1][maxPos1] > 0){ - /* Keep the lower, remove the upper.*/ - pAddHarmSfb[i+1] = 0; - for(est=start; est 0){ - /* Keep the upper, remove the lower.*/ - pAddHarmSfb[i] = 0; - for(est=start; est 0) - pAddHarmSfb[i] = 0; - } - } -} - - -/*****************************************************************************/ -/*! - \brief Detection for one tonality estimate. - - This is the actual missing harmonics detection, using information from the - previous detection. - - If a missing harmonic was detected (in a previous frame) due to too high - tonality differences, but there was not enough tonality difference in the - current frame, the detection algorithm still continues to trace the strongest - tone in the scalefactor band (assuming that this is the tone that is going to - be replaced in the decoder). This is done to avoid abrupt endings of sines - fading out (e.g. in the glockenspiel). - - The function also tries to estimate where one sine is going to be replaced - with multiple sines (due to the patching). This is done by comparing the - tonality flatness measure of the original and the SBR signal. - - The function also tries to estimate (for the scalefactor bands only - containing one qmf subband) when a strong tone in the original will be - replaced by a strong tone in the adjacent QMF subband. - - \return none. - -*/ -/**************************************************************************/ -static void detection(FIXP_DBL *quotaBuffer, - FIXP_DBL *pDiffVecScfb, - INT nSfb, - UCHAR *pHarmVec, - const UCHAR *pFreqBandTable, - FIXP_DBL *sfmOrig, - FIXP_DBL *sfmSbr, - GUIDE_VECTORS guideVectors, - GUIDE_VECTORS newGuideVectors, - THRES_HOLDS mhThresh) -{ - - INT i,j,ll, lu; - FIXP_DBL thresTemp,thresOrig; - - /* - * Do detection on the difference vector, i.e. the difference between - * the original and the transposed. - *********************************************************************/ - for(i=0;i thresTemp){ - pHarmVec[i] = 1; - newGuideVectors.guideVectorDiff[i] = pDiffVecScfb[i]; - } - else{ - /* If the guide wasn't zero, but the current level is to low, - start tracking the decay on the tone in the original rather - than the difference.*/ - if(guideVectors.guideVectorDiff[i] != FL2FXCONST_DBL(0.0f)){ - guideVectors.guideVectorOrig[i] = mhThresh.thresHoldToneGuide; - } - } - } - - /* - * Trace tones in the original signal that at one point - * have been detected because they will be replaced by - * multiple tones in the sbr signal. - ****************************************************/ - - for(i=0;i thresOrig){ - pHarmVec[i] = 1; - newGuideVectors.guideVectorOrig[i] = quotaBuffer[j]; - } - } - } - } - - /* - * Check for multiple sines in the transposed signal, - * where there is only one in the original. - ****************************************************/ - thresOrig = mhThresh.thresHoldTone; - - for(i=0;i 1){ - for(j= ll;j thresOrig && (sfmSbr[i] > mhThresh.sfmThresSbr && sfmOrig[i] < mhThresh.sfmThresOrig)){ - pHarmVec[i] = 1; - newGuideVectors.guideVectorOrig[i] = quotaBuffer[j]; - } - } - } - else{ - if(i < nSfb -1){ - ll = pFreqBandTable[i]; - - if(i>0){ - if(quotaBuffer[ll] > mhThresh.thresHoldTone && (pDiffVecScfb[i+1] < mhThresh.invThresHoldTone || pDiffVecScfb[i-1] < mhThresh.invThresHoldTone)){ - pHarmVec[i] = 1; - newGuideVectors.guideVectorOrig[i] = quotaBuffer[ll]; - } - } - else{ - if(quotaBuffer[ll] > mhThresh.thresHoldTone && pDiffVecScfb[i+1] < mhThresh.invThresHoldTone){ - pHarmVec[i] = 1; - newGuideVectors.guideVectorOrig[i] = quotaBuffer[ll]; - } - } - } - } - } - } -} - - -/**************************************************************************/ -/*! - \brief Do detection for every tonality estimate, using forward prediction. - - - \return none. - -*/ -/**************************************************************************/ -static void detectionWithPrediction(FIXP_DBL **quotaBuffer, - FIXP_DBL **pDiffVecScfb, - INT ** signBuffer, - INT nSfb, - const UCHAR* pFreqBandTable, - FIXP_DBL **sfmOrig, - FIXP_DBL **sfmSbr, - UCHAR **detectionVectors, - UCHAR *pPrevAddHarmSfb, - GUIDE_VECTORS *guideVectors, - INT noEstPerFrame, - INT detectionStart, - INT totNoEst, - INT newDetectionAllowed, - INT *pAddHarmFlag, - UCHAR *pAddHarmSfb, - FIXP_DBL *pNrgVector, - const DETECTOR_PARAMETERS_MH *mhParams) -{ - INT est = 0,i; - INT start; - - FDKmemclear(pAddHarmSfb,nSfb*sizeof(UCHAR)); - - if(newDetectionAllowed){ - - /* Since we don't want to use the transient region for detection (since the tonality values - tend to be a bit unreliable for this region) the guide-values are copied to the current - starting point. */ - if(totNoEst > 1){ - start = detectionStart+1; - - if (start != 0) { - FDKmemcpy(guideVectors[start].guideVectorDiff,guideVectors[0].guideVectorDiff,nSfb*sizeof(FIXP_DBL)); - FDKmemcpy(guideVectors[start].guideVectorOrig,guideVectors[0].guideVectorOrig,nSfb*sizeof(FIXP_DBL)); - FDKmemclear(guideVectors[start-1].guideVectorDetected,nSfb*sizeof(UCHAR)); - } - } - else{ - start = 0; - } - } - else{ - start = 0; - } - - - for(est = start; est < totNoEst; est++){ - - /* - * Do detection on the current frame using - * guide-info from the previous. - *******************************************/ - if(est > 0){ - FDKmemcpy(guideVectors[est].guideVectorDetected,detectionVectors[est-1],nSfb*sizeof(UCHAR)); - } - - FDKmemclear(detectionVectors[est], nSfb*sizeof(UCHAR)); - - if(est < totNoEst-1){ - FDKmemclear(guideVectors[est+1].guideVectorDiff,nSfb*sizeof(FIXP_DBL)); - FDKmemclear(guideVectors[est+1].guideVectorOrig,nSfb*sizeof(FIXP_DBL)); - FDKmemclear(guideVectors[est+1].guideVectorDetected,nSfb*sizeof(UCHAR)); - - detection(quotaBuffer[est], - pDiffVecScfb[est], - nSfb, - detectionVectors[est], - pFreqBandTable, - sfmOrig[est], - sfmSbr[est], - guideVectors[est], - guideVectors[est+1], - mhParams->thresHolds); - } - else{ - FDKmemclear(guideVectors[est].guideVectorDiff,nSfb*sizeof(FIXP_DBL)); - FDKmemclear(guideVectors[est].guideVectorOrig,nSfb*sizeof(FIXP_DBL)); - FDKmemclear(guideVectors[est].guideVectorDetected,nSfb*sizeof(UCHAR)); - - detection(quotaBuffer[est], - pDiffVecScfb[est], - nSfb, - detectionVectors[est], - pFreqBandTable, - sfmOrig[est], - sfmSbr[est], - guideVectors[est], - guideVectors[est], - mhParams->thresHolds); - } - } - - - /* Clean up the detection.*/ - transientCleanUp(quotaBuffer, - nSfb, - detectionVectors, - pAddHarmSfb, - pPrevAddHarmSfb, - signBuffer, - pFreqBandTable, - start, - totNoEst, - newDetectionAllowed, - pNrgVector, - mhParams->thresHolds); - - - /* Set flag... */ - *pAddHarmFlag = 0; - for(i=0; i maxVal){ - maxVal = pTonalityMatrix[est][l]; - maxPosF = l; - maxPosT = est; - } - } - } - - /* - * If the maximum tonality is at the lower border of the - * scalefactor band, we check the sign of the adjacent channels - * to see if this sine is shared by the lower channel. If so, the - * energy of the single sine will be present in two scalefactor bands - * in the SBR data, which will cause problems in the decoder, when we - * add a sine to just one of the channels. - *********************************************************************/ - if(maxPosF == ll && scfBand){ - if(!pAddHarmSfb[scfBand - 1]) { /* No detection below*/ - if (pSignMatrix[maxPosT][maxPosF - 1] > 0 && pSignMatrix[maxPosT][maxPosF] < 0) { - /* The comp value is calulated as the tonallity value, i.e we want to - reduce the envelope data for this channel with as much as the tonality - that is spread from the channel above. (ld64(RELAXATION) = 0.31143075889) */ - tmp = fixp_abs((FIXP_DBL)CalcLdData(pTonalityMatrix[maxPosT][maxPosF - 1]) + RELAXATION_LD64); - tmp = (tmp >> (DFRACT_BITS-1-LD_DATA_SHIFT-1)) + (FIXP_DBL)1; /* shift one bit less for rounding */ - compValue = ((INT)(LONG)tmp) >> 1; - - /* limit the comp-value*/ - if (compValue > maxComp) - compValue = maxComp; - - pEnvComp[scfBand-1] = compValue; - } - } - } - - /* - * Same as above, but for the upper end of the scalefactor-band. - ***************************************************************/ - if(maxPosF == lu-1 && scfBand+1 < nSfb){ /* Upper border*/ - if(!pAddHarmSfb[scfBand + 1]) { - if (pSignMatrix[maxPosT][maxPosF] > 0 && pSignMatrix[maxPosT][maxPosF + 1] < 0) { - tmp = fixp_abs((FIXP_DBL)CalcLdData(pTonalityMatrix[maxPosT][maxPosF + 1]) + RELAXATION_LD64); - tmp = (tmp >> (DFRACT_BITS-1-LD_DATA_SHIFT-1)) + (FIXP_DBL)1; /* shift one bit less for rounding */ - compValue = ((INT)(LONG)tmp) >> 1; - - if (compValue > maxComp) - compValue = maxComp; - - pEnvComp[scfBand+1] = compValue; - } - } - } - } - } - - if(newDetectionAllowed == 0){ - for(scfBand=0;scfBanddetectionVectors; - INT move = h_sbrMHDet->move; - INT noEstPerFrame = h_sbrMHDet->noEstPerFrame; - INT totNoEst = h_sbrMHDet->totNoEst; - INT prevTransientFlag = h_sbrMHDet->previousTransientFlag; - INT prevTransientFrame = h_sbrMHDet->previousTransientFrame; - INT transientPosOffset = h_sbrMHDet->transientPosOffset; - INT prevTransientPos = h_sbrMHDet->previousTransientPos; - GUIDE_VECTORS* guideVectors = h_sbrMHDet->guideVectors; - INT deltaTime = h_sbrMHDet->mhParams->deltaTime; - INT maxComp = h_sbrMHDet->mhParams->maxComp; - - int est; - - /* - Buffer values. - */ - FDK_ASSERT(move<=(MAX_NO_OF_ESTIMATES>>1)); - FDK_ASSERT(noEstPerFrame<=(MAX_NO_OF_ESTIMATES>>1)); - - FIXP_DBL *sfmSbr[MAX_NO_OF_ESTIMATES]; - FIXP_DBL *sfmOrig[MAX_NO_OF_ESTIMATES]; - FIXP_DBL *tonalityDiff[MAX_NO_OF_ESTIMATES]; - - for (est=0; est < MAX_NO_OF_ESTIMATES/2; est++) { - sfmSbr[est] = h_sbrMHDet->sfmSbr[est]; - sfmOrig[est] = h_sbrMHDet->sfmOrig[est]; - tonalityDiff[est] = h_sbrMHDet->tonalityDiff[est]; - } - - C_ALLOC_SCRATCH_START(scratch_mem, FIXP_DBL, (3*MAX_NO_OF_ESTIMATES/2*MAX_FREQ_COEFFS)); - FIXP_DBL *scratch = scratch_mem; - for (; est < MAX_NO_OF_ESTIMATES; est++) { - sfmSbr[est] = scratch; scratch+=MAX_FREQ_COEFFS; - sfmOrig[est] = scratch; scratch+=MAX_FREQ_COEFFS; - tonalityDiff[est] = scratch; scratch+=MAX_FREQ_COEFFS; - } - - - - /* Determine if we're allowed to detect "missing harmonics" that wasn't detected before. - In order to be allowed to do new detection, there must be a transient in the current - frame, or a transient in the previous frame sufficiently close to the current frame. */ - newDetectionAllowed = isDetectionOfNewToneAllowed(pFrameInfo, - &transientDetStart, - noEstPerFrame, - prevTransientFrame, - prevTransientPos, - prevTransientFlag, - transientPosOffset, - transientFlag, - transientPos, - deltaTime, - h_sbrMHDet); - - /* Calulate the variables that will be used subsequently for the actual detection */ - calculateDetectorInput(pQuotaBuffer, - indexVector, - tonalityDiff, - sfmOrig, - sfmSbr, - freqBandTable, - nSfb, - noEstPerFrame, - move); - - /* Do the actual detection using information from previous detections */ - detectionWithPrediction(pQuotaBuffer, - tonalityDiff, - pSignBuffer, - nSfb, - freqBandTable, - sfmOrig, - sfmSbr, - detectionVectors, - h_sbrMHDet->guideScfb, - guideVectors, - noEstPerFrame, - transientDetStart, - totNoEst, - newDetectionAllowed, - pAddHarmonicsFlag, - pAddHarmonicsScaleFactorBands, - pNrgVector, - h_sbrMHDet->mhParams); - - /* Calculate the comp vector, so that the energy can be - compensated for a sine between two QMF-bands. */ - calculateCompVector(pAddHarmonicsScaleFactorBands, - pQuotaBuffer, - pSignBuffer, - envelopeCompensation, - nSfb, - freqBandTable, - totNoEst, - maxComp, - h_sbrMHDet->prevEnvelopeCompensation, - newDetectionAllowed); - - for (est=0; est < move; est++) { - FDKmemcpy(tonalityDiff[est], tonalityDiff[est + noEstPerFrame], sizeof(FIXP_DBL)*MAX_FREQ_COEFFS); - FDKmemcpy(sfmOrig[est], sfmOrig[est + noEstPerFrame], sizeof(FIXP_DBL)*MAX_FREQ_COEFFS); - FDKmemcpy(sfmSbr[est], sfmSbr[est + noEstPerFrame], sizeof(FIXP_DBL)*MAX_FREQ_COEFFS); - } - C_ALLOC_SCRATCH_END(scratch, FIXP_DBL, (3*MAX_NO_OF_ESTIMATES/2*MAX_FREQ_COEFFS)); - - -} - -/**************************************************************************/ -/*! - \brief Initialize an instance of the missing harmonics detector. - - - \return errorCode, noError if OK. - -*/ -/**************************************************************************/ -INT -FDKsbrEnc_CreateSbrMissingHarmonicsDetector ( - HANDLE_SBR_MISSING_HARMONICS_DETECTOR hSbrMHDet, - INT chan) -{ - HANDLE_SBR_MISSING_HARMONICS_DETECTOR hs = hSbrMHDet; - INT i; - - UCHAR* detectionVectors = GetRam_Sbr_detectionVectors(chan); - UCHAR* guideVectorDetected = GetRam_Sbr_guideVectorDetected(chan); - FIXP_DBL* guideVectorDiff = GetRam_Sbr_guideVectorDiff(chan); - FIXP_DBL* guideVectorOrig = GetRam_Sbr_guideVectorOrig(chan); - - FDKmemclear (hs,sizeof(SBR_MISSING_HARMONICS_DETECTOR)); - - hs->prevEnvelopeCompensation = GetRam_Sbr_prevEnvelopeCompensation(chan); - hs->guideScfb = GetRam_Sbr_guideScfb(chan); - - for(i=0; iguideVectors[i].guideVectorDiff = guideVectorDiff + (i*MAX_FREQ_COEFFS); - hs->guideVectors[i].guideVectorOrig = guideVectorOrig + (i*MAX_FREQ_COEFFS); - hs->detectionVectors[i] = detectionVectors + (i*MAX_FREQ_COEFFS); - hs->guideVectors[i].guideVectorDetected = guideVectorDetected + (i*MAX_FREQ_COEFFS); - } - - return 0; -} - - -/**************************************************************************/ -/*! - \brief Initialize an instance of the missing harmonics detector. - - - \return errorCode, noError if OK. - -*/ -/**************************************************************************/ -INT -FDKsbrEnc_InitSbrMissingHarmonicsDetector ( - HANDLE_SBR_MISSING_HARMONICS_DETECTOR hSbrMHDet, - INT sampleFreq, - INT frameSize, - INT nSfb, - INT qmfNoChannels, - INT totNoEst, - INT move, - INT noEstPerFrame, - UINT sbrSyntaxFlags - ) -{ - HANDLE_SBR_MISSING_HARMONICS_DETECTOR hs = hSbrMHDet; - int i; - - FDK_ASSERT(totNoEst <= MAX_NO_OF_ESTIMATES); - - if (sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) - { - switch(frameSize){ - case 1024: - case 512: - hs->transientPosOffset = FRAME_MIDDLE_SLOT_512LD; - hs->timeSlots = 16; - break; - case 960: - case 480: - hs->transientPosOffset = FRAME_MIDDLE_SLOT_512LD; - hs->timeSlots = 15; - break; - default: - return -1; - } - } else - { - switch(frameSize){ - case 2048: - case 1024: - hs->transientPosOffset = FRAME_MIDDLE_SLOT_2048; - hs->timeSlots = NUMBER_TIME_SLOTS_2048; - break; - case 1920: - case 960: - hs->transientPosOffset = FRAME_MIDDLE_SLOT_1920; - hs->timeSlots = NUMBER_TIME_SLOTS_1920; - break; - default: - return -1; - } - } - - if (sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) { - hs->mhParams = ¶msAacLd; - } else - hs->mhParams = ¶msAac; - - hs->qmfNoChannels = qmfNoChannels; - hs->sampleFreq = sampleFreq; - hs->nSfb = nSfb; - - hs->totNoEst = totNoEst; - hs->move = move; - hs->noEstPerFrame = noEstPerFrame; - - for(i=0; iguideVectors[i].guideVectorDiff,sizeof(FIXP_DBL)*MAX_FREQ_COEFFS); - FDKmemclear (hs->guideVectors[i].guideVectorOrig,sizeof(FIXP_DBL)*MAX_FREQ_COEFFS); - FDKmemclear (hs->detectionVectors[i],sizeof(UCHAR)*MAX_FREQ_COEFFS); - FDKmemclear (hs->guideVectors[i].guideVectorDetected,sizeof(UCHAR)*MAX_FREQ_COEFFS); - } - - //for(i=0; itonalityDiff[i],sizeof(FIXP_DBL)*MAX_FREQ_COEFFS); - FDKmemclear (hs->sfmOrig[i],sizeof(FIXP_DBL)*MAX_FREQ_COEFFS); - FDKmemclear (hs->sfmSbr[i],sizeof(FIXP_DBL)*MAX_FREQ_COEFFS); - } - - FDKmemclear ( hs->prevEnvelopeCompensation, sizeof(UCHAR)*MAX_FREQ_COEFFS); - FDKmemclear ( hs->guideScfb, sizeof(UCHAR)*MAX_FREQ_COEFFS); - - hs->previousTransientFlag = 0; - hs->previousTransientFrame = 0; - hs->previousTransientPos = 0; - - return (0); -} - -/**************************************************************************/ -/*! - \brief Deletes an instance of the missing harmonics detector. - - - \return none. - -*/ -/**************************************************************************/ -void -FDKsbrEnc_DeleteSbrMissingHarmonicsDetector(HANDLE_SBR_MISSING_HARMONICS_DETECTOR hSbrMHDet) -{ - if (hSbrMHDet) { - HANDLE_SBR_MISSING_HARMONICS_DETECTOR hs = hSbrMHDet; - - FreeRam_Sbr_detectionVectors(&hs->detectionVectors[0]); - FreeRam_Sbr_guideVectorDetected(&hs->guideVectors[0].guideVectorDetected); - FreeRam_Sbr_guideVectorDiff(&hs->guideVectors[0].guideVectorDiff); - FreeRam_Sbr_guideVectorOrig(&hs->guideVectors[0].guideVectorOrig); - FreeRam_Sbr_prevEnvelopeCompensation(&hs->prevEnvelopeCompensation); - FreeRam_Sbr_guideScfb(&hs->guideScfb); - - } -} - -/**************************************************************************/ -/*! - \brief Resets an instance of the missing harmonics detector. - - - \return error code, noError if OK. - -*/ -/**************************************************************************/ -INT -FDKsbrEnc_ResetSbrMissingHarmonicsDetector (HANDLE_SBR_MISSING_HARMONICS_DETECTOR hSbrMissingHarmonicsDetector, - INT nSfb) -{ - int i; - FIXP_DBL tempGuide[MAX_FREQ_COEFFS]; - UCHAR tempGuideInt[MAX_FREQ_COEFFS]; - INT nSfbPrev; - - nSfbPrev = hSbrMissingHarmonicsDetector->nSfb; - hSbrMissingHarmonicsDetector->nSfb = nSfb; - - FDKmemcpy( tempGuideInt, hSbrMissingHarmonicsDetector->guideScfb, nSfbPrev * sizeof(UCHAR) ); - - if ( nSfb > nSfbPrev ) { - for ( i = 0; i < (nSfb - nSfbPrev); i++ ) { - hSbrMissingHarmonicsDetector->guideScfb[i] = 0; - } - - for ( i = 0; i < nSfbPrev; i++ ) { - hSbrMissingHarmonicsDetector->guideScfb[i + (nSfb - nSfbPrev)] = tempGuideInt[i]; - } - } - else { - for ( i = 0; i < nSfb; i++ ) { - hSbrMissingHarmonicsDetector->guideScfb[i] = tempGuideInt[i + (nSfbPrev-nSfb)]; - } - } - - FDKmemcpy ( tempGuide, hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorDiff, nSfbPrev * sizeof(FIXP_DBL) ); - - if (nSfb > nSfbPrev ) { - for ( i = 0; i < (nSfb - nSfbPrev); i++ ) { - hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorDiff[i] = FL2FXCONST_DBL(0.0f); - } - - for ( i = 0; i < nSfbPrev; i++ ) { - hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorDiff[i + (nSfb - nSfbPrev)] = tempGuide[i]; - } - } - else { - for ( i = 0; i < nSfb; i++ ) { - hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorDiff[i] = tempGuide[i + (nSfbPrev-nSfb)]; - } - } - - FDKmemcpy ( tempGuide, hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorOrig, nSfbPrev * sizeof(FIXP_DBL) ); - - if ( nSfb > nSfbPrev ) { - for ( i = 0; i< (nSfb - nSfbPrev); i++ ) { - hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorOrig[i] = FL2FXCONST_DBL(0.0f); - } - - for ( i = 0; i < nSfbPrev; i++ ) { - hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorOrig[i + (nSfb - nSfbPrev)] = tempGuide[i]; - } - } - else { - for ( i = 0; i < nSfb; i++ ) { - hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorOrig[i] = tempGuide[i + (nSfbPrev-nSfb)]; - } - } - - FDKmemcpy ( tempGuideInt, hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorDetected, nSfbPrev * sizeof(UCHAR) ); - - if ( nSfb > nSfbPrev ) { - for ( i = 0; i < (nSfb - nSfbPrev); i++ ) { - hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorDetected[i] = 0; - } - - for ( i = 0; i < nSfbPrev; i++ ) { - hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorDetected[i + (nSfb - nSfbPrev)] = tempGuideInt[i]; - } - } - else { - for ( i = 0; i < nSfb; i++ ) { - hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorDetected[i] = tempGuideInt[i + (nSfbPrev-nSfb)]; - } - } - - FDKmemcpy ( tempGuideInt, hSbrMissingHarmonicsDetector->prevEnvelopeCompensation, nSfbPrev * sizeof(UCHAR) ); - - if ( nSfb > nSfbPrev ) { - for ( i = 0; i < (nSfb - nSfbPrev); i++ ) { - hSbrMissingHarmonicsDetector->prevEnvelopeCompensation[i] = 0; - } - - for ( i = 0; i < nSfbPrev; i++ ) { - hSbrMissingHarmonicsDetector->prevEnvelopeCompensation[i + (nSfb - nSfbPrev)] = tempGuideInt[i]; - } - } - else { - for ( i = 0; i < nSfb; i++ ) { - hSbrMissingHarmonicsDetector->prevEnvelopeCompensation[i] = tempGuideInt[i + (nSfbPrev-nSfb)]; - } - } - - return 0; -} - diff --git a/libSBRenc/src/mh_det.h b/libSBRenc/src/mh_det.h deleted file mode 100644 index 74c2a99..0000000 --- a/libSBRenc/src/mh_det.h +++ /dev/null @@ -1,196 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief missing harmonics detection header file -*/ - -#ifndef __MH_DETECT_H -#define __MH_DETECT_H - -#include "sbr_encoder.h" -#include "fram_gen.h" - -typedef struct -{ - FIXP_DBL thresHoldDiff; /*!< threshold for tonality difference */ - FIXP_DBL thresHoldDiffGuide; /*!< threshold for tonality difference for the guide */ - FIXP_DBL thresHoldTone; /*!< threshold for tonality for a sine */ - FIXP_DBL invThresHoldTone; - FIXP_DBL thresHoldToneGuide; /*!< threshold for tonality for a sine for the guide */ - FIXP_DBL sfmThresSbr; /*!< tonality flatness measure threshold for the SBR signal.*/ - FIXP_DBL sfmThresOrig; /*!< tonality flatness measure threshold for the original signal.*/ - FIXP_DBL decayGuideOrig; /*!< decay value of the tonality value of the guide for the tone. */ - FIXP_DBL decayGuideDiff; /*!< decay value of the tonality value of the guide for the tonality difference. */ - FIXP_DBL derivThresMaxLD64; /*!< threshold for detecting LP character in a signal. */ - FIXP_DBL derivThresBelowLD64; /*!< threshold for detecting LP character in a signal. */ - FIXP_DBL derivThresAboveLD64; /*!< threshold for detecting LP character in a signal. */ -}THRES_HOLDS; - -typedef struct -{ - INT deltaTime; /*!< maximum allowed transient distance (from frame border in number of qmf subband sample) - for a frame to be considered a transient frame.*/ - THRES_HOLDS thresHolds; /*!< the thresholds used for detection. */ - INT maxComp; /*!< maximum alllowed compensation factor for the envelope data. */ -}DETECTOR_PARAMETERS_MH; - -typedef struct -{ - FIXP_DBL *guideVectorDiff; - FIXP_DBL *guideVectorOrig; - UCHAR* guideVectorDetected; -}GUIDE_VECTORS; - - -typedef struct -{ - INT qmfNoChannels; - INT nSfb; - INT sampleFreq; - INT previousTransientFlag; - INT previousTransientFrame; - INT previousTransientPos; - - INT noVecPerFrame; - INT transientPosOffset; - - INT move; - INT totNoEst; - INT noEstPerFrame; - INT timeSlots; - - UCHAR *guideScfb; - UCHAR *prevEnvelopeCompensation; - UCHAR *detectionVectors[MAX_NO_OF_ESTIMATES]; - FIXP_DBL tonalityDiff[MAX_NO_OF_ESTIMATES/2][MAX_FREQ_COEFFS]; - FIXP_DBL sfmOrig[MAX_NO_OF_ESTIMATES/2][MAX_FREQ_COEFFS]; - FIXP_DBL sfmSbr[MAX_NO_OF_ESTIMATES/2][MAX_FREQ_COEFFS]; - const DETECTOR_PARAMETERS_MH *mhParams; - GUIDE_VECTORS guideVectors[MAX_NO_OF_ESTIMATES]; -} -SBR_MISSING_HARMONICS_DETECTOR; - -typedef SBR_MISSING_HARMONICS_DETECTOR *HANDLE_SBR_MISSING_HARMONICS_DETECTOR; - -void -FDKsbrEnc_SbrMissingHarmonicsDetectorQmf(HANDLE_SBR_MISSING_HARMONICS_DETECTOR h_sbrMissingHarmonicsDetector, - FIXP_DBL ** pQuotaBuffer, - INT ** pSignBuffer, - SCHAR *indexVector, - const SBR_FRAME_INFO *pFrameInfo, - const UCHAR* pTranInfo, - INT* pAddHarmonicsFlag, - UCHAR* pAddHarmonicsScaleFactorBands, - const UCHAR* freqBandTable, - INT nSfb, - UCHAR * envelopeCompensation, - FIXP_DBL *pNrgVector); - -INT -FDKsbrEnc_CreateSbrMissingHarmonicsDetector ( - HANDLE_SBR_MISSING_HARMONICS_DETECTOR hSbrMHDet, - INT chan); - -INT -FDKsbrEnc_InitSbrMissingHarmonicsDetector( - HANDLE_SBR_MISSING_HARMONICS_DETECTOR h_sbrMissingHarmonicsDetector, - INT sampleFreq, - INT frameSize, - INT nSfb, - INT qmfNoChannels, - INT totNoEst, - INT move, - INT noEstPerFrame, - UINT sbrSyntaxFlags); - -void -FDKsbrEnc_DeleteSbrMissingHarmonicsDetector (HANDLE_SBR_MISSING_HARMONICS_DETECTOR h_sbrMissingHarmonicsDetector); - - -INT -FDKsbrEnc_ResetSbrMissingHarmonicsDetector (HANDLE_SBR_MISSING_HARMONICS_DETECTOR hSbrMissingHarmonicsDetector, - INT nSfb); - -#endif diff --git a/libSBRenc/src/nf_est.cpp b/libSBRenc/src/nf_est.cpp deleted file mode 100644 index a4c5574..0000000 --- a/libSBRenc/src/nf_est.cpp +++ /dev/null @@ -1,584 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -#include "nf_est.h" - -#include "sbr_misc.h" - -#include "genericStds.h" - -/* smoothFilter[4] = {0.05857864376269f, 0.2f, 0.34142135623731f, 0.4f}; */ -static const FIXP_DBL smoothFilter[4] = { 0x077f813d, 0x19999995, 0x2bb3b1f5, 0x33333335 }; - -/* static const INT smoothFilterLength = 4; */ - -static const FIXP_DBL QuantOffset = (INT)0xfc000000; /* ld64(0.25) */ - -#ifndef min -#define min(a,b) ( a < b ? a:b) -#endif - -#ifndef max -#define max(a,b) ( a > b ? a:b) -#endif - -#define NOISE_FLOOR_OFFSET_SCALING (4) - - - -/**************************************************************************/ -/*! - \brief The function applies smoothing to the noise levels. - - - - \return none - -*/ -/**************************************************************************/ -static void -smoothingOfNoiseLevels(FIXP_DBL *NoiseLevels, /*!< pointer to noise-floor levels.*/ - INT nEnvelopes, /*!< Number of noise floor envelopes.*/ - INT noNoiseBands, /*!< Number of noise bands for every noise floor envelope. */ - FIXP_DBL prevNoiseLevels[NF_SMOOTHING_LENGTH][MAX_NUM_NOISE_VALUES],/*!< Previous noise floor envelopes. */ - const FIXP_DBL *smoothFilter, /*!< filter used for smoothing the noise floor levels. */ - INT transientFlag) /*!< flag indicating if a transient is present*/ - -{ - INT i,band,env; - FIXP_DBL accu; - - for(env = 0; env < nEnvelopes; env++){ - if(transientFlag){ - for (i = 0; i < NF_SMOOTHING_LENGTH; i++){ - FDKmemcpy(prevNoiseLevels[i],NoiseLevels+env*noNoiseBands,noNoiseBands*sizeof(FIXP_DBL)); - } - } - else { - for (i = 1; i < NF_SMOOTHING_LENGTH; i++){ - FDKmemcpy(prevNoiseLevels[i - 1],prevNoiseLevels[i],noNoiseBands*sizeof(FIXP_DBL)); - } - FDKmemcpy(prevNoiseLevels[NF_SMOOTHING_LENGTH - 1],NoiseLevels+env*noNoiseBands,noNoiseBands*sizeof(FIXP_DBL)); - } - - for (band = 0; band < noNoiseBands; band++){ - accu = FL2FXCONST_DBL(0.0f); - for (i = 0; i < NF_SMOOTHING_LENGTH; i++){ - accu += fMultDiv2(smoothFilter[i], prevNoiseLevels[i][band]); - } - FDK_ASSERT( (band + env*noNoiseBands) < MAX_NUM_NOISE_VALUES); - NoiseLevels[band+ env*noNoiseBands] = accu<<1; - } - } -} - -/**************************************************************************/ -/*! - \brief Does the noise floor level estiamtion. - - The noiseLevel samples are scaled by the factor 0.25 - - \return none - -*/ -/**************************************************************************/ -static void -qmfBasedNoiseFloorDetection(FIXP_DBL *noiseLevel, /*!< Pointer to vector to store the noise levels in.*/ - FIXP_DBL ** quotaMatrixOrig, /*!< Matrix holding the quota values of the original. */ - SCHAR *indexVector, /*!< Index vector to obtain the patched data. */ - INT startIndex, /*!< Start index. */ - INT stopIndex, /*!< Stop index. */ - INT startChannel, /*!< Start channel of the current noise floor band.*/ - INT stopChannel, /*!< Stop channel of the current noise floor band. */ - FIXP_DBL ana_max_level, /*!< Maximum level of the adaptive noise.*/ - FIXP_DBL noiseFloorOffset, /*!< Noise floor offset. */ - INT missingHarmonicFlag, /*!< Flag indicating if a strong tonal component is missing.*/ - FIXP_DBL weightFac, /*!< Weightening factor for the difference between orig and sbr. */ - INVF_MODE diffThres, /*!< Threshold value to control the inverse filtering decision.*/ - INVF_MODE inverseFilteringLevel) /*!< Inverse filtering level of the current band.*/ -{ - INT scale, l, k; - FIXP_DBL meanOrig=FL2FXCONST_DBL(0.0f), meanSbr=FL2FXCONST_DBL(0.0f), diff; - FIXP_DBL invIndex = GetInvInt(stopIndex-startIndex); - FIXP_DBL invChannel = GetInvInt(stopChannel-startChannel); - FIXP_DBL accu; - - /* - Calculate the mean value, over the current time segment, for the original, the HFR - and the difference, over all channels in the current frequency range. - */ - - if(missingHarmonicFlag == 1){ - for(l = startChannel; l < stopChannel;l++){ - /* tonalityOrig */ - accu = FL2FXCONST_DBL(0.0f); - for(k = startIndex ; k < stopIndex; k++){ - accu += fMultDiv2(quotaMatrixOrig[k][l], invIndex); - } - meanOrig = fixMax(meanOrig,(accu<<1)); - - /* tonalitySbr */ - accu = FL2FXCONST_DBL(0.0f); - for(k = startIndex ; k < stopIndex; k++){ - accu += fMultDiv2(quotaMatrixOrig[k][indexVector[l]], invIndex); - } - meanSbr = fixMax(meanSbr,(accu<<1)); - - } - } - else{ - for(l = startChannel; l < stopChannel;l++){ - /* tonalityOrig */ - accu = FL2FXCONST_DBL(0.0f); - for(k = startIndex ; k < stopIndex; k++){ - accu += fMultDiv2(quotaMatrixOrig[k][l], invIndex); - } - meanOrig += fMult((accu<<1), invChannel); - - /* tonalitySbr */ - accu = FL2FXCONST_DBL(0.0f); - for(k = startIndex ; k < stopIndex; k++){ - accu += fMultDiv2(quotaMatrixOrig[k][indexVector[l]], invIndex); - } - meanSbr += fMult((accu<<1), invChannel); - } - } - - /* Small fix to avoid noise during silent passages.*/ - if( meanOrig <= FL2FXCONST_DBL(0.000976562f*RELAXATION_FLOAT) && - meanSbr <= FL2FXCONST_DBL(0.000976562f*RELAXATION_FLOAT) ) - { - meanOrig = FL2FXCONST_DBL(101.5936673f*RELAXATION_FLOAT); - meanSbr = FL2FXCONST_DBL(101.5936673f*RELAXATION_FLOAT); - } - - meanOrig = fixMax(meanOrig,RELAXATION); - meanSbr = fixMax(meanSbr,RELAXATION); - - if (missingHarmonicFlag == 1 || - inverseFilteringLevel == INVF_MID_LEVEL || - inverseFilteringLevel == INVF_LOW_LEVEL || - inverseFilteringLevel == INVF_OFF || - inverseFilteringLevel <= diffThres) - { - diff = RELAXATION; - } - else { - accu = fDivNorm(meanSbr, meanOrig, &scale); - - diff = fixMax( RELAXATION, - fMult(RELAXATION_FRACT,fMult(weightFac,accu)) >>( RELAXATION_SHIFT-scale ) ) ; - } - - /* - * noise Level is now a positive value, i.e. - * the more harmonic the signal is the higher noise level, - * this makes no sense so we change the sign. - *********************************************************/ - accu = fDivNorm(diff, meanOrig, &scale); - scale -= 2; - - if ( (scale>0) && (accu > ((FIXP_DBL)MAXVAL_DBL)>>scale) ) { - *noiseLevel = (FIXP_DBL)MAXVAL_DBL; - } - else { - *noiseLevel = scaleValue(accu, scale); - } - - /* - * Add a noise floor offset to compensate for bias in the detector - *****************************************************************/ - if(!missingHarmonicFlag) { - *noiseLevel = fixMin(fMult(*noiseLevel, noiseFloorOffset), (FIXP_DBL)MAXVAL_DBL>>NOISE_FLOOR_OFFSET_SCALING) << NOISE_FLOOR_OFFSET_SCALING; - } - - /* - * check to see that we don't exceed the maximum allowed level - **************************************************************/ - *noiseLevel = fixMin(*noiseLevel, ana_max_level); /* ana_max_level is scaled with factor 0.25 */ -} - -/**************************************************************************/ -/*! - \brief Does the noise floor level estiamtion. - The function calls the Noisefloor estimation function - for the time segments decided based upon the transient - information. The block is always divided into one or two segments. - - - \return none - -*/ -/**************************************************************************/ -void -FDKsbrEnc_sbrNoiseFloorEstimateQmf(HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */ - const SBR_FRAME_INFO *frame_info, /*!< Time frequency grid of the current frame. */ - FIXP_DBL *noiseLevels, /*!< Pointer to vector to store the noise levels in.*/ - FIXP_DBL **quotaMatrixOrig, /*!< Matrix holding the quota values of the original. */ - SCHAR *indexVector, /*!< Index vector to obtain the patched data. */ - INT missingHarmonicsFlag, /*!< Flag indicating if a strong tonal component will be missing. */ - INT startIndex, /*!< Start index. */ - UINT numberOfEstimatesPerFrame, /*!< The number of tonality estimates per frame. */ - int transientFrame, /*!< A flag indicating if a transient is present. */ - INVF_MODE* pInvFiltLevels, /*!< Pointer to the vector holding the inverse filtering levels. */ - UINT sbrSyntaxFlags - ) - -{ - - INT nNoiseEnvelopes, startPos[2], stopPos[2], env, band; - - INT noNoiseBands = h_sbrNoiseFloorEstimate->noNoiseBands; - INT *freqBandTable = h_sbrNoiseFloorEstimate->freqBandTableQmf; - - nNoiseEnvelopes = frame_info->nNoiseEnvelopes; - - if (sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) { - nNoiseEnvelopes = 1; - startPos[0] = startIndex; - stopPos[0] = startIndex + min(numberOfEstimatesPerFrame,2); - } else - if(nNoiseEnvelopes == 1){ - startPos[0] = startIndex; - stopPos[0] = startIndex + 2; - } - else{ - startPos[0] = startIndex; - stopPos[0] = startIndex + 1; - startPos[1] = startIndex + 1; - stopPos[1] = startIndex + 2; - } - - /* - * Estimate the noise floor. - **************************************/ - for(env = 0; env < nNoiseEnvelopes; env++){ - for(band = 0; band < noNoiseBands; band++){ - FDK_ASSERT( (band + env*noNoiseBands) < MAX_NUM_NOISE_VALUES); - qmfBasedNoiseFloorDetection(&noiseLevels[band + env*noNoiseBands], - quotaMatrixOrig, - indexVector, - startPos[env], - stopPos[env], - freqBandTable[band], - freqBandTable[band+1], - h_sbrNoiseFloorEstimate->ana_max_level, - h_sbrNoiseFloorEstimate->noiseFloorOffset[band], - missingHarmonicsFlag, - h_sbrNoiseFloorEstimate->weightFac, - h_sbrNoiseFloorEstimate->diffThres, - pInvFiltLevels[band]); - } - } - - - /* - * Smoothing of the values. - **************************/ - smoothingOfNoiseLevels(noiseLevels, - nNoiseEnvelopes, - h_sbrNoiseFloorEstimate->noNoiseBands, - h_sbrNoiseFloorEstimate->prevNoiseLevels, - h_sbrNoiseFloorEstimate->smoothFilter, - transientFrame); - - - /* quantisation*/ - for(env = 0; env < nNoiseEnvelopes; env++){ - for(band = 0; band < noNoiseBands; band++){ - FDK_ASSERT( (band + env*noNoiseBands) < MAX_NUM_NOISE_VALUES); - noiseLevels[band + env*noNoiseBands] = - (FIXP_DBL)NOISE_FLOOR_OFFSET_64 - (FIXP_DBL)CalcLdData(noiseLevels[band + env*noNoiseBands]+(FIXP_DBL)1) + QuantOffset; - } - } -} - -/**************************************************************************/ -/*! - \brief - - - \return errorCode, noError if successful - -*/ -/**************************************************************************/ -static INT -downSampleLoRes(INT *v_result, /*!< */ - INT num_result, /*!< */ - const UCHAR *freqBandTableRef,/*!< */ - INT num_Ref) /*!< */ -{ - INT step; - INT i,j; - INT org_length,result_length; - INT v_index[MAX_FREQ_COEFFS/2]; - - /* init */ - org_length=num_Ref; - result_length=num_result; - - v_index[0]=0; /* Always use left border */ - i=0; - while(org_length > 0) /* Create downsample vector */ - { - i++; - step=org_length/result_length; /* floor; */ - org_length=org_length - step; - result_length--; - v_index[i]=v_index[i-1]+step; - } - - if(i != num_result ) /* Should never happen */ - return (1);/* error downsampling */ - - for(j=0;j<=i;j++) /* Use downsample vector to index LoResolution vector. */ - { - v_result[j]=freqBandTableRef[v_index[j]]; - } - - return (0); -} - -/**************************************************************************/ -/*! - \brief Initialize an instance of the noise floor level estimation module. - - - \return errorCode, noError if successful - -*/ -/**************************************************************************/ -INT -FDKsbrEnc_InitSbrNoiseFloorEstimate (HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */ - INT ana_max_level, /*!< Maximum level of the adaptive noise. */ - const UCHAR *freqBandTable, /*!< Frequany band table. */ - INT nSfb, /*!< Number of frequency bands. */ - INT noiseBands, /*!< Number of noise bands per octave. */ - INT noiseFloorOffset, /*!< Noise floor offset. */ - INT timeSlots, /*!< Number of time slots in a frame. */ - UINT useSpeechConfig /*!< Flag: adapt tuning parameters according to speech */ - ) -{ - INT i, qexp, qtmp; - FIXP_DBL tmp, exp; - - FDKmemclear(h_sbrNoiseFloorEstimate,sizeof(SBR_NOISE_FLOOR_ESTIMATE)); - - h_sbrNoiseFloorEstimate->smoothFilter = smoothFilter; - if (useSpeechConfig) { - h_sbrNoiseFloorEstimate->weightFac = (FIXP_DBL)MAXVAL_DBL; - h_sbrNoiseFloorEstimate->diffThres = INVF_LOW_LEVEL; - } - else { - h_sbrNoiseFloorEstimate->weightFac = FL2FXCONST_DBL(0.25f); - h_sbrNoiseFloorEstimate->diffThres = INVF_MID_LEVEL; - } - - h_sbrNoiseFloorEstimate->timeSlots = timeSlots; - h_sbrNoiseFloorEstimate->noiseBands = noiseBands; - - /* h_sbrNoiseFloorEstimate->ana_max_level is scaled by 0.25 */ - switch(ana_max_level) - { - case 6: - h_sbrNoiseFloorEstimate->ana_max_level = (FIXP_DBL)MAXVAL_DBL; - break; - case 3: - h_sbrNoiseFloorEstimate->ana_max_level = FL2FXCONST_DBL(0.5); - break; - case -3: - h_sbrNoiseFloorEstimate->ana_max_level = FL2FXCONST_DBL(0.125); - break; - default: - /* Should not enter here */ - h_sbrNoiseFloorEstimate->ana_max_level = (FIXP_DBL)MAXVAL_DBL; - break; - } - - /* - calculate number of noise bands and allocate - */ - if(FDKsbrEnc_resetSbrNoiseFloorEstimate(h_sbrNoiseFloorEstimate,freqBandTable,nSfb)) - return(1); - - if(noiseFloorOffset == 0) { - tmp = ((FIXP_DBL)MAXVAL_DBL)>>NOISE_FLOOR_OFFSET_SCALING; - } - else { - /* noiseFloorOffset has to be smaller than 12, because - the result of the calculation below must be smaller than 1: - (2^(noiseFloorOffset/3))*2^4<1 */ - FDK_ASSERT(noiseFloorOffset<12); - - /* Assumes the noise floor offset in tuning table are in q31 */ - /* Change the qformat here when non-zero values would be filled */ - exp = fDivNorm((FIXP_DBL)noiseFloorOffset, 3, &qexp); - tmp = fPow(2, DFRACT_BITS-1, exp, qexp, &qtmp); - tmp = scaleValue(tmp, qtmp-NOISE_FLOOR_OFFSET_SCALING); - } - - for(i=0;inoNoiseBands;i++) { - h_sbrNoiseFloorEstimate->noiseFloorOffset[i] = tmp; - } - - return (0); -} - -/**************************************************************************/ -/*! - \brief Resets the current instance of the noise floor estiamtion - module. - - - \return errorCode, noError if successful - -*/ -/**************************************************************************/ -INT -FDKsbrEnc_resetSbrNoiseFloorEstimate (HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */ - const UCHAR *freqBandTable, /*!< Frequany band table. */ - INT nSfb) /*!< Number of bands in the frequency band table. */ -{ - INT k2,kx; - - /* - * Calculate number of noise bands - ***********************************/ - k2=freqBandTable[nSfb]; - kx=freqBandTable[0]; - if(h_sbrNoiseFloorEstimate->noiseBands == 0){ - h_sbrNoiseFloorEstimate->noNoiseBands = 1; - } - else{ - /* - * Calculate number of noise bands 1,2 or 3 bands/octave - ********************************************************/ - FIXP_DBL tmp, ratio, lg2; - INT ratio_e, qlg2, nNoiseBands; - - ratio = fDivNorm(k2, kx, &ratio_e); - lg2 = fLog2(ratio, ratio_e, &qlg2); - tmp = fMult((FIXP_DBL)(h_sbrNoiseFloorEstimate->noiseBands<<24), lg2); - tmp = scaleValue(tmp, qlg2-23); - - nNoiseBands = (INT)((tmp + (FIXP_DBL)1) >> 1); - - - if (nNoiseBands > MAX_NUM_NOISE_COEFFS ) { - nNoiseBands = MAX_NUM_NOISE_COEFFS; - } - - if( nNoiseBands == 0 ) { - nNoiseBands = 1; - } - - h_sbrNoiseFloorEstimate->noNoiseBands = nNoiseBands; - - } - - - return(downSampleLoRes(h_sbrNoiseFloorEstimate->freqBandTableQmf, - h_sbrNoiseFloorEstimate->noNoiseBands, - freqBandTable,nSfb)); -} - -/**************************************************************************/ -/*! - \brief Deletes the current instancce of the noise floor level - estimation module. - - - \return none - -*/ -/**************************************************************************/ -void -FDKsbrEnc_deleteSbrNoiseFloorEstimate (HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate) /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */ -{ - - if (h_sbrNoiseFloorEstimate) { - /* - nothing to do - */ - } -} diff --git a/libSBRenc/src/nf_est.h b/libSBRenc/src/nf_est.h deleted file mode 100644 index f26f74f..0000000 --- a/libSBRenc/src/nf_est.h +++ /dev/null @@ -1,147 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief Noise floor estimation structs and prototypes -*/ - -#ifndef __NF_EST_H -#define __NF_EST_H - -#include "sbr_encoder.h" -#include "fram_gen.h" - -#define NF_SMOOTHING_LENGTH 4 /*!< Smoothing length of the noise floors. */ - -typedef struct -{ - FIXP_DBL prevNoiseLevels[NF_SMOOTHING_LENGTH][MAX_NUM_NOISE_VALUES]; /*!< The previous noise levels. */ - FIXP_DBL noiseFloorOffset[MAX_NUM_NOISE_VALUES]; /*!< Noise floor offset, scaled with NOISE_FLOOR_OFFSET_SCALING */ - const FIXP_DBL *smoothFilter; /*!< Smoothing filter to use. */ - FIXP_DBL ana_max_level; /*!< Max level allowed. */ - FIXP_DBL weightFac; /*!< Weightening factor for the difference between orig and sbr. */ - INT freqBandTableQmf[MAX_NUM_NOISE_VALUES + 1]; /*!< Frequncy band table for the noise floor bands.*/ - INT noNoiseBands; /*!< Number of noisebands. */ - INT noiseBands; /*!< NoiseBands switch 4 bit.*/ - INT timeSlots; /*!< Number of timeslots in a frame. */ - INVF_MODE diffThres; /*!< Threshold value to control the inverse filtering decision */ -} -SBR_NOISE_FLOOR_ESTIMATE; - -typedef SBR_NOISE_FLOOR_ESTIMATE *HANDLE_SBR_NOISE_FLOOR_ESTIMATE; - -void -FDKsbrEnc_sbrNoiseFloorEstimateQmf(HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */ - const SBR_FRAME_INFO *frame_info, /*!< Time frequency grid of the current frame. */ - FIXP_DBL *noiseLevels, /*!< Pointer to vector to store the noise levels in.*/ - FIXP_DBL **quotaMatrixOrig, /*!< Matrix holding the quota values of the original. */ - SCHAR* indexVector, /*!< Index vector to obtain the patched data. */ - INT missingHarmonicsFlag, /*!< Flag indicating if a strong tonal component will be missing. */ - INT startIndex, /*!< Start index. */ - UINT numberOfEstimatesPerFrame, /*!< The number of tonality estimates per frame. */ - INT transientFrame, /*!< A flag indicating if a transient is present. */ - INVF_MODE* pInvFiltLevels, /*!< Pointer to the vector holding the inverse filtering levels. */ - UINT sbrSyntaxFlags - ); - -INT -FDKsbrEnc_InitSbrNoiseFloorEstimate (HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */ - INT ana_max_level, /*!< Maximum level of the adaptive noise. */ - const UCHAR *freqBandTable, /*!< Frequany band table. */ - INT nSfb, /*!< Number of frequency bands. */ - INT noiseBands, /*!< Number of noise bands per octave. */ - INT noiseFloorOffset, /*!< Noise floor offset. */ - INT timeSlots, /*!< Number of time slots in a frame. */ - UINT useSpeechConfig /*!< Flag: adapt tuning parameters according to speech */ - ); - -INT -FDKsbrEnc_resetSbrNoiseFloorEstimate (HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */ - const UCHAR *freqBandTable, /*!< Frequany band table. */ - INT nSfb); /*!< Number of bands in the frequency band table. */ - -void -FDKsbrEnc_deleteSbrNoiseFloorEstimate (HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate); /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */ - -#endif diff --git a/libSBRenc/src/ps_bitenc.cpp b/libSBRenc/src/ps_bitenc.cpp deleted file mode 100644 index 420ea15..0000000 --- a/libSBRenc/src/ps_bitenc.cpp +++ /dev/null @@ -1,698 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/***************************** MPEG Audio Encoder *************************** - - Initial author: N. Rettelbach - contents/description: Parametric Stereo bitstream encoder - -******************************************************************************/ - -#include "ps_main.h" - - -#include "ps_const.h" -#include "ps_bitenc.h" - -static -inline UCHAR FDKsbrEnc_WriteBits_ps(HANDLE_FDK_BITSTREAM hBitStream, UINT value, - const UINT numberOfBits) -{ - /* hBitStream == NULL happens here intentionally */ - if(hBitStream!=NULL){ - FDKwriteBits(hBitStream, value, numberOfBits); - } - return numberOfBits; -} - -#define SI_SBR_EXTENSION_SIZE_BITS 4 -#define SI_SBR_EXTENSION_ESC_COUNT_BITS 8 -#define SI_SBR_EXTENSION_ID_BITS 2 -#define EXTENSION_ID_PS_CODING 2 -#define PS_EXT_ID_V0 0 - -static const INT iidDeltaCoarse_Offset = 14; -static const INT iidDeltaCoarse_MaxVal = 28; -static const INT iidDeltaFine_Offset = 30; -static const INT iidDeltaFine_MaxVal = 60; - -/* PS Stereo Huffmantable: iidDeltaFreqCoarse */ -static const UINT iidDeltaFreqCoarse_Length[] = -{ - 17, 17, 17, 17, 16, 15, 13, 10, 9, 7, - 6, 5, 4, 3, 1, 3, 4, 5, 6, 6, - 8, 11, 13, 14, 14, 15, 17, 18, 18 -}; -static const UINT iidDeltaFreqCoarse_Code[] = -{ - 0x0001fffb, 0x0001fffc, 0x0001fffd, 0x0001fffa, 0x0000fffc, 0x00007ffc, 0x00001ffd, 0x000003fe, 0x000001fe, 0x0000007e, - 0x0000003c, 0x0000001d, 0x0000000d, 0x00000005, 0000000000, 0x00000004, 0x0000000c, 0x0000001c, 0x0000003d, 0x0000003e, - 0x000000fe, 0x000007fe, 0x00001ffc, 0x00003ffc, 0x00003ffd, 0x00007ffd, 0x0001fffe, 0x0003fffe, 0x0003ffff -}; - -/* PS Stereo Huffmantable: iidDeltaFreqFine */ -static const UINT iidDeltaFreqFine_Length[] = -{ - 18, 18, 18, 18, 18, 18, 18, 18, 18, 17, - 18, 17, 17, 16, 16, 15, 14, 14, 13, 12, - 12, 11, 10, 10, 8, 7, 6, 5, 4, 3, - 1, 3, 4, 5, 6, 7, 8, 9, 10, 11, - 11, 12, 13, 14, 14, 15, 16, 16, 17, 17, - 18, 17, 18, 18, 18, 18, 18, 18, 18, 18, - 18 -}; -static const UINT iidDeltaFreqFine_Code[] = -{ - 0x0001feb4, 0x0001feb5, 0x0001fd76, 0x0001fd77, 0x0001fd74, 0x0001fd75, 0x0001fe8a, 0x0001fe8b, 0x0001fe88, 0x0000fe80, - 0x0001feb6, 0x0000fe82, 0x0000feb8, 0x00007f42, 0x00007fae, 0x00003faf, 0x00001fd1, 0x00001fe9, 0x00000fe9, 0x000007ea, - 0x000007fb, 0x000003fb, 0x000001fb, 0x000001ff, 0x0000007c, 0x0000003c, 0x0000001c, 0x0000000c, 0000000000, 0x00000001, - 0x00000001, 0x00000002, 0x00000001, 0x0000000d, 0x0000001d, 0x0000003d, 0x0000007d, 0x000000fc, 0x000001fc, 0x000003fc, - 0x000003f4, 0x000007eb, 0x00000fea, 0x00001fea, 0x00001fd6, 0x00003fd0, 0x00007faf, 0x00007f43, 0x0000feb9, 0x0000fe83, - 0x0001feb7, 0x0000fe81, 0x0001fe89, 0x0001fe8e, 0x0001fe8f, 0x0001fe8c, 0x0001fe8d, 0x0001feb2, 0x0001feb3, 0x0001feb0, - 0x0001feb1 -}; - -/* PS Stereo Huffmantable: iidDeltaTimeCoarse */ -static const UINT iidDeltaTimeCoarse_Length[] = -{ - 19, 19, 19, 20, 20, 20, 17, 15, 12, 10, - 8, 6, 4, 2, 1, 3, 5, 7, 9, 11, - 13, 14, 17, 19, 20, 20, 20, 20, 20 -}; -static const UINT iidDeltaTimeCoarse_Code[] = -{ - 0x0007fff9, 0x0007fffa, 0x0007fffb, 0x000ffff8, 0x000ffff9, 0x000ffffa, 0x0001fffd, 0x00007ffe, 0x00000ffe, 0x000003fe, - 0x000000fe, 0x0000003e, 0x0000000e, 0x00000002, 0000000000, 0x00000006, 0x0000001e, 0x0000007e, 0x000001fe, 0x000007fe, - 0x00001ffe, 0x00003ffe, 0x0001fffc, 0x0007fff8, 0x000ffffb, 0x000ffffc, 0x000ffffd, 0x000ffffe, 0x000fffff -}; - -/* PS Stereo Huffmantable: iidDeltaTimeFine */ -static const UINT iidDeltaTimeFine_Length[] = -{ - 16, 16, 16, 16, 16, 16, 16, 16, 16, 15, - 15, 15, 15, 15, 15, 14, 14, 13, 13, 13, - 12, 12, 11, 10, 9, 9, 7, 6, 5, 3, - 1, 2, 5, 6, 7, 8, 9, 10, 11, 11, - 12, 12, 13, 13, 14, 14, 15, 15, 15, 15, - 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, - 16 -}; -static const UINT iidDeltaTimeFine_Code[] = -{ - 0x00004ed4, 0x00004ed5, 0x00004ece, 0x00004ecf, 0x00004ecc, 0x00004ed6, 0x00004ed8, 0x00004f46, 0x00004f60, 0x00002718, - 0x00002719, 0x00002764, 0x00002765, 0x0000276d, 0x000027b1, 0x000013b7, 0x000013d6, 0x000009c7, 0x000009e9, 0x000009ed, - 0x000004ee, 0x000004f7, 0x00000278, 0x00000139, 0x0000009a, 0x0000009f, 0x00000020, 0x00000011, 0x0000000a, 0x00000003, - 0x00000001, 0000000000, 0x0000000b, 0x00000012, 0x00000021, 0x0000004c, 0x0000009b, 0x0000013a, 0x00000279, 0x00000270, - 0x000004ef, 0x000004e2, 0x000009ea, 0x000009d8, 0x000013d7, 0x000013d0, 0x000027b2, 0x000027a2, 0x0000271a, 0x0000271b, - 0x00004f66, 0x00004f67, 0x00004f61, 0x00004f47, 0x00004ed9, 0x00004ed7, 0x00004ecd, 0x00004ed2, 0x00004ed3, 0x00004ed0, - 0x00004ed1 -}; - -static const INT iccDelta_Offset = 7; -static const INT iccDelta_MaxVal = 14; -/* PS Stereo Huffmantable: iccDeltaFreq */ -static const UINT iccDeltaFreq_Length[] = -{ - 14, 14, 12, 10, 7, 5, 3, 1, 2, 4, - 6, 8, 9, 11, 13 -}; -static const UINT iccDeltaFreq_Code[] = -{ - 0x00003fff, 0x00003ffe, 0x00000ffe, 0x000003fe, 0x0000007e, 0x0000001e, 0x00000006, 0000000000, 0x00000002, 0x0000000e, - 0x0000003e, 0x000000fe, 0x000001fe, 0x000007fe, 0x00001ffe -}; - -/* PS Stereo Huffmantable: iccDeltaTime */ -static const UINT iccDeltaTime_Length[] = -{ - 14, 13, 11, 9, 7, 5, 3, 1, 2, 4, - 6, 8, 10, 12, 14 -}; -static const UINT iccDeltaTime_Code[] = -{ - 0x00003ffe, 0x00001ffe, 0x000007fe, 0x000001fe, 0x0000007e, 0x0000001e, 0x00000006, 0000000000, 0x00000002, 0x0000000e, - 0x0000003e, 0x000000fe, 0x000003fe, 0x00000ffe, 0x00003fff -}; - - - -static const INT ipdDelta_Offset = 0; -static const INT ipdDelta_MaxVal = 7; -/* PS Stereo Huffmantable: ipdDeltaFreq */ -static const UINT ipdDeltaFreq_Length[] = -{ - 1, 3, 4, 4, 4, 4, 4, 4 -}; -static const UINT ipdDeltaFreq_Code[] = -{ - 0x00000001, 0000000000, 0x00000006, 0x00000004, 0x00000002, 0x00000003, 0x00000005, 0x00000007 -}; - -/* PS Stereo Huffmantable: ipdDeltaTime */ -static const UINT ipdDeltaTime_Length[] = -{ - 1, 3, 4, 5, 5, 4, 4, 3 -}; -static const UINT ipdDeltaTime_Code[] = -{ - 0x00000001, 0x00000002, 0x00000002, 0x00000003, 0x00000002, 0000000000, 0x00000003, 0x00000003 -}; - - -static const INT opdDelta_Offset = 0; -static const INT opdDelta_MaxVal = 7; -/* PS Stereo Huffmantable: opdDeltaFreq */ -static const UINT opdDeltaFreq_Length[] = -{ - 1, 3, 4, 4, 5, 5, 4, 3 -}; -static const UINT opdDeltaFreq_Code[] = -{ - 0x00000001, 0x00000001, 0x00000006, 0x00000004, 0x0000000f, 0x0000000e, 0x00000005, 0000000000, -}; - -/* PS Stereo Huffmantable: opdDeltaTime */ -static const UINT opdDeltaTime_Length[] = -{ - 1, 3, 4, 5, 5, 4, 4, 3 -}; -static const UINT opdDeltaTime_Code[] = -{ - 0x00000001, 0x00000002, 0x00000001, 0x00000007, 0x00000006, 0000000000, 0x00000002, 0x00000003 -}; - -static INT getNoBands(const INT mode) -{ - INT noBands = 0; - - switch (mode) { - case 0: case 3: /* coarse */ - noBands = PS_BANDS_COARSE; - break; - case 1: case 4: /* mid */ - noBands = PS_BANDS_MID; - break; - case 2: case 5: /* fine not supported */ - default: /* coarse as default */ - noBands = PS_BANDS_COARSE; - } - - return noBands; -} - -static INT getIIDRes(INT iidMode) -{ - if(iidMode<3) - return PS_IID_RES_COARSE; - else - return PS_IID_RES_FINE; -} - -static INT -encodeDeltaFreq(HANDLE_FDK_BITSTREAM hBitBuf, - const INT *val, - const INT nBands, - const UINT *codeTable, - const UINT *lengthTable, - const INT tableOffset, - const INT maxVal, - INT *error) -{ - INT bitCnt = 0; - INT lastVal = 0; - INT band; - - for(band=0;bandmaxVal) || (delta<0) ) { - *error = 1; - delta = delta>0?maxVal:0; - } - bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, codeTable[delta], lengthTable[delta]); - } - - return bitCnt; -} - -static INT -encodeDeltaTime(HANDLE_FDK_BITSTREAM hBitBuf, - const INT *val, - const INT *valLast, - const INT nBands, - const UINT *codeTable, - const UINT *lengthTable, - const INT tableOffset, - const INT maxVal, - INT *error) -{ - INT bitCnt = 0; - INT band; - - for(band=0;bandmaxVal) || (delta<0) ) { - *error = 1; - delta = delta>0?maxVal:0; - } - bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, codeTable[delta], lengthTable[delta]); - } - - return bitCnt; -} - -INT FDKsbrEnc_EncodeIid(HANDLE_FDK_BITSTREAM hBitBuf, - const INT *iidVal, - const INT *iidValLast, - const INT nBands, - const PS_IID_RESOLUTION res, - const PS_DELTA mode, - INT *error) -{ - const UINT *codeTable; - const UINT *lengthTable; - INT bitCnt = 0; - - bitCnt = 0; - - switch(mode) { - case PS_DELTA_FREQ: - switch(res) { - case PS_IID_RES_COARSE: - codeTable = iidDeltaFreqCoarse_Code; - lengthTable = iidDeltaFreqCoarse_Length; - bitCnt += encodeDeltaFreq(hBitBuf, iidVal, nBands, codeTable, - lengthTable, iidDeltaCoarse_Offset, - iidDeltaCoarse_MaxVal, error); - break; - case PS_IID_RES_FINE: - codeTable = iidDeltaFreqFine_Code; - lengthTable = iidDeltaFreqFine_Length; - bitCnt += encodeDeltaFreq(hBitBuf, iidVal, nBands, codeTable, - lengthTable, iidDeltaFine_Offset, - iidDeltaFine_MaxVal, error); - break; - default: - *error = 1; - } - break; - - case PS_DELTA_TIME: - switch(res) { - case PS_IID_RES_COARSE: - codeTable = iidDeltaTimeCoarse_Code; - lengthTable = iidDeltaTimeCoarse_Length; - bitCnt += encodeDeltaTime(hBitBuf, iidVal, iidValLast, nBands, codeTable, - lengthTable, iidDeltaCoarse_Offset, - iidDeltaCoarse_MaxVal, error); - break; - case PS_IID_RES_FINE: - codeTable = iidDeltaTimeFine_Code; - lengthTable = iidDeltaTimeFine_Length; - bitCnt += encodeDeltaTime(hBitBuf, iidVal, iidValLast, nBands, codeTable, - lengthTable, iidDeltaFine_Offset, - iidDeltaFine_MaxVal, error); - break; - default: - *error = 1; - } - break; - - default: - *error = 1; - } - - return bitCnt; -} - - -INT FDKsbrEnc_EncodeIcc(HANDLE_FDK_BITSTREAM hBitBuf, - const INT *iccVal, - const INT *iccValLast, - const INT nBands, - const PS_DELTA mode, - INT *error) -{ - const UINT *codeTable; - const UINT *lengthTable; - INT bitCnt = 0; - - switch(mode) { - case PS_DELTA_FREQ: - codeTable = iccDeltaFreq_Code; - lengthTable = iccDeltaFreq_Length; - bitCnt += encodeDeltaFreq(hBitBuf, iccVal, nBands, codeTable, - lengthTable, iccDelta_Offset, iccDelta_MaxVal, error); - break; - - case PS_DELTA_TIME: - codeTable = iccDeltaTime_Code; - lengthTable = iccDeltaTime_Length; - - bitCnt += encodeDeltaTime(hBitBuf, iccVal, iccValLast, nBands, codeTable, - lengthTable, iccDelta_Offset, iccDelta_MaxVal, error); - break; - - default: - *error = 1; - } - - return bitCnt; -} - -INT FDKsbrEnc_EncodeIpd(HANDLE_FDK_BITSTREAM hBitBuf, - const INT *ipdVal, - const INT *ipdValLast, - const INT nBands, - const PS_DELTA mode, - INT *error) -{ - const UINT *codeTable; - const UINT *lengthTable; - INT bitCnt = 0; - - switch(mode) { - case PS_DELTA_FREQ: - codeTable = ipdDeltaFreq_Code; - lengthTable = ipdDeltaFreq_Length; - bitCnt += encodeDeltaFreq(hBitBuf, ipdVal, nBands, codeTable, - lengthTable, ipdDelta_Offset, ipdDelta_MaxVal, error); - break; - - case PS_DELTA_TIME: - codeTable = ipdDeltaTime_Code; - lengthTable = ipdDeltaTime_Length; - - bitCnt += encodeDeltaTime(hBitBuf, ipdVal, ipdValLast, nBands, codeTable, - lengthTable, ipdDelta_Offset, ipdDelta_MaxVal, error); - break; - - default: - *error = 1; - } - - return bitCnt; -} - -INT FDKsbrEnc_EncodeOpd(HANDLE_FDK_BITSTREAM hBitBuf, - const INT *opdVal, - const INT *opdValLast, - const INT nBands, - const PS_DELTA mode, - INT *error) -{ - const UINT *codeTable; - const UINT *lengthTable; - INT bitCnt = 0; - - switch(mode) { - case PS_DELTA_FREQ: - codeTable = opdDeltaFreq_Code; - lengthTable = opdDeltaFreq_Length; - bitCnt += encodeDeltaFreq(hBitBuf, opdVal, nBands, codeTable, - lengthTable, opdDelta_Offset, opdDelta_MaxVal, error); - break; - - case PS_DELTA_TIME: - codeTable = opdDeltaTime_Code; - lengthTable = opdDeltaTime_Length; - - bitCnt += encodeDeltaTime(hBitBuf, opdVal, opdValLast, nBands, codeTable, - lengthTable, opdDelta_Offset, opdDelta_MaxVal, error); - break; - - default: - *error = 1; - } - - return bitCnt; -} - -static INT encodeIpdOpd(HANDLE_PS_OUT psOut, - HANDLE_FDK_BITSTREAM hBitBuf ) -{ - INT bitCnt = 0; - INT error = 0; - INT env; - - FDKsbrEnc_WriteBits_ps(hBitBuf, psOut->enableIpdOpd, 1); - - if(psOut->enableIpdOpd==1) { - INT *ipdLast = psOut->ipdLast; - INT *opdLast = psOut->opdLast; - - for(env=0; envnEnvelopes; env++) { - bitCnt += FDKsbrEnc_WriteBits_ps( hBitBuf, psOut->deltaIPD[env], 1); - bitCnt += FDKsbrEnc_EncodeIpd( hBitBuf, - psOut->ipd[env], - ipdLast, - getNoBands(psOut->iidMode), - psOut->deltaIPD[env], - &error); - - bitCnt += FDKsbrEnc_WriteBits_ps( hBitBuf, psOut->deltaOPD[env], 1); - bitCnt += FDKsbrEnc_EncodeOpd( hBitBuf, - psOut->opd[env], - opdLast, - getNoBands(psOut->iidMode), - psOut->deltaOPD[env], - &error ); - } - /* reserved bit */ - bitCnt += FDKsbrEnc_WriteBits_ps( hBitBuf, 0, 1); - } - - - return bitCnt; -} - -static INT getEnvIdx(const INT nEnvelopes, const INT frameClass) -{ - INT envIdx = 0; - - switch(nEnvelopes) { - case 0: - envIdx = 0; - break; - - case 1: - if (frameClass==0) - envIdx = 1; - else - envIdx = 0; - break; - - case 2: - if (frameClass==0) - envIdx = 2; - else - envIdx = 1; - break; - - case 3: - envIdx = 2; - break; - - case 4: - envIdx = 3; - break; - - default: - /* unsupported number of envelopes */ - envIdx = 0; - } - - return envIdx; -} - - -static INT encodePSExtension(const HANDLE_PS_OUT psOut, - HANDLE_FDK_BITSTREAM hBitBuf ) -{ - INT bitCnt = 0; - - if(psOut->enableIpdOpd==1) { - INT ipdOpdBits = 0; - INT extSize = (2 + encodeIpdOpd(psOut,NULL)+7)>>3; - - if(extSize<15) { - bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, extSize, 4); - } - else { - bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, 15 , 4); - bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, (extSize-15), 8); - } - - /* write ipd opd data */ - ipdOpdBits += FDKsbrEnc_WriteBits_ps(hBitBuf, PS_EXT_ID_V0, 2); - ipdOpdBits += encodeIpdOpd(psOut, hBitBuf ); - - /* byte align the ipd opd data */ - if(ipdOpdBits%8) - ipdOpdBits += FDKsbrEnc_WriteBits_ps(hBitBuf, 0, (8-(ipdOpdBits%8)) ); - - bitCnt += ipdOpdBits; - } - - return (bitCnt); -} - -INT FDKsbrEnc_WritePSBitstream(const HANDLE_PS_OUT psOut, - HANDLE_FDK_BITSTREAM hBitBuf ) -{ - INT psExtEnable = 0; - INT bitCnt = 0; - INT error = 0; - INT env; - - if(psOut != NULL){ - - /* PS HEADER */ - bitCnt += FDKsbrEnc_WriteBits_ps( hBitBuf, psOut->enablePSHeader, 1); - - if(psOut->enablePSHeader) { - - bitCnt += FDKsbrEnc_WriteBits_ps( hBitBuf, psOut->enableIID, 1); - if(psOut->enableIID) { - bitCnt += FDKsbrEnc_WriteBits_ps( hBitBuf, psOut->iidMode, 3); - } - bitCnt += FDKsbrEnc_WriteBits_ps( hBitBuf, psOut->enableICC, 1); - if(psOut->enableICC) { - bitCnt += FDKsbrEnc_WriteBits_ps( hBitBuf, psOut->iccMode, 3); - } - if(psOut->enableIpdOpd) { - psExtEnable = 1; - } - bitCnt += FDKsbrEnc_WriteBits_ps( hBitBuf, psExtEnable, 1); - } - - /* Frame class, number of envelopes */ - bitCnt += FDKsbrEnc_WriteBits_ps( hBitBuf, psOut->frameClass, 1); - bitCnt += FDKsbrEnc_WriteBits_ps( hBitBuf, getEnvIdx(psOut->nEnvelopes, psOut->frameClass), 2); - - if(psOut->frameClass==1) { - for(env=0; envnEnvelopes; env++) { - bitCnt += FDKsbrEnc_WriteBits_ps( hBitBuf, psOut->frameBorder[env], 5); - } - } - - if(psOut->enableIID==1) { - INT *iidLast = psOut->iidLast; - for(env=0; envnEnvelopes; env++) { - bitCnt += FDKsbrEnc_WriteBits_ps( hBitBuf, psOut->deltaIID[env], 1); - bitCnt += FDKsbrEnc_EncodeIid( hBitBuf, - psOut->iid[env], - iidLast, - getNoBands(psOut->iidMode), - (PS_IID_RESOLUTION)getIIDRes(psOut->iidMode), - psOut->deltaIID[env], - &error ); - - iidLast = psOut->iid[env]; - } - } - - if(psOut->enableICC==1) { - INT *iccLast = psOut->iccLast; - for(env=0; envnEnvelopes; env++) { - bitCnt += FDKsbrEnc_WriteBits_ps( hBitBuf, psOut->deltaICC[env], 1); - bitCnt += FDKsbrEnc_EncodeIcc( hBitBuf, - psOut->icc[env], - iccLast, - getNoBands(psOut->iccMode), - psOut->deltaICC[env], - &error); - - iccLast = psOut->icc[env]; - } - } - - if(psExtEnable!=0) { - bitCnt += encodePSExtension(psOut, hBitBuf); - } - - } /* if(psOut != NULL) */ - - return bitCnt; -} - diff --git a/libSBRenc/src/ps_bitenc.h b/libSBRenc/src/ps_bitenc.h deleted file mode 100644 index e98fe58..0000000 --- a/libSBRenc/src/ps_bitenc.h +++ /dev/null @@ -1,177 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/***************************** MPEG Audio Encoder *************************** - - Initial author: N. Rettelbach - contents/description: Parametric Stereo bitstream encoder - -******************************************************************************/ - -#include "ps_main.h" -#include "ps_const.h" -#include "FDK_bitstream.h" - -#ifndef PS_BITENC_H -#define PS_BITENC_H - -typedef struct T_PS_OUT { - - INT enablePSHeader; - INT enableIID; - INT iidMode; - INT enableICC; - INT iccMode; - INT enableIpdOpd; - - INT frameClass; - INT nEnvelopes; - /* ENV data */ - INT frameBorder[PS_MAX_ENVELOPES]; - - /* iid data */ - PS_DELTA deltaIID[PS_MAX_ENVELOPES]; - INT iid[PS_MAX_ENVELOPES][PS_MAX_BANDS]; - INT iidLast[PS_MAX_BANDS]; - - /* icc data */ - PS_DELTA deltaICC[PS_MAX_ENVELOPES]; - INT icc[PS_MAX_ENVELOPES][PS_MAX_BANDS]; - INT iccLast[PS_MAX_BANDS]; - - /* ipd data */ - PS_DELTA deltaIPD[PS_MAX_ENVELOPES]; - INT ipd[PS_MAX_ENVELOPES][PS_MAX_BANDS]; - INT ipdLast[PS_MAX_BANDS]; - - /* opd data */ - PS_DELTA deltaOPD[PS_MAX_ENVELOPES]; - INT opd[PS_MAX_ENVELOPES][PS_MAX_BANDS]; - INT opdLast[PS_MAX_BANDS]; - -} PS_OUT, *HANDLE_PS_OUT; - - -#ifdef __cplusplus -extern "C" { -#endif /* __cplusplus */ - -INT FDKsbrEnc_EncodeIid(HANDLE_FDK_BITSTREAM hBitBuf, - const INT *iidVal, - const INT *iidValLast, - const INT nBands, - const PS_IID_RESOLUTION res, - const PS_DELTA mode, - INT *error); - -INT FDKsbrEnc_EncodeIcc(HANDLE_FDK_BITSTREAM hBitBuf, - const INT *iccVal, - const INT *iccValLast, - const INT nBands, - const PS_DELTA mode, - INT *error); - -INT FDKsbrEnc_EncodeIpd(HANDLE_FDK_BITSTREAM hBitBuf, - const INT *ipdVal, - const INT *ipdValLast, - const INT nBands, - const PS_DELTA mode, - INT *error); - -INT FDKsbrEnc_EncodeOpd(HANDLE_FDK_BITSTREAM hBitBuf, - const INT *opdVal, - const INT *opdValLast, - const INT nBands, - const PS_DELTA mode, - INT *error); - -INT FDKsbrEnc_WritePSBitstream(const HANDLE_PS_OUT psOut, - HANDLE_FDK_BITSTREAM hBitBuf); - - -#ifdef __cplusplus -} -#endif /* __cplusplus */ - - -#endif /* #ifndef PS_BITENC_H */ diff --git a/libSBRenc/src/ps_const.h b/libSBRenc/src/ps_const.h deleted file mode 100644 index 633d210..0000000 --- a/libSBRenc/src/ps_const.h +++ /dev/null @@ -1,148 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/***************************** MPEG Audio Encoder *************************** - - Initial author: N. Rettelbach - contents/description: Parametric Stereo constants - -******************************************************************************/ - -#ifndef PS_CONST_H -#define PS_CONST_H - -#define MAX_PS_CHANNELS ( 2 ) -#define HYBRID_MAX_QMF_BANDS ( 3 ) -#define HYBRID_FILTER_LENGTH ( 13 ) -#define HYBRID_FILTER_DELAY ( (HYBRID_FILTER_LENGTH-1)/2 ) - -#define HYBRID_FRAMESIZE ( QMF_MAX_TIME_SLOTS ) -#define HYBRID_READ_OFFSET ( 10 ) - -#define MAX_HYBRID_BANDS ( (QMF_CHANNELS-HYBRID_MAX_QMF_BANDS+10) ) - - -typedef enum { - PS_RES_COARSE = 0, - PS_RES_MID = 1, - PS_RES_FINE = 2 -} PS_RESOLUTION; - -typedef enum { - PS_BANDS_COARSE = 10, - PS_BANDS_MID = 20, - PS_MAX_BANDS = PS_BANDS_MID -} PS_BANDS; - -typedef enum { - PS_IID_RES_COARSE=0, - PS_IID_RES_FINE -} PS_IID_RESOLUTION; - -typedef enum { - PS_ICC_ROT_A=0, - PS_ICC_ROT_B -} PS_ICC_ROTATION_MODE; - -typedef enum { - PS_DELTA_FREQ, - PS_DELTA_TIME -} PS_DELTA; - - -typedef enum { - PS_MAX_ENVELOPES = 4 - -} PS_CONSTS; - -typedef enum { - PSENC_OK = 0x0000, /*!< No error happened. All fine. */ - PSENC_INVALID_HANDLE = 0x0020, /*!< Handle passed to function call was invalid. */ - PSENC_MEMORY_ERROR = 0x0021, /*!< Memory allocation failed. */ - PSENC_INIT_ERROR = 0x0040, /*!< General initialization error. */ - PSENC_ENCODE_ERROR = 0x0060 /*!< The encoding process was interrupted by an unexpected error. */ - -} FDK_PSENC_ERROR; - - -#endif diff --git a/libSBRenc/src/ps_encode.cpp b/libSBRenc/src/ps_encode.cpp deleted file mode 100644 index fec39e8..0000000 --- a/libSBRenc/src/ps_encode.cpp +++ /dev/null @@ -1,1054 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/***************************** MPEG Audio Encoder *************************** - - Initial Authors: M. Neuendorf, N. Rettelbach, M. Multrus - Contents/Description: PS parameter extraction, encoding - -******************************************************************************/ -/*! - \file - \brief PS parameter extraction, encoding functions -*/ - -#include "ps_main.h" - - -#include "sbr_ram.h" -#include "ps_encode.h" - -#include "qmf.h" - -#include "ps_const.h" -#include "sbr_misc.h" - -#include "genericStds.h" - -inline void FDKsbrEnc_addFIXP_DBL(const FIXP_DBL *X, const FIXP_DBL *Y, FIXP_DBL *Z, INT n) -{ - for (INT i=0; i>1) + (Y[i]>>1); -} - -#define LOG10_2_10 3.01029995664f /* 10.0f*log10(2.f) */ - -static const INT iidGroupBordersLoRes[QMF_GROUPS_LO_RES + SUBQMF_GROUPS_LO_RES + 1] = -{ - 0, 1, 2, 3, 4, 5, /* 6 subqmf subbands - 0th qmf subband */ - 6, 7, /* 2 subqmf subbands - 1st qmf subband */ - 8, 9, /* 2 subqmf subbands - 2nd qmf subband */ - 10, 11, 12, 13, 14, 15, 16, 18, 21, 25, 30, 42, 71 -}; - -static const UCHAR iidGroupWidthLdLoRes[QMF_GROUPS_LO_RES + SUBQMF_GROUPS_LO_RES] = -{ - 0, 0, 0, 0, 0, 0, - 0, 0, - 0, 0, - 0, 0, 0, 0, 0, 0, 1, 2, 2, 3, 4, 5 -}; - - -static const INT subband2parameter20[QMF_GROUPS_LO_RES + SUBQMF_GROUPS_LO_RES] = -{ - 1, 0, 0, 1, 2, 3, /* 6 subqmf subbands - 0th qmf subband */ - 4, 5, /* 2 subqmf subbands - 1st qmf subband */ - 6, 7, /* 2 subqmf subbands - 2nd qmf subband */ - 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19 -}; - - -typedef enum { - MAX_TIME_DIFF_FRAMES = 20, - MAX_PS_NOHEADER_CNT = 10, - MAX_NOENV_CNT = 10, - DO_NOT_USE_THIS_MODE = 0x7FFFFF -} __PS_CONSTANTS; - - - -static const FIXP_DBL iidQuant_fx[15] = { - (FIXP_DBL)0xce000000, (FIXP_DBL)0xdc000000, (FIXP_DBL)0xe4000000, (FIXP_DBL)0xec000000, (FIXP_DBL)0xf2000000, (FIXP_DBL)0xf8000000, (FIXP_DBL)0xfc000000, (FIXP_DBL)0x00000000, - (FIXP_DBL)0x04000000, (FIXP_DBL)0x08000000, (FIXP_DBL)0x0e000000, (FIXP_DBL)0x14000000, (FIXP_DBL)0x1c000000, (FIXP_DBL)0x24000000, (FIXP_DBL)0x32000000 -}; - -static const FIXP_DBL iidQuantFine_fx[31] = { - (FIXP_DBL)0x9c000001, (FIXP_DBL)0xa6000001, (FIXP_DBL)0xb0000001, (FIXP_DBL)0xba000001, (FIXP_DBL)0xc4000000, (FIXP_DBL)0xce000000, (FIXP_DBL)0xd4000000, (FIXP_DBL)0xda000000, - (FIXP_DBL)0xe0000000, (FIXP_DBL)0xe6000000, (FIXP_DBL)0xec000000, (FIXP_DBL)0xf0000000, (FIXP_DBL)0xf4000000, (FIXP_DBL)0xf8000000, (FIXP_DBL)0xfc000000, (FIXP_DBL)0x00000000, - (FIXP_DBL)0x04000000, (FIXP_DBL)0x08000000, (FIXP_DBL)0x0c000000, (FIXP_DBL)0x10000000, (FIXP_DBL)0x14000000, (FIXP_DBL)0x1a000000, (FIXP_DBL)0x20000000, (FIXP_DBL)0x26000000, - (FIXP_DBL)0x2c000000, (FIXP_DBL)0x32000000, (FIXP_DBL)0x3c000000, (FIXP_DBL)0x45ffffff, (FIXP_DBL)0x4fffffff, (FIXP_DBL)0x59ffffff, (FIXP_DBL)0x63ffffff -}; - - - -static const FIXP_DBL iccQuant[8] = { - (FIXP_DBL)0x7fffffff, (FIXP_DBL)0x77ef9d7f, (FIXP_DBL)0x6babc97f, (FIXP_DBL)0x4ceaf27f, (FIXP_DBL)0x2f0ed3c0, (FIXP_DBL)0x00000000, (FIXP_DBL)0xb49ba601, (FIXP_DBL)0x80000000 -}; - -static FDK_PSENC_ERROR InitPSData( - HANDLE_PS_DATA hPsData - ) -{ - FDK_PSENC_ERROR error = PSENC_OK; - - if(hPsData == NULL) { - error = PSENC_INVALID_HANDLE; - } - else { - int i, env; - FDKmemclear(hPsData,sizeof(PS_DATA)); - - for (i=0; iiidIdxLast[i] = 0; - hPsData->iccIdxLast[i] = 0; - } - - hPsData->iidEnable = hPsData->iidEnableLast = 0; - hPsData->iccEnable = hPsData->iccEnableLast = 0; - hPsData->iidQuantMode = hPsData->iidQuantModeLast = PS_IID_RES_COARSE; - hPsData->iccQuantMode = hPsData->iccQuantModeLast = PS_ICC_ROT_A; - - for(env=0; enviccDiffMode[env] = PS_DELTA_FREQ; - hPsData->iccDiffMode[env] = PS_DELTA_FREQ; - - for (i=0; iiidIdx[env][i] = 0; - hPsData->iccIdx[env][i] = 0; - } - } - - hPsData->nEnvelopesLast = 0; - - hPsData->headerCnt = MAX_PS_NOHEADER_CNT; - hPsData->iidTimeCnt = MAX_TIME_DIFF_FRAMES; - hPsData->iccTimeCnt = MAX_TIME_DIFF_FRAMES; - hPsData->noEnvCnt = MAX_NOENV_CNT; - } - - return error; -} - -static FIXP_DBL quantizeCoef( const FIXP_DBL *RESTRICT input, - const INT nBands, - const FIXP_DBL *RESTRICT quantTable, - const INT idxOffset, - const INT nQuantSteps, - INT *RESTRICT quantOut) -{ - INT idx, band; - FIXP_DBL quantErr = FL2FXCONST_DBL(0.f); - - for (band=0; band>1)-(quantTable[idx+1]>>1)) > - fixp_abs((input[band]>>1)-(quantTable[idx]>>1)) ) - { - break; - } - } - quantErr += (fixp_abs(input[band]-quantTable[idx])>>PS_QUANT_SCALE); /* don't scale before subtraction; diff smaller (64-25)/64 */ - quantOut[band] = idx - idxOffset; - } - - return quantErr; -} - -static INT getICCMode(const INT nBands, - const INT rotType) -{ - INT mode = 0; - - switch(nBands) { - case PS_BANDS_COARSE: - mode = PS_RES_COARSE; - break; - case PS_BANDS_MID: - mode = PS_RES_MID; - break; - default: - mode = 0; - } - if(rotType==PS_ICC_ROT_B){ - mode += 3; - } - - return mode; -} - - -static INT getIIDMode(const INT nBands, - const INT iidRes) -{ - INT mode = 0; - - switch(nBands) { - case PS_BANDS_COARSE: - mode = PS_RES_COARSE; - break; - case PS_BANDS_MID: - mode = PS_RES_MID; - break; - default: - mode = 0; - break; - } - - if(iidRes == PS_IID_RES_FINE){ - mode += 3; - } - - return mode; -} - - -static INT envelopeReducible(FIXP_DBL iid[PS_MAX_ENVELOPES][PS_MAX_BANDS], - FIXP_DBL icc[PS_MAX_ENVELOPES][PS_MAX_BANDS], - INT psBands, - INT nEnvelopes) -{ - #define THRESH_SCALE 7 - - INT reducible = 1; /* true */ - INT e = 0, b = 0; - FIXP_DBL dIid = FL2FXCONST_DBL(0.f); - FIXP_DBL dIcc = FL2FXCONST_DBL(0.f); - - FIXP_DBL iidErrThreshold, iccErrThreshold; - FIXP_DBL iidMeanError, iccMeanError; - - /* square values to prevent sqrt, - multiply bands to prevent division; bands shifted DFRACT_BITS instead (DFRACT_BITS-1) because fMultDiv2 used*/ - iidErrThreshold = fMultDiv2 ( FL2FXCONST_DBL(6.5f*6.5f/(IID_SCALE_FT*IID_SCALE_FT)), (FIXP_DBL)(psBands<<((DFRACT_BITS)-THRESH_SCALE)) ); - iccErrThreshold = fMultDiv2 ( FL2FXCONST_DBL(0.75f*0.75f), (FIXP_DBL)(psBands<<((DFRACT_BITS)-THRESH_SCALE)) ); - - if (nEnvelopes <= 1) { - reducible = 0; - } else { - - /* mean error criterion */ - for (e=0; (e < nEnvelopes/2) && (reducible!=0 ) ; e++) { - iidMeanError = iccMeanError = FL2FXCONST_DBL(0.f); - for(b=0; b>1) - (iid[2*e+1][b]>>1); /* scale 1 bit; squared -> 2 bit */ - dIcc = (icc[2*e][b]>>1) - (icc[2*e+1][b]>>1); - iidMeanError += fPow2Div2(dIid)>>(5-1); /* + (bands=20) scale = 5 */ - iccMeanError += fPow2Div2(dIcc)>>(5-1); - } /* --> scaling = 7 bit = THRESH_SCALE !! */ - - /* instead sqrt values are squared! - instead of division, multiply threshold with psBands - scaling necessary!! */ - - /* quit as soon as threshold is reached */ - if ( (iidMeanError > (iidErrThreshold)) || - (iccMeanError > (iccErrThreshold)) ) { - reducible = 0; - } - } - } /* nEnvelopes != 1 */ - - return reducible; -} - - -static void processIidData(PS_DATA *psData, - FIXP_DBL iid[PS_MAX_ENVELOPES][PS_MAX_BANDS], - const INT psBands, - const INT nEnvelopes, - const FIXP_DBL quantErrorThreshold) -{ - INT iidIdxFine [PS_MAX_ENVELOPES][PS_MAX_BANDS]; - INT iidIdxCoarse[PS_MAX_ENVELOPES][PS_MAX_BANDS]; - - FIXP_DBL errIID = FL2FXCONST_DBL(0.f); - FIXP_DBL errIIDFine = FL2FXCONST_DBL(0.f); - INT bitsIidFreq = 0; - INT bitsIidTime = 0; - INT bitsFineTot = 0; - INT bitsCoarseTot = 0; - INT error = 0; - INT env, band; - INT diffMode[PS_MAX_ENVELOPES], diffModeFine[PS_MAX_ENVELOPES]; - INT loudnDiff = 0; - INT iidTransmit = 0; - - bitsIidFreq = bitsIidTime = 0; - - /* Quantize IID coefficients */ - for(env=0;enviidEnable = 0; - for(env=0;env fMultI(FL2FXCONST_DBL(0.7f),iidTransmit)){ /* 0.7f empiric value */ - psData->iidEnable = 1; - } - - /* if iid not active -> RESET data */ - if(psData->iidEnable==0) { - psData->iidTimeCnt = MAX_TIME_DIFF_FRAMES; - for(env=0;enviidDiffMode[env] = PS_DELTA_FREQ; - FDKmemclear(psData->iidIdx[env], sizeof(INT)*psBands); - } - return; - } - - /* count COARSE quantization bits for first envelope*/ - bitsIidFreq = FDKsbrEnc_EncodeIid(NULL, iidIdxCoarse[0], NULL, psBands, PS_IID_RES_COARSE, PS_DELTA_FREQ, &error); - - if( (psData->iidTimeCnt>=MAX_TIME_DIFF_FRAMES) || (psData->iidQuantModeLast==PS_IID_RES_FINE) ) { - bitsIidTime = DO_NOT_USE_THIS_MODE; - } - else { - bitsIidTime = FDKsbrEnc_EncodeIid(NULL, iidIdxCoarse[0], psData->iidIdxLast, psBands, PS_IID_RES_COARSE, PS_DELTA_TIME, &error); - } - - /* decision DELTA_FREQ vs DELTA_TIME */ - if(bitsIidTime>bitsIidFreq) { - diffMode[0] = PS_DELTA_FREQ; - bitsCoarseTot = bitsIidFreq; - } - else { - diffMode[0] = PS_DELTA_TIME; - bitsCoarseTot = bitsIidTime; - } - - /* count COARSE quantization bits for following envelopes*/ - for(env=1;envbitsIidFreq) { - diffMode[env] = PS_DELTA_FREQ; - bitsCoarseTot += bitsIidFreq; - } - else { - diffMode[env] = PS_DELTA_TIME; - bitsCoarseTot += bitsIidTime; - } - } - - - /* count FINE quantization bits for first envelope*/ - bitsIidFreq = FDKsbrEnc_EncodeIid(NULL, iidIdxFine[0], NULL, psBands, PS_IID_RES_FINE, PS_DELTA_FREQ, &error); - - if( (psData->iidTimeCnt>=MAX_TIME_DIFF_FRAMES) || (psData->iidQuantModeLast==PS_IID_RES_COARSE) ) { - bitsIidTime = DO_NOT_USE_THIS_MODE; - } - else { - bitsIidTime = FDKsbrEnc_EncodeIid(NULL, iidIdxFine[0], psData->iidIdxLast, psBands, PS_IID_RES_FINE, PS_DELTA_TIME, &error); - } - - /* decision DELTA_FREQ vs DELTA_TIME */ - if(bitsIidTime>bitsIidFreq) { - diffModeFine[0] = PS_DELTA_FREQ; - bitsFineTot = bitsIidFreq; - } - else { - diffModeFine[0] = PS_DELTA_TIME; - bitsFineTot = bitsIidTime; - } - - /* count FINE quantization bits for following envelopes*/ - for(env=1;envbitsIidFreq) { - diffModeFine[env] = PS_DELTA_FREQ; - bitsFineTot += bitsIidFreq; - } - else { - diffModeFine[env] = PS_DELTA_TIME; - bitsFineTot += bitsIidTime; - } - } - - if(bitsFineTot == bitsCoarseTot){ - /* if same number of bits is needed, use the quantization with lower error */ - if(errIIDFine < errIID){ - bitsCoarseTot = DO_NOT_USE_THIS_MODE; - } else { - bitsFineTot = DO_NOT_USE_THIS_MODE; - } - } else { - /* const FIXP_DBL minThreshold = FL2FXCONST_DBL(0.2f/(IID_SCALE_FT*PS_QUANT_SCALE_FT)*(psBands*nEnvelopes)); */ - const FIXP_DBL minThreshold = (FIXP_DBL)((LONG)0x00019999 * (psBands*nEnvelopes)); - - /* decision RES_FINE vs RES_COARSE */ - /* test if errIIDFine*quantErrorThreshold < errIID */ - /* shiftVal 2 comes from scaling of quantErrorThreshold */ - if(fixMax(((errIIDFine>>1)+(minThreshold>>1))>>1, fMult(quantErrorThreshold,errIIDFine)) < (errIID>>2) ) { - bitsCoarseTot = DO_NOT_USE_THIS_MODE; - } - else if(fixMax(((errIID>>1)+(minThreshold>>1))>>1, fMult(quantErrorThreshold,errIID)) < (errIIDFine>>2) ) { - bitsFineTot = DO_NOT_USE_THIS_MODE; - } - } - - /* decision RES_FINE vs RES_COARSE */ - if(bitsFineTotiidQuantMode = PS_IID_RES_FINE; - for(env=0;enviidDiffMode[env] = diffModeFine[env]; - FDKmemcpy(psData->iidIdx[env], iidIdxFine[env], psBands*sizeof(INT)); - } - } - else { - psData->iidQuantMode = PS_IID_RES_COARSE; - for(env=0;enviidDiffMode[env] = diffMode[env]; - FDKmemcpy(psData->iidIdx[env], iidIdxCoarse[env], psBands*sizeof(INT)); - } - } - - /* Count DELTA_TIME encoding streaks */ - for(env=0;enviidDiffMode[env]==PS_DELTA_TIME) - psData->iidTimeCnt++; - else - psData->iidTimeCnt=0; - } -} - - -static INT similarIid(PS_DATA *psData, - const INT psBands, - const INT nEnvelopes) -{ - const INT diffThr = (psData->iidQuantMode == PS_IID_RES_COARSE) ? 2 : 3; - const INT sumDiffThr = diffThr * psBands/4; - INT similar = 0; - INT diff = 0; - INT sumDiff = 0; - INT env = 0; - INT b = 0; - if ((nEnvelopes == psData->nEnvelopesLast) && (nEnvelopes==1)) { - similar = 1; - for (env=0; enviidIdx[env][b] - psData->iidIdxLast[b]); - sumDiff += diff; - if ( (diff > diffThr) /* more than x quantization steps in any band */ - || (sumDiff > sumDiffThr) ) { /* more than x quantisations steps overall difference */ - similar = 0; - } - b++; - } while ((b0)); - } - } /* nEnvelopes==1 */ - - return similar; -} - - -static INT similarIcc(PS_DATA *psData, - const INT psBands, - const INT nEnvelopes) -{ - const INT diffThr = 2; - const INT sumDiffThr = diffThr * psBands/4; - INT similar = 0; - INT diff = 0; - INT sumDiff = 0; - INT env = 0; - INT b = 0; - if ((nEnvelopes == psData->nEnvelopesLast) && (nEnvelopes==1)) { - similar = 1; - for (env=0; enviccIdx[env][b] - psData->iccIdxLast[b]); - sumDiff += diff; - if ( (diff > diffThr) /* more than x quantisation step in any band */ - || (sumDiff > sumDiffThr) ) { /* more than x quantisations steps overall difference */ - similar = 0; - } - b++; - } while ((b0)); - } - } /* nEnvelopes==1 */ - - return similar; -} - -static void processIccData(PS_DATA *psData, - FIXP_DBL icc[PS_MAX_ENVELOPES][PS_MAX_BANDS], /* const input values: unable to declare as const, since it does not poINT to const memory */ - const INT psBands, - const INT nEnvelopes) -{ - FIXP_DBL errICC = FL2FXCONST_DBL(0.f); - INT env, band; - INT bitsIccFreq, bitsIccTime; - INT error = 0; - INT inCoherence=0, iccTransmit=0; - INT *iccIdxLast; - - iccIdxLast = psData->iccIdxLast; - - /* Quantize ICC coefficients */ - for(env=0;enviccIdx[env]); - } - - /* Check if ICC coefficients should be used */ - psData->iccEnable = 0; - for(env=0;enviccIdx[env][band]; - iccTransmit ++; - } - } - if(inCoherence > fMultI(FL2FXCONST_DBL(0.5f),iccTransmit)){ /* 0.5f empiric value */ - psData->iccEnable = 1; - } - - if(psData->iccEnable==0) { - psData->iccTimeCnt = MAX_TIME_DIFF_FRAMES; - for(env=0;enviccDiffMode[env] = PS_DELTA_FREQ; - FDKmemclear(psData->iccIdx[env], sizeof(INT)*psBands); - } - return; - } - - for(env=0;enviccIdx[env], NULL, psBands, PS_DELTA_FREQ, &error); - - if(psData->iccTimeCnticcIdx[env], iccIdxLast, psBands, PS_DELTA_TIME, &error); - } - else { - bitsIccTime = DO_NOT_USE_THIS_MODE; - } - - if(bitsIccFreq>bitsIccTime) { - psData->iccDiffMode[env] = PS_DELTA_TIME; - psData->iccTimeCnt++; - } - else { - psData->iccDiffMode[env] = PS_DELTA_FREQ; - psData->iccTimeCnt=0; - } - iccIdxLast = psData->iccIdx[env]; - } -} - -static void calculateIID(FIXP_DBL ldPwrL[PS_MAX_ENVELOPES][PS_MAX_BANDS], - FIXP_DBL ldPwrR[PS_MAX_ENVELOPES][PS_MAX_BANDS], - FIXP_DBL iid[PS_MAX_ENVELOPES][PS_MAX_BANDS], - INT nEnvelopes, - INT psBands) -{ - INT i=0; - INT env=0; - for(env=0; env>(LD_DATA_SHIFT+1)) ); - IID = fixMax( IID, (FIXP_DBL)(MINVAL_DBL>>(LD_DATA_SHIFT+1)) ); - iid[env][i] = IID << (LD_DATA_SHIFT+1); - } - } -} - -static void calculateICC(FIXP_DBL ldPwrL[PS_MAX_ENVELOPES][PS_MAX_BANDS], - FIXP_DBL ldPwrR[PS_MAX_ENVELOPES][PS_MAX_BANDS], - FIXP_DBL pwrCr[PS_MAX_ENVELOPES][PS_MAX_BANDS], - FIXP_DBL pwrCi[PS_MAX_ENVELOPES][PS_MAX_BANDS], - FIXP_DBL icc[PS_MAX_ENVELOPES][PS_MAX_BANDS], - INT nEnvelopes, - INT psBands) -{ - INT i = 0; - INT env = 0; - INT border = psBands; - - switch (psBands) { - case PS_BANDS_COARSE: - border = 5; - break; - case PS_BANDS_MID: - border = 11; - break; - default: - break; - } - - for(env=0; env>1) + (ldPwrR[env][i]>>1) + (FIXP_DBL)1) ); - INT scale, invScale = CountLeadingBits(invNrg); - - scale = (DFRACT_BITS-1) - invScale; - ICC = fMult(pwrCr[env][i], invNrg<>1)>>1) - (FIXP_DBL)((sc1-1)<<(DFRACT_BITS-1-LD_DATA_SHIFT)) ); - - FIXP_DBL invNrg = CalcInvLdData ( -((ldPwrL[env][i]>>1) + (ldPwrR[env][i]>>1) + (FIXP_DBL)1) ); - sc1 = CountLeadingBits(invNrg); - invNrg <<= sc1; - - sc2 = CountLeadingBits(ICC); - ICC = fMult(ICC<>= -sc1; - } - else { - if (ICC >= ((FIXP_DBL)MAXVAL_DBL>>sc1) ) - ICC = (FIXP_DBL)MAXVAL_DBL; - else - ICC <<= sc1; - } - - icc[env][i] = ICC; - } - } -} - -void FDKsbrEnc_initPsBandNrgScale(HANDLE_PS_ENCODE hPsEncode) -{ - INT group, bin; - INT nIidGroups = hPsEncode->nQmfIidGroups + hPsEncode->nSubQmfIidGroups; - - FDKmemclear(hPsEncode->psBandNrgScale, PS_MAX_BANDS*sizeof(SCHAR)); - - for (group=0; group < nIidGroups; group++) { - /* Translate group to bin */ - bin = hPsEncode->subband2parameterIndex[group]; - - /* Translate from 20 bins to 10 bins */ - if (hPsEncode->psEncMode == PS_BANDS_COARSE) { - bin = bin>>1; - } - - hPsEncode->psBandNrgScale[bin] = (hPsEncode->psBandNrgScale[bin]==0) - ? (hPsEncode->iidGroupWidthLd[group] + 5) - : (fixMax(hPsEncode->iidGroupWidthLd[group],hPsEncode->psBandNrgScale[bin]) + 1) ; - - } -} - -FDK_PSENC_ERROR FDKsbrEnc_CreatePSEncode( - HANDLE_PS_ENCODE *phPsEncode - ) -{ - FDK_PSENC_ERROR error = PSENC_OK; - - if (phPsEncode==NULL) { - error = PSENC_INVALID_HANDLE; - } - else { - HANDLE_PS_ENCODE hPsEncode = NULL; - if (NULL==(hPsEncode = GetRam_PsEncode())) { - error = PSENC_MEMORY_ERROR; - goto bail; - } - FDKmemclear(hPsEncode,sizeof(PS_ENCODE)); - *phPsEncode = hPsEncode; /* return allocated handle */ - } -bail: - return error; -} - -FDK_PSENC_ERROR FDKsbrEnc_InitPSEncode( - HANDLE_PS_ENCODE hPsEncode, - const PS_BANDS psEncMode, - const FIXP_DBL iidQuantErrorThreshold - ) -{ - FDK_PSENC_ERROR error = PSENC_OK; - - if (NULL==hPsEncode) { - error = PSENC_INVALID_HANDLE; - } - else { - if (PSENC_OK != (InitPSData(&hPsEncode->psData))) { - goto bail; - } - - switch(psEncMode){ - case PS_BANDS_COARSE: - case PS_BANDS_MID: - hPsEncode->nQmfIidGroups = QMF_GROUPS_LO_RES; - hPsEncode->nSubQmfIidGroups = SUBQMF_GROUPS_LO_RES; - FDKmemcpy(hPsEncode->iidGroupBorders, iidGroupBordersLoRes, (hPsEncode->nQmfIidGroups + hPsEncode->nSubQmfIidGroups + 1)*sizeof(INT)); - FDKmemcpy(hPsEncode->subband2parameterIndex, subband2parameter20, (hPsEncode->nQmfIidGroups + hPsEncode->nSubQmfIidGroups) *sizeof(INT)); - FDKmemcpy(hPsEncode->iidGroupWidthLd, iidGroupWidthLdLoRes, (hPsEncode->nQmfIidGroups + hPsEncode->nSubQmfIidGroups) *sizeof(UCHAR)); - break; - default: - error = PSENC_INIT_ERROR; - goto bail; - } - - hPsEncode->psEncMode = psEncMode; - hPsEncode->iidQuantErrorThreshold = iidQuantErrorThreshold; - FDKsbrEnc_initPsBandNrgScale(hPsEncode); - } -bail: - return error; -} - - -FDK_PSENC_ERROR FDKsbrEnc_DestroyPSEncode( - HANDLE_PS_ENCODE *phPsEncode - ) -{ - FDK_PSENC_ERROR error = PSENC_OK; - - if (NULL !=phPsEncode) { - FreeRam_PsEncode(phPsEncode); - } - - return error; -} - -typedef struct { - FIXP_DBL pwrL[PS_MAX_ENVELOPES][PS_MAX_BANDS]; - FIXP_DBL pwrR[PS_MAX_ENVELOPES][PS_MAX_BANDS]; - FIXP_DBL ldPwrL[PS_MAX_ENVELOPES][PS_MAX_BANDS]; - FIXP_DBL ldPwrR[PS_MAX_ENVELOPES][PS_MAX_BANDS]; - FIXP_DBL pwrCr[PS_MAX_ENVELOPES][PS_MAX_BANDS]; - FIXP_DBL pwrCi[PS_MAX_ENVELOPES][PS_MAX_BANDS]; - -} PS_PWR_DATA; - - -FDK_PSENC_ERROR FDKsbrEnc_PSEncode( - HANDLE_PS_ENCODE hPsEncode, - HANDLE_PS_OUT hPsOut, - UCHAR *dynBandScale, - UINT maxEnvelopes, - FIXP_DBL *hybridData[HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2], - const INT frameSize, - const INT sendHeader - ) -{ - FDK_PSENC_ERROR error = PSENC_OK; - - HANDLE_PS_DATA hPsData = &hPsEncode->psData; - FIXP_DBL iid [PS_MAX_ENVELOPES][PS_MAX_BANDS]; - FIXP_DBL icc [PS_MAX_ENVELOPES][PS_MAX_BANDS]; - int envBorder[PS_MAX_ENVELOPES+1]; - - int group, bin, col, subband, band; - int i = 0; - - int env = 0; - int psBands = (int) hPsEncode->psEncMode; - int nIidGroups = hPsEncode->nQmfIidGroups + hPsEncode->nSubQmfIidGroups; - int nEnvelopes = fixMin(maxEnvelopes, (UINT)PS_MAX_ENVELOPES); - - C_ALLOC_SCRATCH_START(pwrData, PS_PWR_DATA, 1); - - for(env=0; envpwrL[env][band] = pwrData->pwrR[env][band] = pwrData->pwrCr[env][band] = pwrData->pwrCi[env][band] = FIXP_DBL(1); - } - - /**** calculate energies and correlation ****/ - - /* start with hybrid data */ - for (group=0; group < nIidGroups; group++) { - /* Translate group to bin */ - bin = hPsEncode->subband2parameterIndex[group]; - - /* Translate from 20 bins to 10 bins */ - if (hPsEncode->psEncMode == PS_BANDS_COARSE) { - bin >>= 1; - } - - /* determine group border */ - int bScale = hPsEncode->psBandNrgScale[bin]; - - FIXP_DBL pwrL_env_bin = pwrData->pwrL[env][bin]; - FIXP_DBL pwrR_env_bin = pwrData->pwrR[env][bin]; - FIXP_DBL pwrCr_env_bin = pwrData->pwrCr[env][bin]; - FIXP_DBL pwrCi_env_bin = pwrData->pwrCi[env][bin]; - - int scale = (int)dynBandScale[bin]; - for (col=envBorder[env]; coliidGroupBorders[group]; subband < hPsEncode->iidGroupBorders[group+1]; subband++) { - FIXP_QMF l_real = (hybridData[col][0][0][subband]) << scale; - FIXP_QMF l_imag = (hybridData[col][0][1][subband]) << scale; - FIXP_QMF r_real = (hybridData[col][1][0][subband]) << scale; - FIXP_QMF r_imag = (hybridData[col][1][1][subband]) << scale; - - pwrL_env_bin += (fPow2Div2(l_real) + fPow2Div2(l_imag)) >> bScale; - pwrR_env_bin += (fPow2Div2(r_real) + fPow2Div2(r_imag)) >> bScale; - pwrCr_env_bin += (fMultDiv2(l_real, r_real) + fMultDiv2(l_imag, r_imag)) >> bScale; - pwrCi_env_bin += (fMultDiv2(r_real, l_imag) - fMultDiv2(l_real, r_imag)) >> bScale; - } - } - /* assure, nrg's of left and right channel are not negative; necessary on 16 bit multiply units */ - pwrData->pwrL[env][bin] = fixMax((FIXP_DBL)0,pwrL_env_bin); - pwrData->pwrR[env][bin] = fixMax((FIXP_DBL)0,pwrR_env_bin); - - pwrData->pwrCr[env][bin] = pwrCr_env_bin; - pwrData->pwrCi[env][bin] = pwrCi_env_bin; - - } /* nIidGroups */ - - /* calc logarithmic energy */ - LdDataVector(pwrData->pwrL[env], pwrData->ldPwrL[env], psBands); - LdDataVector(pwrData->pwrR[env], pwrData->ldPwrR[env], psBands); - - } /* nEnvelopes */ - - /* calculate iid and icc */ - calculateIID(pwrData->ldPwrL, pwrData->ldPwrR, iid, nEnvelopes, psBands); - calculateICC(pwrData->ldPwrL, pwrData->ldPwrR, pwrData->pwrCr, pwrData->pwrCi, icc, nEnvelopes, psBands); - - /*** Envelope Reduction ***/ - while (envelopeReducible(iid,icc,psBands,nEnvelopes)) { - int e=0; - /* sum energies of two neighboring envelopes */ - nEnvelopes >>= 1; - for (e=0; epwrL[2*e], pwrData->pwrL[2*e+1], pwrData->pwrL[e], psBands); - FDKsbrEnc_addFIXP_DBL(pwrData->pwrR[2*e], pwrData->pwrR[2*e+1], pwrData->pwrR[e], psBands); - FDKsbrEnc_addFIXP_DBL(pwrData->pwrCr[2*e],pwrData->pwrCr[2*e+1],pwrData->pwrCr[e],psBands); - FDKsbrEnc_addFIXP_DBL(pwrData->pwrCi[2*e],pwrData->pwrCi[2*e+1],pwrData->pwrCi[e],psBands); - - /* calc logarithmic energy */ - LdDataVector(pwrData->pwrL[e], pwrData->ldPwrL[e], psBands); - LdDataVector(pwrData->pwrR[e], pwrData->ldPwrR[e], psBands); - - /* reduce number of envelopes and adjust borders */ - envBorder[e] = envBorder[2*e]; - } - envBorder[nEnvelopes] = envBorder[2*nEnvelopes]; - - /* re-calculate iid and icc */ - calculateIID(pwrData->ldPwrL, pwrData->ldPwrR, iid, nEnvelopes, psBands); - calculateICC(pwrData->ldPwrL, pwrData->ldPwrR, pwrData->pwrCr, pwrData->pwrCi, icc, nEnvelopes, psBands); - } - - - /* */ - if(sendHeader) { - hPsData->headerCnt = MAX_PS_NOHEADER_CNT; - hPsData->iidTimeCnt = MAX_TIME_DIFF_FRAMES; - hPsData->iccTimeCnt = MAX_TIME_DIFF_FRAMES; - hPsData->noEnvCnt = MAX_NOENV_CNT; - } - - /*** Parameter processing, quantisation etc ***/ - processIidData(hPsData, iid, psBands, nEnvelopes, hPsEncode->iidQuantErrorThreshold); - processIccData(hPsData, icc, psBands, nEnvelopes); - - - /*** Initialize output struct ***/ - - /* PS Header on/off ? */ - if( (hPsData->headerCntiidQuantMode == hPsData->iidQuantModeLast) && (hPsData->iccQuantMode == hPsData->iccQuantModeLast) ) - && ( (hPsData->iidEnable == hPsData->iidEnableLast) && (hPsData->iccEnable == hPsData->iccEnableLast) ) ) { - hPsOut->enablePSHeader = 0; - } - else { - hPsOut->enablePSHeader = 1; - hPsData->headerCnt = 0; - } - - /* nEnvelopes = 0 ? */ - if ( (hPsData->noEnvCnt < MAX_NOENV_CNT) - && (similarIid(hPsData, psBands, nEnvelopes)) - && (similarIcc(hPsData, psBands, nEnvelopes)) ) { - hPsOut->nEnvelopes = nEnvelopes = 0; - hPsData->noEnvCnt++; - } else { - hPsData->noEnvCnt = 0; - } - - - if (nEnvelopes>0) { - - hPsOut->enableIID = hPsData->iidEnable; - hPsOut->iidMode = getIIDMode(psBands, hPsData->iidQuantMode); - - hPsOut->enableICC = hPsData->iccEnable; - hPsOut->iccMode = getICCMode(psBands, hPsData->iccQuantMode); - - hPsOut->enableIpdOpd = 0; - hPsOut->frameClass = 0; - hPsOut->nEnvelopes = nEnvelopes; - - for(env=0; envframeBorder[env] = envBorder[env+1]; - } - - for(env=0; envnEnvelopes; env++) { - hPsOut->deltaIID[env] = (PS_DELTA)hPsData->iidDiffMode[env]; - - for(band=0; bandiid[env][band] = hPsData->iidIdx[env][band]; - } - } - - for(env=0; envnEnvelopes; env++) { - hPsOut->deltaICC[env] = (PS_DELTA)hPsData->iccDiffMode[env]; - for(band=0; bandicc[env][band] = hPsData->iccIdx[env][band]; - } - } - - /* IPD OPD not supported right now */ - FDKmemclear(hPsOut->ipd, PS_MAX_ENVELOPES*PS_MAX_BANDS*sizeof(PS_DELTA)); - for(env=0; envdeltaIPD[env] = PS_DELTA_FREQ; - hPsOut->deltaOPD[env] = PS_DELTA_FREQ; - } - - FDKmemclear(hPsOut->ipdLast, PS_MAX_BANDS*sizeof(INT)); - FDKmemclear(hPsOut->opdLast, PS_MAX_BANDS*sizeof(INT)); - - for(band=0; bandiidLast[band] = hPsData->iidIdxLast[band]; - hPsOut->iccLast[band] = hPsData->iccIdxLast[band]; - } - - /* save iids and iccs for differential time coding in the next frame */ - hPsData->nEnvelopesLast = nEnvelopes; - hPsData->iidEnableLast = hPsData->iidEnable; - hPsData->iccEnableLast = hPsData->iccEnable; - hPsData->iidQuantModeLast = hPsData->iidQuantMode; - hPsData->iccQuantModeLast = hPsData->iccQuantMode; - for (i=0; iiidIdxLast[i] = hPsData->iidIdx[nEnvelopes-1][i]; - hPsData->iccIdxLast[i] = hPsData->iccIdx[nEnvelopes-1][i]; - } - } /* Envelope > 0 */ - - C_ALLOC_SCRATCH_END(pwrData, PS_PWR_DATA, 1) - - return error; -} - diff --git a/libSBRenc/src/ps_encode.h b/libSBRenc/src/ps_encode.h deleted file mode 100644 index f728d47..0000000 --- a/libSBRenc/src/ps_encode.h +++ /dev/null @@ -1,187 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/***************************** MPEG Audio Encoder *************************** - - Initial author: M. Neuendorf, N. Rettelbach, M. Multrus - contents/description: PS Parameter extraction, encoding - -******************************************************************************/ -/*! - \file - \brief PS parameter extraction, encoding functions -*/ - -#ifndef __INCLUDED_PS_ENCODE_H -#define __INCLUDED_PS_ENCODE_H - -#include "ps_const.h" -#include "ps_bitenc.h" - - -#define IID_SCALE_FT (64.f) /* maxVal in Quant tab is +/- 50 */ -#define IID_SCALE 6 /* maxVal in Quant tab is +/- 50 */ -#define IID_MAXVAL (1< QuantScale 64 */ -#define PS_QUANT_SCALE 6 /* error smaller (64-25)/64 * 20 bands * 4 env -> QuantScale 6 bit */ - - -#define QMF_GROUPS_LO_RES 12 -#define SUBQMF_GROUPS_LO_RES 10 -#define QMF_GROUPS_HI_RES 18 -#define SUBQMF_GROUPS_HI_RES 30 - - -typedef struct T_PS_DATA { - - INT iidEnable; - INT iidEnableLast; - INT iidQuantMode; - INT iidQuantModeLast; - INT iidDiffMode[PS_MAX_ENVELOPES]; - INT iidIdx [PS_MAX_ENVELOPES][PS_MAX_BANDS]; - INT iidIdxLast [PS_MAX_BANDS]; - - INT iccEnable; - INT iccEnableLast; - INT iccQuantMode; - INT iccQuantModeLast; - INT iccDiffMode[PS_MAX_ENVELOPES]; - INT iccIdx [PS_MAX_ENVELOPES][PS_MAX_BANDS]; - INT iccIdxLast [PS_MAX_BANDS]; - - INT nEnvelopesLast; - - INT headerCnt; - INT iidTimeCnt; - INT iccTimeCnt; - INT noEnvCnt; - -} PS_DATA, *HANDLE_PS_DATA; - - -typedef struct T_PS_ENCODE{ - - PS_DATA psData; - - PS_BANDS psEncMode; - INT nQmfIidGroups; - INT nSubQmfIidGroups; - INT iidGroupBorders[QMF_GROUPS_HI_RES + SUBQMF_GROUPS_HI_RES + 1]; - INT subband2parameterIndex[QMF_GROUPS_HI_RES + SUBQMF_GROUPS_HI_RES]; - UCHAR iidGroupWidthLd[QMF_GROUPS_HI_RES + SUBQMF_GROUPS_HI_RES]; - FIXP_DBL iidQuantErrorThreshold; - - UCHAR psBandNrgScale [PS_MAX_BANDS]; - -} PS_ENCODE; - - -typedef struct T_PS_ENCODE *HANDLE_PS_ENCODE; - -FDK_PSENC_ERROR FDKsbrEnc_CreatePSEncode( - HANDLE_PS_ENCODE *phPsEncode - ); - -FDK_PSENC_ERROR FDKsbrEnc_InitPSEncode( - HANDLE_PS_ENCODE hPsEncode, - const PS_BANDS psEncMode, - const FIXP_DBL iidQuantErrorThreshold - ); - -FDK_PSENC_ERROR FDKsbrEnc_DestroyPSEncode( - HANDLE_PS_ENCODE *phPsEncode - ); - -FDK_PSENC_ERROR FDKsbrEnc_PSEncode( - HANDLE_PS_ENCODE hPsEncode, - HANDLE_PS_OUT hPsOut, - UCHAR *dynBandScale, - UINT maxEnvelopes, - FIXP_DBL *hybridData[HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2], - const INT frameSize, - const INT sendHeader - ); - -#endif diff --git a/libSBRenc/src/ps_main.cpp b/libSBRenc/src/ps_main.cpp deleted file mode 100644 index ab183e2..0000000 --- a/libSBRenc/src/ps_main.cpp +++ /dev/null @@ -1,618 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/***************************** MPEG Audio Encoder *************************** - - Initial Authors: M. Multrus - Contents/Description: PS Wrapper, Downmix - -******************************************************************************/ - -#include "ps_main.h" - - -/* Includes ******************************************************************/ - -#include "ps_const.h" -#include "ps_bitenc.h" - -#include "sbr_ram.h" - -/*--------------- function declarations --------------------*/ -static void psFindBestScaling( - HANDLE_PARAMETRIC_STEREO hParametricStereo, - FIXP_DBL *hybridData[HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2], - UCHAR *dynBandScale, - FIXP_QMF *maxBandValue, - SCHAR *dmxScale - ); - -/*------------- function definitions ----------------*/ -FDK_PSENC_ERROR PSEnc_Create( - HANDLE_PARAMETRIC_STEREO *phParametricStereo - ) -{ - FDK_PSENC_ERROR error = PSENC_OK; - - if (phParametricStereo==NULL) { - error = PSENC_INVALID_HANDLE; - } - else { - int i; - HANDLE_PARAMETRIC_STEREO hParametricStereo = NULL; - - if (NULL==(hParametricStereo = GetRam_ParamStereo())) { - error = PSENC_MEMORY_ERROR; - goto bail; - } - FDKmemclear(hParametricStereo, sizeof(PARAMETRIC_STEREO)); - - if (PSENC_OK != (error = FDKsbrEnc_CreatePSEncode(&hParametricStereo->hPsEncode))) { - goto bail; - } - - for (i=0; ifdkHybAnaFilter[i], - hParametricStereo->__staticHybAnaStatesLF[i], - sizeof(hParametricStereo->__staticHybAnaStatesLF[i]), - hParametricStereo->__staticHybAnaStatesHF[i], - sizeof(hParametricStereo->__staticHybAnaStatesHF[i]) - ) !=0 ) - { - error = PSENC_MEMORY_ERROR; - goto bail; - } - } - - *phParametricStereo = hParametricStereo; /* return allocated handle */ - } -bail: - return error; -} - -FDK_PSENC_ERROR PSEnc_Init( - HANDLE_PARAMETRIC_STEREO hParametricStereo, - const HANDLE_PSENC_CONFIG hPsEncConfig, - INT noQmfSlots, - INT noQmfBands - ,UCHAR *dynamic_RAM - ) -{ - FDK_PSENC_ERROR error = PSENC_OK; - - if ( (NULL==hParametricStereo) || (NULL==hPsEncConfig) ) { - error = PSENC_INVALID_HANDLE; - } - else { - int ch, i; - - hParametricStereo->initPS = 1; - hParametricStereo->noQmfSlots = noQmfSlots; - hParametricStereo->noQmfBands = noQmfBands; - - /* clear delay lines */ - FDKmemclear(hParametricStereo->qmfDelayLines, sizeof(hParametricStereo->qmfDelayLines)); - - hParametricStereo->qmfDelayScale = FRACT_BITS-1; - - /* create configuration for hybrid filter bank */ - for (ch=0; chfdkHybAnaFilter[ch], - THREE_TO_TEN, - QMF_CHANNELS, - QMF_CHANNELS, - 1 - ); - } /* ch */ - - FDKhybridSynthesisInit( - &hParametricStereo->fdkHybSynFilter, - THREE_TO_TEN, - QMF_CHANNELS, - QMF_CHANNELS - ); - - /* determine average delay */ - hParametricStereo->psDelay = (HYBRID_FILTER_DELAY*hParametricStereo->noQmfBands); - - if ( (hPsEncConfig->maxEnvelopes < PSENC_NENV_1) || (hPsEncConfig->maxEnvelopes > PSENC_NENV_MAX) ) { - hPsEncConfig->maxEnvelopes = PSENC_NENV_DEFAULT; - } - hParametricStereo->maxEnvelopes = hPsEncConfig->maxEnvelopes; - - if (PSENC_OK != (error = FDKsbrEnc_InitPSEncode(hParametricStereo->hPsEncode, (PS_BANDS) hPsEncConfig->nStereoBands, hPsEncConfig->iidQuantErrorThreshold))){ - goto bail; - } - - for (ch = 0; chpHybridData[i+HYBRID_READ_OFFSET][ch][0] = &pDynReal[i*MAX_HYBRID_BANDS]; - hParametricStereo->pHybridData[i+HYBRID_READ_OFFSET][ch][1] = &pDynImag[i*MAX_HYBRID_BANDS];; - } - - for (i=0; ipHybridData[i][ch][0] = hParametricStereo->__staticHybridData[i][ch][0]; - hParametricStereo->pHybridData[i][ch][1] = hParametricStereo->__staticHybridData[i][ch][1]; - } - } /* ch */ - - /* clear static hybrid buffer */ - FDKmemclear(hParametricStereo->__staticHybridData, sizeof(hParametricStereo->__staticHybridData)); - - /* clear bs buffer */ - FDKmemclear(hParametricStereo->psOut, sizeof(hParametricStereo->psOut)); - - hParametricStereo->psOut[0].enablePSHeader = 1; /* write ps header in first frame */ - - /* clear scaling buffer */ - FDKmemclear(hParametricStereo->dynBandScale, sizeof(UCHAR)*PS_MAX_BANDS); - FDKmemclear(hParametricStereo->maxBandValue, sizeof(FIXP_QMF)*PS_MAX_BANDS); - - } /* valid handle */ -bail: - return error; -} - - -FDK_PSENC_ERROR PSEnc_Destroy( - HANDLE_PARAMETRIC_STEREO *phParametricStereo - ) -{ - FDK_PSENC_ERROR error = PSENC_OK; - - if (NULL!=phParametricStereo) { - HANDLE_PARAMETRIC_STEREO hParametricStereo = *phParametricStereo; - if(hParametricStereo != NULL){ - FDKsbrEnc_DestroyPSEncode(&hParametricStereo->hPsEncode); - FreeRam_ParamStereo(phParametricStereo); - } - } - - return error; -} - -static FDK_PSENC_ERROR ExtractPSParameters( - HANDLE_PARAMETRIC_STEREO hParametricStereo, - const int sendHeader, - FIXP_DBL *hybridData[HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2] - ) -{ - FDK_PSENC_ERROR error = PSENC_OK; - - if (hParametricStereo == NULL) { - error = PSENC_INVALID_HANDLE; - } - else { - /* call ps encode function */ - if (hParametricStereo->initPS){ - hParametricStereo->psOut[1] = hParametricStereo->psOut[0]; - } - hParametricStereo->psOut[0] = hParametricStereo->psOut[1]; - - if (PSENC_OK != (error = FDKsbrEnc_PSEncode( - hParametricStereo->hPsEncode, - &hParametricStereo->psOut[1], - hParametricStereo->dynBandScale, - hParametricStereo->maxEnvelopes, - hybridData, - hParametricStereo->noQmfSlots, - sendHeader))) - { - goto bail; - } - - if (hParametricStereo->initPS) { - hParametricStereo->psOut[0] = hParametricStereo->psOut[1]; - hParametricStereo->initPS = 0; - } - } -bail: - return error; -} - - -static FDK_PSENC_ERROR DownmixPSQmfData( - HANDLE_PARAMETRIC_STEREO hParametricStereo, - HANDLE_QMF_FILTER_BANK sbrSynthQmf, - FIXP_QMF **RESTRICT mixRealQmfData, - FIXP_QMF **RESTRICT mixImagQmfData, - INT_PCM *downsampledOutSignal, - FIXP_DBL *hybridData[HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2], - const INT noQmfSlots, - const INT psQmfScale[MAX_PS_CHANNELS], - SCHAR *qmfScale - ) -{ - FDK_PSENC_ERROR error = PSENC_OK; - - if(hParametricStereo == NULL){ - error = PSENC_INVALID_HANDLE; - } - else { - int n, k; - C_AALLOC_SCRATCH_START(pWorkBuffer, FIXP_QMF, 2*QMF_CHANNELS) - - /* define scalings */ - int dynQmfScale = fixMax(0, hParametricStereo->dmxScale-1); /* scale one bit more for addition of left and right */ - int downmixScale = psQmfScale[0] - dynQmfScale; - const FIXP_DBL maxStereoScaleFactor = MAXVAL_DBL; /* 2.f/2.f */ - - for (n = 0; n>1) < fMult(maxStereoScaleFactor,tmpScaleFactor) ) { - - int sc_num = CountLeadingBits(stereoScaleFactor) ; - int sc_denum = CountLeadingBits(tmpScaleFactor) ; - sc = -(sc_num-sc_denum); - - tmpScaleFactor = schur_div((stereoScaleFactor<<(sc_num))>>1, - tmpScaleFactor<>=1; - } - stereoScaleFactor = sqrtFixp(tmpScaleFactor); - stereoScaleFactor <<= (sc>>1); - } - else { - stereoScaleFactor = maxStereoScaleFactor; - } - - /* write data to hybrid output */ - tmpHybrid[0][k] = fMultDiv2(stereoScaleFactor, (FIXP_QMF)(tmpLeftReal + tmpRightReal))>>dynScale; - tmpHybrid[1][k] = fMultDiv2(stereoScaleFactor, (FIXP_QMF)(tmpLeftImag + tmpRightImag))>>dynScale; - - } /* hybrid bands - k */ - - FDKhybridSynthesisApply( - &hParametricStereo->fdkHybSynFilter, - tmpHybrid[0], - tmpHybrid[1], - mixRealQmfData[n], - mixImagQmfData[n]); - - qmfSynthesisFilteringSlot( - sbrSynthQmf, - mixRealQmfData[n], - mixImagQmfData[n], - downmixScale-7, - downmixScale-7, - downsampledOutSignal+(n*sbrSynthQmf->no_channels), - 1, - pWorkBuffer); - - } /* slots */ - - *qmfScale = -downmixScale + 7; - - C_AALLOC_SCRATCH_END(pWorkBuffer, FIXP_QMF, 2*QMF_CHANNELS) - - { - const INT noQmfSlots2 = hParametricStereo->noQmfSlots>>1; - const int noQmfBands = hParametricStereo->noQmfBands; - - INT scale, i, j, slotOffset; - - FIXP_QMF tmp[2][QMF_CHANNELS]; - - for (i=0; iqmfDelayLines[0][i], noQmfBands*sizeof(FIXP_QMF)); - FDKmemcpy(tmp[1], hParametricStereo->qmfDelayLines[1][i], noQmfBands*sizeof(FIXP_QMF)); - - FDKmemcpy(hParametricStereo->qmfDelayLines[0][i], mixRealQmfData[i+noQmfSlots2], noQmfBands*sizeof(FIXP_QMF)); - FDKmemcpy(hParametricStereo->qmfDelayLines[1][i], mixImagQmfData[i+noQmfSlots2], noQmfBands*sizeof(FIXP_QMF)); - - FDKmemcpy(mixRealQmfData[i+noQmfSlots2], mixRealQmfData[i], noQmfBands*sizeof(FIXP_QMF)); - FDKmemcpy(mixImagQmfData[i+noQmfSlots2], mixImagQmfData[i], noQmfBands*sizeof(FIXP_QMF)); - - FDKmemcpy(mixRealQmfData[i], tmp[0], noQmfBands*sizeof(FIXP_QMF)); - FDKmemcpy(mixImagQmfData[i], tmp[1], noQmfBands*sizeof(FIXP_QMF)); - } - - if (hParametricStereo->qmfDelayScale > *qmfScale) { - scale = hParametricStereo->qmfDelayScale - *qmfScale; - slotOffset = 0; - } - else { - scale = *qmfScale - hParametricStereo->qmfDelayScale; - slotOffset = noQmfSlots2; - } - - for (i=0; i>= scale; - mixImagQmfData[i+slotOffset][j] >>= scale; - } - } - - scale = *qmfScale; - *qmfScale = FDKmin(*qmfScale, hParametricStereo->qmfDelayScale); - hParametricStereo->qmfDelayScale = scale; - } - - } /* valid handle */ - - return error; -} - - -INT FDKsbrEnc_PSEnc_WritePSData( - HANDLE_PARAMETRIC_STEREO hParametricStereo, - HANDLE_FDK_BITSTREAM hBitstream - ) -{ - return ( (hParametricStereo!=NULL) ? FDKsbrEnc_WritePSBitstream(&hParametricStereo->psOut[0], hBitstream) : 0 ); -} - - -FDK_PSENC_ERROR FDKsbrEnc_PSEnc_ParametricStereoProcessing( - HANDLE_PARAMETRIC_STEREO hParametricStereo, - INT_PCM *samples[2], - UINT timeInStride, - QMF_FILTER_BANK **hQmfAnalysis, - FIXP_QMF **RESTRICT downmixedRealQmfData, - FIXP_QMF **RESTRICT downmixedImagQmfData, - INT_PCM *downsampledOutSignal, - HANDLE_QMF_FILTER_BANK sbrSynthQmf, - SCHAR *qmfScale, - const int sendHeader - ) -{ - FDK_PSENC_ERROR error = PSENC_OK; - INT psQmfScale[MAX_PS_CHANNELS] = {0}; - int psCh, i; - C_AALLOC_SCRATCH_START(pWorkBuffer, FIXP_QMF, 4*QMF_CHANNELS) - - for (psCh = 0; psChno_col; i++) { - - qmfAnalysisFilteringSlot( - hQmfAnalysis[psCh], - &pWorkBuffer[2*QMF_CHANNELS], /* qmfReal[QMF_CHANNELS] */ - &pWorkBuffer[3*QMF_CHANNELS], /* qmfImag[QMF_CHANNELS] */ - samples[psCh]+i*(hQmfAnalysis[psCh]->no_channels*timeInStride), - timeInStride, - &pWorkBuffer[0*QMF_CHANNELS] /* qmf workbuffer 2*QMF_CHANNELS */ - ); - - FDKhybridAnalysisApply( - &hParametricStereo->fdkHybAnaFilter[psCh], - &pWorkBuffer[2*QMF_CHANNELS], /* qmfReal[QMF_CHANNELS] */ - &pWorkBuffer[3*QMF_CHANNELS], /* qmfImag[QMF_CHANNELS] */ - hParametricStereo->pHybridData[i+HYBRID_READ_OFFSET][psCh][0], - hParametricStereo->pHybridData[i+HYBRID_READ_OFFSET][psCh][1] - ); - - } /* no_col loop i */ - - psQmfScale[psCh] = hQmfAnalysis[psCh]->outScalefactor; - - } /* for psCh */ - - C_AALLOC_SCRATCH_END(pWorkBuffer, FIXP_QMF, 4*QMF_CHANNELS) - - /* find best scaling in new QMF and Hybrid data */ - psFindBestScaling( hParametricStereo, - &hParametricStereo->pHybridData[HYBRID_READ_OFFSET], - hParametricStereo->dynBandScale, - hParametricStereo->maxBandValue, - &hParametricStereo->dmxScale ) ; - - - /* extract the ps parameters */ - if(PSENC_OK != (error = ExtractPSParameters(hParametricStereo, sendHeader, &hParametricStereo->pHybridData[0]))){ - goto bail; - } - - /* save hybrid date for next frame */ - for (i=0; ipHybridData[i][0][0], hParametricStereo->pHybridData[HYBRID_FRAMESIZE+i][0][0], MAX_HYBRID_BANDS*sizeof(FIXP_DBL)); /* left, real */ - FDKmemcpy(hParametricStereo->pHybridData[i][0][1], hParametricStereo->pHybridData[HYBRID_FRAMESIZE+i][0][1], MAX_HYBRID_BANDS*sizeof(FIXP_DBL)); /* left, imag */ - FDKmemcpy(hParametricStereo->pHybridData[i][1][0], hParametricStereo->pHybridData[HYBRID_FRAMESIZE+i][1][0], MAX_HYBRID_BANDS*sizeof(FIXP_DBL)); /* right, real */ - FDKmemcpy(hParametricStereo->pHybridData[i][1][1], hParametricStereo->pHybridData[HYBRID_FRAMESIZE+i][1][1], MAX_HYBRID_BANDS*sizeof(FIXP_DBL)); /* right, imag */ - } - - /* downmix and hybrid synthesis */ - if (PSENC_OK != (error = DownmixPSQmfData(hParametricStereo, sbrSynthQmf, downmixedRealQmfData, downmixedImagQmfData, downsampledOutSignal, &hParametricStereo->pHybridData[HYBRID_READ_OFFSET], hParametricStereo->noQmfSlots, psQmfScale, qmfScale))) { - goto bail; - } - -bail: - - return error; -} - -static void psFindBestScaling( - HANDLE_PARAMETRIC_STEREO hParametricStereo, - FIXP_DBL *hybridData[HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2], - UCHAR *dynBandScale, - FIXP_QMF *maxBandValue, - SCHAR *dmxScale - ) -{ - HANDLE_PS_ENCODE hPsEncode = hParametricStereo->hPsEncode; - - INT group, bin, col, band; - const INT frameSize = hParametricStereo->noQmfSlots; - const INT psBands = (INT) hPsEncode->psEncMode; - const INT nIidGroups = hPsEncode->nQmfIidGroups + hPsEncode->nSubQmfIidGroups; - - /* group wise scaling */ - FIXP_QMF maxVal [2][PS_MAX_BANDS]; - FIXP_QMF maxValue = FL2FXCONST_DBL(0.f); - - FDKmemclear(maxVal, sizeof(maxVal)); - - /* start with hybrid data */ - for (group=0; group < nIidGroups; group++) { - /* Translate group to bin */ - bin = hPsEncode->subband2parameterIndex[group]; - - /* Translate from 20 bins to 10 bins */ - if (hPsEncode->psEncMode == PS_BANDS_COARSE) { - bin >>= 1; - } - - /* QMF downmix scaling */ - { - FIXP_QMF tmp = maxVal[0][bin]; - int i; - for (col=0; coliidGroupBorders[group]; i < hPsEncode->iidGroupBorders[group+1]; i++) { - tmp = fixMax(tmp, (FIXP_QMF)fixp_abs(hybridData[col][0][0][i])); - tmp = fixMax(tmp, (FIXP_QMF)fixp_abs(hybridData[col][0][1][i])); - tmp = fixMax(tmp, (FIXP_QMF)fixp_abs(hybridData[col][1][0][i])); - tmp = fixMax(tmp, (FIXP_QMF)fixp_abs(hybridData[col][1][1][i])); - } - } - maxVal[0][bin] = tmp; - - tmp = maxVal[1][bin]; - for (col=frameSize-HYBRID_READ_OFFSET; coliidGroupBorders[group]; i < hPsEncode->iidGroupBorders[group+1]; i++) { - tmp = fixMax(tmp, (FIXP_QMF)fixp_abs(hybridData[col][0][0][i])); - tmp = fixMax(tmp, (FIXP_QMF)fixp_abs(hybridData[col][0][1][i])); - tmp = fixMax(tmp, (FIXP_QMF)fixp_abs(hybridData[col][1][0][i])); - tmp = fixMax(tmp, (FIXP_QMF)fixp_abs(hybridData[col][1][1][i])); - } - } - maxVal[1][bin] = tmp; - } - } /* nIidGroups */ - - /* convert maxSpec to maxScaling, find scaling space */ - for (band=0; band>1][QMF_CHANNELS]; - int qmfDelayScale; - - INT psDelay; - UINT maxEnvelopes; - UCHAR dynBandScale[PS_MAX_BANDS]; - FIXP_DBL maxBandValue[PS_MAX_BANDS]; - SCHAR dmxScale; - INT initPS; - INT noQmfSlots; - INT noQmfBands; - - FIXP_DBL __staticHybAnaStatesLF[MAX_PS_CHANNELS][2*HYBRID_FILTER_LENGTH*HYBRID_MAX_QMF_BANDS]; - FIXP_DBL __staticHybAnaStatesHF[MAX_PS_CHANNELS][2*HYBRID_FILTER_DELAY*(QMF_CHANNELS-HYBRID_MAX_QMF_BANDS)]; - FDK_ANA_HYB_FILTER fdkHybAnaFilter[MAX_PS_CHANNELS]; - FDK_SYN_HYB_FILTER fdkHybSynFilter; - -} PARAMETRIC_STEREO; - - -typedef struct T_PSENC_CONFIG { - INT frameSize; - INT qmfFilterMode; - INT sbrPsDelay; - PSENC_STEREO_BANDS_CONFIG nStereoBands; - PSENC_NENV_CONFIG maxEnvelopes; - FIXP_DBL iidQuantErrorThreshold; - -} PSENC_CONFIG, *HANDLE_PSENC_CONFIG; - -typedef struct T_PARAMETRIC_STEREO *HANDLE_PARAMETRIC_STEREO; - - -/** - * \brief Create a parametric stereo encoder instance. - * - * \param phParametricStereo A pointer to a parametric stereo handle to be allocated. Initialized on return. - * - * \return - * - PSENC_OK, on succes. - * - PSENC_INVALID_HANDLE, PSENC_MEMORY_ERROR, on failure. - */ -FDK_PSENC_ERROR PSEnc_Create( - HANDLE_PARAMETRIC_STEREO *phParametricStereo - ); - - -/** - * \brief Initialize a parametric stereo encoder instance. - * - * \param hParametricStereo Meta Data handle. - * \param hPsEncConfig Filled parametric stereo configuration structure. - * \param noQmfSlots Number of slots within one audio frame. - * \param noQmfBands Number of QMF bands. - * \param dynamic_RAM Pointer to preallocated workbuffer. - * - * \return - * - PSENC_OK, on succes. - * - PSENC_INVALID_HANDLE, PSENC_INIT_ERROR, on failure. - */ -FDK_PSENC_ERROR PSEnc_Init( - HANDLE_PARAMETRIC_STEREO hParametricStereo, - const HANDLE_PSENC_CONFIG hPsEncConfig, - INT noQmfSlots, - INT noQmfBands - ,UCHAR *dynamic_RAM - ); - - -/** - * \brief Destroy parametric stereo encoder instance. - * - * Deallocate instance and free whole memory. - * - * \param phParametricStereo Pointer to the parametric stereo handle to be deallocated. - * - * \return - * - PSENC_OK, on succes. - * - PSENC_INVALID_HANDLE, on failure. - */ -FDK_PSENC_ERROR PSEnc_Destroy( - HANDLE_PARAMETRIC_STEREO *phParametricStereo - ); - - -/** - * \brief Apply parametric stereo processing. - * - * \param hParametricStereo Meta Data handle. - * \param samples Pointer to 2 channel audio input signal. - * \param timeInStride, Stride factor of input buffer. - * \param hQmfAnalysis, Pointer to QMF analysis filterbanks. - * \param downmixedRealQmfData Pointer to real QMF buffer to be written to. - * \param downmixedImagQmfData Pointer to imag QMF buffer to be written to. - * \param downsampledOutSignal Pointer to buffer where to write downmixed timesignal. - * \param sbrSynthQmf Pointer to QMF synthesis filterbank. - * \param qmfScale Return scaling factor of the qmf data. - * \param sendHeader Signal whether to write header data. - * - * \return - * - PSENC_OK, on succes. - * - PSENC_INVALID_HANDLE, PSENC_ENCODE_ERROR, on failure. - */ -FDK_PSENC_ERROR FDKsbrEnc_PSEnc_ParametricStereoProcessing( - HANDLE_PARAMETRIC_STEREO hParametricStereo, - INT_PCM *samples[2], - UINT timeInStride, - QMF_FILTER_BANK **hQmfAnalysis, - FIXP_QMF **RESTRICT downmixedRealQmfData, - FIXP_QMF **RESTRICT downmixedImagQmfData, - INT_PCM *downsampledOutSignal, - HANDLE_QMF_FILTER_BANK sbrSynthQmf, - SCHAR *qmfScale, - const int sendHeader - ); - - -/** - * \brief Write parametric stereo bitstream. - * - * Write ps_data() element to bitstream and return number of written bits. - * Returns number of written bits only, if hBitstream == NULL. - * - * \param hParametricStereo Meta Data handle. - * \param hBitstream Bitstream buffer handle. - * - * \return - * - number of written bits. - */ -INT FDKsbrEnc_PSEnc_WritePSData( - HANDLE_PARAMETRIC_STEREO hParametricStereo, - HANDLE_FDK_BITSTREAM hBitstream - ); - -#endif /* __INCLUDED_PS_MAIN_H */ diff --git a/libSBRenc/src/resampler.cpp b/libSBRenc/src/resampler.cpp deleted file mode 100644 index 4adb243..0000000 --- a/libSBRenc/src/resampler.cpp +++ /dev/null @@ -1,507 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief FDK resampler tool box: - \author M. Werner -*/ - -#include "resampler.h" - -#include "genericStds.h" - - -/**************************************************************************/ -/* BIQUAD Filter Specifications */ -/**************************************************************************/ - -#define B1 0 -#define B2 1 -#define A1 2 -#define A2 3 - -#define BQC(x) FL2FXCONST_SGL(x/2) - - -struct FILTER_PARAM { - const FIXP_SGL *coeffa; /*! SOS matrix One row/section. Scaled using BQC(). Order of coefficients: B1,B2,A1,A2. B0=A0=1.0 */ - FIXP_DBL g; /*! overall gain */ - int Wc; /*! normalized passband bandwidth at input samplerate * 1000 */ - int noCoeffs; /*! number of filter coeffs */ - int delay; /*! delay in samples at input samplerate */ -}; - -#define BIQUAD_COEFSTEP 4 - -/** - *\brief Low Pass - Wc = 0,5, order 30, Stop Band -96dB. Wc criteria is "almost 0dB passband", not the usual -3db gain point. - [b,a]=cheby2(30,96,0.505) - [sos,g]=tf2sos(b,a) - bandwidth 0.48 - */ -static const FIXP_SGL sos48[] = { - BQC(1.98941075681938), BQC(0.999999996890811), BQC(0.863264527201963), BQC( 0.189553799960663), - BQC(1.90733804822445), BQC(1.00000001736189), BQC(0.836321575841691), BQC( 0.203505809266564), - BQC(1.75616665495325), BQC(0.999999946079721), BQC(0.784699225121588), BQC( 0.230471265506986), - BQC(1.55727745512726), BQC(1.00000011737815), BQC(0.712515423588351), BQC( 0.268752723900498), - BQC(1.33407591943643), BQC(0.999999795953228), BQC(0.625059117330989), BQC( 0.316194685288965), - BQC(1.10689898412458), BQC(1.00000035057114), BQC(0.52803514366398), BQC( 0.370517843224669), - BQC(0.89060371078454), BQC(0.999999343962822), BQC(0.426920462165257), BQC( 0.429608200207746), - BQC(0.694438261209433), BQC( 1.0000008629792), BQC(0.326530699561716), BQC( 0.491714450654174), - BQC(0.523237800935322), BQC(1.00000101349782), BQC(0.230829556274851), BQC( 0.555559034843281), - BQC(0.378631165929563), BQC(0.99998986482665), BQC(0.142906422036095), BQC( 0.620338874442411), - BQC(0.260786911308437), BQC(1.00003261460178), BQC(0.0651008576256505), BQC( 0.685759923926262), - BQC(0.168409429188098), BQC(0.999933049695828), BQC(-0.000790067789975562), BQC( 0.751905896602325), - BQC(0.100724533818628), BQC(1.00009472669872), BQC(-0.0533772830257041), BQC( 0.81930744384525), - BQC(0.0561434357867363), BQC(0.999911636304276), BQC(-0.0913550299236405), BQC( 0.88883625875915), - BQC(0.0341680678662057), BQC(1.00003667508676), BQC(-0.113405185536697), BQC( 0.961756638268446) -}; - -#ifdef RS_BIQUAD_SCATTERGAIN -static const FIXP_DBL g48 = FL2FXCONST_DBL(0.67436532061161992682404480717671 - 0.001); -#else -static const FIXP_DBL g48 = FL2FXCONST_DBL(0.002712866530047) - (FIXP_DBL)0x8000; -#endif - -static const struct FILTER_PARAM param_set48 = { - sos48, - g48, - 480, - 15, - 4 /* LF 2 */ -}; - -/** - *\brief Low Pass - Wc = 0,5, order 24, Stop Band -96dB. Wc criteria is "almost 0dB passband", not the usual -3db gain point. - [b,a]=cheby2(24,96,0.5) - [sos,g]=tf2sos(b,a) - bandwidth 0.45 - */ -static const FIXP_SGL sos45[] = { - BQC(1.982962601444), BQC(1.00000000007504), BQC(0.646113303737836), BQC( 0.10851149979981), - BQC(1.85334094281111), BQC(0.999999999677192), BQC(0.612073220102006), BQC( 0.130022141698044), - BQC(1.62541051415425), BQC(1.00000000080398), BQC(0.547879702855959), BQC( 0.171165825133192), - BQC(1.34554656923247), BQC(0.9999999980169), BQC(0.460373914508491), BQC( 0.228677463376354), - BQC(1.05656568503116), BQC(1.00000000569363), BQC(0.357891894038287), BQC( 0.298676843912185), - BQC(0.787967587877312), BQC(0.999999984415017), BQC(0.248826893211877), BQC( 0.377441803512978), - BQC(0.555480971120497), BQC(1.00000003583307), BQC(0.140614263345315), BQC( 0.461979302213679), - BQC(0.364986207070964), BQC(0.999999932084303), BQC(0.0392669446074516), BQC( 0.55033451180825), - BQC(0.216827267631558), BQC(1.00000010534682), BQC(-0.0506232228865103), BQC( 0.641691581560946), - BQC(0.108951672277119), BQC(0.999999871167516), BQC(-0.125584840183225), BQC( 0.736367748771803), - BQC(0.0387988607229035), BQC(1.00000011205574), BQC(-0.182814849097974), BQC( 0.835802108714964), - BQC(0.0042866175809225), BQC(0.999999954830813), BQC(-0.21965740617151), BQC( 0.942623047782363) -}; - -#ifdef RS_BIQUAD_SCATTERGAIN -static const FIXP_DBL g45 = FL2FXCONST_DBL(0.60547428891341319051142629706723 - 0.001); -#else -static const FIXP_DBL g45 = FL2FXCONST_DBL(0.00242743980909524) - (FIXP_DBL)0x8000; -#endif - -static const struct FILTER_PARAM param_set45 = { - sos45, - g45, - 450, - 12, - 4 /* LF 2 */ -}; - -/* - Created by Octave 2.1.73, Mon Oct 13 17:31:32 2008 CEST - Wc = 0,5, order 16, Stop Band -96dB damping. - [b,a]=cheby2(16,96,0.5) - [sos,g]=tf2sos(b,a) - bandwidth = 0.41 - */ - -static const FIXP_SGL sos41[] = -{ - BQC(1.96193625292), BQC(0.999999999999964), BQC(0.169266178786789), BQC(0.0128823300475907), - BQC(1.68913437662092), BQC(1.00000000000053), BQC(0.124751503206552), BQC(0.0537472273950989), - BQC(1.27274692366017), BQC(0.999999999995674), BQC(0.0433108625178357), BQC(0.131015753236317), - BQC(0.85214175088395), BQC(1.00000000001813), BQC(-0.0625658152550408), BQC(0.237763778993806), - BQC(0.503841579939009), BQC(0.999999999953223), BQC(-0.179176128722865), BQC(0.367475236424474), - BQC(0.249990711986162), BQC(1.00000000007952), BQC(-0.294425165824676), BQC(0.516594857170212), - BQC(0.087971668680286), BQC(0.999999999915528), BQC(-0.398956566777928), BQC(0.686417767801123), - BQC(0.00965373325350294), BQC(1.00000000003744), BQC(-0.48579173764817), BQC(0.884931534239068) -}; - -#ifdef RS_BIQUAD_SCATTERGAIN -static const FIXP_DBL g41 = FL2FXCONST_DBL(0.44578514476476679750811222123569); -#else -static const FIXP_DBL g41 = FL2FXCONST_DBL(0.00155956951169248); -#endif - -static const struct FILTER_PARAM param_set41 = { - sos41, - g41, - 410, - 8, - 5 /* LF 3 */ -}; - -/* - # Created by Octave 2.1.73, Mon Oct 13 17:55:33 2008 CEST - Wc = 0,5, order 12, Stop Band -96dB damping. - [b,a]=cheby2(12,96,0.5); - [sos,g]=tf2sos(b,a) -*/ -static const FIXP_SGL sos35[] = -{ - BQC(1.93299325235762), BQC(0.999999999999985), BQC(-0.140733187246596), BQC(0.0124139497836062), - BQC(1.4890416764109), BQC(1.00000000000011), BQC(-0.198215402588504), BQC(0.0746730616584138), - BQC(0.918450161309795), BQC(0.999999999999619), BQC(-0.30133912791941), BQC(0.192276468839529), - BQC(0.454877024246818), BQC(1.00000000000086), BQC(-0.432337328809815), BQC(0.356852933642815), - BQC(0.158017147118507), BQC(0.999999999998876), BQC(-0.574817494249777), BQC(0.566380436970833), - BQC(0.0171834649478749), BQC(1.00000000000055), BQC(-0.718581178041165), BQC(0.83367484487889) -}; - -#ifdef RS_BIQUAD_SCATTERGAIN -static const FIXP_DBL g35 = FL2FXCONST_DBL(0.34290853574973898694521267606792); -#else -static const FIXP_DBL g35 = FL2FXCONST_DBL(0.00162580994125131); -#endif - -static const struct FILTER_PARAM param_set35 = { - sos35, - g35, - 350, - 6, - 4 -}; - -/* - # Created by Octave 2.1.73, Mon Oct 13 18:15:38 2008 CEST - Wc = 0,5, order 8, Stop Band -96dB damping. - [b,a]=cheby2(8,96,0.5); - [sos,g]=tf2sos(b,a) -*/ -static const FIXP_SGL sos25[] = -{ - BQC(1.85334094301225), BQC(1.0), BQC(-0.702127214212663), BQC(0.132452403998767), - BQC(1.056565682167), BQC(0.999999999999997), BQC(-0.789503667880785), BQC(0.236328693569128), - BQC(0.364986307455489), BQC(0.999999999999996), BQC(-0.955191189843375), BQC(0.442966457936379), - BQC(0.0387985751642125), BQC(1.0), BQC(-1.19817786088084), BQC(0.770493895456328) -}; - -#ifdef RS_BIQUAD_SCATTERGAIN -static const FIXP_DBL g25 = FL2FXCONST_DBL(0.17533917408936346960080259950471); -#else -static const FIXP_DBL g25 = FL2FXCONST_DBL(0.000945182835294559); -#endif - -static const struct FILTER_PARAM param_set25 = { - sos25, - g25, - 250, - 4, - 5 -}; - -/* Must be sorted in descending order */ -static const struct FILTER_PARAM *const filter_paramSet[] = { - ¶m_set48, - ¶m_set45, - ¶m_set41, - ¶m_set35, - ¶m_set25 -}; - - -/**************************************************************************/ -/* Resampler Functions */ -/**************************************************************************/ - - -/*! - \brief Reset downsampler instance and clear delay lines - - \return success of operation -*/ - -INT FDKaacEnc_InitDownsampler(DOWNSAMPLER *DownSampler, /*!< pointer to downsampler instance */ - int Wc, /*!< normalized cutoff freq * 1000* */ - int ratio) /*!< downsampler ratio (only 2 supported at the momment) */ - -{ - UINT i; - const struct FILTER_PARAM *currentSet=NULL; - - FDK_ASSERT(ratio == 2); - FDKmemclear(DownSampler->downFilter.states, sizeof(DownSampler->downFilter.states)); - DownSampler->downFilter.ptr = 0; - - /* - find applicable parameter set - */ - currentSet = filter_paramSet[0]; - for(i=1;iWc <= Wc) { - break; - } - currentSet = filter_paramSet[i]; - } - - DownSampler->downFilter.coeffa = currentSet->coeffa; - - - DownSampler->downFilter.gain = currentSet->g; - FDK_ASSERT(currentSet->noCoeffs <= MAXNR_SECTIONS*2); - - DownSampler->downFilter.noCoeffs = currentSet->noCoeffs; - DownSampler->delay = currentSet->delay; - DownSampler->downFilter.Wc = currentSet->Wc; - - DownSampler->ratio = ratio; - DownSampler->pending = ratio-1; - return(1); -} - - -/*! - \brief faster simple folding operation - Filter: - H(z) = A(z)/B(z) - with - A(z) = a[0]*z^0 + a[1]*z^1 + a[2]*z^2 ... a[n]*z^n - - \return filtered value -*/ - -static inline INT_PCM AdvanceFilter(LP_FILTER *downFilter, /*!< pointer to iir filter instance */ - INT_PCM *pInput, /*!< input of filter */ - int downRatio, - int inStride) -{ - INT_PCM output; - int i, n; - - -#ifdef RS_BIQUAD_SCATTERGAIN -#define BIQUAD_SCALE 3 -#else -#define BIQUAD_SCALE 12 -#endif - - FIXP_DBL y = FL2FXCONST_DBL(0.0f); - FIXP_DBL input; - - for (n=0; nstates; - const FIXP_SGL *coeff = downFilter->coeffa; - int s1,s2; - - s1 = downFilter->ptr; - s2 = s1 ^ 1; - -#if (SAMPLE_BITS == 16) - input = ((FIXP_DBL)pInput[n*inStride]) << (DFRACT_BITS-SAMPLE_BITS-BIQUAD_SCALE); -#elif (SAMPLE_BITS == 32) - input = pInput[n*inStride] >> BIQUAD_SCALE; -#else -#error NOT IMPLEMENTED -#endif - -#ifndef RS_BIQUAD_SCATTERGAIN /* Merged Direct form I */ - - FIXP_BQS state1, state2, state1b, state2b; - - state1 = states[0][s1]; - state2 = states[0][s2]; - - /* Loop over sections */ - for (i=0; inoCoeffs; i++) - { - FIXP_DBL state0; - - /* Load merged states (from next section) */ - state1b = states[i+1][s1]; - state2b = states[i+1][s2]; - - state0 = input + fMult(state1, coeff[B1]) + fMult(state2, coeff[B2]); - y = state0 - fMult(state1b, coeff[A1]) - fMult(state2b, coeff[A2]); - - /* Store new feed forward merge state */ - states[i+1][s2] = y<<1; - /* Store new feed backward state */ - states[i][s2] = input<<1; - - /* Feedback output to next section. */ - input = y; - - /* Transfer merged states */ - state1 = state1b; - state2 = state2b; - - /* Step to next coef set */ - coeff += BIQUAD_COEFSTEP; - } - downFilter->ptr ^= 1; - } - /* Apply global gain */ - y = fMult(y, downFilter->gain); - -#else /* Direct form II */ - - /* Loop over sections */ - for (i=0; inoCoeffs; i++) - { - FIXP_BQS state1, state2; - FIXP_DBL state0; - - /* Load states */ - state1 = states[i][s1]; - state2 = states[i][s2]; - - state0 = input - fMult(state1, coeff[A1]) - fMult(state2, coeff[A2]); - y = state0 + fMult(state1, coeff[B1]) + fMult(state2, coeff[B2]); - /* Apply scattered gain */ - y = fMult(y, downFilter->gain); - - /* Store new state in normalized form */ -#ifdef RS_BIQUAD_STATES16 - /* Do not saturate any state value ! The result would be unacceptable. Rounding makes SNR around 10dB better. */ - states[i][s2] = (FIXP_BQS)(LONG)((state0 + (FIXP_DBL)(1<<(DFRACT_BITS-FRACT_BITS-2))) >> (DFRACT_BITS-FRACT_BITS-1)); -#else - states[i][s2] = state0<<1; -#endif - - /* Feedback output to next section. */ - input=y; - - /* Step to next coef set */ - coeff += BIQUAD_COEFSTEP; - } - downFilter->ptr ^= 1; - } - -#endif - - /* Apply final gain/scaling to output */ -#if (SAMPLE_BITS == 16) - output = (INT_PCM) SATURATE_RIGHT_SHIFT(y+(FIXP_DBL)(1<<(DFRACT_BITS-SAMPLE_BITS-BIQUAD_SCALE-1)), DFRACT_BITS-SAMPLE_BITS-BIQUAD_SCALE, SAMPLE_BITS); - //output = (INT_PCM) SATURATE_RIGHT_SHIFT(y, DFRACT_BITS-SAMPLE_BITS-BIQUAD_SCALE, SAMPLE_BITS); -#else - output = SATURATE_LEFT_SHIFT(y, BIQUAD_SCALE, SAMPLE_BITS); -#endif - - - return output; -} - - - - -/*! - \brief FDKaacEnc_Downsample numInSamples of type INT_PCM - Returns number of output samples in numOutSamples - - \return success of operation -*/ - -INT FDKaacEnc_Downsample(DOWNSAMPLER *DownSampler, /*!< pointer to downsampler instance */ - INT_PCM *inSamples, /*!< pointer to input samples */ - INT numInSamples, /*!< number of input samples */ - INT inStride, /*!< increment of input samples */ - INT_PCM *outSamples, /*!< pointer to output samples */ - INT *numOutSamples, /*!< pointer tp number of output samples */ - INT outStride /*!< increment of output samples */ - ) -{ - INT i; - *numOutSamples=0; - - for(i=0; iratio) - { - *outSamples = AdvanceFilter(&(DownSampler->downFilter), &inSamples[i*inStride], DownSampler->ratio, inStride); - outSamples += outStride; - } - *numOutSamples = numInSamples/DownSampler->ratio; - - return 0; -} - diff --git a/libSBRenc/src/resampler.h b/libSBRenc/src/resampler.h deleted file mode 100644 index 0192970..0000000 --- a/libSBRenc/src/resampler.h +++ /dev/null @@ -1,151 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -#ifndef __RESAMPLER_H -#define __RESAMPLER_H -/*! - \file - \brief Fixed Point Resampler Tool Box -*/ - -#include "common_fix.h" - - -/**************************************************************************/ -/* BIQUAD Filter Structure */ -/**************************************************************************/ - -#define MAXNR_SECTIONS (15) - -#ifdef RS_BIQUAD_STATES16 -typedef FIXP_SGL FIXP_BQS; -#else -typedef FIXP_DBL FIXP_BQS; -#endif - -typedef struct -{ - FIXP_BQS states[MAXNR_SECTIONS+1][2]; /*! state buffer */ - const FIXP_SGL *coeffa; /*! pointer to filter coeffs */ - FIXP_DBL gain; /*! overall gain factor */ - int Wc; /*! normalized cutoff freq * 1000 */ - int noCoeffs; /*! number of filter coeffs sets */ - int ptr; /*! index to rinbuffers */ -} LP_FILTER; - - -/**************************************************************************/ -/* Downsampler Structure */ -/**************************************************************************/ - -typedef struct -{ - LP_FILTER downFilter; /*! filter instance */ - int ratio; /*! downsampling ration */ - int delay; /*! downsampling delay (source fs) */ - int pending; /*! number of pending output samples */ -} DOWNSAMPLER; - - -/** - * \brief Initialized a given downsampler structure. - */ -INT FDKaacEnc_InitDownsampler(DOWNSAMPLER *DownSampler, /*!< pointer to downsampler instance */ - INT Wc, /*!< normalized cutoff freq * 1000 */ - INT ratio); /*!< downsampler ratio */ - -/** - * \brief Downsample a set of audio samples. numInSamples must be at least equal to the - * downsampler ratio. - */ -INT FDKaacEnc_Downsample(DOWNSAMPLER *DownSampler, /*!< pointer to downsampler instance */ - INT_PCM *inSamples, /*!< pointer to input samples */ - INT numInSamples, /*!< number of input samples */ - INT inStride, /*!< increment of input samples */ - INT_PCM *outSamples, /*!< pointer to output samples */ - INT *numOutSamples, /*!< pointer tp number of output samples */ - INT outstride); /*!< increment of output samples */ - - - -#endif /* __RESAMPLER_H */ diff --git a/libSBRenc/src/sbr.h b/libSBRenc/src/sbr.h deleted file mode 100644 index c74ad2a..0000000 --- a/libSBRenc/src/sbr.h +++ /dev/null @@ -1,166 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief Main SBR structs definitions -*/ - -#ifndef __SBR_H -#define __SBR_H - -#include "fram_gen.h" -#include "bit_sbr.h" -#include "tran_det.h" -#include "code_env.h" -#include "env_est.h" -#include "cmondata.h" - -#include "qmf.h" -#include "resampler.h" - -#include "ton_corr.h" - - -/* SBR bitstream delay */ - #define DELAY_FRAMES 2 - - -typedef struct SBR_CHANNEL { - struct ENV_CHANNEL hEnvChannel; - //INT_PCM *pDSOutBuffer; /**< Pointer to downsampled audio output of SBR encoder */ - DOWNSAMPLER downSampler; - -} SBR_CHANNEL; -typedef SBR_CHANNEL* HANDLE_SBR_CHANNEL; - -typedef struct SBR_ELEMENT { - HANDLE_SBR_CHANNEL sbrChannel[2]; - QMF_FILTER_BANK *hQmfAnalysis[2]; - SBR_CONFIG_DATA sbrConfigData; - SBR_HEADER_DATA sbrHeaderData; - SBR_BITSTREAM_DATA sbrBitstreamData; - COMMON_DATA CmonData; - INT dynXOverFreqDelay[5]; /**< to delay a frame (I don't like it that much that way - hrc) */ - SBR_ELEMENT_INFO elInfo; - - UCHAR payloadDelayLine[1+DELAY_FRAMES][MAX_PAYLOAD_SIZE]; - UINT payloadDelayLineSize[1+DELAY_FRAMES]; /* Sizes in bits */ - -} SBR_ELEMENT, *HANDLE_SBR_ELEMENT; - -typedef struct SBR_ENCODER -{ - HANDLE_SBR_ELEMENT sbrElement[(8)]; - HANDLE_SBR_CHANNEL pSbrChannel[(8)]; - QMF_FILTER_BANK QmfAnalysis[(8)]; - DOWNSAMPLER lfeDownSampler; - int lfeChIdx; /* -1 default for no lfe, else assign channel index */ - int noElements; /* Number of elements */ - int nChannels; /* Total channel count across all elements. */ - int frameSize; /* SBR framelength. */ - int bufferOffset; /* Offset for SBR parameter extraction in time domain input buffer. */ - int downsampledOffset; /* Offset of downsampled/mixed output for core encoder. */ - int downmixSize; /* Size in samples of downsampled/mixed output for core encoder. */ - INT downSampleFactor; /* Sampling rate relation between the SBR and the core encoder. */ - int fTimeDomainDownsampling; /* Flag signalling time domain downsampling instead of QMF downsampling. */ - int nBitstrDelay; /* Amount of SBR frames to be delayed in bitstream domain. */ - INT estimateBitrate; /* estimate bitrate of SBR encoder */ - INT inputDataDelay; /* delay caused by downsampler, in/out buffer at sbrEncoder_EncodeFrame */ - - UCHAR* dynamicRam; - UCHAR* pSBRdynamic_RAM; - - HANDLE_PARAMETRIC_STEREO hParametricStereo; - QMF_FILTER_BANK qmfSynthesisPS; - - /* parameters describing allocation volume of present instance */ - INT maxElements; - INT maxChannels; - INT supportPS; - - -} SBR_ENCODER; - - -#endif /* __SBR_H */ diff --git a/libSBRenc/src/sbr_def.h b/libSBRenc/src/sbr_def.h deleted file mode 100644 index 85ac587..0000000 --- a/libSBRenc/src/sbr_def.h +++ /dev/null @@ -1,275 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief SBR main definitions -*/ -#ifndef __SBR_DEF_H -#define __SBR_DEF_H - -#include "common_fix.h" - -#define noError 0 -#define HANDLE_ERROR_INFO INT -#define ERROR(a,b) 1 -#define handBack - -/* #define SBR_ENV_STATISTICS_BITRATE */ -#undef SBR_ENV_STATISTICS_BITRATE - -/* #define SBR_ENV_STATISTICS */ -#undef SBR_ENV_STATISTICS - -/* #define SBR_PAYLOAD_MONITOR */ -#undef SBR_PAYLOAD_MONITOR - -#define SWAP(a,b) tempr=a, a=b, b=tempr -#define TRUE 1 -#define FALSE 0 - - -/* Constants */ -#define EPS 1e-12 -#define LOG2 0.69314718056f /* natural logarithm of 2 */ -#define ILOG2 1.442695041f /* 1/LOG2 */ -#define RELAXATION_FLOAT (1e-6f) -#define RELAXATION (FL2FXCONST_DBL(RELAXATION_FLOAT)) -#define RELAXATION_FRACT (FL2FXCONST_DBL(0.524288f)) /* 0.524288f is fractional part of RELAXATION */ -#define RELAXATION_SHIFT (19) -#define RELAXATION_LD64 (FL2FXCONST_DBL(0.31143075889f))/* (ld64(RELAXATION) */ - -/************ Definitions ***************/ -#define SBR_COMP_MODE_DELTA 0 -#define SBR_COMP_MODE_CTS 1 -#define SBR_MAX_ENERGY_VALUES 5 -#define SBR_GLOBAL_TONALITY_VALUES 2 - -#define MAX_NUM_CHANNELS 2 - -#define MAX_NOISE_ENVELOPES 2 -#define MAX_NUM_NOISE_COEFFS 5 -#define MAX_NUM_NOISE_VALUES (MAX_NUM_NOISE_COEFFS*MAX_NOISE_ENVELOPES) - -#define MAX_NUM_ENVELOPE_VALUES (MAX_ENVELOPES * MAX_FREQ_COEFFS) -#define MAX_ENVELOPES 5 -#define MAX_FREQ_COEFFS 48 - -#define MAX_FREQ_COEFFS_FS44100 35 -#define MAX_FREQ_COEFFS_FS48000 32 - - -#define QMF_CHANNELS 64 -#define QMF_FILTER_LENGTH 640 -#define QMF_MAX_TIME_SLOTS 32 -#define NO_OF_ESTIMATES_LC 4 -#define NO_OF_ESTIMATES_LD 3 -#define MAX_NO_OF_ESTIMATES 4 - - -#define NOISE_FLOOR_OFFSET 6 -#define NOISE_FLOOR_OFFSET_64 (FL2FXCONST_DBL(0.09375f)) - -#define LOW_RES 0 -#define HIGH_RES 1 - -#define LO 0 -#define HI 1 - -#define LENGTH_SBR_FRAME_INFO 35 /* 19 */ - -#define SBR_NSFB_LOW_RES 9 /* 8 */ -#define SBR_NSFB_HIGH_RES 18 /* 16 */ - - -#define SBR_XPOS_CTRL_DEFAULT 2 - -#define SBR_FREQ_SCALE_DEFAULT 2 -#define SBR_ALTER_SCALE_DEFAULT 1 -#define SBR_NOISE_BANDS_DEFAULT 2 - -#define SBR_LIMITER_BANDS_DEFAULT 2 -#define SBR_LIMITER_GAINS_DEFAULT 2 -#define SBR_LIMITER_GAINS_INFINITE 3 -#define SBR_INTERPOL_FREQ_DEFAULT 1 -#define SBR_SMOOTHING_LENGTH_DEFAULT 0 - - -/* sbr_header */ -#define SI_SBR_AMP_RES_BITS 1 -#define SI_SBR_COUPLING_BITS 1 -#define SI_SBR_START_FREQ_BITS 4 -#define SI_SBR_STOP_FREQ_BITS 4 -#define SI_SBR_XOVER_BAND_BITS 3 -#define SI_SBR_RESERVED_BITS 2 -#define SI_SBR_DATA_EXTRA_BITS 1 -#define SI_SBR_HEADER_EXTRA_1_BITS 1 -#define SI_SBR_HEADER_EXTRA_2_BITS 1 - -/* sbr_header extra 1 */ -#define SI_SBR_FREQ_SCALE_BITS 2 -#define SI_SBR_ALTER_SCALE_BITS 1 -#define SI_SBR_NOISE_BANDS_BITS 2 - -/* sbr_header extra 2 */ -#define SI_SBR_LIMITER_BANDS_BITS 2 -#define SI_SBR_LIMITER_GAINS_BITS 2 -#define SI_SBR_INTERPOL_FREQ_BITS 1 -#define SI_SBR_SMOOTHING_LENGTH_BITS 1 - -/* sbr_grid */ -#define SBR_CLA_BITS 2 /*!< size of bs_frame_class */ -#define SBR_CLA_BITS_LD 1 /*!< size of bs_frame_class */ -#define SBR_ENV_BITS 2 /*!< size of bs_num_env_raw */ -#define SBR_ABS_BITS 2 /*!< size of bs_abs_bord_raw for HE-AAC */ -#define SBR_NUM_BITS 2 /*!< size of bs_num_rel */ -#define SBR_REL_BITS 2 /*!< size of bs_rel_bord_raw */ -#define SBR_RES_BITS 1 /*!< size of bs_freq_res_flag */ -#define SBR_DIR_BITS 1 /*!< size of bs_df_flag */ - - -/* sbr_data */ -#define SI_SBR_INVF_MODE_BITS 2 - - -#define SI_SBR_START_ENV_BITS_AMP_RES_3_0 6 -#define SI_SBR_START_ENV_BITS_BALANCE_AMP_RES_3_0 5 -#define SI_SBR_START_NOISE_BITS_AMP_RES_3_0 5 -#define SI_SBR_START_NOISE_BITS_BALANCE_AMP_RES_3_0 5 - -#define SI_SBR_START_ENV_BITS_AMP_RES_1_5 7 -#define SI_SBR_START_ENV_BITS_BALANCE_AMP_RES_1_5 6 - - -#define SI_SBR_EXTENDED_DATA_BITS 1 -#define SI_SBR_EXTENSION_SIZE_BITS 4 -#define SI_SBR_EXTENSION_ESC_COUNT_BITS 8 -#define SI_SBR_EXTENSION_ID_BITS 2 - -#define SBR_EXTENDED_DATA_MAX_CNT (15+255) - -#define EXTENSION_ID_PS_CODING 2 - -/* Envelope coding constants */ -#define FREQ 0 -#define TIME 1 - -/* qmf data scaling */ -#define QMF_SCALE_OFFSET 7 - -/* huffman tables */ -#define CODE_BOOK_SCF_LAV00 60 -#define CODE_BOOK_SCF_LAV01 31 -#define CODE_BOOK_SCF_LAV10 60 -#define CODE_BOOK_SCF_LAV11 31 -#define CODE_BOOK_SCF_LAV_BALANCE11 12 -#define CODE_BOOK_SCF_LAV_BALANCE10 24 - -typedef enum -{ - SBR_AMP_RES_1_5=0, - SBR_AMP_RES_3_0 -} -AMP_RES; - -typedef enum -{ - XPOS_MDCT, - XPOS_MDCT_CROSS, - XPOS_LC, - XPOS_RESERVED, - XPOS_SWITCHED /* not a real choice but used here to control behaviour */ -} -XPOS_MODE; - -typedef enum -{ - INVF_OFF = 0, - INVF_LOW_LEVEL, - INVF_MID_LEVEL, - INVF_HIGH_LEVEL, - INVF_SWITCHED /* not a real choice but used here to control behaviour */ -} -INVF_MODE; - -#endif diff --git a/libSBRenc/src/sbr_encoder.cpp b/libSBRenc/src/sbr_encoder.cpp deleted file mode 100644 index 71aab78..0000000 --- a/libSBRenc/src/sbr_encoder.cpp +++ /dev/null @@ -1,2443 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*************************** Fraunhofer IIS FDK Tools *********************** - - Author(s): Andreas Ehret, Tobias Chalupka - Description: SBR encoder top level processing. - -******************************************************************************/ - -#include "sbr_encoder.h" - -#include "sbr_ram.h" -#include "sbr_rom.h" -#include "sbrenc_freq_sca.h" -#include "env_bit.h" -#include "cmondata.h" -#include "sbr_misc.h" -#include "sbr.h" -#include "qmf.h" - -#include "ps_main.h" - -#define SBRENCODER_LIB_VL0 3 -#define SBRENCODER_LIB_VL1 3 -#define SBRENCODER_LIB_VL2 12 - - - -/***************************************************************************/ -/* - * SBR Delay balancing definitions. - */ - -/* - input buffer (1ch) - - |------------ 1537 -------------|-----|---------- 2048 -------------| - (core2sbr delay ) ds (read, core and ds area) -*/ - -#define SFB(dwnsmp) (32 << (dwnsmp-1)) /* SBR Frequency bands: 64 for dual-rate, 32 for single-rate */ -#define STS(fl) (((fl)==1024)?32:30) /* SBR Time Slots: 32 for core frame length 1024, 30 for core frame length 960 */ - -#define DELAY_QMF_ANA(dwnsmp) ((320<<((dwnsmp)-1)) - (32<<((dwnsmp)-1))) /* Full bandwidth */ -#define DELAY_HYB_ANA (10*64) /* + 0.5 */ /* */ -#define DELAY_HYB_SYN (6*64 - 32) /* */ -#define DELAY_QMF_POSTPROC(dwnsmp) (32*(dwnsmp)) /* QMF postprocessing delay */ -#define DELAY_DEC_QMF(dwnsmp) (6 * SFB(dwnsmp) ) /* Decoder QMF overlap */ -#define DELAY_QMF_SYN (2) /* NO_POLY/2=2.5, rounded down to 2 */ -#define DELAY_QMF_DS (32) /* QMF synthesis for downsampled time signal */ - -/* Delay in QMF paths */ -#define DELAY_SBR(fl,dwnsmp) (DELAY_QMF_ANA(dwnsmp) + (SFB(dwnsmp)*STS(fl) - 1) + DELAY_QMF_SYN) -#define DELAY_PS(fl,dwnsmp) (DELAY_QMF_ANA(dwnsmp) + DELAY_HYB_ANA + DELAY_DEC_QMF(dwnsmp) + (SFB(dwnsmp)*STS(fl)-1) + DELAY_HYB_SYN + DELAY_QMF_SYN) -#define DELAY_ELDSBR(fl,dwnsmp) ( ( ((fl)/2)*(dwnsmp) ) - 1 + DELAY_QMF_POSTPROC(dwnsmp) ) - -/* Delay differences for SBR and SBR+PS */ -#define MAX_DS_FILTER_DELAY (5) /* the additional max downsampler filter delay (source fs) */ -#define DELAY_AAC2SBR(fl,dwnsmp) ((DELAY_QMF_ANA(dwnsmp) + DELAY_DEC_QMF(dwnsmp) + DELAY_QMF_SYN) - DELAY_SBR((fl),(dwnsmp))) -#define DELAY_ELD2SBR(fl,dwnsmp) ((DELAY_QMF_POSTPROC(dwnsmp)) - DELAY_ELDSBR(fl,dwnsmp)) -#define DELAY_AAC2PS(fl,dwnsmp) ((DELAY_QMF_ANA(dwnsmp) + DELAY_QMF_DS + /*(DELAY_AAC(fl)*2) + */ DELAY_QMF_ANA(dwnsmp) + DELAY_DEC_QMF(dwnsmp) + DELAY_HYB_SYN + DELAY_QMF_SYN) - DELAY_PS(fl,dwnsmp)) /* 2048 - 463*2 */ - -/* Assumption: The sample delay resulting of of DELAY_AAC2PS is always smaller than the sample delay implied by DELAY_AAC2SBR */ -#define MAX_SAMPLE_DELAY (DELAY_AAC2SBR(1024,2) + MAX_DS_FILTER_DELAY) /* maximum delay: frame length of 1024 and dual-rate sbr */ - -/***************************************************************************/ - - - -#define INVALID_TABLE_IDX -1 - -/***************************************************************************/ -/*! - - \brief Selects the SBR tuning settings to use dependent on number of - channels, bitrate, sample rate and core coder - - \return Index to the appropriate table - -****************************************************************************/ -#define DISTANCE_CEIL_VALUE 5000000 -static INT -getSbrTuningTableIndex(UINT bitrate, /*! the total bitrate in bits/sec */ - UINT numChannels,/*! the number of channels for the core coder */ - UINT sampleRate, /*! the sampling rate of the core coder */ - AUDIO_OBJECT_TYPE core, - UINT *pBitRateClosest - ) -{ - int i, bitRateClosestLowerIndex=-1, bitRateClosestUpperIndex=-1, found = 0; - UINT bitRateClosestUpper = 0, bitRateClosestLower=DISTANCE_CEIL_VALUE; - - #define isForThisCore(i) \ - ( ( sbrTuningTable[i].coreCoder == CODEC_AACLD && core == AOT_ER_AAC_ELD ) || \ - ( sbrTuningTable[i].coreCoder == CODEC_AAC && core != AOT_ER_AAC_ELD ) ) - - for (i=0; i < sbrTuningTableSize ; i++) { - if ( isForThisCore(i) ) /* tuning table is for this core codec */ - { - if ( numChannels == sbrTuningTable [i].numChannels - && sampleRate == sbrTuningTable [i].sampleRate ) - { - found = 1; - if ((bitrate >= sbrTuningTable [i].bitrateFrom) && - (bitrate < sbrTuningTable [i].bitrateTo)) { - bitRateClosestLower = bitrate; - bitRateClosestUpper = bitrate; - //FDKprintf("entry %d\n", i); - return i ; - } else { - if ( sbrTuningTable [i].bitrateFrom > bitrate ) { - if (sbrTuningTable [i].bitrateFrom < bitRateClosestLower) { - bitRateClosestLower = sbrTuningTable [i].bitrateFrom; - bitRateClosestLowerIndex = i; - } - } - if ( sbrTuningTable [i].bitrateTo <= bitrate ) { - if (sbrTuningTable [i].bitrateTo > bitRateClosestUpper) { - bitRateClosestUpper = sbrTuningTable [i].bitrateTo-1; - bitRateClosestUpperIndex = i; - } - } - } - } - } - } - - if (pBitRateClosest != NULL) - { - /* If there was at least one matching tuning entry found then pick the least distance bit rate */ - if (found) - { - int distanceUpper=DISTANCE_CEIL_VALUE, distanceLower=DISTANCE_CEIL_VALUE; - if (bitRateClosestLowerIndex >= 0) { - distanceLower = sbrTuningTable [bitRateClosestLowerIndex].bitrateFrom - bitrate; - } - if (bitRateClosestUpperIndex >= 0) { - distanceUpper = bitrate - sbrTuningTable [bitRateClosestUpperIndex].bitrateTo; - } - if ( distanceUpper < distanceLower ) - { - *pBitRateClosest = bitRateClosestUpper; - } else { - *pBitRateClosest = bitRateClosestLower; - } - } else { - *pBitRateClosest = 0; - } - } - - return INVALID_TABLE_IDX; -} - -/***************************************************************************/ -/*! - - \brief Selects the PS tuning settings to use dependent on bitrate - and core coder - - \return Index to the appropriate table - -****************************************************************************/ -static INT -getPsTuningTableIndex(UINT bitrate, UINT *pBitRateClosest){ - - INT i, paramSets = sizeof (psTuningTable) / sizeof (psTuningTable [0]); - int bitRateClosestLowerIndex=-1, bitRateClosestUpperIndex=-1; - UINT bitRateClosestUpper = 0, bitRateClosestLower=DISTANCE_CEIL_VALUE; - - for (i = 0 ; i < paramSets ; i++) { - if ((bitrate >= psTuningTable [i].bitrateFrom) && - (bitrate < psTuningTable [i].bitrateTo)) { - return i ; - } else { - if ( psTuningTable [i].bitrateFrom > bitrate ) { - if (psTuningTable [i].bitrateFrom < bitRateClosestLower) { - bitRateClosestLower = psTuningTable [i].bitrateFrom; - bitRateClosestLowerIndex = i; - } - } - if ( psTuningTable [i].bitrateTo <= bitrate ) { - if (psTuningTable [i].bitrateTo > bitRateClosestUpper) { - bitRateClosestUpper = psTuningTable [i].bitrateTo-1; - bitRateClosestUpperIndex = i; - } - } - } - } - - if (pBitRateClosest != NULL) - { - int distanceUpper=DISTANCE_CEIL_VALUE, distanceLower=DISTANCE_CEIL_VALUE; - if (bitRateClosestLowerIndex >= 0) { - distanceLower = sbrTuningTable [bitRateClosestLowerIndex].bitrateFrom - bitrate; - } - if (bitRateClosestUpperIndex >= 0) { - distanceUpper = bitrate - sbrTuningTable [bitRateClosestUpperIndex].bitrateTo; - } - if ( distanceUpper < distanceLower ) - { - *pBitRateClosest = bitRateClosestUpper; - } else { - *pBitRateClosest = bitRateClosestLower; - } - } - - return INVALID_TABLE_IDX; -} - -/***************************************************************************/ -/*! - - \brief In case of downsampled SBR we may need to lower the stop freq - of a tuning setting to fit into the lower half of the - spectrum ( which is sampleRate/4 ) - - \return the adapted stop frequency index (-1 -> error) - - \ingroup SbrEncCfg - -****************************************************************************/ -static INT -FDKsbrEnc_GetDownsampledStopFreq ( - const INT sampleRateCore, - const INT startFreq, - INT stopFreq, - const INT downSampleFactor - ) -{ - INT maxStopFreqRaw = sampleRateCore / 2; - INT startBand, stopBand; - HANDLE_ERROR_INFO err; - - while (stopFreq > 0 && FDKsbrEnc_getSbrStopFreqRAW(stopFreq, sampleRateCore) > maxStopFreqRaw) { - stopFreq--; - } - - if (FDKsbrEnc_getSbrStopFreqRAW( stopFreq, sampleRateCore) > maxStopFreqRaw) - return -1; - - err = FDKsbrEnc_FindStartAndStopBand ( - sampleRateCore<<(downSampleFactor-1), - sampleRateCore, - 32<<(downSampleFactor-1), - startFreq, - stopFreq, - &startBand, - &stopBand - ); - if (err) - return -1; - - return stopFreq; -} - - -/***************************************************************************/ -/*! - - \brief tells us, if for the given coreCoder, bitrate, number of channels - and input sampling rate an SBR setting is available. If yes, it - tells us also the core sampling rate we would need to run with - - \return a flag indicating success: yes (1) or no (0) - -****************************************************************************/ -static UINT -FDKsbrEnc_IsSbrSettingAvail ( - UINT bitrate, /*! the total bitrate in bits/sec */ - UINT vbrMode, /*! the vbr paramter, 0 means constant bitrate */ - UINT numOutputChannels, /*! the number of channels for the core coder */ - UINT sampleRateInput, /*! the input sample rate [in Hz] */ - UINT sampleRateCore, /*! the core's sampling rate */ - AUDIO_OBJECT_TYPE core - ) -{ - INT idx = INVALID_TABLE_IDX; - - if (sampleRateInput < 16000) - return 0; - - if (bitrate==0) { - /* map vbr quality to bitrate */ - if (vbrMode < 30) - bitrate = 24000; - else if (vbrMode < 40) - bitrate = 28000; - else if (vbrMode < 60) - bitrate = 32000; - else if (vbrMode < 75) - bitrate = 40000; - else - bitrate = 48000; - bitrate *= numOutputChannels; - } - - idx = getSbrTuningTableIndex(bitrate, numOutputChannels, sampleRateCore, core, NULL); - - return (idx == INVALID_TABLE_IDX ? 0 : 1); -} - - -/***************************************************************************/ -/*! - - \brief Adjusts the SBR settings according to the chosen core coder - settings which are accessible via config->codecSettings - - \return A flag indicating success: yes (1) or no (0) - -****************************************************************************/ -static UINT -FDKsbrEnc_AdjustSbrSettings (const sbrConfigurationPtr config, /*! output, modified */ - UINT bitRate, /*! the total bitrate in bits/sec */ - UINT numChannels, /*! the core coder number of channels */ - UINT sampleRateCore, /*! the core coder sampling rate in Hz */ - UINT sampleRateSbr, /*! the sbr coder sampling rate in Hz */ - UINT transFac, /*! the short block to long block ratio */ - UINT standardBitrate, /*! the standard bitrate per channel in bits/sec */ - UINT vbrMode, /*! the vbr paramter, 0 poor quality .. 100 high quality*/ - UINT useSpeechConfig, /*!< adapt tuning parameters for speech ? */ - UINT lcsMode, /*! the low complexity stereo mode */ - UINT bParametricStereo, /*!< use parametric stereo */ - AUDIO_OBJECT_TYPE core) /* Core audio codec object type */ -{ - INT idx = INVALID_TABLE_IDX; - /* set the core codec settings */ - config->codecSettings.bitRate = bitRate; - config->codecSettings.nChannels = numChannels; - config->codecSettings.sampleFreq = sampleRateCore; - config->codecSettings.transFac = transFac; - config->codecSettings.standardBitrate = standardBitrate; - - if (bitRate < 28000) { - config->threshold_AmpRes_FF_m = (FIXP_DBL)MAXVAL_DBL; - config->threshold_AmpRes_FF_e = 7; - } - else if (bitRate >= 28000 && bitRate <= 48000) { - /* The float threshold is 75 - 0.524288f is fractional part of RELAXATION, the quotaMatrix and therefore tonality are scaled by this - 2/3 is because the original implementation divides the tonality values by 3, here it's divided by 2 - 128 compensates the necessary shiftfactor of 7 */ - config->threshold_AmpRes_FF_m = FL2FXCONST_DBL(75.0f*0.524288f/(2.0f/3.0f)/128.0f); - config->threshold_AmpRes_FF_e = 7; - } - else if (bitRate > 48000) { - config->threshold_AmpRes_FF_m = FL2FXCONST_DBL(0); - config->threshold_AmpRes_FF_e = 0; - } - - if (bitRate==0) { - /* map vbr quality to bitrate */ - if (vbrMode < 30) - bitRate = 24000; - else if (vbrMode < 40) - bitRate = 28000; - else if (vbrMode < 60) - bitRate = 32000; - else if (vbrMode < 75) - bitRate = 40000; - else - bitRate = 48000; - bitRate *= numChannels; - /* fix to enable mono vbrMode<40 @ 44.1 of 48kHz */ - if (numChannels==1) { - if (sampleRateSbr==44100 || sampleRateSbr==48000) { - if (vbrMode<40) bitRate = 32000; - } - } - } - - idx = getSbrTuningTableIndex(bitRate,numChannels,sampleRateCore, core, NULL); - - if (idx != INVALID_TABLE_IDX) { - config->startFreq = sbrTuningTable[idx].startFreq ; - config->stopFreq = sbrTuningTable[idx].stopFreq ; - if (useSpeechConfig) { - config->startFreq = sbrTuningTable[idx].startFreqSpeech; - config->stopFreq = sbrTuningTable[idx].stopFreqSpeech; - } - - /* Adapt stop frequency in case of downsampled SBR - only 32 bands then */ - if (1 == config->downSampleFactor) { - INT dsStopFreq = FDKsbrEnc_GetDownsampledStopFreq( - sampleRateCore, - config->startFreq, - config->stopFreq, - config->downSampleFactor - ); - if (dsStopFreq < 0) { - return 0; - } - - config->stopFreq = dsStopFreq; - } - - config->sbr_noise_bands = sbrTuningTable[idx].numNoiseBands ; - if (core == AOT_ER_AAC_ELD) - config->init_amp_res_FF = SBR_AMP_RES_1_5; - config->noiseFloorOffset= sbrTuningTable[idx].noiseFloorOffset; - - config->ana_max_level = sbrTuningTable[idx].noiseMaxLevel ; - config->stereoMode = sbrTuningTable[idx].stereoMode ; - config->freqScale = sbrTuningTable[idx].freqScale ; - - if (numChannels == 1) { - /* stereo case */ - switch (core) { - case AOT_AAC_LC: - if (bitRate <= (useSpeechConfig?24000U:20000U)) { - config->freq_res_fixfix[0] = FREQ_RES_LOW; /* set low frequency resolution for non-split frames */ - config->freq_res_fixfix[1] = FREQ_RES_LOW; /* set low frequency resolution for split frames */ - } - break; - case AOT_ER_AAC_ELD: - if (bitRate < 36000) - config->freq_res_fixfix[1] = FREQ_RES_LOW; /* set low frequency resolution for split frames */ - if (bitRate < 26000) { - config->freq_res_fixfix[0] = FREQ_RES_LOW; /* set low frequency resolution for non-split frames */ - config->fResTransIsLow = 1; /* for transient frames, set low frequency resolution */ - } - break; - default: - break; - } - } - else { - /* stereo case */ - switch (core) { - case AOT_AAC_LC: - if (bitRate <= 28000) { - config->freq_res_fixfix[0] = FREQ_RES_LOW; /* set low frequency resolution for non-split frames */ - config->freq_res_fixfix[1] = FREQ_RES_LOW; /* set low frequency resolution for split frames */ - } - break; - case AOT_ER_AAC_ELD: - if (bitRate < 72000) { - config->freq_res_fixfix[1] = FREQ_RES_LOW; /* set low frequency resolution for split frames */ - } - if (bitRate < 52000) { - config->freq_res_fixfix[0] = FREQ_RES_LOW; /* set low frequency resolution for non-split frames */ - config->fResTransIsLow = 1; /* for transient frames, set low frequency resolution */ - } - break; - default: - break; - } - if (bitRate <= 28000) { - /* - additionally restrict frequency resolution in FIXFIX frames - to further reduce SBR payload size */ - config->freq_res_fixfix[0] = FREQ_RES_LOW; - config->freq_res_fixfix[1] = FREQ_RES_LOW; - } - } - - /* adjust usage of parametric coding dependent on bitrate and speech config flag */ - if (useSpeechConfig) - config->parametricCoding = 0; - - if (core == AOT_ER_AAC_ELD) { - if (bitRate < 28000) - config->init_amp_res_FF = SBR_AMP_RES_3_0; - config->SendHeaderDataTime = -1; - } - - if (numChannels == 1) { - if (bitRate < 16000) { - config->parametricCoding = 0; - } - } - else { - if (bitRate < 20000) { - config->parametricCoding = 0; - } - } - - config->useSpeechConfig = useSpeechConfig; - - /* PS settings */ - config->bParametricStereo = bParametricStereo; - - return 1 ; - } - else { - return 0 ; - } -} - -/***************************************************************************** - - functionname: FDKsbrEnc_InitializeSbrDefaults - description: initializes the SBR confifuration - returns: error status - input: - core codec type, - - factor of SBR to core frame length, - - core frame length - output: initialized SBR configuration - -*****************************************************************************/ -static UINT -FDKsbrEnc_InitializeSbrDefaults (sbrConfigurationPtr config, - INT downSampleFactor, - UINT codecGranuleLen - ,const INT isLowDelay - ) -{ - if ( (downSampleFactor < 1 || downSampleFactor > 2) || - (codecGranuleLen*downSampleFactor > QMF_CHANNELS*QMF_MAX_TIME_SLOTS) ) - return(0); /* error */ - - config->SendHeaderDataTime = 1000; - config->useWaveCoding = 0; - config->crcSbr = 0; - config->dynBwSupported = 1; - if (isLowDelay) - config->tran_thr = 6000; - else - config->tran_thr = 13000; - - config->parametricCoding = 1; - - config->sbrFrameSize = codecGranuleLen * downSampleFactor; - config->downSampleFactor = downSampleFactor; - - /* sbr default parameters */ - config->sbr_data_extra = 0; - config->amp_res = SBR_AMP_RES_3_0 ; - config->tran_fc = 0 ; - config->tran_det_mode = 1 ; - config->spread = 1 ; - config->stat = 0 ; - config->e = 1 ; - config->deltaTAcrossFrames = 1 ; - config->dF_edge_1stEnv = FL2FXCONST_DBL(0.3f) ; - config->dF_edge_incr = FL2FXCONST_DBL(0.3f) ; - - config->sbr_invf_mode = INVF_SWITCHED; - config->sbr_xpos_mode = XPOS_LC; - config->sbr_xpos_ctrl = SBR_XPOS_CTRL_DEFAULT; - config->sbr_xpos_level = 0; - config->useSaPan = 0; - config->dynBwEnabled = 0; - - - /* the following parameters are overwritten by the FDKsbrEnc_AdjustSbrSettings() function since - they are included in the tuning table */ - config->stereoMode = SBR_SWITCH_LRC; - config->ana_max_level = 6; - config->noiseFloorOffset = 0; - config->startFreq = 5; /* 5.9 respectively 6.0 kHz at fs = 44.1/48 kHz */ - config->stopFreq = 9; /* 16.2 respectively 16.8 kHz at fs = 44.1/48 kHz */ - config->freq_res_fixfix[0] = FREQ_RES_HIGH; /* non-split case */ - config->freq_res_fixfix[1] = FREQ_RES_HIGH; /* split case */ - config->fResTransIsLow = 0; /* for transient frames, set variable frequency resolution according to freqResTable */ - - /* header_extra_1 */ - config->freqScale = SBR_FREQ_SCALE_DEFAULT; - config->alterScale = SBR_ALTER_SCALE_DEFAULT; - config->sbr_noise_bands = SBR_NOISE_BANDS_DEFAULT; - - /* header_extra_2 */ - config->sbr_limiter_bands = SBR_LIMITER_BANDS_DEFAULT; - config->sbr_limiter_gains = SBR_LIMITER_GAINS_DEFAULT; - config->sbr_interpol_freq = SBR_INTERPOL_FREQ_DEFAULT; - config->sbr_smoothing_length = SBR_SMOOTHING_LENGTH_DEFAULT; - - return 1; -} - - -/***************************************************************************** - - functionname: DeleteEnvChannel - description: frees memory of one SBR channel - returns: - - input: handle of channel - output: released handle - -*****************************************************************************/ -static void -deleteEnvChannel (HANDLE_ENV_CHANNEL hEnvCut) -{ - if (hEnvCut) { - - FDKsbrEnc_DeleteTonCorrParamExtr(&hEnvCut->TonCorr); - - FDKsbrEnc_deleteExtractSbrEnvelope (&hEnvCut->sbrExtractEnvelope); - } - -} - - -/***************************************************************************** - - functionname: sbrEncoder_ChannelClose - description: close the channel coding handle - returns: - input: phSbrChannel - output: - -*****************************************************************************/ -static void -sbrEncoder_ChannelClose(HANDLE_SBR_CHANNEL hSbrChannel) -{ - if (hSbrChannel != NULL) - { - deleteEnvChannel (&hSbrChannel->hEnvChannel); - } -} - -/***************************************************************************** - - functionname: sbrEncoder_ElementClose - description: close the channel coding handle - returns: - input: phSbrChannel - output: - -*****************************************************************************/ -static void -sbrEncoder_ElementClose(HANDLE_SBR_ELEMENT *phSbrElement) -{ - HANDLE_SBR_ELEMENT hSbrElement = *phSbrElement; - - if (hSbrElement!=NULL) { - if (hSbrElement->sbrConfigData.v_k_master) - FreeRam_Sbr_v_k_master(&hSbrElement->sbrConfigData.v_k_master); - if (hSbrElement->sbrConfigData.freqBandTable[LO]) - FreeRam_Sbr_freqBandTableLO(&hSbrElement->sbrConfigData.freqBandTable[LO]); - if (hSbrElement->sbrConfigData.freqBandTable[HI]) - FreeRam_Sbr_freqBandTableHI(&hSbrElement->sbrConfigData.freqBandTable[HI]); - - FreeRam_SbrElement(phSbrElement); - } - return ; - -} - - -void sbrEncoder_Close (HANDLE_SBR_ENCODER *phSbrEncoder) -{ - HANDLE_SBR_ENCODER hSbrEncoder = *phSbrEncoder; - - if (hSbrEncoder != NULL) - { - int el, ch; - - for (el=0; el<(8); el++) - { - if (hSbrEncoder->sbrElement[el]!=NULL) { - sbrEncoder_ElementClose(&hSbrEncoder->sbrElement[el]); - } - } - - /* Close sbr Channels */ - for (ch=0; ch<(8); ch++) - { - if (hSbrEncoder->pSbrChannel[ch]) { - sbrEncoder_ChannelClose(hSbrEncoder->pSbrChannel[ch]); - FreeRam_SbrChannel(&hSbrEncoder->pSbrChannel[ch]); - } - - if (hSbrEncoder->QmfAnalysis[ch].FilterStates) - FreeRam_Sbr_QmfStatesAnalysis((FIXP_QAS**)&hSbrEncoder->QmfAnalysis[ch].FilterStates); - - - } - - if (hSbrEncoder->hParametricStereo) - PSEnc_Destroy(&hSbrEncoder->hParametricStereo); - if (hSbrEncoder->qmfSynthesisPS.FilterStates) - FreeRam_PsQmfStatesSynthesis((FIXP_DBL**)&hSbrEncoder->qmfSynthesisPS.FilterStates); - - /* Release Overlay */ - FreeRam_SbrDynamic_RAM((FIXP_DBL**)&hSbrEncoder->pSBRdynamic_RAM); - - - FreeRam_SbrEncoder(phSbrEncoder); - } - -} - -/***************************************************************************** - - functionname: updateFreqBandTable - description: updates vk_master - returns: - - input: config handle - output: error info - -*****************************************************************************/ -static INT updateFreqBandTable( - HANDLE_SBR_CONFIG_DATA sbrConfigData, - HANDLE_SBR_HEADER_DATA sbrHeaderData, - const INT downSampleFactor - ) -{ - INT k0, k2; - - if( FDKsbrEnc_FindStartAndStopBand ( - sbrConfigData->sampleFreq, - sbrConfigData->sampleFreq >> (downSampleFactor-1), - sbrConfigData->noQmfBands, - sbrHeaderData->sbr_start_frequency, - sbrHeaderData->sbr_stop_frequency, - &k0, - &k2 - ) - ) - return(1); - - - if( FDKsbrEnc_UpdateFreqScale( - sbrConfigData->v_k_master, - &sbrConfigData->num_Master, - k0, - k2, - sbrHeaderData->freqScale, - sbrHeaderData->alterScale - ) - ) - return(1); - - - sbrHeaderData->sbr_xover_band=0; - - - if( FDKsbrEnc_UpdateHiRes( - sbrConfigData->freqBandTable[HI], - &sbrConfigData->nSfb[HI], - sbrConfigData->v_k_master, - sbrConfigData->num_Master, - &sbrHeaderData->sbr_xover_band - ) - ) - return(1); - - - FDKsbrEnc_UpdateLoRes( - sbrConfigData->freqBandTable[LO], - &sbrConfigData->nSfb[LO], - sbrConfigData->freqBandTable[HI], - sbrConfigData->nSfb[HI] - ); - - - sbrConfigData->xOverFreq = (sbrConfigData->freqBandTable[LOW_RES][0] * sbrConfigData->sampleFreq / sbrConfigData->noQmfBands+1)>>1; - - return (0); -} - - -/***************************************************************************** - - functionname: resetEnvChannel - description: resets parameters and allocates memory - returns: error status - input: - output: hEnv - -*****************************************************************************/ -static INT resetEnvChannel (HANDLE_SBR_CONFIG_DATA sbrConfigData, - HANDLE_SBR_HEADER_DATA sbrHeaderData, - HANDLE_ENV_CHANNEL hEnv) -{ - /* note !!! hEnv->encEnvData.noOfnoisebands will be updated later in function FDKsbrEnc_extractSbrEnvelope !!!*/ - hEnv->TonCorr.sbrNoiseFloorEstimate.noiseBands = sbrHeaderData->sbr_noise_bands; - - - if(FDKsbrEnc_ResetTonCorrParamExtr(&hEnv->TonCorr, - sbrConfigData->xposCtrlSwitch, - sbrConfigData->freqBandTable[HI][0], - sbrConfigData->v_k_master, - sbrConfigData->num_Master, - sbrConfigData->sampleFreq, - sbrConfigData->freqBandTable, - sbrConfigData->nSfb, - sbrConfigData->noQmfBands)) - return(1); - - hEnv->sbrCodeNoiseFloor.nSfb[LO] = hEnv->TonCorr.sbrNoiseFloorEstimate.noNoiseBands; - hEnv->sbrCodeNoiseFloor.nSfb[HI] = hEnv->TonCorr.sbrNoiseFloorEstimate.noNoiseBands; - - hEnv->sbrCodeEnvelope.nSfb[LO] = sbrConfigData->nSfb[LO]; - hEnv->sbrCodeEnvelope.nSfb[HI] = sbrConfigData->nSfb[HI]; - - hEnv->encEnvData.noHarmonics = sbrConfigData->nSfb[HI]; - - hEnv->sbrCodeEnvelope.upDate = 0; - hEnv->sbrCodeNoiseFloor.upDate = 0; - - return (0); -} - -/* ****************************** FDKsbrEnc_SbrGetXOverFreq ******************************/ -/** - * @fn - * @brief calculates the closest possible crossover frequency - * @return the crossover frequency SBR accepts - * - */ -static INT -FDKsbrEnc_SbrGetXOverFreq(HANDLE_SBR_ELEMENT hEnv, /*!< handle to SBR encoder instance */ - INT xoverFreq) /*!< from core coder suggested crossover frequency */ -{ - INT band; - INT lastDiff, newDiff; - INT cutoffSb; - - UCHAR *RESTRICT pVKMaster = hEnv->sbrConfigData.v_k_master; - - /* Check if there is a matching cutoff frequency in the master table */ - cutoffSb = (4*xoverFreq * hEnv->sbrConfigData.noQmfBands / hEnv->sbrConfigData.sampleFreq + 1)>>1; - lastDiff = cutoffSb; - for (band = 0; band < hEnv->sbrConfigData.num_Master; band++) { - - newDiff = fixp_abs((INT)pVKMaster[band] - cutoffSb); - - if(newDiff >= lastDiff) { - band--; - break; - } - - lastDiff = newDiff; - } - - return ((pVKMaster[band] * hEnv->sbrConfigData.sampleFreq/hEnv->sbrConfigData.noQmfBands+1)>>1); -} - -/***************************************************************************** - - functionname: FDKsbrEnc_EnvEncodeFrame - description: performs the sbr envelope calculation for one element - returns: - input: - output: - -*****************************************************************************/ -INT -FDKsbrEnc_EnvEncodeFrame(HANDLE_SBR_ENCODER hEnvEncoder, - int iElement, - INT_PCM *samples, /*!< time samples, always interleaved */ - UINT timeInStride, /*!< time buffer channel interleaving stride */ - UINT *sbrDataBits, /*!< Size of SBR payload */ - UCHAR *sbrData, /*!< SBR payload */ - int clearOutput /*!< Do not consider any input signal */ - ) -{ - HANDLE_SBR_ELEMENT hSbrElement = NULL; - FDK_CRCINFO crcInfo; - INT crcReg; - INT ch; - INT band; - INT cutoffSb; - INT newXOver; - - if (hEnvEncoder == NULL) - return -1; - - hSbrElement = hEnvEncoder->sbrElement[iElement]; - - if (hSbrElement == NULL) - return -1; - - - /* header bitstream handling */ - HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData = &hSbrElement->sbrBitstreamData; - - INT psHeaderActive = 0; - sbrBitstreamData->HeaderActive = 0; - - /* Anticipate PS header because of internal PS bitstream delay in order to be in sync with SBR header. */ - if ( sbrBitstreamData->CountSendHeaderData==(sbrBitstreamData->NrSendHeaderData-1) ) - { - psHeaderActive = 1; - } - - /* Signal SBR header to be written into bitstream */ - if ( sbrBitstreamData->CountSendHeaderData==0 ) - { - sbrBitstreamData->HeaderActive = 1; - } - - /* Increment header interval counter */ - if (sbrBitstreamData->NrSendHeaderData == 0) { - sbrBitstreamData->CountSendHeaderData = 1; - } - else { - if (sbrBitstreamData->CountSendHeaderData >= 0) { - sbrBitstreamData->CountSendHeaderData++; - sbrBitstreamData->CountSendHeaderData %= sbrBitstreamData->NrSendHeaderData; - } - } - - if (hSbrElement->CmonData.dynBwEnabled ) { - INT i; - for ( i = 4; i > 0; i-- ) - hSbrElement->dynXOverFreqDelay[i] = hSbrElement->dynXOverFreqDelay[i-1]; - - hSbrElement->dynXOverFreqDelay[0] = hSbrElement->CmonData.dynXOverFreqEnc; - if (hSbrElement->dynXOverFreqDelay[1] > hSbrElement->dynXOverFreqDelay[2]) - newXOver = hSbrElement->dynXOverFreqDelay[2]; - else - newXOver = hSbrElement->dynXOverFreqDelay[1]; - - /* has the crossover frequency changed? */ - if ( hSbrElement->sbrConfigData.dynXOverFreq != newXOver ) { - - /* get corresponding master band */ - cutoffSb = ((4* newXOver * hSbrElement->sbrConfigData.noQmfBands - / hSbrElement->sbrConfigData.sampleFreq)+1)>>1; - - for ( band = 0; band < hSbrElement->sbrConfigData.num_Master; band++ ) { - if ( cutoffSb == hSbrElement->sbrConfigData.v_k_master[band] ) - break; - } - FDK_ASSERT( band < hSbrElement->sbrConfigData.num_Master ); - - hSbrElement->sbrConfigData.dynXOverFreq = newXOver; - hSbrElement->sbrHeaderData.sbr_xover_band = band; - hSbrElement->sbrBitstreamData.HeaderActive=1; - psHeaderActive = 1; /* ps header is one frame delayed */ - - /* - update vk_master table - */ - if(updateFreqBandTable(&hSbrElement->sbrConfigData, - &hSbrElement->sbrHeaderData, - hEnvEncoder->downSampleFactor - )) - return(1); - - - /* reset SBR channels */ - INT nEnvCh = hSbrElement->sbrConfigData.nChannels; - for ( ch = 0; ch < nEnvCh; ch++ ) { - if(resetEnvChannel (&hSbrElement->sbrConfigData, - &hSbrElement->sbrHeaderData, - &hSbrElement->sbrChannel[ch]->hEnvChannel)) - return(1); - - } - } - } - - /* - allocate space for dummy header and crc - */ - crcReg = FDKsbrEnc_InitSbrBitstream(&hSbrElement->CmonData, - hSbrElement->payloadDelayLine[hEnvEncoder->nBitstrDelay], - MAX_PAYLOAD_SIZE*sizeof(UCHAR), - &crcInfo, - hSbrElement->sbrConfigData.sbrSyntaxFlags); - - /* Temporal Envelope Data */ - SBR_FRAME_TEMP_DATA _fData; - SBR_FRAME_TEMP_DATA *fData = &_fData; - SBR_ENV_TEMP_DATA eData[MAX_NUM_CHANNELS]; - - /* Init Temporal Envelope Data */ - { - int i; - - FDKmemclear(&eData[0], sizeof(SBR_ENV_TEMP_DATA)); - FDKmemclear(&eData[1], sizeof(SBR_ENV_TEMP_DATA)); - FDKmemclear(fData, sizeof(SBR_FRAME_TEMP_DATA)); - - for(i=0; ires[i] = FREQ_RES_HIGH; - } - - - if (!clearOutput) - { - /* - * Transform audio data into QMF domain - */ - for(ch = 0; ch < hSbrElement->sbrConfigData.nChannels; ch++) - { - HANDLE_ENV_CHANNEL h_envChan = &hSbrElement->sbrChannel[ch]->hEnvChannel; - HANDLE_SBR_EXTRACT_ENVELOPE sbrExtrEnv = &h_envChan->sbrExtractEnvelope; - - if(hSbrElement->elInfo.fParametricStereo == 0) - { - QMF_SCALE_FACTOR tmpScale; - FIXP_DBL **pQmfReal, **pQmfImag; - C_AALLOC_SCRATCH_START(qmfWorkBuffer, FIXP_DBL, QMF_CHANNELS*2) - - - /* Obtain pointers to QMF buffers. */ - pQmfReal = sbrExtrEnv->rBuffer; - pQmfImag = sbrExtrEnv->iBuffer; - - qmfAnalysisFiltering( hSbrElement->hQmfAnalysis[ch], - pQmfReal, - pQmfImag, - &tmpScale, - samples + hSbrElement->elInfo.ChannelIndex[ch], - timeInStride, - qmfWorkBuffer ); - - h_envChan->qmfScale = tmpScale.lb_scale + 7; - - - C_AALLOC_SCRATCH_END(qmfWorkBuffer, FIXP_DBL, QMF_CHANNELS*2) - - } /* fParametricStereo == 0 */ - - - /* - Parametric Stereo processing - */ - if (hSbrElement->elInfo.fParametricStereo) - { - INT error = noError; - - - /* Limit Parametric Stereo to one instance */ - FDK_ASSERT(ch == 0); - - - if(error == noError){ - /* parametric stereo processing: - - input: - o left and right time domain samples - - processing: - o stereo qmf analysis - o stereo hybrid analysis - o ps parameter extraction - o downmix + hybrid synthesis - - output: - o downmixed qmf data is written to sbrExtrEnv->rBuffer and sbrExtrEnv->iBuffer - */ - SCHAR qmfScale; - INT_PCM* pSamples[2] = {samples + hSbrElement->elInfo.ChannelIndex[0],samples + hSbrElement->elInfo.ChannelIndex[1]}; - error = FDKsbrEnc_PSEnc_ParametricStereoProcessing( hEnvEncoder->hParametricStereo, - pSamples, - timeInStride, - hSbrElement->hQmfAnalysis, - sbrExtrEnv->rBuffer, - sbrExtrEnv->iBuffer, - samples + hSbrElement->elInfo.ChannelIndex[ch], - &hEnvEncoder->qmfSynthesisPS, - &qmfScale, - psHeaderActive ); - if (noError != error) - { - error = handBack(error); - } - h_envChan->qmfScale = (int)qmfScale; - } - - - } /* if (hEnvEncoder->hParametricStereo) */ - - /* - - Extract Envelope relevant things from QMF data - - */ - FDKsbrEnc_extractSbrEnvelope1( - &hSbrElement->sbrConfigData, - &hSbrElement->sbrHeaderData, - &hSbrElement->sbrBitstreamData, - h_envChan, - &hSbrElement->CmonData, - &eData[ch], - fData - ); - - } /* hEnvEncoder->sbrConfigData.nChannels */ - } - - /* - Process Envelope relevant things and calculate envelope data and write payload - */ - FDKsbrEnc_extractSbrEnvelope2( - &hSbrElement->sbrConfigData, - &hSbrElement->sbrHeaderData, - (hSbrElement->elInfo.fParametricStereo) ? hEnvEncoder->hParametricStereo : NULL, - &hSbrElement->sbrBitstreamData, - &hSbrElement->sbrChannel[0]->hEnvChannel, - &hSbrElement->sbrChannel[1]->hEnvChannel, - &hSbrElement->CmonData, - eData, - fData, - clearOutput - ); - - /* - format payload, calculate crc - */ - FDKsbrEnc_AssembleSbrBitstream(&hSbrElement->CmonData, &crcInfo, crcReg, hSbrElement->sbrConfigData.sbrSyntaxFlags); - - /* - save new payload, set to zero length if greater than MAX_PAYLOAD_SIZE - */ - hSbrElement->payloadDelayLineSize[hEnvEncoder->nBitstrDelay] = FDKgetValidBits(&hSbrElement->CmonData.sbrBitbuf); - - if(hSbrElement->payloadDelayLineSize[hEnvEncoder->nBitstrDelay] > (MAX_PAYLOAD_SIZE<<3)) - hSbrElement->payloadDelayLineSize[hEnvEncoder->nBitstrDelay]=0; - - /* While filling the Delay lines, sbrData is NULL */ - if (sbrData) { - *sbrDataBits = hSbrElement->payloadDelayLineSize[0]; - FDKmemcpy(sbrData, hSbrElement->payloadDelayLine[0], (hSbrElement->payloadDelayLineSize[0]+7)>>3); - - - } - - -/*******************************/ - - if (hEnvEncoder->fTimeDomainDownsampling) - { - int ch; - int nChannels = hSbrElement->sbrConfigData.nChannels; - - for (ch=0; ch < nChannels; ch++) - { - INT nOutSamples; - - FDKaacEnc_Downsample(&hSbrElement->sbrChannel[ch]->downSampler, - samples + hSbrElement->elInfo.ChannelIndex[ch] + hEnvEncoder->bufferOffset, - hSbrElement->sbrConfigData.frameSize, - timeInStride, - samples + hSbrElement->elInfo.ChannelIndex[ch], - &nOutSamples, - hEnvEncoder->nChannels); - } - } /* downsample */ - - - return (0); -} - -/***************************************************************************** - - functionname: createEnvChannel - description: initializes parameters and allocates memory - returns: error status - input: - output: hEnv - -*****************************************************************************/ - -static INT -createEnvChannel (HANDLE_ENV_CHANNEL hEnv, - INT channel - ,UCHAR* dynamic_RAM - ) -{ - FDKmemclear(hEnv,sizeof (struct ENV_CHANNEL)); - - if ( FDKsbrEnc_CreateTonCorrParamExtr(&hEnv->TonCorr, - channel) ) - { - return(1); - } - - if ( FDKsbrEnc_CreateExtractSbrEnvelope (&hEnv->sbrExtractEnvelope, - channel - ,/*chan*/0 - ,dynamic_RAM - ) ) - { - return(1); - } - - return 0; -} - -/***************************************************************************** - - functionname: initEnvChannel - description: initializes parameters - returns: error status - input: - output: - -*****************************************************************************/ -static INT -initEnvChannel ( HANDLE_SBR_CONFIG_DATA sbrConfigData, - HANDLE_SBR_HEADER_DATA sbrHeaderData, - HANDLE_ENV_CHANNEL hEnv, - sbrConfigurationPtr params, - ULONG statesInitFlag - ,INT chanInEl - ,UCHAR* dynamic_RAM - ) -{ - int frameShift, tran_off=0; - INT e; - INT tran_fc; - INT timeSlots, timeStep, startIndex; - INT noiseBands[2] = { 3, 3 }; - - e = 1 << params->e; - - FDK_ASSERT(params->e >= 0); - - hEnv->encEnvData.freq_res_fixfix[0] = params->freq_res_fixfix[0]; - hEnv->encEnvData.freq_res_fixfix[1] = params->freq_res_fixfix[1]; - hEnv->encEnvData.fResTransIsLow = params->fResTransIsLow; - - hEnv->fLevelProtect = 0; - - hEnv->encEnvData.ldGrid = (sbrConfigData->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) ? 1 : 0; - - hEnv->encEnvData.sbr_xpos_mode = (XPOS_MODE)params->sbr_xpos_mode; - - if (hEnv->encEnvData.sbr_xpos_mode == XPOS_SWITCHED) { - /* - no other type than XPOS_MDCT or XPOS_SPEECH allowed, - but enable switching - */ - sbrConfigData->switchTransposers = TRUE; - hEnv->encEnvData.sbr_xpos_mode = XPOS_MDCT; - } - else { - sbrConfigData->switchTransposers = FALSE; - } - - hEnv->encEnvData.sbr_xpos_ctrl = params->sbr_xpos_ctrl; - - - /* extended data */ - if(params->parametricCoding) { - hEnv->encEnvData.extended_data = 1; - } - else { - hEnv->encEnvData.extended_data = 0; - } - - hEnv->encEnvData.extension_size = 0; - - startIndex = QMF_FILTER_PROTOTYPE_SIZE - sbrConfigData->noQmfBands; - - switch (params->sbrFrameSize) { - case 2304: - timeSlots = 18; - break; - case 2048: - case 1024: - case 512: - timeSlots = 16; - break; - case 1920: - case 960: - case 480: - timeSlots = 15; - break; - case 1152: - timeSlots = 9; - break; - default: - return (1); /* Illegal frame size */ - } - - timeStep = sbrConfigData->noQmfSlots / timeSlots; - - if ( FDKsbrEnc_InitTonCorrParamExtr(params->sbrFrameSize, - &hEnv->TonCorr, - sbrConfigData, - timeSlots, - params->sbr_xpos_ctrl, - params->ana_max_level, - sbrHeaderData->sbr_noise_bands, - params->noiseFloorOffset, - params->useSpeechConfig) ) - return(1); - - hEnv->encEnvData.noOfnoisebands = hEnv->TonCorr.sbrNoiseFloorEstimate.noNoiseBands; - - noiseBands[0] = hEnv->encEnvData.noOfnoisebands; - noiseBands[1] = hEnv->encEnvData.noOfnoisebands; - - hEnv->encEnvData.sbr_invf_mode = (INVF_MODE)params->sbr_invf_mode; - - if (hEnv->encEnvData.sbr_invf_mode == INVF_SWITCHED) { - hEnv->encEnvData.sbr_invf_mode = INVF_MID_LEVEL; - hEnv->TonCorr.switchInverseFilt = TRUE; - } - else { - hEnv->TonCorr.switchInverseFilt = FALSE; - } - - - tran_fc = params->tran_fc; - - if (tran_fc == 0) { - tran_fc = fixMin (5000, FDKsbrEnc_getSbrStartFreqRAW (sbrHeaderData->sbr_start_frequency,params->codecSettings.sampleFreq)); - } - - tran_fc = (tran_fc*4*sbrConfigData->noQmfBands/sbrConfigData->sampleFreq + 1)>>1; - - if (sbrConfigData->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) { - frameShift = LD_PRETRAN_OFF; - tran_off = LD_PRETRAN_OFF + FRAME_MIDDLE_SLOT_512LD*timeStep; - } else - { - frameShift = 0; - switch (timeSlots) { - /* The factor of 2 is by definition. */ - case NUMBER_TIME_SLOTS_2048: tran_off = 8 + FRAME_MIDDLE_SLOT_2048 * timeStep; break; - case NUMBER_TIME_SLOTS_1920: tran_off = 7 + FRAME_MIDDLE_SLOT_1920 * timeStep; break; - default: return 1; - } - } - if ( FDKsbrEnc_InitExtractSbrEnvelope (&hEnv->sbrExtractEnvelope, - sbrConfigData->noQmfSlots, - sbrConfigData->noQmfBands, startIndex, - timeSlots, timeStep, tran_off, - statesInitFlag - ,chanInEl - ,dynamic_RAM - ,sbrConfigData->sbrSyntaxFlags - ) ) - return(1); - - if(FDKsbrEnc_InitSbrCodeEnvelope (&hEnv->sbrCodeEnvelope, - sbrConfigData->nSfb, - params->deltaTAcrossFrames, - params->dF_edge_1stEnv, - params->dF_edge_incr)) - return(1); - - if(FDKsbrEnc_InitSbrCodeEnvelope (&hEnv->sbrCodeNoiseFloor, - noiseBands, - params->deltaTAcrossFrames, - 0,0)) - return(1); - - sbrConfigData->initAmpResFF = params->init_amp_res_FF; - - if(FDKsbrEnc_InitSbrHuffmanTables (&hEnv->encEnvData, - &hEnv->sbrCodeEnvelope, - &hEnv->sbrCodeNoiseFloor, - sbrHeaderData->sbr_amp_res)) - return(1); - - FDKsbrEnc_initFrameInfoGenerator (&hEnv->SbrEnvFrame, - params->spread, - e, - params->stat, - timeSlots, - hEnv->encEnvData.freq_res_fixfix, - hEnv->encEnvData.fResTransIsLow, - hEnv->encEnvData.ldGrid - ); - - if(sbrConfigData->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) - { - INT bandwidth_qmf_slot = (sbrConfigData->sampleFreq>>1) / (sbrConfigData->noQmfBands); - if(FDKsbrEnc_InitSbrFastTransientDetector( - &hEnv->sbrFastTransientDetector, - sbrConfigData->noQmfSlots, - bandwidth_qmf_slot, - sbrConfigData->noQmfBands, - sbrConfigData->freqBandTable[0][0] - )) - return(1); - } - - /* The transient detector has to be initialized also if the fast transient - detector was active, because the values from the transient detector - structure are used. */ - if(FDKsbrEnc_InitSbrTransientDetector (&hEnv->sbrTransientDetector, - sbrConfigData->sbrSyntaxFlags, - sbrConfigData->frameSize, - sbrConfigData->sampleFreq, - params, - tran_fc, - sbrConfigData->noQmfSlots, - sbrConfigData->noQmfBands, - hEnv->sbrExtractEnvelope.YBufferWriteOffset, - hEnv->sbrExtractEnvelope.YBufferSzShift, - frameShift, - tran_off - )) - return(1); - - - sbrConfigData->xposCtrlSwitch = params->sbr_xpos_ctrl; - - hEnv->encEnvData.noHarmonics = sbrConfigData->nSfb[HI]; - hEnv->encEnvData.addHarmonicFlag = 0; - - return (0); -} - -INT sbrEncoder_Open( - HANDLE_SBR_ENCODER *phSbrEncoder, - INT nElements, - INT nChannels, - INT supportPS - ) -{ - INT i; - INT errorStatus = 1; - HANDLE_SBR_ENCODER hSbrEncoder = NULL; - - if (phSbrEncoder==NULL - ) - { - goto bail; - } - - hSbrEncoder = GetRam_SbrEncoder(); - if (hSbrEncoder==NULL) { - goto bail; - } - FDKmemclear(hSbrEncoder, sizeof(SBR_ENCODER)); - - hSbrEncoder->pSBRdynamic_RAM = (UCHAR*)GetRam_SbrDynamic_RAM(); - hSbrEncoder->dynamicRam = hSbrEncoder->pSBRdynamic_RAM; - - for (i=0; isbrElement[i] = GetRam_SbrElement(i); - if (hSbrEncoder->sbrElement[i]==NULL) { - goto bail; - } - FDKmemclear(hSbrEncoder->sbrElement[i], sizeof(SBR_ELEMENT)); - hSbrEncoder->sbrElement[i]->sbrConfigData.freqBandTable[LO] = GetRam_Sbr_freqBandTableLO(i); - hSbrEncoder->sbrElement[i]->sbrConfigData.freqBandTable[HI] = GetRam_Sbr_freqBandTableHI(i); - hSbrEncoder->sbrElement[i]->sbrConfigData.v_k_master = GetRam_Sbr_v_k_master(i); - if ( (hSbrEncoder->sbrElement[i]->sbrConfigData.freqBandTable[LO]==NULL) || - (hSbrEncoder->sbrElement[i]->sbrConfigData.freqBandTable[HI]==NULL) || - (hSbrEncoder->sbrElement[i]->sbrConfigData.v_k_master==NULL) ) - { - goto bail; - } - } - - for (i=0; ipSbrChannel[i] = GetRam_SbrChannel(i); - if (hSbrEncoder->pSbrChannel[i]==NULL) { - goto bail; - } - - if ( createEnvChannel(&hSbrEncoder->pSbrChannel[i]->hEnvChannel, - i - ,hSbrEncoder->dynamicRam - ) ) - { - goto bail; - } - - } - - for (i=0; iQmfAnalysis[i].FilterStates = GetRam_Sbr_QmfStatesAnalysis(i); - if (hSbrEncoder->QmfAnalysis[i].FilterStates==NULL) { - goto bail; - } - } - - if (supportPS) { - if (PSEnc_Create(&hSbrEncoder->hParametricStereo)) - { - goto bail; - } - - hSbrEncoder->qmfSynthesisPS.FilterStates = GetRam_PsQmfStatesSynthesis(); - if (hSbrEncoder->qmfSynthesisPS.FilterStates==NULL) { - goto bail; - } - } /* supportPS */ - - *phSbrEncoder = hSbrEncoder; - - errorStatus = 0; - return errorStatus; - -bail: - /* Close SBR encoder instance */ - sbrEncoder_Close(&hSbrEncoder); - return errorStatus; -} - -static -INT FDKsbrEnc_Reallocate( - HANDLE_SBR_ENCODER hSbrEncoder, - SBR_ELEMENT_INFO elInfo[(8)], - const INT noElements) -{ - INT totalCh = 0; - INT totalQmf = 0; - INT coreEl; - INT el=-1; - - hSbrEncoder->lfeChIdx = -1; /* default value, until lfe found */ - - for (coreEl=0; coreEllfeChIdx = elInfo[coreEl].ChannelIndex[0]; - } - continue; - } - - SBR_ELEMENT_INFO *pelInfo = &elInfo[coreEl]; - HANDLE_SBR_ELEMENT hSbrElement = hSbrEncoder->sbrElement[el]; - - int ch; - for ( ch = 0; ch < pelInfo->nChannelsInEl; ch++ ) { - hSbrElement->sbrChannel[ch] = hSbrEncoder->pSbrChannel[totalCh]; - totalCh++; - } - /* analysis QMF */ - for ( ch = 0; ch < ((pelInfo->fParametricStereo)?2:pelInfo->nChannelsInEl); ch++ ) { - hSbrElement->elInfo.ChannelIndex[ch] = pelInfo->ChannelIndex[ch]; - hSbrElement->hQmfAnalysis[ch] = &hSbrEncoder->QmfAnalysis[totalQmf++]; - } - - /* Copy Element info */ - hSbrElement->elInfo.elType = pelInfo->elType; - hSbrElement->elInfo.instanceTag = pelInfo->instanceTag; - hSbrElement->elInfo.nChannelsInEl = pelInfo->nChannelsInEl; - hSbrElement->elInfo.fParametricStereo = pelInfo->fParametricStereo; - } /* coreEl */ - - return 0; -} - - - -/***************************************************************************** - - functionname: FDKsbrEnc_EnvInit - description: initializes parameters - returns: error status - input: - output: hEnv - -*****************************************************************************/ -static -INT FDKsbrEnc_EnvInit ( - HANDLE_SBR_ELEMENT hSbrElement, - sbrConfigurationPtr params, - INT *coreBandWith, - AUDIO_OBJECT_TYPE aot, - int nBitstrDelay, - int nElement, - const int headerPeriod, - ULONG statesInitFlag, - int fTimeDomainDownsampling - ,UCHAR *dynamic_RAM - ) -{ - UCHAR *bitstreamBuffer; - int ch, i; - - if ((params->codecSettings.nChannels < 1) || (params->codecSettings.nChannels > MAX_NUM_CHANNELS)){ - return(1); - } - - /* initialize the encoder handle and structs*/ - bitstreamBuffer = hSbrElement->payloadDelayLine[nBitstrDelay]; - - /* init and set syntax flags */ - hSbrElement->sbrConfigData.sbrSyntaxFlags = 0; - - switch (aot) { - case AOT_ER_AAC_ELD: - hSbrElement->sbrConfigData.sbrSyntaxFlags |= SBR_SYNTAX_LOW_DELAY; - break; - default: - break; - } - if (params->crcSbr) { - hSbrElement->sbrConfigData.sbrSyntaxFlags |= SBR_SYNTAX_CRC; - } - - hSbrElement->sbrConfigData.noQmfBands = QMF_CHANNELS>>(2-params->downSampleFactor); - switch (hSbrElement->sbrConfigData.noQmfBands) - { - case 64: hSbrElement->sbrConfigData.noQmfSlots = params->sbrFrameSize>>6; - break; - case 32: hSbrElement->sbrConfigData.noQmfSlots = params->sbrFrameSize>>5; - break; - default: hSbrElement->sbrConfigData.noQmfSlots = params->sbrFrameSize>>6; - return(2); - } - - FDKinitBitStream(&hSbrElement->CmonData.sbrBitbuf, bitstreamBuffer, MAX_PAYLOAD_SIZE*sizeof(UCHAR), 0, BS_WRITER); - - /* - now initialize sbrConfigData, sbrHeaderData and sbrBitstreamData, - */ - hSbrElement->sbrConfigData.nChannels = params->codecSettings.nChannels; - - if(params->codecSettings.nChannels == 2) - hSbrElement->sbrConfigData.stereoMode = params->stereoMode; - else - hSbrElement->sbrConfigData.stereoMode = SBR_MONO; - - hSbrElement->sbrConfigData.frameSize = params->sbrFrameSize; - - hSbrElement->sbrConfigData.sampleFreq = params->downSampleFactor * params->codecSettings.sampleFreq; - - hSbrElement->sbrBitstreamData.CountSendHeaderData = 0; - if (params->SendHeaderDataTime > 0 ) { - - if (headerPeriod==-1) { - - hSbrElement->sbrBitstreamData.NrSendHeaderData = (INT)(params->SendHeaderDataTime * hSbrElement->sbrConfigData.sampleFreq - / (1000 * hSbrElement->sbrConfigData.frameSize)); - hSbrElement->sbrBitstreamData.NrSendHeaderData = fixMax(hSbrElement->sbrBitstreamData.NrSendHeaderData,1); - } - else { - /* assure header period at least once per second */ - hSbrElement->sbrBitstreamData.NrSendHeaderData = fixMin(fixMax(headerPeriod,1),(hSbrElement->sbrConfigData.sampleFreq/hSbrElement->sbrConfigData.frameSize)); - } - } - else { - hSbrElement->sbrBitstreamData.NrSendHeaderData = 0; - } - - hSbrElement->sbrHeaderData.sbr_data_extra = params->sbr_data_extra; - hSbrElement->sbrBitstreamData.HeaderActive = 0; - hSbrElement->sbrHeaderData.sbr_start_frequency = params->startFreq; - hSbrElement->sbrHeaderData.sbr_stop_frequency = params->stopFreq; - hSbrElement->sbrHeaderData.sbr_xover_band = 0; - hSbrElement->sbrHeaderData.sbr_lc_stereo_mode = 0; - - /* data_extra */ - if (params->sbr_xpos_ctrl!= SBR_XPOS_CTRL_DEFAULT) - hSbrElement->sbrHeaderData.sbr_data_extra = 1; - - hSbrElement->sbrHeaderData.sbr_amp_res = (AMP_RES)params->amp_res; - - /* header_extra_1 */ - hSbrElement->sbrHeaderData.freqScale = params->freqScale; - hSbrElement->sbrHeaderData.alterScale = params->alterScale; - hSbrElement->sbrHeaderData.sbr_noise_bands = params->sbr_noise_bands; - hSbrElement->sbrHeaderData.header_extra_1 = 0; - - if ((params->freqScale != SBR_FREQ_SCALE_DEFAULT) || - (params->alterScale != SBR_ALTER_SCALE_DEFAULT) || - (params->sbr_noise_bands != SBR_NOISE_BANDS_DEFAULT)) - { - hSbrElement->sbrHeaderData.header_extra_1 = 1; - } - - /* header_extra_2 */ - hSbrElement->sbrHeaderData.sbr_limiter_bands = params->sbr_limiter_bands; - hSbrElement->sbrHeaderData.sbr_limiter_gains = params->sbr_limiter_gains; - - if ((hSbrElement->sbrConfigData.sampleFreq > 48000) && - (hSbrElement->sbrHeaderData.sbr_start_frequency >= 9)) - { - hSbrElement->sbrHeaderData.sbr_limiter_gains = SBR_LIMITER_GAINS_INFINITE; - } - - hSbrElement->sbrHeaderData.sbr_interpol_freq = params->sbr_interpol_freq; - hSbrElement->sbrHeaderData.sbr_smoothing_length = params->sbr_smoothing_length; - hSbrElement->sbrHeaderData.header_extra_2 = 0; - - if ((params->sbr_limiter_bands != SBR_LIMITER_BANDS_DEFAULT) || - (params->sbr_limiter_gains != SBR_LIMITER_GAINS_DEFAULT) || - (params->sbr_interpol_freq != SBR_INTERPOL_FREQ_DEFAULT) || - (params->sbr_smoothing_length != SBR_SMOOTHING_LENGTH_DEFAULT)) - { - hSbrElement->sbrHeaderData.header_extra_2 = 1; - } - - /* other switches */ - hSbrElement->sbrConfigData.useWaveCoding = params->useWaveCoding; - hSbrElement->sbrConfigData.useParametricCoding = params->parametricCoding; - hSbrElement->sbrConfigData.thresholdAmpResFF_m = params->threshold_AmpRes_FF_m; - hSbrElement->sbrConfigData.thresholdAmpResFF_e = params->threshold_AmpRes_FF_e; - - /* init freq band table */ - if(updateFreqBandTable(&hSbrElement->sbrConfigData, - &hSbrElement->sbrHeaderData, - params->downSampleFactor - )) - { - return(1); - } - - /* now create envelope ext and QMF for each available channel */ - for ( ch = 0; ch < hSbrElement->sbrConfigData.nChannels; ch++ ) { - - if ( initEnvChannel(&hSbrElement->sbrConfigData, - &hSbrElement->sbrHeaderData, - &hSbrElement->sbrChannel[ch]->hEnvChannel, - params, - statesInitFlag - ,ch - ,dynamic_RAM - ) ) - { - return(1); - } - - - } /* nChannels */ - - /* reset and intialize analysis qmf */ - for ( ch = 0; ch < ((hSbrElement->elInfo.fParametricStereo)?2:hSbrElement->sbrConfigData.nChannels); ch++ ) - { - int err; - UINT qmfFlags = (hSbrElement->sbrConfigData.sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) ? QMF_FLAG_CLDFB : 0; - if (statesInitFlag) - qmfFlags &= ~QMF_FLAG_KEEP_STATES; - else - qmfFlags |= QMF_FLAG_KEEP_STATES; - - err = qmfInitAnalysisFilterBank( hSbrElement->hQmfAnalysis[ch], - (FIXP_QAS*)hSbrElement->hQmfAnalysis[ch]->FilterStates, - hSbrElement->sbrConfigData.noQmfSlots, - hSbrElement->sbrConfigData.noQmfBands, - hSbrElement->sbrConfigData.noQmfBands, - hSbrElement->sbrConfigData.noQmfBands, - qmfFlags ); - if (0!=err) { - return err; - } - } - - /* */ - hSbrElement->CmonData.xOverFreq = hSbrElement->sbrConfigData.xOverFreq; - hSbrElement->CmonData.dynBwEnabled = (params->dynBwSupported && params->dynBwEnabled); - hSbrElement->CmonData.dynXOverFreqEnc = FDKsbrEnc_SbrGetXOverFreq( hSbrElement, hSbrElement->CmonData.xOverFreq); - for ( i = 0; i < 5; i++ ) - hSbrElement->dynXOverFreqDelay[i] = hSbrElement->CmonData.dynXOverFreqEnc; - hSbrElement->CmonData.sbrNumChannels = hSbrElement->sbrConfigData.nChannels; - hSbrElement->sbrConfigData.dynXOverFreq = hSbrElement->CmonData.xOverFreq; - - /* Update Bandwith to be passed to the core encoder */ - *coreBandWith = hSbrElement->CmonData.xOverFreq; - - return(0); - } - -INT sbrEncoder_GetInBufferSize(int noChannels) -{ - INT temp; - - temp = (2048); - temp += 1024 + MAX_SAMPLE_DELAY; - temp *= noChannels; - temp *= sizeof(INT_PCM); - return temp; -} - -/* - * Encode Dummy SBR payload frames to fill the delay lines. - */ -static -INT FDKsbrEnc_DelayCompensation ( - HANDLE_SBR_ENCODER hEnvEnc, - INT_PCM *timeBuffer - ) -{ - int n, el; - - for (n=hEnvEnc->nBitstrDelay; n>0; n--) - { - for (el=0; elnoElements; el++) - { - if (FDKsbrEnc_EnvEncodeFrame( - hEnvEnc, - el, - timeBuffer + hEnvEnc->downsampledOffset, - hEnvEnc->sbrElement[el]->sbrConfigData.nChannels, - NULL, - NULL, - 1 - )) - return -1; - } - sbrEncoder_UpdateBuffers(hEnvEnc, timeBuffer); - } - return 0; -} - -UINT sbrEncoder_LimitBitRate(UINT bitRate, UINT numChannels, UINT coreSampleRate, AUDIO_OBJECT_TYPE aot) -{ - UINT newBitRate; - INT index; - - FDK_ASSERT(numChannels > 0 && numChannels <= 2); - if (aot == AOT_PS) { - if (numChannels == 2) { - index = getPsTuningTableIndex(bitRate, &newBitRate); - if (index == INVALID_TABLE_IDX) { - bitRate = newBitRate; - } - /* Set numChannels to 1 because for PS we need a SBR SCE (mono) element. */ - numChannels = 1; - } else { - return 0; - } - } - index = getSbrTuningTableIndex(bitRate, numChannels, coreSampleRate, aot, &newBitRate); - if (index != INVALID_TABLE_IDX) { - newBitRate = bitRate; - } - - return newBitRate; -} - -UINT sbrEncoder_IsSingleRatePossible(AUDIO_OBJECT_TYPE aot) -{ - UINT isPossible=(AOT_PS==aot)?0:1; - return isPossible; -} - -INT sbrEncoder_Init( - HANDLE_SBR_ENCODER hSbrEncoder, - SBR_ELEMENT_INFO elInfo[(8)], - int noElements, - INT_PCM *inputBuffer, - INT *coreBandwidth, - INT *inputBufferOffset, - INT *numChannels, - INT *coreSampleRate, - UINT *downSampleFactor, - INT *frameLength, - AUDIO_OBJECT_TYPE aot, - int *delay, - int transformFactor, - const int headerPeriod, - ULONG statesInitFlag - ) -{ - HANDLE_ERROR_INFO errorInfo = noError; - sbrConfiguration sbrConfig[(8)]; - INT error = 0; - INT lowestBandwidth; - /* Save input parameters */ - INT inputSampleRate = *coreSampleRate; - int coreFrameLength = *frameLength; - int inputBandWidth = *coreBandwidth; - int inputChannels = *numChannels; - - int downsampledOffset = 0; - int sbrOffset = 0; - int downsamplerDelay = 0; - int timeDomainDownsample = 0; - int nBitstrDelay = 0; - int highestSbrStartFreq, highestSbrStopFreq; - int lowDelay = 0; - int usePs = 0; - - /* check whether SBR setting is available for the current encoder configuration (bitrate, samplerate) */ - if (!sbrEncoder_IsSingleRatePossible(aot)) { - *downSampleFactor = 2; - } - - - - if ( aot==AOT_PS ) { - usePs = 1; - } - if ( aot==AOT_ER_AAC_ELD ) { - lowDelay = 1; - } - else if ( aot==AOT_ER_AAC_LD ) { - error = 1; - goto bail; - } - - /* Parametric Stereo */ - if ( usePs ) { - if ( *numChannels == 2 && noElements == 1) { - /* Override Element type in case of Parametric stereo */ - elInfo[0].elType = ID_SCE; - elInfo[0].fParametricStereo = 1; - elInfo[0].nChannelsInEl = 1; - /* core encoder gets downmixed mono signal */ - *numChannels = 1; - } else { - error = 1; - goto bail; - } - } /* usePs */ - - /* set the core's sample rate */ - switch (*downSampleFactor) { - case 1: - *coreSampleRate = inputSampleRate; - break; - case 2: - *coreSampleRate = inputSampleRate>>1; - break; - default: - *coreSampleRate = inputSampleRate>>1; - return 0; /* return error */ - } - - /* check whether SBR setting is available for the current encoder configuration (bitrate, coreSampleRate) */ - { - int delayDiff = 0; - int el, coreEl; - - /* Check if every element config is feasible */ - for (coreEl=0; coreEl 0) { - /* - * We must tweak the balancing into a situation where the downsampled path - * is the one to be delayed, because delaying the QMF domain input, also delays - * the downsampled audio, counteracting to the purpose of delay balancing. - */ - while ( delayDiff > 0 ) - { - /* Encoder delay increases */ - { - *delay += coreFrameLength * *downSampleFactor; - /* Add one frame delay to SBR path */ - delayDiff -= coreFrameLength * *downSampleFactor; - } - nBitstrDelay += 1; - } - } else - { - *delay += fixp_abs(delayDiff); - } - - if (delayDiff < 0) { - /* Delay AAC data */ - delayDiff = -delayDiff; - /* Multiply downsampled offset by AAC core channels. Divide by 2 because of half samplerate of downsampled data. */ - FDK_ASSERT(*downSampleFactor>0 && *downSampleFactor<=2); - downsampledOffset = (delayDiff*(*numChannels))>>(*downSampleFactor-1); - sbrOffset = 0; - } else { - /* Delay SBR input */ - if ( delayDiff > (int)coreFrameLength * (int)*downSampleFactor ) - { - /* Do bitstream frame-wise delay balancing if we have more than SBR framelength samples delay difference */ - delayDiff -= coreFrameLength * *downSampleFactor; - nBitstrDelay = 1; - } - /* Multiply input offset by input channels */ - sbrOffset = delayDiff*(*numChannels); - downsampledOffset = 0; - } - hSbrEncoder->nBitstrDelay = nBitstrDelay; - hSbrEncoder->nChannels = *numChannels; - hSbrEncoder->frameSize = coreFrameLength * *downSampleFactor; - hSbrEncoder->fTimeDomainDownsampling = timeDomainDownsample; - hSbrEncoder->downSampleFactor = *downSampleFactor; - hSbrEncoder->estimateBitrate = 0; - hSbrEncoder->inputDataDelay = 0; - - - /* Open SBR elements */ - el = -1; - highestSbrStartFreq = highestSbrStopFreq = 0; - lowestBandwidth = 99999; - - /* Loop through each core encoder element and get a matching SBR element config */ - for (coreEl=0; coreElnoElements = el+1; - - FDKsbrEnc_Reallocate(hSbrEncoder, - elInfo, - noElements); - - for (el=0; elnoElements; el++) { - - int bandwidth = *coreBandwidth; - - /* Use lowest common bandwidth */ - sbrConfig[el].startFreq = highestSbrStartFreq; - sbrConfig[el].stopFreq = highestSbrStopFreq; - - /* initialize SBR element, and get core bandwidth */ - error = FDKsbrEnc_EnvInit(hSbrEncoder->sbrElement[el], - &sbrConfig[el], - &bandwidth, - aot, - nBitstrDelay, - el, - headerPeriod, - statesInitFlag, - hSbrEncoder->fTimeDomainDownsampling - ,hSbrEncoder->dynamicRam - ); - - if (error != 0) { - error = 2; - goto bail; - } - - /* Get lowest core encoder bandwidth to be returned later. */ - lowestBandwidth = fixMin(lowestBandwidth, bandwidth); - - } /* second element loop */ - - /* Initialize a downsampler for each channel in each SBR element */ - if (hSbrEncoder->fTimeDomainDownsampling) - { - for (el=0; elnoElements; el++) - { - HANDLE_SBR_ELEMENT hSbrEl = hSbrEncoder->sbrElement[el]; - INT Wc, ch; - - /* Calculated required normalized cutoff frequency (Wc = 1.0 -> lowestBandwidth = inputSampleRate/2) */ - Wc = (2*lowestBandwidth)*1000 / inputSampleRate; - - for (ch=0; chelInfo.nChannelsInEl; ch++) - { - FDKaacEnc_InitDownsampler (&hSbrEl->sbrChannel[ch]->downSampler, Wc, *downSampleFactor); - FDK_ASSERT (hSbrEl->sbrChannel[ch]->downSampler.delay <=MAX_DS_FILTER_DELAY); - } - - downsamplerDelay = hSbrEl->sbrChannel[0]->downSampler.delay; - } /* third element loop */ - - /* lfe */ - FDKaacEnc_InitDownsampler (&hSbrEncoder->lfeDownSampler, 0, *downSampleFactor); - - /* Add the resampler additional delay to get the final delay and buffer offset values. */ - if (sbrOffset > 0 || downsampledOffset <= ((downsamplerDelay * (*numChannels))>>(*downSampleFactor-1))) { - sbrOffset += (downsamplerDelay - downsampledOffset) * (*numChannels) ; - *delay += downsamplerDelay - downsampledOffset; - downsampledOffset = 0; - } else { - downsampledOffset -= (downsamplerDelay * (*numChannels))>>(*downSampleFactor-1); - sbrOffset = 0; - } - - hSbrEncoder->inputDataDelay = downsamplerDelay; - } - - /* Assign core encoder Bandwidth */ - *coreBandwidth = lowestBandwidth; - - /* Estimate sbr bitrate, 2.5 kBit/s per sbr channel */ - hSbrEncoder->estimateBitrate += 2500 * (*numChannels); - - /* initialize parametric stereo */ - if (usePs) - { - PSENC_CONFIG psEncConfig; - FDK_ASSERT(hSbrEncoder->noElements == 1); - INT psTuningTableIdx = getPsTuningTableIndex(elInfo[0].bitRate, NULL); - - psEncConfig.frameSize = coreFrameLength; //sbrConfig.sbrFrameSize; - psEncConfig.qmfFilterMode = 0; - psEncConfig.sbrPsDelay = 0; - - /* tuning parameters */ - if (psTuningTableIdx != INVALID_TABLE_IDX) { - psEncConfig.nStereoBands = psTuningTable[psTuningTableIdx].nStereoBands; - psEncConfig.maxEnvelopes = psTuningTable[psTuningTableIdx].nEnvelopes; - psEncConfig.iidQuantErrorThreshold = (FIXP_DBL)psTuningTable[psTuningTableIdx].iidQuantErrorThreshold; - - /* calculation is not quite linear, increased number of envelopes causes more bits */ - /* assume avg. 50 bits per frame for 10 stereo bands / 1 envelope configuration */ - hSbrEncoder->estimateBitrate += ( (((*coreSampleRate) * 5 * psEncConfig.nStereoBands * psEncConfig.maxEnvelopes) / hSbrEncoder->frameSize)); - - } else { - error = ERROR(CDI, "Invalid ps tuning table index."); - goto bail; - } - - qmfInitSynthesisFilterBank(&hSbrEncoder->qmfSynthesisPS, - (FIXP_DBL*)hSbrEncoder->qmfSynthesisPS.FilterStates, - hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfSlots, - hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfBands>>1, - hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfBands>>1, - hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfBands>>1, - (statesInitFlag) ? 0 : QMF_FLAG_KEEP_STATES); - - if(errorInfo == noError){ - /* update delay */ - psEncConfig.sbrPsDelay = FDKsbrEnc_GetEnvEstDelay(&hSbrEncoder->sbrElement[0]->sbrChannel[0]->hEnvChannel.sbrExtractEnvelope); - - if(noError != (errorInfo = PSEnc_Init( hSbrEncoder->hParametricStereo, - &psEncConfig, - hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfSlots, - hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfBands - ,hSbrEncoder->dynamicRam - ))) - { - errorInfo = handBack(errorInfo); - } - } - - /* QMF analysis + Hybrid analysis + Hybrid synthesis + QMF synthesis + downsampled input buffer delay */ - hSbrEncoder->inputDataDelay = (64*10/2) + (6*64) + (0) + (64*10/2-64+1) + ((*downSampleFactor)*downsampledOffset); - } - - hSbrEncoder->downsampledOffset = downsampledOffset; - { - hSbrEncoder->downmixSize = coreFrameLength*(*numChannels); - } - - hSbrEncoder->bufferOffset = sbrOffset; - /* Delay Compensation: fill bitstream delay buffer with zero input signal */ - if ( hSbrEncoder->nBitstrDelay > 0 ) - { - error = FDKsbrEnc_DelayCompensation (hSbrEncoder, inputBuffer); - if (error != 0) - goto bail; - } - - /* Set Output frame length */ - *frameLength = coreFrameLength * *downSampleFactor; - /* Input buffer offset */ - *inputBufferOffset = fixMax(sbrOffset, downsampledOffset); - - - } - - return error; - -bail: - /* Restore input settings */ - *coreSampleRate = inputSampleRate; - *frameLength = coreFrameLength; - *numChannels = inputChannels; - *coreBandwidth = inputBandWidth; - - return error; - } - - -INT -sbrEncoder_EncodeFrame( HANDLE_SBR_ENCODER hSbrEncoder, - INT_PCM *samples, - UINT timeInStride, - UINT sbrDataBits[(8)], - UCHAR sbrData[(8)][MAX_PAYLOAD_SIZE] - ) -{ - INT error; - int el; - - for (el=0; elnoElements; el++) - { - if (hSbrEncoder->sbrElement[el] != NULL) - { - error = FDKsbrEnc_EnvEncodeFrame( - hSbrEncoder, - el, - samples + hSbrEncoder->downsampledOffset, - timeInStride, - &sbrDataBits[el], - sbrData[el], - 0 - ); - if (error) - return error; - } - } - - if ( ( hSbrEncoder->lfeChIdx!=-1) && (hSbrEncoder->downSampleFactor > 1) ) - { /* lfe downsampler */ - INT nOutSamples; - - FDKaacEnc_Downsample(&hSbrEncoder->lfeDownSampler, - samples + hSbrEncoder->downsampledOffset + hSbrEncoder->bufferOffset + hSbrEncoder->lfeChIdx, - hSbrEncoder->frameSize, - timeInStride, - samples + hSbrEncoder->downsampledOffset + hSbrEncoder->lfeChIdx, - &nOutSamples, - hSbrEncoder->nChannels); - - - } - - return 0; -} - - -INT sbrEncoder_UpdateBuffers( - HANDLE_SBR_ENCODER hSbrEncoder, - INT_PCM *timeBuffer - ) - { - if ( hSbrEncoder->downsampledOffset > 0 ) { - /* Move delayed downsampled data */ - FDKmemcpy ( timeBuffer, - timeBuffer + hSbrEncoder->downmixSize, - sizeof(INT_PCM) * (hSbrEncoder->downsampledOffset) ); - } else { - /* Move delayed input data */ - FDKmemcpy ( timeBuffer, - timeBuffer + hSbrEncoder->nChannels * hSbrEncoder->frameSize, - sizeof(INT_PCM) * hSbrEncoder->bufferOffset ); - } - if ( hSbrEncoder->nBitstrDelay > 0 ) - { - int el; - - for (el=0; elnoElements; el++) - { - FDKmemmove ( hSbrEncoder->sbrElement[el]->payloadDelayLine[0], - hSbrEncoder->sbrElement[el]->payloadDelayLine[1], - sizeof(UCHAR) * (hSbrEncoder->nBitstrDelay*MAX_PAYLOAD_SIZE) ); - - FDKmemmove( &hSbrEncoder->sbrElement[el]->payloadDelayLineSize[0], - &hSbrEncoder->sbrElement[el]->payloadDelayLineSize[1], - sizeof(UINT) * (hSbrEncoder->nBitstrDelay) ); - } - } - return 0; - } - - -INT sbrEncoder_GetEstimateBitrate(HANDLE_SBR_ENCODER hSbrEncoder) -{ - INT estimateBitrate = 0; - - if(hSbrEncoder) { - estimateBitrate += hSbrEncoder->estimateBitrate; - } - - return estimateBitrate; -} - -INT sbrEncoder_GetInputDataDelay(HANDLE_SBR_ENCODER hSbrEncoder) -{ - INT delay = -1; - - if(hSbrEncoder) { - delay = hSbrEncoder->inputDataDelay; - } - return delay; -} - - -INT sbrEncoder_GetLibInfo( LIB_INFO *info ) -{ - int i; - - if (info == NULL) { - return -1; - } - /* search for next free tab */ - for (i = 0; i < FDK_MODULE_LAST; i++) { - if (info[i].module_id == FDK_NONE) break; - } - if (i == FDK_MODULE_LAST) { - return -1; - } - info += i; - - info->module_id = FDK_SBRENC; - info->version = LIB_VERSION(SBRENCODER_LIB_VL0, SBRENCODER_LIB_VL1, SBRENCODER_LIB_VL2); - LIB_VERSION_STRING(info); -#ifdef __ANDROID__ - info->build_date = ""; - info->build_time = ""; -#else - info->build_date = __DATE__; - info->build_time = __TIME__; -#endif - info->title = "SBR Encoder"; - - /* Set flags */ - info->flags = 0 - | CAPF_SBR_HQ - | CAPF_SBR_PS_MPEG - ; - /* End of flags */ - - return 0; -} diff --git a/libSBRenc/src/sbr_misc.cpp b/libSBRenc/src/sbr_misc.cpp deleted file mode 100644 index c673b81..0000000 --- a/libSBRenc/src/sbr_misc.cpp +++ /dev/null @@ -1,272 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief Sbr miscellaneous helper functions -*/ -#include "sbr_misc.h" - - -void FDKsbrEnc_Shellsort_fract (FIXP_DBL *in, INT n) -{ - FIXP_DBL v; - INT i, j; - INT inc = 1; - - do - inc = 3 * inc + 1; - while (inc <= n); - - do { - inc = inc / 3; - for (i = inc + 1; i <= n; i++) { - v = in[i-1]; - j = i; - while (in[j-inc-1] > v) { - in[j-1] = in[j-inc-1]; - j -= inc; - if (j <= inc) - break; - } - in[j-1] = v; - } - } while (inc > 1); - -} - -/* Sorting routine */ -void FDKsbrEnc_Shellsort_int (INT *in, INT n) -{ - - INT i, j, v; - INT inc = 1; - - do - inc = 3 * inc + 1; - while (inc <= n); - - do { - inc = inc / 3; - for (i = inc + 1; i <= n; i++) { - v = in[i-1]; - j = i; - while (in[j-inc-1] > v) { - in[j-1] = in[j-inc-1]; - j -= inc; - if (j <= inc) - break; - } - in[j-1] = v; - } - } while (inc > 1); - -} - - - -/******************************************************************************* - Functionname: FDKsbrEnc_AddVecLeft - ******************************************************************************* - - Description: - - Arguments: INT* dst, INT* length_dst, INT* src, INT length_src - - Return: none - -*******************************************************************************/ -void -FDKsbrEnc_AddVecLeft (INT *dst, INT *length_dst, INT *src, INT length_src) -{ - INT i; - - for (i = length_src - 1; i >= 0; i--) - FDKsbrEnc_AddLeft (dst, length_dst, src[i]); -} - - -/******************************************************************************* - Functionname: FDKsbrEnc_AddLeft - ******************************************************************************* - - Description: - - Arguments: INT* vector, INT* length_vector, INT value - - Return: none - -*******************************************************************************/ -void -FDKsbrEnc_AddLeft (INT *vector, INT *length_vector, INT value) -{ - INT i; - - for (i = *length_vector; i > 0; i--) - vector[i] = vector[i - 1]; - vector[0] = value; - (*length_vector)++; -} - - -/******************************************************************************* - Functionname: FDKsbrEnc_AddRight - ******************************************************************************* - - Description: - - Arguments: INT* vector, INT* length_vector, INT value - - Return: none - -*******************************************************************************/ -void -FDKsbrEnc_AddRight (INT *vector, INT *length_vector, INT value) -{ - vector[*length_vector] = value; - (*length_vector)++; -} - - - -/******************************************************************************* - Functionname: FDKsbrEnc_AddVecRight - ******************************************************************************* - - Description: - - Arguments: INT* dst, INT* length_dst, INT* src, INT length_src) - - Return: none - -*******************************************************************************/ -void -FDKsbrEnc_AddVecRight (INT *dst, INT *length_dst, INT *src, INT length_src) -{ - INT i; - for (i = 0; i < length_src; i++) - FDKsbrEnc_AddRight (dst, length_dst, src[i]); -} - - -/***************************************************************************** - - functionname: FDKsbrEnc_LSI_divide_scale_fract - - description: Calculates division with best precision and scales the result. - - return: num*scale/denom - -*****************************************************************************/ -FIXP_DBL FDKsbrEnc_LSI_divide_scale_fract(FIXP_DBL num, FIXP_DBL denom, FIXP_DBL scale) -{ - FIXP_DBL tmp = FL2FXCONST_DBL(0.0f); - if (num != FL2FXCONST_DBL(0.0f)) { - - INT shiftCommon; - INT shiftNum = CountLeadingBits(num); - INT shiftDenom = CountLeadingBits(denom); - INT shiftScale = CountLeadingBits(scale); - - num = num << shiftNum; - scale = scale << shiftScale; - - tmp = fMultDiv2(num,scale); - - if ( denom > (tmp >> fixMin(shiftNum+shiftScale-1,(DFRACT_BITS-1))) ) { - denom = denom << shiftDenom; - tmp = schur_div(tmp,denom,15); - shiftCommon = fixMin((shiftNum-shiftDenom+shiftScale-1),(DFRACT_BITS-1)); - if (shiftCommon < 0) - tmp <<= -shiftCommon; - else - tmp >>= shiftCommon; - } - else { - tmp = /*FL2FXCONST_DBL(1.0)*/ (FIXP_DBL)MAXVAL_DBL; - } - } - - return (tmp); -} - diff --git a/libSBRenc/src/sbr_misc.h b/libSBRenc/src/sbr_misc.h deleted file mode 100644 index f471974..0000000 --- a/libSBRenc/src/sbr_misc.h +++ /dev/null @@ -1,106 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief Sbr miscellaneous helper functions prototypes - \author -*/ - -#ifndef _SBR_MISC_H -#define _SBR_MISC_H - -#include "sbr_encoder.h" - -/* Sorting routines */ -void FDKsbrEnc_Shellsort_fract (FIXP_DBL *in, INT n); -void FDKsbrEnc_Shellsort_int (INT *in, INT n); - -void FDKsbrEnc_AddLeft (INT *vector, INT *length_vector, INT value); -void FDKsbrEnc_AddRight (INT *vector, INT *length_vector, INT value); -void FDKsbrEnc_AddVecLeft (INT *dst, INT *length_dst, INT *src, INT length_src); -void FDKsbrEnc_AddVecRight (INT *dst, INT *length_vector_dst, INT *src, INT length_src); - -FIXP_DBL FDKsbrEnc_LSI_divide_scale_fract(FIXP_DBL num, FIXP_DBL denom, FIXP_DBL scale); - -#endif diff --git a/libSBRenc/src/sbr_ram.cpp b/libSBRenc/src/sbr_ram.cpp deleted file mode 100644 index ee6c37f..0000000 --- a/libSBRenc/src/sbr_ram.cpp +++ /dev/null @@ -1,222 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief Memory layout - - - This module declares all static and dynamic memory spaces -*/ -#include "sbr_ram.h" - -#include "sbr.h" -#include "genericStds.h" - -C_ALLOC_MEM (Ram_SbrDynamic_RAM, FIXP_DBL, ((SBR_ENC_DYN_RAM_SIZE)/sizeof(FIXP_DBL))) - -/*! - \name StaticSbrData - - Static memory areas, must not be overwritten in other sections of the encoder -*/ -/* @{ */ - -/*! static sbr encoder instance for one encoder (2 channels) - all major static and dynamic memory areas are located - in module sbr_ram and sbr rom -*/ -C_ALLOC_MEM (Ram_SbrEncoder, SBR_ENCODER, 1) -C_ALLOC_MEM2(Ram_SbrChannel, SBR_CHANNEL, 1, (8)) -C_ALLOC_MEM2(Ram_SbrElement, SBR_ELEMENT, 1, (8)) - -/*! Filter states for QMF-analysis.
- Dimension: #MAXNRSBRCHANNELS * #SBR_QMF_FILTER_LENGTH -*/ -C_AALLOC_MEM2_L (Ram_Sbr_QmfStatesAnalysis, FIXP_QAS, QMF_FILTER_LENGTH, (8), SECT_DATA_L1) - - -/*! Matrix holding the quota values for all estimates, all channels - Dimension #MAXNRSBRCHANNELS * +#SBR_QMF_CHANNELS* #MAX_NO_OF_ESTIMATES -*/ -C_ALLOC_MEM2_L (Ram_Sbr_quotaMatrix, FIXP_DBL, (MAX_NO_OF_ESTIMATES*QMF_CHANNELS), (8), SECT_DATA_L1) - -/*! Matrix holding the sign values for all estimates, all channels - Dimension #MAXNRSBRCHANNELS * +#SBR_QMF_CHANNELS* #MAX_NO_OF_ESTIMATES -*/ -C_ALLOC_MEM2 (Ram_Sbr_signMatrix, INT, (MAX_NO_OF_ESTIMATES*QMF_CHANNELS), (8)) - -/*! Frequency band table (low res)
- Dimension #MAX_FREQ_COEFFS/2+1 -*/ -C_ALLOC_MEM2 (Ram_Sbr_freqBandTableLO, UCHAR, (MAX_FREQ_COEFFS/2+1), (8)) - -/*! Frequency band table (high res)
- Dimension #MAX_FREQ_COEFFS +1 -*/ -C_ALLOC_MEM2 (Ram_Sbr_freqBandTableHI, UCHAR, (MAX_FREQ_COEFFS+1), (8)) - -/*! vk matser table
- Dimension #MAX_FREQ_COEFFS +1 -*/ -C_ALLOC_MEM2 (Ram_Sbr_v_k_master, UCHAR, (MAX_FREQ_COEFFS+1), (8)) - - -/* - Missing harmonics detection -*/ - -/*! sbr_detectionVectors
- Dimension #MAX_NUM_CHANNELS*#MAX_NO_OF_ESTIMATES*#MAX_FREQ_COEFFS] -*/ -C_ALLOC_MEM2 (Ram_Sbr_detectionVectors, UCHAR, (MAX_NO_OF_ESTIMATES*MAX_FREQ_COEFFS), (8)) - -/*! sbr_prevCompVec[
- Dimension #MAX_NUM_CHANNELS*#MAX_FREQ_COEFFS] -*/ -C_ALLOC_MEM2 (Ram_Sbr_prevEnvelopeCompensation, UCHAR, MAX_FREQ_COEFFS, (8)) -/*! sbr_guideScfb[
- Dimension #MAX_NUM_CHANNELS*#MAX_FREQ_COEFFS] -*/ -C_ALLOC_MEM2 (Ram_Sbr_guideScfb, UCHAR, MAX_FREQ_COEFFS, (8)) - -/*! sbr_guideVectorDetected
- Dimension #MAX_NUM_CHANNELS*#MAX_NO_OF_ESTIMATES*#MAX_FREQ_COEFFS] -*/ -C_ALLOC_MEM2 (Ram_Sbr_guideVectorDetected, UCHAR, (MAX_NO_OF_ESTIMATES*MAX_FREQ_COEFFS), (8)) -C_ALLOC_MEM2 (Ram_Sbr_guideVectorDiff, FIXP_DBL, (MAX_NO_OF_ESTIMATES*MAX_FREQ_COEFFS), (8)) -C_ALLOC_MEM2 (Ram_Sbr_guideVectorOrig, FIXP_DBL, (MAX_NO_OF_ESTIMATES*MAX_FREQ_COEFFS), (8)) - -/* - Static Parametric Stereo memory -*/ -C_AALLOC_MEM_L(Ram_PsQmfStatesSynthesis, FIXP_DBL, QMF_FILTER_LENGTH/2, SECT_DATA_L1) - -C_ALLOC_MEM_L (Ram_PsEncode, PS_ENCODE, 1, SECT_DATA_L1) -C_ALLOC_MEM (Ram_ParamStereo, PARAMETRIC_STEREO, 1) - - - -/* @} */ - - -/*! - \name DynamicSbrData - - Dynamic memory areas, might be reused in other algorithm sections, - e.g. the core encoder. -*/ -/* @{ */ - - /*! Energy buffer for envelope extraction
- Dimension #MAXNRSBRCHANNELS * +#SBR_QMF_SLOTS * #SBR_QMF_CHANNELS - */ - C_ALLOC_MEM2 (Ram_Sbr_envYBuffer, FIXP_DBL, (QMF_MAX_TIME_SLOTS/2 * QMF_CHANNELS), (8)) - - FIXP_DBL* GetRam_Sbr_envYBuffer (int n, UCHAR* dynamic_RAM) { - FDK_ASSERT(dynamic_RAM!=0); - return ((FIXP_DBL*) (dynamic_RAM + OFFSET_NRG + (n*Y_2_BUF_BYTE) )); - } - - /* - * QMF data - */ - /* The SBR encoder uses a single channel overlapping buffer set (always n=0), but PS does not. */ - FIXP_DBL* GetRam_Sbr_envRBuffer (int n, UCHAR* dynamic_RAM) { - FDK_ASSERT(dynamic_RAM!=0); - return ((FIXP_DBL*) (dynamic_RAM + OFFSET_QMF + (n*(ENV_R_BUFF_BYTE+ENV_I_BUFF_BYTE)) )); - } - FIXP_DBL* GetRam_Sbr_envIBuffer (int n, UCHAR* dynamic_RAM) { - FDK_ASSERT(dynamic_RAM!=0); - return ((FIXP_DBL*) (dynamic_RAM + OFFSET_QMF + (ENV_R_BUFF_BYTE) + (n*(ENV_R_BUFF_BYTE+ENV_I_BUFF_BYTE)))); - } - - - - -/* @} */ - - - - - diff --git a/libSBRenc/src/sbr_ram.h b/libSBRenc/src/sbr_ram.h deleted file mode 100644 index 7e3d0c8..0000000 --- a/libSBRenc/src/sbr_ram.h +++ /dev/null @@ -1,187 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! -\file -\brief Memory layout - -*/ -#ifndef __SBR_RAM_H -#define __SBR_RAM_H - -#include "sbr_def.h" -#include "env_est.h" -#include "sbr_encoder.h" -#include "sbr.h" - - - -#include "ps_main.h" -#include "ps_encode.h" - - -#define ENV_TRANSIENTS_BYTE ( (sizeof(FIXP_DBL)*(MAX_NUM_CHANNELS*3*QMF_MAX_TIME_SLOTS)) ) - - #define ENV_R_BUFF_BYTE ( (sizeof(FIXP_DBL)*((QMF_MAX_TIME_SLOTS) * MAX_HYBRID_BANDS)) ) - #define ENV_I_BUFF_BYTE ( (sizeof(FIXP_DBL)*((QMF_MAX_TIME_SLOTS) * MAX_HYBRID_BANDS)) ) - #define Y_BUF_CH_BYTE ( (2*sizeof(FIXP_DBL)*((QMF_MAX_TIME_SLOTS) * MAX_HYBRID_BANDS)) ) - - -#define ENV_R_BUF_PS_BYTE ( (sizeof(FIXP_DBL)*QMF_MAX_TIME_SLOTS * QMF_CHANNELS / 2) ) -#define ENV_I_BUF_PS_BYTE ( (sizeof(FIXP_DBL)*QMF_MAX_TIME_SLOTS * QMF_CHANNELS / 2) ) - -#define TON_BUF_CH_BYTE ( (sizeof(FIXP_DBL)*(MAX_NO_OF_ESTIMATES*MAX_FREQ_COEFFS)) ) - -#define Y_2_BUF_BYTE ( Y_BUF_CH_BYTE>>1 ) - - -/* Workbuffer RAM - Allocation */ -/* - ++++++++++++++++++++++++++++++++++++++++++++++++++++ - | OFFSET_QMF | OFFSET_NRG | - ++++++++++++++++++++++++++++++++++++++++++++++++++++ - ------------------------- ------------------------- - | | 0.5 * | - | sbr_envRBuffer | sbr_envYBuffer_size | - | sbr_envIBuffer | | - ------------------------- ------------------------- - -*/ - #define BUF_NRG_SIZE ( (MAX_NUM_CHANNELS * Y_2_BUF_BYTE) ) - #define BUF_QMF_SIZE (ENV_R_BUFF_BYTE + ENV_I_BUFF_BYTE) - - /* Size of the shareable memory region than can be reused */ - #define SBR_ENC_DYN_RAM_SIZE ( BUF_QMF_SIZE + BUF_NRG_SIZE ) - - #define OFFSET_QMF ( 0 ) - #define OFFSET_NRG ( OFFSET_QMF + BUF_QMF_SIZE ) - - -/* - ***************************************************************************************************** - */ - - H_ALLOC_MEM(Ram_SbrDynamic_RAM, FIXP_DBL) - - H_ALLOC_MEM(Ram_SbrEncoder, SBR_ENCODER) - H_ALLOC_MEM(Ram_SbrChannel, SBR_CHANNEL) - H_ALLOC_MEM(Ram_SbrElement, SBR_ELEMENT) - - H_ALLOC_MEM(Ram_Sbr_quotaMatrix, FIXP_DBL) - H_ALLOC_MEM(Ram_Sbr_signMatrix, INT) - - H_ALLOC_MEM(Ram_Sbr_QmfStatesAnalysis, FIXP_QAS) - - H_ALLOC_MEM(Ram_Sbr_freqBandTableLO, UCHAR) - H_ALLOC_MEM(Ram_Sbr_freqBandTableHI, UCHAR) - H_ALLOC_MEM(Ram_Sbr_v_k_master, UCHAR) - - H_ALLOC_MEM(Ram_Sbr_detectionVectors, UCHAR) - H_ALLOC_MEM(Ram_Sbr_prevEnvelopeCompensation, UCHAR) - H_ALLOC_MEM(Ram_Sbr_guideScfb, UCHAR) - H_ALLOC_MEM(Ram_Sbr_guideVectorDetected, UCHAR) - - /* Dynamic Memory Allocation */ - - H_ALLOC_MEM(Ram_Sbr_envYBuffer, FIXP_DBL) - FIXP_DBL* GetRam_Sbr_envYBuffer (int n, UCHAR* dynamic_RAM); - FIXP_DBL* GetRam_Sbr_envRBuffer (int n, UCHAR* dynamic_RAM); - FIXP_DBL* GetRam_Sbr_envIBuffer (int n, UCHAR* dynamic_RAM); - - H_ALLOC_MEM(Ram_Sbr_guideVectorDiff, FIXP_DBL) - H_ALLOC_MEM(Ram_Sbr_guideVectorOrig, FIXP_DBL) - - - H_ALLOC_MEM(Ram_PsQmfStatesSynthesis, FIXP_DBL) - - H_ALLOC_MEM(Ram_PsEncode, PS_ENCODE) - - FIXP_DBL* FDKsbrEnc_SliceRam_PsRqmf (FIXP_DBL* rQmfData, UCHAR* dynamic_RAM, int n, int i, int qmfSlots); - FIXP_DBL* FDKsbrEnc_SliceRam_PsIqmf (FIXP_DBL* iQmfData, UCHAR* dynamic_RAM, int n, int i, int qmfSlots); - - H_ALLOC_MEM(Ram_ParamStereo, PARAMETRIC_STEREO) - - - -#endif - diff --git a/libSBRenc/src/sbr_rom.cpp b/libSBRenc/src/sbr_rom.cpp deleted file mode 100644 index 7a51668..0000000 --- a/libSBRenc/src/sbr_rom.cpp +++ /dev/null @@ -1,795 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief Definition of constant tables - - - This module contains most of the constant data that can be stored in ROM. -*/ - -#include "sbr_rom.h" -#include "genericStds.h" - -//@{ -/******************************************************************************* - - Table Overview: - - o envelope level, 1.5 dB: - 1a) v_Huff_envelopeLevelC10T[121] - 1b) v_Huff_envelopeLevelL10T[121] - 2a) v_Huff_envelopeLevelC10F[121] - 2b) v_Huff_envelopeLevelL10F[121] - - o envelope balance, 1.5 dB: - 3a) bookSbrEnvBalanceC10T[49] - 3b) bookSbrEnvBalanceL10T[49] - 4a) bookSbrEnvBalanceC10F[49] - 4b) bookSbrEnvBalanceL10F[49] - - o envelope level, 3.0 dB: - 5a) v_Huff_envelopeLevelC11T[63] - 5b) v_Huff_envelopeLevelL11T[63] - 6a) v_Huff_envelopeLevelC11F[63] - 6b) v_Huff_envelopeLevelC11F[63] - - o envelope balance, 3.0 dB: - 7a) bookSbrEnvBalanceC11T[25] - 7b) bookSbrEnvBalanceL11T[25] - 8a) bookSbrEnvBalanceC11F[25] - 8b) bookSbrEnvBalanceL11F[25] - - o noise level, 3.0 dB: - 9a) v_Huff_NoiseLevelC11T[63] - 9b) v_Huff_NoiseLevelL11T[63] - - ) (v_Huff_envelopeLevelC11F[63] is used for freq dir) - - ) (v_Huff_envelopeLevelL11F[63] is used for freq dir) - - o noise balance, 3.0 dB: - 10a) bookSbrNoiseBalanceC11T[25] - 10b) bookSbrNoiseBalanceL11T[25] - - ) (bookSbrEnvBalanceC11F[25] is used for freq dir) - - ) (bookSbrEnvBalanceL11F[25] is used for freq dir) - - - (1.5 dB is never used for noise) - -********************************************************************************/ - - -/*******************************************************************************/ -/* table : envelope level, 1.5 dB */ -/* theor range : [-58,58], CODE_BOOK_SCF_LAV = 58 */ -/* implem range: [-60,60], CODE_BOOK_SCF_LAV10 = 60 */ -/* raw stats : envelopeLevel_00 (yes, wrong suffix in name) KK 01-03-09 */ -/*******************************************************************************/ - -/* direction: time - contents : codewords - raw table: HuffCode3C2FIX.m/envelopeLevel_00T_cF.mat/v_nChex_cF - built by : FH 01-07-05 */ - -const INT v_Huff_envelopeLevelC10T[121] = -{ - 0x0003FFD6, 0x0003FFD7, 0x0003FFD8, 0x0003FFD9, 0x0003FFDA, 0x0003FFDB, 0x0007FFB8, 0x0007FFB9, - 0x0007FFBA, 0x0007FFBB, 0x0007FFBC, 0x0007FFBD, 0x0007FFBE, 0x0007FFBF, 0x0007FFC0, 0x0007FFC1, - 0x0007FFC2, 0x0007FFC3, 0x0007FFC4, 0x0007FFC5, 0x0007FFC6, 0x0007FFC7, 0x0007FFC8, 0x0007FFC9, - 0x0007FFCA, 0x0007FFCB, 0x0007FFCC, 0x0007FFCD, 0x0007FFCE, 0x0007FFCF, 0x0007FFD0, 0x0007FFD1, - 0x0007FFD2, 0x0007FFD3, 0x0001FFE6, 0x0003FFD4, 0x0000FFF0, 0x0001FFE9, 0x0003FFD5, 0x0001FFE7, - 0x0000FFF1, 0x0000FFEC, 0x0000FFED, 0x0000FFEE, 0x00007FF4, 0x00003FF9, 0x00003FF7, 0x00001FFA, - 0x00001FF9, 0x00000FFB, 0x000007FC, 0x000003FC, 0x000001FD, 0x000000FD, 0x0000007D, 0x0000003D, - 0x0000001D, 0x0000000D, 0x00000005, 0x00000001, 0x00000000, 0x00000004, 0x0000000C, 0x0000001C, - 0x0000003C, 0x0000007C, 0x000000FC, 0x000001FC, 0x000003FD, 0x00000FFA, 0x00001FF8, 0x00003FF6, - 0x00003FF8, 0x00007FF5, 0x0000FFEF, 0x0001FFE8, 0x0000FFF2, 0x0007FFD4, 0x0007FFD5, 0x0007FFD6, - 0x0007FFD7, 0x0007FFD8, 0x0007FFD9, 0x0007FFDA, 0x0007FFDB, 0x0007FFDC, 0x0007FFDD, 0x0007FFDE, - 0x0007FFDF, 0x0007FFE0, 0x0007FFE1, 0x0007FFE2, 0x0007FFE3, 0x0007FFE4, 0x0007FFE5, 0x0007FFE6, - 0x0007FFE7, 0x0007FFE8, 0x0007FFE9, 0x0007FFEA, 0x0007FFEB, 0x0007FFEC, 0x0007FFED, 0x0007FFEE, - 0x0007FFEF, 0x0007FFF0, 0x0007FFF1, 0x0007FFF2, 0x0007FFF3, 0x0007FFF4, 0x0007FFF5, 0x0007FFF6, - 0x0007FFF7, 0x0007FFF8, 0x0007FFF9, 0x0007FFFA, 0x0007FFFB, 0x0007FFFC, 0x0007FFFD, 0x0007FFFE, - 0x0007FFFF -}; - - -/* direction: time - contents : codeword lengths - raw table: HuffCode3C2FIX.m/envelopeLevel_00T_cF.mat/v_nLhex_cF - built by : FH 01-07-05 */ - -const UCHAR v_Huff_envelopeLevelL10T[121] = -{ - 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, - 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, - 0x13, 0x13, 0x11, 0x12, 0x10, 0x11, 0x12, 0x11, 0x10, 0x10, 0x10, 0x10, 0x0F, 0x0E, 0x0E, 0x0D, - 0x0D, 0x0C, 0x0B, 0x0A, 0x09, 0x08, 0x07, 0x06, 0x05, 0x04, 0x03, 0x02, 0x02, 0x03, 0x04, 0x05, - 0x06, 0x07, 0x08, 0x09, 0x0A, 0x0C, 0x0D, 0x0E, 0x0E, 0x0F, 0x10, 0x11, 0x10, 0x13, 0x13, 0x13, - 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, - 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, - 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13 -}; - - -/* direction: freq - contents : codewords - raw table: HuffCode3C2FIX.m/envelopeLevel_00F_cF.mat/v_nChex_cF - built by : FH 01-07-05 */ - -const INT v_Huff_envelopeLevelC10F[121] = -{ - 0x0007FFE7, 0x0007FFE8, 0x000FFFD2, 0x000FFFD3, 0x000FFFD4, 0x000FFFD5, 0x000FFFD6, 0x000FFFD7, - 0x000FFFD8, 0x0007FFDA, 0x000FFFD9, 0x000FFFDA, 0x000FFFDB, 0x000FFFDC, 0x0007FFDB, 0x000FFFDD, - 0x0007FFDC, 0x0007FFDD, 0x000FFFDE, 0x0003FFE4, 0x000FFFDF, 0x000FFFE0, 0x000FFFE1, 0x0007FFDE, - 0x000FFFE2, 0x000FFFE3, 0x000FFFE4, 0x0007FFDF, 0x000FFFE5, 0x0007FFE0, 0x0003FFE8, 0x0007FFE1, - 0x0003FFE0, 0x0003FFE9, 0x0001FFEF, 0x0003FFE5, 0x0001FFEC, 0x0001FFED, 0x0001FFEE, 0x0000FFF4, - 0x0000FFF3, 0x0000FFF0, 0x00007FF7, 0x00007FF6, 0x00003FFA, 0x00001FFA, 0x00001FF9, 0x00000FFA, - 0x00000FF8, 0x000007F9, 0x000003FB, 0x000001FC, 0x000001FA, 0x000000FB, 0x0000007C, 0x0000003C, - 0x0000001C, 0x0000000C, 0x00000005, 0x00000001, 0x00000000, 0x00000004, 0x0000000D, 0x0000001D, - 0x0000003D, 0x000000FA, 0x000000FC, 0x000001FB, 0x000003FA, 0x000007F8, 0x000007FA, 0x000007FB, - 0x00000FF9, 0x00000FFB, 0x00001FF8, 0x00001FFB, 0x00003FF8, 0x00003FF9, 0x0000FFF1, 0x0000FFF2, - 0x0001FFEA, 0x0001FFEB, 0x0003FFE1, 0x0003FFE2, 0x0003FFEA, 0x0003FFE3, 0x0003FFE6, 0x0003FFE7, - 0x0003FFEB, 0x000FFFE6, 0x0007FFE2, 0x000FFFE7, 0x000FFFE8, 0x000FFFE9, 0x000FFFEA, 0x000FFFEB, - 0x000FFFEC, 0x0007FFE3, 0x000FFFED, 0x000FFFEE, 0x000FFFEF, 0x000FFFF0, 0x0007FFE4, 0x000FFFF1, - 0x0003FFEC, 0x000FFFF2, 0x000FFFF3, 0x0007FFE5, 0x0007FFE6, 0x000FFFF4, 0x000FFFF5, 0x000FFFF6, - 0x000FFFF7, 0x000FFFF8, 0x000FFFF9, 0x000FFFFA, 0x000FFFFB, 0x000FFFFC, 0x000FFFFD, 0x000FFFFE, - 0x000FFFFF -}; - - -/* direction: freq - contents : codeword lengths - raw table: HuffCode3C2FIX.m/envelopeLevel_00F_cF.mat/v_nLhex_cF - built by : FH 01-07-05 */ - -const UCHAR v_Huff_envelopeLevelL10F[121] = -{ - 0x13, 0x13, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x13, 0x14, 0x14, 0x14, 0x14, 0x13, 0x14, - 0x13, 0x13, 0x14, 0x12, 0x14, 0x14, 0x14, 0x13, 0x14, 0x14, 0x14, 0x13, 0x14, 0x13, 0x12, 0x13, - 0x12, 0x12, 0x11, 0x12, 0x11, 0x11, 0x11, 0x10, 0x10, 0x10, 0x0F, 0x0F, 0x0E, 0x0D, 0x0D, 0x0C, - 0x0C, 0x0B, 0x0A, 0x09, 0x09, 0x08, 0x07, 0x06, 0x05, 0x04, 0x03, 0x02, 0x02, 0x03, 0x04, 0x05, - 0x06, 0x08, 0x08, 0x09, 0x0A, 0x0B, 0x0B, 0x0B, 0x0C, 0x0C, 0x0D, 0x0D, 0x0E, 0x0E, 0x10, 0x10, - 0x11, 0x11, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x14, 0x13, 0x14, 0x14, 0x14, 0x14, 0x14, - 0x14, 0x13, 0x14, 0x14, 0x14, 0x14, 0x13, 0x14, 0x12, 0x14, 0x14, 0x13, 0x13, 0x14, 0x14, 0x14, - 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14 -}; - - -/*******************************************************************************/ -/* table : envelope balance, 1.5 dB */ -/* theor range : [-48,48], CODE_BOOK_SCF_LAV = 48 */ -/* implem range: same but mapped to [-24,24], CODE_BOOK_SCF_LAV_BALANCE10 = 24 */ -/* raw stats : envelopePan_00 (yes, wrong suffix in name) KK 01-03-09 */ -/*******************************************************************************/ - -/* direction: time - contents : codewords - raw table: HuffCode3C.m/envelopePan_00T.mat/v_nBhex - built by : FH 01-05-15 */ - -const INT bookSbrEnvBalanceC10T[49] = -{ - 0x0000FFE4, 0x0000FFE5, 0x0000FFE6, 0x0000FFE7, 0x0000FFE8, 0x0000FFE9, 0x0000FFEA, 0x0000FFEB, - 0x0000FFEC, 0x0000FFED, 0x0000FFEE, 0x0000FFEF, 0x0000FFF0, 0x0000FFF1, 0x0000FFF2, 0x0000FFF3, - 0x0000FFF4, 0x0000FFE2, 0x00000FFC, 0x000007FC, 0x000001FE, 0x0000007E, 0x0000001E, 0x00000006, - 0x00000000, 0x00000002, 0x0000000E, 0x0000003E, 0x000000FE, 0x000007FD, 0x00000FFD, 0x00007FF0, - 0x0000FFE3, 0x0000FFF5, 0x0000FFF6, 0x0000FFF7, 0x0000FFF8, 0x0000FFF9, 0x0000FFFA, 0x0001FFF6, - 0x0001FFF7, 0x0001FFF8, 0x0001FFF9, 0x0001FFFA, 0x0001FFFB, 0x0001FFFC, 0x0001FFFD, 0x0001FFFE, - 0x0001FFFF -}; - - -/* direction: time - contents : codeword lengths - raw table: HuffCode3C.m/envelopePan_00T.mat/v_nLhex - built by : FH 01-05-15 */ - -const UCHAR bookSbrEnvBalanceL10T[49] = -{ - 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, - 0x10, 0x10, 0x0C, 0x0B, 0x09, 0x07, 0x05, 0x03, 0x01, 0x02, 0x04, 0x06, 0x08, 0x0B, 0x0C, 0x0F, - 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x11, 0x11, 0x11, 0x11, 0x11, 0x11, 0x11, 0x11, 0x11, - 0x11 -}; - - -/* direction: freq - contents : codewords - raw table: HuffCode3C.m/envelopePan_00F.mat/v_nBhex - built by : FH 01-05-15 */ - -const INT bookSbrEnvBalanceC10F[49] = -{ - 0x0003FFE2, 0x0003FFE3, 0x0003FFE4, 0x0003FFE5, 0x0003FFE6, 0x0003FFE7, 0x0003FFE8, 0x0003FFE9, - 0x0003FFEA, 0x0003FFEB, 0x0003FFEC, 0x0003FFED, 0x0003FFEE, 0x0003FFEF, 0x0003FFF0, 0x0000FFF7, - 0x0001FFF0, 0x00003FFC, 0x000007FE, 0x000007FC, 0x000000FE, 0x0000007E, 0x0000000E, 0x00000002, - 0x00000000, 0x00000006, 0x0000001E, 0x0000003E, 0x000001FE, 0x000007FD, 0x00000FFE, 0x00007FFA, - 0x0000FFF6, 0x0003FFF1, 0x0003FFF2, 0x0003FFF3, 0x0003FFF4, 0x0003FFF5, 0x0003FFF6, 0x0003FFF7, - 0x0003FFF8, 0x0003FFF9, 0x0003FFFA, 0x0003FFFB, 0x0003FFFC, 0x0003FFFD, 0x0003FFFE, 0x0007FFFE, - 0x0007FFFF -}; - - -/* direction: freq - contents : codeword lengths - raw table: HuffCode3C.m/envelopePan_00F.mat/v_nLhex - built by : FH 01-05-15 */ - -const UCHAR bookSbrEnvBalanceL10F[49] = -{ - 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x10, - 0x11, 0x0E, 0x0B, 0x0B, 0x08, 0x07, 0x04, 0x02, 0x01, 0x03, 0x05, 0x06, 0x09, 0x0B, 0x0C, 0x0F, - 0x10, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x13, - 0x13 -}; - - -/*******************************************************************************/ -/* table : envelope level, 3.0 dB */ -/* theor range : [-29,29], CODE_BOOK_SCF_LAV = 29 */ -/* implem range: [-31,31], CODE_BOOK_SCF_LAV11 = 31 */ -/* raw stats : envelopeLevel_11 KK 00-02-03 */ -/*******************************************************************************/ - -/* direction: time - contents : codewords - raw table: HuffCode2.m - built by : FH 00-02-04 */ - -const INT v_Huff_envelopeLevelC11T[63] = { - 0x0003FFED, 0x0003FFEE, 0x0007FFDE, 0x0007FFDF, 0x0007FFE0, 0x0007FFE1, 0x0007FFE2, 0x0007FFE3, - 0x0007FFE4, 0x0007FFE5, 0x0007FFE6, 0x0007FFE7, 0x0007FFE8, 0x0007FFE9, 0x0007FFEA, 0x0007FFEB, - 0x0007FFEC, 0x0001FFF4, 0x0000FFF7, 0x0000FFF9, 0x0000FFF8, 0x00003FFB, 0x00003FFA, 0x00003FF8, - 0x00001FFA, 0x00000FFC, 0x000007FC, 0x000000FE, 0x0000003E, 0x0000000E, 0x00000002, 0x00000000, - 0x00000006, 0x0000001E, 0x0000007E, 0x000001FE, 0x000007FD, 0x00001FFB, 0x00003FF9, 0x00003FFC, - 0x00007FFA, 0x0000FFF6, 0x0001FFF5, 0x0003FFEC, 0x0007FFED, 0x0007FFEE, 0x0007FFEF, 0x0007FFF0, - 0x0007FFF1, 0x0007FFF2, 0x0007FFF3, 0x0007FFF4, 0x0007FFF5, 0x0007FFF6, 0x0007FFF7, 0x0007FFF8, - 0x0007FFF9, 0x0007FFFA, 0x0007FFFB, 0x0007FFFC, 0x0007FFFD, 0x0007FFFE, 0x0007FFFF -}; - - -/* direction: time - contents : codeword lengths - raw table: HuffCode2.m - built by : FH 00-02-04 */ - -const UCHAR v_Huff_envelopeLevelL11T[63] = { - 0x12, 0x12, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, - 0x13, 0x11, 0x10, 0x10, 0x10, 0x0E, 0x0E, 0x0E, 0x0D, 0x0C, 0x0B, 0x08, 0x06, 0x04, 0x02, 0x01, - 0x03, 0x05, 0x07, 0x09, 0x0B, 0x0D, 0x0E, 0x0E, 0x0F, 0x10, 0x11, 0x12, 0x13, 0x13, 0x13, 0x13, - 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13 -}; - - -/* direction: freq - contents : codewords - raw table: HuffCode2.m - built by : FH 00-02-04 */ - -const INT v_Huff_envelopeLevelC11F[63] = { - 0x000FFFF0, 0x000FFFF1, 0x000FFFF2, 0x000FFFF3, 0x000FFFF4, 0x000FFFF5, 0x000FFFF6, 0x0003FFF3, - 0x0007FFF5, 0x0007FFEE, 0x0007FFEF, 0x0007FFF6, 0x0003FFF4, 0x0003FFF2, 0x000FFFF7, 0x0007FFF0, - 0x0001FFF5, 0x0003FFF0, 0x0001FFF4, 0x0000FFF7, 0x0000FFF6, 0x00007FF8, 0x00003FFB, 0x00000FFD, - 0x000007FD, 0x000003FD, 0x000001FD, 0x000000FD, 0x0000003E, 0x0000000E, 0x00000002, 0x00000000, - 0x00000006, 0x0000001E, 0x000000FC, 0x000001FC, 0x000003FC, 0x000007FC, 0x00000FFC, 0x00001FFC, - 0x00003FFA, 0x00007FF9, 0x00007FFA, 0x0000FFF8, 0x0000FFF9, 0x0001FFF6, 0x0001FFF7, 0x0003FFF5, - 0x0003FFF6, 0x0003FFF1, 0x000FFFF8, 0x0007FFF1, 0x0007FFF2, 0x0007FFF3, 0x000FFFF9, 0x0007FFF7, - 0x0007FFF4, 0x000FFFFA, 0x000FFFFB, 0x000FFFFC, 0x000FFFFD, 0x000FFFFE, 0x000FFFFF -}; - - -/* direction: freq - contents : codeword lengths - raw table: HuffCode2.m - built by : FH 00-02-04 */ - -const UCHAR v_Huff_envelopeLevelL11F[63] = { - 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x12, 0x13, 0x13, 0x13, 0x13, 0x12, 0x12, 0x14, 0x13, - 0x11, 0x12, 0x11, 0x10, 0x10, 0x0F, 0x0E, 0x0C, 0x0B, 0x0A, 0x09, 0x08, 0x06, 0x04, 0x02, 0x01, - 0x03, 0x05, 0x08, 0x09, 0x0A, 0x0B, 0x0C, 0x0D, 0x0E, 0x0F, 0x0F, 0x10, 0x10, 0x11, 0x11, 0x12, - 0x12, 0x12, 0x14, 0x13, 0x13, 0x13, 0x14, 0x13, 0x13, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14 -}; - - - -/*******************************************************************************/ -/* table : envelope balance, 3.0 dB */ -/* theor range : [-24,24], CODE_BOOK_SCF_LAV = 24 */ -/* implem range: same but mapped to [-12,12], CODE_BOOK_SCF_LAV_BALANCE11 = 12 */ -/* raw stats : envelopeBalance_11 KK 00-02-03 */ -/*******************************************************************************/ - -/* direction: time - contents : codewords - raw table: HuffCode3C.m/envelopeBalance_11T.mat/v_nBhex - built by : FH 01-05-15 */ - -const INT bookSbrEnvBalanceC11T[25] = -{ - 0x00001FF2, 0x00001FF3, 0x00001FF4, 0x00001FF5, 0x00001FF6, 0x00001FF7, 0x00001FF8, 0x00000FF8, - 0x000000FE, 0x0000007E, 0x0000000E, 0x00000006, 0x00000000, 0x00000002, 0x0000001E, 0x0000003E, - 0x000001FE, 0x00001FF9, 0x00001FFA, 0x00001FFB, 0x00001FFC, 0x00001FFD, 0x00001FFE, 0x00003FFE, - 0x00003FFF -}; - - -/* direction: time - contents : codeword lengths - raw table: HuffCode3C.m/envelopeBalance_11T.mat/v_nLhex - built by : FH 01-05-15 */ - -const UCHAR bookSbrEnvBalanceL11T[25] = -{ - 0x0D, 0x0D, 0x0D, 0x0D, 0x0D, 0x0D, 0x0D, 0x0C, 0x08, 0x07, 0x04, 0x03, 0x01, 0x02, 0x05, 0x06, - 0x09, 0x0D, 0x0D, 0x0D, 0x0D, 0x0D, 0x0D, 0x0E, 0x0E -}; - - -/* direction: freq - contents : codewords - raw table: HuffCode3C.m/envelopeBalance_11F.mat/v_nBhex - built by : FH 01-05-15 */ - -const INT bookSbrEnvBalanceC11F[25] = -{ - 0x00001FF7, 0x00001FF8, 0x00001FF9, 0x00001FFA, 0x00001FFB, 0x00003FF8, 0x00003FF9, 0x000007FC, - 0x000000FE, 0x0000007E, 0x0000000E, 0x00000002, 0x00000000, 0x00000006, 0x0000001E, 0x0000003E, - 0x000001FE, 0x00000FFA, 0x00001FF6, 0x00003FFA, 0x00003FFB, 0x00003FFC, 0x00003FFD, 0x00003FFE, - 0x00003FFF -}; - - -/* direction: freq - contents : codeword lengths - raw table: HuffCode3C.m/envelopeBalance_11F.mat/v_nLhex - built by : FH 01-05-15 */ - -const UCHAR bookSbrEnvBalanceL11F[25] = -{ - 0x0D, 0x0D, 0x0D, 0x0D, 0x0D, 0x0E, 0x0E, 0x0B, 0x08, 0x07, 0x04, 0x02, 0x01, 0x03, 0x05, 0x06, - 0x09, 0x0C, 0x0D, 0x0E, 0x0E, 0x0E, 0x0E, 0x0E, 0x0E -}; - - -/*******************************************************************************/ -/* table : noise level, 3.0 dB */ -/* theor range : [-29,29], CODE_BOOK_SCF_LAV = 29 */ -/* implem range: [-31,31], CODE_BOOK_SCF_LAV11 = 31 */ -/* raw stats : noiseLevel_11 KK 00-02-03 */ -/*******************************************************************************/ - -/* direction: time - contents : codewords - raw table: HuffCode2.m - built by : FH 00-02-04 */ - -const INT v_Huff_NoiseLevelC11T[63] = { - 0x00001FCE, 0x00001FCF, 0x00001FD0, 0x00001FD1, 0x00001FD2, 0x00001FD3, 0x00001FD4, 0x00001FD5, - 0x00001FD6, 0x00001FD7, 0x00001FD8, 0x00001FD9, 0x00001FDA, 0x00001FDB, 0x00001FDC, 0x00001FDD, - 0x00001FDE, 0x00001FDF, 0x00001FE0, 0x00001FE1, 0x00001FE2, 0x00001FE3, 0x00001FE4, 0x00001FE5, - 0x00001FE6, 0x00001FE7, 0x000007F2, 0x000000FD, 0x0000003E, 0x0000000E, 0x00000006, 0x00000000, - 0x00000002, 0x0000001E, 0x000000FC, 0x000003F8, 0x00001FCC, 0x00001FE8, 0x00001FE9, 0x00001FEA, - 0x00001FEB, 0x00001FEC, 0x00001FCD, 0x00001FED, 0x00001FEE, 0x00001FEF, 0x00001FF0, 0x00001FF1, - 0x00001FF2, 0x00001FF3, 0x00001FF4, 0x00001FF5, 0x00001FF6, 0x00001FF7, 0x00001FF8, 0x00001FF9, - 0x00001FFA, 0x00001FFB, 0x00001FFC, 0x00001FFD, 0x00001FFE, 0x00003FFE, 0x00003FFF -}; - - -/* direction: time - contents : codeword lengths - raw table: HuffCode2.m - built by : FH 00-02-04 */ - -const UCHAR v_Huff_NoiseLevelL11T[63] = { - 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, - 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, - 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, - 0x0000000D, 0x0000000D, 0x0000000B, 0x00000008, 0x00000006, 0x00000004, 0x00000003, 0x00000001, - 0x00000002, 0x00000005, 0x00000008, 0x0000000A, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, - 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, - 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, - 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000E, 0x0000000E -}; - - -/*******************************************************************************/ -/* table : noise balance, 3.0 dB */ -/* theor range : [-24,24], CODE_BOOK_SCF_LAV = 24 */ -/* implem range: same but mapped to [-12,12], CODE_BOOK_SCF_LAV_BALANCE11 = 12 */ -/* raw stats : noiseBalance_11 KK 00-02-03 */ -/*******************************************************************************/ - -/* direction: time - contents : codewords - raw table: HuffCode3C.m/noiseBalance_11.mat/v_nBhex - built by : FH 01-05-15 */ - -const INT bookSbrNoiseBalanceC11T[25] = -{ - 0x000000EC, 0x000000ED, 0x000000EE, 0x000000EF, 0x000000F0, 0x000000F1, 0x000000F2, 0x000000F3, - 0x000000F4, 0x000000F5, 0x0000001C, 0x00000002, 0x00000000, 0x00000006, 0x0000003A, 0x000000F6, - 0x000000F7, 0x000000F8, 0x000000F9, 0x000000FA, 0x000000FB, 0x000000FC, 0x000000FD, 0x000000FE, - 0x000000FF -}; - - -/* direction: time - contents : codeword lengths - raw table: HuffCode3C.m/noiseBalance_11.mat/v_nLhex - built by : FH 01-05-15 */ - -const UCHAR bookSbrNoiseBalanceL11T[25] = -{ - 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x05, 0x02, 0x01, 0x03, 0x06, 0x08, - 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08 -}; - -/* - tuningTable -*/ -const sbrTuningTable_t sbrTuningTable[] = -{ - /* Some of the low bitrates are commented out here, this is because the - encoder could lose frames at those bitrates and throw an error because - it has insufficient bits to encode for some test items. - */ - - /*** HE-AAC section ***/ - /* sf,sfsp,sf,sfsp,nnb,nfo,saml,SM,FS*/ - - /*** mono ***/ - - /* 8/16 kHz dual rate */ - { CODEC_AAC, 8000, 10000, 8000, 1, 7, 6, 11,10, 1, 0, 6, SBR_MONO, 3 }, - { CODEC_AAC, 10000, 12000, 8000, 1, 11, 7, 13,12, 1, 0, 6, SBR_MONO, 3 }, - { CODEC_AAC, 12000, 16001, 8000, 1, 14,10, 13,13, 1, 0, 6, SBR_MONO, 3 }, - { CODEC_AAC, 16000, 24000, 8000, 1, 14,10, 14,14, 2, 0, 3, SBR_MONO, 2 }, /* placebo */ - { CODEC_AAC, 24000, 32000, 8000, 1, 14,10, 14,14, 2, 0, 3, SBR_MONO, 2 }, /* placebo */ - { CODEC_AAC, 32000, 48001, 8000, 1, 14,11, 15,15, 2, 0, 3, SBR_MONO, 2 }, /* placebo */ /* bitrates higher than 48000 not supported by AAC core */ - - /* 11/22 kHz dual rate */ - { CODEC_AAC, 8000, 10000, 11025, 1, 5, 4, 6, 6, 1, 0, 6, SBR_MONO, 3 }, - { CODEC_AAC, 10000, 12000, 11025, 1, 8, 5, 12, 9, 1, 0, 6, SBR_MONO, 3 }, - { CODEC_AAC, 12000, 16000, 11025, 1, 12, 8, 13, 8, 1, 0, 6, SBR_MONO, 3 }, - { CODEC_AAC, 16000, 20000, 11025, 1, 12, 8, 13, 8, 1, 0, 6, SBR_MONO, 3 }, /* at such "high" bitrates it's better to upsample the input */ - { CODEC_AAC, 20000, 24001, 11025, 1, 13, 9, 13, 8, 1, 0, 6, SBR_MONO, 3 }, /* signal by a factor of 2 before sending it into the encoder */ - { CODEC_AAC, 24000, 32000, 11025, 1, 14,10, 14, 9, 2, 0, 3, SBR_MONO, 2 }, /* placebo */ - { CODEC_AAC, 32000, 48000, 11025, 1, 15,11, 15,10, 2, 0, 3, SBR_MONO, 2 }, /* placebo */ - { CODEC_AAC, 48000, 64001, 11025, 1, 15,11, 15,10, 2, 0, 3, SBR_MONO, 1 }, /* placebo */ - - /* 12/24 kHz dual rate */ - { CODEC_AAC, 8000, 10000, 12000, 1, 4, 3, 6, 6, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 8 kbit/s */ - { CODEC_AAC, 10000, 12000, 12000, 1, 7, 4, 11, 8, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 10 kbit/s */ - { CODEC_AAC, 12000, 16000, 12000, 1, 11, 7, 12, 8, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 12 kbit/s */ - { CODEC_AAC, 16000, 20000, 12000, 1, 11, 7, 12, 8, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 16 kbit/s */ /* at such "high" bitrates it's better to upsample the input */ - { CODEC_AAC, 20000, 24001, 12000, 1, 12, 8, 12, 8, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 20 kbit/s */ /* signal by a factor of 2 before sending it into the encoder */ - { CODEC_AAC, 24000, 32000, 12000, 1, 13, 9, 13, 9, 2, 0, 3, SBR_MONO, 2 }, /* placebo */ - { CODEC_AAC, 32000, 48000, 12000, 1, 14,10, 14,10, 2, 0, 3, SBR_MONO, 2 }, /* placebo */ - { CODEC_AAC, 48000, 64001, 12000, 1, 14,11, 15,11, 2, 0, 3, SBR_MONO, 1 }, /* placebo */ - - /* 16/32 kHz dual rate */ - { CODEC_AAC, 8000, 10000, 16000, 1, 1, 1, 0, 0, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 8 kbit/s */ - { CODEC_AAC, 10000, 12000, 16000, 1, 2, 1, 6, 0, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 10 kbit/s */ - { CODEC_AAC, 12000, 16000, 16000, 1, 4, 2, 6, 0, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 12 kbit/s */ - { CODEC_AAC, 16000, 18000, 16000, 1, 4, 2, 8, 3, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 16 kbit/s */ - { CODEC_AAC, 18000, 22000, 16000, 1, 6, 5,11, 7, 2, 0, 6, SBR_MONO, 2 }, /* nominal: 20 kbit/s */ - { CODEC_AAC, 22000, 28000, 16000, 1, 10, 9,12, 8, 2, 0, 6, SBR_MONO, 2 }, /* nominal: 24 kbit/s */ - { CODEC_AAC, 28000, 36000, 16000, 1, 12,12,13,13, 2, 0, 3, SBR_MONO, 2 }, /* nominal: 32 kbit/s */ - { CODEC_AAC, 36000, 44000, 16000, 1, 14,14,13,13, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */ - { CODEC_AAC, 44000, 64001, 16000, 1, 14,14,13,13, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 48 kbit/s */ - - /* 22.05/44.1 kHz dual rate */ - /* { CODEC_AAC, 8000, 11369, 22050, 1, 1, 1, 1, 1, 1, 0, 6, SBR_MONO, 3 }, */ /* nominal: 8 kbit/s */ /* encoder can not work stable at this extremely low bitrate */ - { CODEC_AAC, 11369, 16000, 22050, 1, 3, 1, 4, 4, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 12 kbit/s */ - { CODEC_AAC, 16000, 18000, 22050, 1, 3, 1, 5, 4, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 16 kbit/s */ - { CODEC_AAC, 18000, 22000, 22050, 1, 4, 4, 8, 5, 2, 0, 6, SBR_MONO, 2 }, /* nominal: 20 kbit/s */ - { CODEC_AAC, 22000, 28000, 22050, 1, 7, 6, 8, 6, 2, 0, 6, SBR_MONO, 2 }, /* nominal: 24 kbit/s */ - { CODEC_AAC, 28000, 36000, 22050, 1, 10,10, 9, 9, 2, 0, 3, SBR_MONO, 2 }, /* nominal: 32 kbit/s */ - { CODEC_AAC, 36000, 44000, 22050, 1, 11,11,10,10, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */ - { CODEC_AAC, 44000, 64001, 22050, 1, 13,13,12,12, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 48 kbit/s */ - - /* 24/48 kHz dual rate */ - /* { CODEC_AAC, 8000, 12000, 24000, 1, 1, 1, 1, 1, 1, 0, 6, SBR_MONO, 3 }, */ /* nominal: 8 kbit/s */ /* encoder can not work stable at this extremely low bitrate */ - { CODEC_AAC, 12000, 16000, 24000, 1, 3, 1, 4, 4, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 12 kbit/s */ - { CODEC_AAC, 16000, 18000, 24000, 1, 3, 1, 5, 4, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 16 kbit/s */ - { CODEC_AAC, 18000, 22000, 24000, 1, 4, 3, 8, 5, 2, 0, 6, SBR_MONO, 2 }, /* nominal: 20 kbit/s */ - { CODEC_AAC, 22000, 28000, 24000, 1, 7, 6, 8, 6, 2, 0, 6, SBR_MONO, 2 }, /* nominal: 24 kbit/s */ - { CODEC_AAC, 28000, 36000, 24000, 1, 10,10, 9, 9, 2, 0, 3, SBR_MONO, 2 }, /* nominal: 32 kbit/s */ - { CODEC_AAC, 36000, 44000, 24000, 1, 11,11,10,10, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */ - { CODEC_AAC, 44000, 64001, 24000, 1, 13,13,11,11, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 48 kbit/s */ - - /* 32/64 kHz dual rate */ /* placebo settings */ - { CODEC_AAC, 24000, 36000, 32000, 1, 4, 4, 4, 4, 2, 0, 3, SBR_MONO, 3 }, /* lowest range */ - { CODEC_AAC, 36000, 60000, 32000, 1, 7, 7, 6, 6, 2, 0, 3, SBR_MONO, 2 }, /* lowest range */ - { CODEC_AAC, 60000, 72000, 32000, 1, 9, 9, 8, 8, 2, 0, 3, SBR_MONO, 1 }, /* low range */ - { CODEC_AAC, 72000,100000, 32000, 1, 11,11,10,10, 2, 0, 3, SBR_MONO, 1 }, /* SBR sweet spot */ - { CODEC_AAC, 100000,160001, 32000, 1, 13,13,11,11, 2, 0, 3, SBR_MONO, 1 }, /* backwards compatible */ - - /* 44.1/88.2 kHz dual rate */ /* placebo settings */ - { CODEC_AAC, 24000, 36000, 44100, 1, 4, 4, 4, 4, 2, 0, 3, SBR_MONO, 3 }, /* lowest range (multichannel rear) */ - { CODEC_AAC, 36000, 60000, 44100, 1, 7, 7, 6, 6, 2, 0, 3, SBR_MONO, 2 }, /* lowest range (multichannel rear) */ - { CODEC_AAC, 60000, 72000, 44100, 1, 9, 9, 8, 8, 2, 0, 3, SBR_MONO, 1 }, /* low range */ - { CODEC_AAC, 72000,100000, 44100, 1, 11,11,10,10, 2, 0, 3, SBR_MONO, 1 }, /* SBR sweet spot */ - { CODEC_AAC, 100000,160001, 44100, 1, 13,13,11,11, 2, 0, 3, SBR_MONO, 1 }, /* backwards compatible */ - - /* 48/96 kHz dual rate */ /* not yet finally tuned */ - { CODEC_AAC, 32000, 36000, 48000, 1, 4, 4, 9, 9, 2, 0, 3, SBR_MONO, 3 }, /* lowest range (multichannel rear) */ - { CODEC_AAC, 36000, 60000, 48000, 1, 7, 7,10,10, 2, 0, 3, SBR_MONO, 2 }, /* nominal: 40 */ - { CODEC_AAC, 60000, 72000, 48000, 1, 9, 9,10,10, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 64 */ - { CODEC_AAC, 72000,100000, 48000, 1, 11,11,11,11, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 80 */ - { CODEC_AAC, 100000,160001, 48000, 1, 13,13,11,11, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 128 */ - - /*** stereo ***/ - /* 08/16 kHz dual rate */ - { CODEC_AAC, 16000, 24000, 8000, 2, 6, 6, 9, 7, 1, 0,-3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */ /* placebo */ - { CODEC_AAC, 24000, 28000, 8000, 2, 9, 9, 11, 9, 1, 0,-3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */ - { CODEC_AAC, 28000, 36000, 8000, 2, 11, 9, 11, 9, 2, 0,-3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */ - { CODEC_AAC, 36000, 44000, 8000, 2, 13,11, 13,11, 2, 0,-3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */ - { CODEC_AAC, 44000, 52000, 8000, 2, 14,12, 13,12, 2, 0,-3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */ - { CODEC_AAC, 52000, 60000, 8000, 2, 14,14, 13,13, 3, 0,-3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */ - { CODEC_AAC, 60000, 76000, 8000, 2, 14,14, 13,13, 3, 0,-3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */ - { CODEC_AAC, 76000,128001, 8000, 2, 14,14, 13,13, 3, 0,-3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */ - - /* 11/22 kHz dual rate */ - { CODEC_AAC, 16000, 24000, 11025, 2, 7, 5, 9, 7, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */ /* placebo */ - { CODEC_AAC, 24000, 28000, 11025, 2, 10, 8,10, 8, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */ - { CODEC_AAC, 28000, 36000, 11025, 2, 12, 8,12, 8, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */ - { CODEC_AAC, 36000, 44000, 11025, 2, 13, 9,13, 9, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */ - { CODEC_AAC, 44000, 52000, 11025, 2, 14,11,13,11, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */ - { CODEC_AAC, 52000, 60000, 11025, 2, 15,15,13,13, 3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */ - { CODEC_AAC, 60000, 76000, 11025, 2, 15,15,13,13, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */ - { CODEC_AAC, 76000,128001, 11025, 2, 15,15,13,13, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */ - - /* 12/24 kHz dual rate */ - { CODEC_AAC, 16000, 24000, 12000, 2, 6, 4, 9, 7, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */ /* placebo */ - { CODEC_AAC, 24000, 28000, 12000, 2, 9, 7,10, 8, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */ - { CODEC_AAC, 28000, 36000, 12000, 2, 11, 7,12, 8, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */ - { CODEC_AAC, 36000, 44000, 12000, 2, 12, 9,12, 9, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */ - { CODEC_AAC, 44000, 52000, 12000, 2, 13,12,13,12, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */ - { CODEC_AAC, 52000, 60000, 12000, 2, 14,14,13,13, 3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */ - { CODEC_AAC, 60000, 76000, 12000, 2, 14,14,13,13, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */ - { CODEC_AAC, 76000,128001, 12000, 2, 14,14,13,13, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */ - - /* 16/32 kHz dual rate */ - { CODEC_AAC, 16000, 24000, 16000, 2, 4, 2, 1, 0, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */ - { CODEC_AAC, 24000, 28000, 16000, 2, 8, 7,10, 8, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */ - { CODEC_AAC, 28000, 36000, 16000, 2, 10, 9,12,11, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */ - { CODEC_AAC, 36000, 44000, 16000, 2, 13,13,13,13, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */ - { CODEC_AAC, 44000, 52000, 16000, 2, 14,14,13,13, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */ - { CODEC_AAC, 52000, 60000, 16000, 2, 14,14,13,13, 3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */ - { CODEC_AAC, 60000, 76000, 16000, 2, 14,14,13,13, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */ - { CODEC_AAC, 76000,128001, 16000, 2, 14,14,13,13, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */ - - /* 22.05/44.1 kHz dual rate */ - { CODEC_AAC, 16000, 24000, 22050, 2, 2, 1, 1, 0, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */ - { CODEC_AAC, 24000, 28000, 22050, 2, 5, 4, 6, 5, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */ - { CODEC_AAC, 28000, 32000, 22050, 2, 5, 4, 8, 7, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 28 kbit/s */ - { CODEC_AAC, 32000, 36000, 22050, 2, 7, 6, 8, 7, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */ - { CODEC_AAC, 36000, 44000, 22050, 2, 10,10, 9, 9, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */ - { CODEC_AAC, 44000, 52000, 22050, 2, 12,12, 9, 9, 3, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */ - { CODEC_AAC, 52000, 60000, 22050, 2, 13,13,10,10, 3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */ - { CODEC_AAC, 60000, 76000, 22050, 2, 14,14,12,12, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */ - { CODEC_AAC, 76000,128001, 22050, 2, 14,14,12,12, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */ - - /* 24/48 kHz dual rate */ - { CODEC_AAC, 16000, 24000, 24000, 2, 2, 1, 1, 0, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */ - { CODEC_AAC, 24000, 28000, 24000, 2, 5, 5, 6, 6, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */ - { CODEC_AAC, 28000, 36000, 24000, 2, 7, 6, 8, 7, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */ - { CODEC_AAC, 36000, 44000, 24000, 2, 10,10, 9, 9, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */ - { CODEC_AAC, 44000, 52000, 24000, 2, 12,12, 9, 9, 3, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */ - { CODEC_AAC, 52000, 60000, 24000, 2, 13,13,10,10, 3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */ - { CODEC_AAC, 60000, 76000, 24000, 2, 14,14,12,12, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */ - { CODEC_AAC, 76000,128001, 24000, 2, 14,14,12,12, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */ - - /* 32/64 kHz dual rate */ /* placebo settings */ - { CODEC_AAC, 32000, 60000, 32000, 2, 4, 4, 4, 4, 2, 0, -3, SBR_SWITCH_LRC, 3 }, /* lowest range (multichannel rear) */ - { CODEC_AAC, 60000, 80000, 32000, 2, 7, 7, 6, 6, 3, 0, -3, SBR_SWITCH_LRC, 2 }, /* lowest range (multichannel rear) */ - { CODEC_AAC, 80000,112000, 32000, 2, 9, 9, 8, 8, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* low range */ - { CODEC_AAC, 112000,144000, 32000, 2, 11,11,10,10, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* SBR sweet spot */ - { CODEC_AAC, 144000,256001, 32000, 2, 13,13,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* backwards compatible */ - - /* 44.1/88.2 kHz dual rate */ /* placebo settings */ - { CODEC_AAC, 32000, 60000, 44100, 2, 4, 4, 4, 4, 2, 0, -3, SBR_SWITCH_LRC, 3 }, /* lowest range (multichannel rear) */ - { CODEC_AAC, 60000, 80000, 44100, 2, 7, 7, 6, 6, 3, 0, -3, SBR_SWITCH_LRC, 2 }, /* lowest range (multichannel rear) */ - { CODEC_AAC, 80000,112000, 44100, 2, 9, 9, 8, 8, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* low range */ - { CODEC_AAC, 112000,144000, 44100, 2, 11,11,10,10, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* SBR sweet spot */ - { CODEC_AAC, 144000,256001, 44100, 2, 13,13,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* backwards compatible */ - - /* 48/96 kHz dual rate */ /* not yet finally tuned */ - { CODEC_AAC, 36000, 60000, 48000, 2, 4, 4, 9, 9, 2, 0, -3, SBR_SWITCH_LRC, 3 }, /* lowest range (multichannel rear) */ - { CODEC_AAC, 60000, 80000, 48000, 2, 7, 7, 9, 9, 3, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 64 */ - { CODEC_AAC, 80000,112000, 48000, 2, 9, 9,10,10, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 96 */ - { CODEC_AAC, 112000,144000, 48000, 2, 11,11,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 128 */ - { CODEC_AAC, 144000,256001, 48000, 2, 13,13,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 192 */ - - - /** AAC LOW DELAY SECTION **/ - - /* 24 kHz dual rate - 12kHz singlerate is not allowed (deactivated in FDKsbrEnc_IsSbrSettingAvail()) */ - { CODEC_AACLD, 8000, 32000, 12000, 1, 1, 1, 0, 0, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 8 kbit/s */ - - /*** mono ***/ - /* 16/32 kHz dual rate not yet tuned ->alb copied from non LD tables*/ - { CODEC_AACLD, 16000, 18000, 16000, 1, 4, 5, 9, 7, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 16 kbit/s wrr: tuned */ - { CODEC_AACLD, 18000, 22000, 16000, 1, 7, 7,12,12, 1, 6, 9, SBR_MONO, 3 }, /* nominal: 20 kbit/s wrr: tuned */ - { CODEC_AACLD, 22000, 28000, 16000, 1, 6, 6, 9, 9, 2, 3, 6, SBR_MONO, 3 }, /* nominal: 24 kbit/s wrr: tuned */ - { CODEC_AACLD, 28000, 36000, 16000, 1, 8, 8,12, 7, 2, 9,12, SBR_MONO, 3 }, /* jgr: special */ /* wrr: tuned */ - { CODEC_AACLD, 36000, 44000, 16000, 1, 10,14,12,13, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */ - { CODEC_AACLD, 44000, 64001, 16000, 1, 11,14,13,13, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 48 kbit/s */ - - /* 22.05/44.1 kHz dual rate */ - { CODEC_AACLD, 18000, 22000, 22050, 1, 4, 4, 5, 5, 2, 0, 6, SBR_MONO, 3 }, /* nominal: 20 kbit/s */ - { CODEC_AACLD, 22000, 28000, 22050, 1, 5, 5, 6, 6, 2, 0, 6, SBR_MONO, 2 }, /* nominal: 24 kbit/s */ - { CODEC_AACLD, 28000, 36000, 22050, 1, 7, 8, 8, 8, 2, 0, 3, SBR_MONO, 2 }, /* nominal: 32 kbit/s */ - { CODEC_AACLD, 36000, 44000, 22050, 1, 9, 9, 9, 9, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */ - { CODEC_AACLD, 44000, 52000, 22050, 1, 12,11,11,11, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 48 kbit/s */ - { CODEC_AACLD, 52000, 64001, 22050, 1, 13,11,11,10, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 56 kbit/s */ - - /* 24/48 kHz dual rate */ - { CODEC_AACLD, 20000, 22000, 24000, 1, 3, 4, 8, 8, 2, 0, 6, SBR_MONO, 2 }, /* nominal: 20 kbit/s */ - { CODEC_AACLD, 22000, 28000, 24000, 1, 3, 8, 8, 7, 2, 0, 3, SBR_MONO, 2 }, /* nominal: 24 kbit/s */ - { CODEC_AACLD, 28000, 36000, 24000, 1, 4, 8, 8, 7, 2, 0, 3, SBR_MONO, 2 }, /* nominal: 32 kbit/s */ - { CODEC_AACLD, 36000, 56000, 24000, 1, 8, 9, 9, 8, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */ - { CODEC_AACLD, 56000, 64001, 24000, 1, 13,11,11,10, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 64 kbit/s */ - - /* 32/64 kHz dual rate */ /* placebo settings */ /*jgr: new, copy from CODEC_AAC */ - { CODEC_AACLD, 24000, 36000, 32000, 1, 4, 4, 4, 4, 2, 0, 3, SBR_MONO, 3 }, /* lowest range */ - { CODEC_AACLD, 36000, 60000, 32000, 1, 7, 7, 6, 6, 2, 0, 3, SBR_MONO, 2 }, /* lowest range */ - { CODEC_AACLD, 60000, 72000, 32000, 1, 9, 9, 8, 8, 2, 0, 3, SBR_MONO, 1 }, /* low range */ - { CODEC_AACLD, 72000,100000, 32000, 1, 11,11,10,10, 2, 0, 3, SBR_MONO, 1 }, /* SBR sweet spot */ - { CODEC_AACLD, 100000,160001, 32000, 1, 13,13,11,11, 2, 0, 3, SBR_MONO, 1 }, /* backwards compatible */ - - /* 44/88 kHz dual rate */ /* not yet finally tuned */ - { CODEC_AACLD, 36000, 60000, 44100, 1, 8, 7, 6, 9, 2, 0, 3, SBR_MONO, 2 }, /* nominal: 40 */ - { CODEC_AACLD, 60000, 72000, 44100, 1, 9, 9,10,10, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 64 */ - { CODEC_AACLD, 72000,100000, 44100, 1, 11,11,11,11, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 80 */ - { CODEC_AACLD, 100000,160001, 44100, 1, 13,13,11,11, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 128 */ - - /* 48/96 kHz dual rate */ /* 32 and 40kbps line tuned for dual-rate SBR */ - { CODEC_AACLD, 36000, 60000, 48000, 1, 4, 7, 4, 4, 2, 0, 3, SBR_MONO, 3 }, /* nominal: 40 */ - { CODEC_AACLD, 60000, 72000, 48000, 1, 9, 9,10,10, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 64 */ - { CODEC_AACLD, 72000,100000, 48000, 1, 11,11,11,11, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 80 */ - { CODEC_AACLD, 100000,160001, 48000, 1, 13,13,11,11, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 128 */ - - /*** stereo ***/ - /* 16/32 kHz dual rate not yet tuned ->alb copied from non LD tables*/ - { CODEC_AACLD, 32000, 36000, 16000, 2, 10, 9,12,11, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */ - { CODEC_AACLD, 36000, 44000, 16000, 2, 13,13,13,13, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */ - { CODEC_AACLD, 44000, 52000, 16000, 2, 10, 9,11, 9, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* tune12 nominal: 48 kbit/s */ - { CODEC_AACLD, 52000, 60000, 16000, 2, 14,14,13,13, 3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */ - { CODEC_AACLD, 60000, 76000, 16000, 2, 14,14,13,13, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */ - { CODEC_AACLD, 76000,128001, 16000, 2, 14,14,13,13, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */ - - /* 22.05/44.1 kHz dual rate */ - { CODEC_AACLD, 32000, 36000, 22050, 2, 5, 4, 7, 6, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */ - { CODEC_AACLD, 36000, 44000, 22050, 2, 5, 8, 8, 8, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */ - { CODEC_AACLD, 44000, 52000, 22050, 2, 7,10, 8, 8, 3, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */ - { CODEC_AACLD, 52000, 60000, 22050, 2, 9,11, 9, 9, 3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */ - { CODEC_AACLD, 60000, 76000, 22050, 2, 10,12,10,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */ - { CODEC_AACLD, 76000, 82000, 22050, 2, 12,12,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */ - { CODEC_AACLD, 82000,128001, 22050, 2, 13,12,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */ - - /* 24/48 kHz dual rate */ - { CODEC_AACLD, 32000, 36000, 24000, 2, 5, 4, 7, 6, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */ - { CODEC_AACLD, 36000, 44000, 24000, 2, 4, 8, 8, 8, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */ - { CODEC_AACLD, 44000, 52000, 24000, 2, 6,10, 8, 8, 3, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */ - { CODEC_AACLD, 52000, 60000, 24000, 2, 9,11, 9, 9, 3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */ - { CODEC_AACLD, 60000, 76000, 24000, 2, 11,12,10,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */ - { CODEC_AACLD, 76000, 88000, 24000, 2, 12,13,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */ - { CODEC_AACLD, 88000,128001, 24000, 2, 13,13,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 92 kbit/s */ - - /* 32/64 kHz dual rate */ /* placebo settings */ /*jgr: new, copy from CODEC_AAC */ - { CODEC_AACLD, 60000, 80000, 32000, 2, 7, 7, 6, 6, 3, 0, -3, SBR_SWITCH_LRC, 2 }, /* lowest range (multichannel rear) */ - { CODEC_AACLD, 80000,112000, 32000, 2, 9, 9, 8, 8, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* low range */ - { CODEC_AACLD, 112000,144000, 32000, 2, 11,11,10,10, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* SBR sweet spot */ - { CODEC_AACLD, 144000,256001, 32000, 2, 13,13,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* backwards compatible */ - - /* 44.1/88.2 kHz dual rate */ /* placebo settings */ /*wrr: new, copy from CODEC_AAC */ - { CODEC_AACLD, 60000, 80000, 44100, 2, 7, 7, 6, 6, 3, 0, -3, SBR_SWITCH_LRC, 2 }, /* lowest range (multichannel rear) */ - { CODEC_AACLD, 80000,112000, 44100, 2, 10,10, 8, 8, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* hlm 11-08-29 */ - { CODEC_AACLD, 112000,144000, 44100, 2, 12,12,10,10, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* hlm 11-08-29 */ - { CODEC_AACLD, 144000,256001, 44100, 2, 13,13,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* backwards compatible */ - - /* 48/96 kHz dual rate */ /* not yet finally tuned */ /*wrr: new, copy from CODEC_AAC */ - { CODEC_AACLD, 60000, 80000, 48000, 2, 7, 7,10,10, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 64 */ - { CODEC_AACLD, 80000,112000, 48000, 2, 9, 9,10,10, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 96 */ - { CODEC_AACLD, 112000,144000, 48000, 2, 11,11,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 128 */ - { CODEC_AACLD, 144000,176000, 48000, 2, 12,12,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* hlm 09-10-19 */ - { CODEC_AACLD, 176000,256001, 48000, 2, 13,13,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* hlm 09-10-19 */ - -}; - -const int sbrTuningTableSize = sizeof(sbrTuningTable)/sizeof(sbrTuningTable[0]); - -const psTuningTable_t psTuningTable[4] = -{ - { 8000, 22000, PSENC_STEREO_BANDS_10, PSENC_NENV_1, FL2FXCONST_DBL(3.0f/4.0f) }, - { 22000, 28000, PSENC_STEREO_BANDS_20, PSENC_NENV_1, FL2FXCONST_DBL(2.0f/4.0f) }, - { 28000, 36000, PSENC_STEREO_BANDS_20, PSENC_NENV_2, FL2FXCONST_DBL(1.5f/4.0f) }, - { 36000, 160001, PSENC_STEREO_BANDS_20, PSENC_NENV_4, FL2FXCONST_DBL(1.1f/4.0f) }, -}; - - -//@} - - - diff --git a/libSBRenc/src/sbr_rom.h b/libSBRenc/src/sbr_rom.h deleted file mode 100644 index afa924e..0000000 --- a/libSBRenc/src/sbr_rom.h +++ /dev/null @@ -1,127 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! -\file -\brief Declaration of constant tables - -*/ -#ifndef __SBR_ROM_H -#define __SBR_ROM_H - -#include "sbr_def.h" -#include "sbr_encoder.h" - -#include "ps_main.h" - -/* - huffman tables -*/ -extern const INT v_Huff_envelopeLevelC10T[121]; -extern const UCHAR v_Huff_envelopeLevelL10T[121]; -extern const INT v_Huff_envelopeLevelC10F[121]; -extern const UCHAR v_Huff_envelopeLevelL10F[121]; -extern const INT bookSbrEnvBalanceC10T[49]; -extern const UCHAR bookSbrEnvBalanceL10T[49]; -extern const INT bookSbrEnvBalanceC10F[49]; -extern const UCHAR bookSbrEnvBalanceL10F[49]; -extern const INT v_Huff_envelopeLevelC11T[63]; -extern const UCHAR v_Huff_envelopeLevelL11T[63]; -extern const INT v_Huff_envelopeLevelC11F[63]; -extern const UCHAR v_Huff_envelopeLevelL11F[63]; -extern const INT bookSbrEnvBalanceC11T[25]; -extern const UCHAR bookSbrEnvBalanceL11T[25]; -extern const INT bookSbrEnvBalanceC11F[25]; -extern const UCHAR bookSbrEnvBalanceL11F[25]; -extern const INT v_Huff_NoiseLevelC11T[63]; -extern const UCHAR v_Huff_NoiseLevelL11T[63]; -extern const INT bookSbrNoiseBalanceC11T[25]; -extern const UCHAR bookSbrNoiseBalanceL11T[25]; - -extern const sbrTuningTable_t sbrTuningTable[]; -extern const int sbrTuningTableSize; - -extern const psTuningTable_t psTuningTable[4]; - - -#endif diff --git a/libSBRenc/src/sbrenc_freq_sca.cpp b/libSBRenc/src/sbrenc_freq_sca.cpp deleted file mode 100644 index 30bc5ca..0000000 --- a/libSBRenc/src/sbrenc_freq_sca.cpp +++ /dev/null @@ -1,691 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief frequency scale - \author Tobias Chalupka -*/ - -#include "sbrenc_freq_sca.h" -#include "sbr_misc.h" - -#include "genericStds.h" - -/* StartFreq */ -static INT getStartFreq(INT fsCore, const INT start_freq); - -/* StopFreq */ -static INT getStopFreq(INT fsCore, const INT stop_freq); - -static INT numberOfBands(INT b_p_o, INT start, INT stop, FIXP_DBL warp_factor); -static void CalcBands(INT * diff, INT start , INT stop , INT num_bands); -static INT modifyBands(INT max_band, INT * diff, INT length); -static void cumSum(INT start_value, INT* diff, INT length, UCHAR *start_adress); - - - -/******************************************************************************* - Functionname: FDKsbrEnc_getSbrStartFreqRAW - ******************************************************************************* - Description: - - Arguments: - - Return: - *******************************************************************************/ - -INT -FDKsbrEnc_getSbrStartFreqRAW (INT startFreq, INT fsCore) -{ - INT result; - - if ( startFreq < 0 || startFreq > 15) { - return -1; - } - /* Update startFreq struct */ - result = getStartFreq(fsCore, startFreq); - - result = (result*(fsCore>>5)+1)>>1; /* (result*fsSBR/QMFbands+1)>>1; */ - - return (result); - -} /* End FDKsbrEnc_getSbrStartFreqRAW */ - - -/******************************************************************************* - Functionname: getSbrStopFreq - ******************************************************************************* - Description: - - Arguments: - - Return: - *******************************************************************************/ -INT FDKsbrEnc_getSbrStopFreqRAW (INT stopFreq, INT fsCore) -{ - INT result; - - if ( stopFreq < 0 || stopFreq > 13) - return -1; - - /* Uppdate stopFreq struct */ - result = getStopFreq(fsCore, stopFreq); - result = (result*(fsCore>>5)+1)>>1; /* (result*fsSBR/QMFbands+1)>>1; */ - - return (result); -} /* End getSbrStopFreq */ - - -/******************************************************************************* - Functionname: getStartFreq - ******************************************************************************* - Description: - - Arguments: fsCore - core sampling rate - - - Return: - *******************************************************************************/ -static INT -getStartFreq(INT fsCore, const INT start_freq) -{ - INT k0_min; - - switch(fsCore){ - case 8000: k0_min = 24; /* (3000 * nQmfChannels / fsSBR ) + 0.5 */ - break; - case 11025: k0_min = 17; /* (3000 * nQmfChannels / fsSBR ) + 0.5 */ - break; - case 12000: k0_min = 16; /* (3000 * nQmfChannels / fsSBR ) + 0.5 */ - break; - case 16000: k0_min = 16; /* (4000 * nQmfChannels / fsSBR ) + 0.5 */ - break; - case 22050: k0_min = 12; /* (4000 * nQmfChannels / fsSBR ) + 0.5 */ - break; - case 24000: k0_min = 11; /* (4000 * nQmfChannels / fsSBR ) + 0.5 */ - break; - case 32000: k0_min = 10; /* (5000 * nQmfChannels / fsSBR ) + 0.5 */ - break; - case 44100: k0_min = 7; /* (5000 * nQmfChannels / fsSBR ) + 0.5 */ - break; - case 48000: k0_min = 7; /* (5000 * nQmfChannels / fsSBR ) + 0.5 */ - break; - case 96000: k0_min = 3; /* (5000 * nQmfChannels / fsSBR ) + 0.5 */ - break; - default: - k0_min=11; /* illegal fs */ - } - - - switch (fsCore) { - - case 8000: - { - INT v_offset[]= {-8, -7, -6, -5, -4, -3, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7}; - return (k0_min + v_offset[start_freq]); - } - case 11025: - { - INT v_offset[]= {-5, -4, -3, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13}; - return (k0_min + v_offset[start_freq]); - } - case 12000: - { - INT v_offset[]= {-5, -3, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13, 16}; - return (k0_min + v_offset[start_freq]); - } - case 16000: - { - INT v_offset[]= {-6, -4, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13, 16}; - return (k0_min + v_offset[start_freq]); - } - case 22050: - case 24000: - case 32000: - { - INT v_offset[]= {-4, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13, 16, 20}; - return (k0_min + v_offset[start_freq]); - } - case 44100: - case 48000: - case 96000: - { - INT v_offset[]= {-2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13, 16, 20, 24}; - return (k0_min + v_offset[start_freq]); - } - default: - { - INT v_offset[]= {0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13, 16, 20, 24, 28, 33}; - return (k0_min + v_offset[start_freq]); - } - } -} /* End getStartFreq */ - - -/******************************************************************************* - Functionname: getStopFreq - ******************************************************************************* - Description: - - Arguments: - - Return: - *******************************************************************************/ - static INT -getStopFreq(INT fsCore, const INT stop_freq) -{ - INT result,i; - INT k1_min; - INT v_dstop[13]; - - INT *v_stop_freq = NULL; - INT v_stop_freq_16[14] = {48,49,50,51,52,54,55,56,57,59,60,61,63,64}; - INT v_stop_freq_22[14] = {35,37,38,40,42,44,46,48,51,53,56,58,61,64}; - INT v_stop_freq_24[14] = {32,34,36,38,40,42,44,46,49,52,55,58,61,64}; - INT v_stop_freq_32[14] = {32,34,36,38,40,42,44,46,49,52,55,58,61,64}; - INT v_stop_freq_44[14] = {23,25,27,29,32,34,37,40,43,47,51,55,59,64}; - INT v_stop_freq_48[14] = {21,23,25,27,30,32,35,38,42,45,49,54,59,64}; - INT v_stop_freq_64[14] = {20,22,24,26,29,31,34,37,41,45,49,54,59,64}; - INT v_stop_freq_88[14] = {15,17,19,21,23,26,29,33,37,41,46,51,57,64}; - INT v_stop_freq_96[14] = {13,15,17,19,21,24,27,31,35,39,44,50,57,64}; - INT v_stop_freq_192[14] = {7, 8,10,12,14,16,19,23,27,32,38,46,54,64}; - - switch(fsCore){ - case 8000: k1_min = 48; - v_stop_freq =v_stop_freq_16; - break; - case 11025: k1_min = 35; - v_stop_freq =v_stop_freq_22; - break; - case 12000: k1_min = 32; - v_stop_freq =v_stop_freq_24; - break; - case 16000: k1_min = 32; - v_stop_freq =v_stop_freq_32; - break; - case 22050: k1_min = 23; - v_stop_freq =v_stop_freq_44; - break; - case 24000: k1_min = 21; - v_stop_freq =v_stop_freq_48; - break; - case 32000: k1_min = 20; - v_stop_freq =v_stop_freq_64; - break; - case 44100: k1_min = 15; - v_stop_freq =v_stop_freq_88; - break; - case 48000: k1_min = 13; - v_stop_freq =v_stop_freq_96; - break; - case 96000: k1_min = 7; - v_stop_freq =v_stop_freq_192; - break; - default: - k1_min = 21; /* illegal fs */ - } - - /* if no valid core samplingrate is used this loop produces - a segfault, because v_stop_freq is not initialized */ - /* Ensure increasing bandwidth */ - for(i = 0; i <= 12; i++) { - v_dstop[i] = v_stop_freq[i+1] - v_stop_freq[i]; - } - - FDKsbrEnc_Shellsort_int(v_dstop, 13); /* Sort bandwidth changes */ - - result = k1_min; - for(i = 0; i < stop_freq; i++) { - result = result + v_dstop[i]; - } - - return(result); - -}/* End getStopFreq */ - - -/******************************************************************************* - Functionname: FDKsbrEnc_FindStartAndStopBand - ******************************************************************************* - Description: - - Arguments: srSbr SBR sampling freqency - srCore AAC core sampling freqency - noChannels Number of QMF channels - startFreq SBR start frequency in QMF bands - stopFreq SBR start frequency in QMF bands - - *k0 Output parameter - *k2 Output parameter - - Return: Error code (0 is OK) - *******************************************************************************/ -INT -FDKsbrEnc_FindStartAndStopBand( - const INT srSbr, - const INT srCore, - const INT noChannels, - const INT startFreq, - const INT stopFreq, - INT *k0, - INT *k2 - ) -{ - - /* Update startFreq struct */ - *k0 = getStartFreq(srCore, startFreq); - - /* Test if start freq is outside corecoder range */ - if( srSbr*noChannels < *k0 * srCore ) { - return (1); /* raise the cross-over frequency and/or lower the number - of target bands per octave (or lower the sampling frequency) */ - } - - /*Update stopFreq struct */ - if ( stopFreq < 14 ) { - *k2 = getStopFreq(srCore, stopFreq); - } else if( stopFreq == 14 ) { - *k2 = 2 * *k0; - } else { - *k2 = 3 * *k0; - } - - /* limit to Nyqvist */ - if (*k2 > noChannels) { - *k2 = noChannels; - } - - - - /* Test for invalid k0 k2 combinations */ - if ( (srCore == 22050) && ( (*k2 - *k0) > MAX_FREQ_COEFFS_FS44100 ) ) - return (1); /* Number of bands exceeds valid range of MAX_FREQ_COEFFS for fs=44.1kHz */ - - if ( (srCore >= 24000) && ( (*k2 - *k0) > MAX_FREQ_COEFFS_FS48000 ) ) - return (1); /* Number of bands exceeds valid range of MAX_FREQ_COEFFS for fs>=48kHz */ - - if ((*k2 - *k0) > MAX_FREQ_COEFFS) - return (1);/*Number of bands exceeds valid range of MAX_FREQ_COEFFS */ - - if ((*k2 - *k0) < 0) - return (1);/* Number of bands is negative */ - - - return(0); -} - -/******************************************************************************* - Functionname: FDKsbrEnc_UpdateFreqScale - ******************************************************************************* - Description: - - Arguments: - - Return: - *******************************************************************************/ -INT -FDKsbrEnc_UpdateFreqScale( - UCHAR *v_k_master, - INT *h_num_bands, - const INT k0, - const INT k2, - const INT freqScale, - const INT alterScale - ) - -{ - - INT b_p_o = 0; /* bands_per_octave */ - FIXP_DBL warp = FL2FXCONST_DBL(0.0f); - INT dk = 0; - - /* Internal variables */ - INT k1 = 0, i; - INT num_bands0; - INT num_bands1; - INT diff_tot[MAX_OCTAVE + MAX_SECOND_REGION]; - INT *diff0 = diff_tot; - INT *diff1 = diff_tot+MAX_OCTAVE; - INT k2_achived; - INT k2_diff; - INT incr = 0; - - /* Init */ - if (freqScale==1) b_p_o = 12; - if (freqScale==2) b_p_o = 10; - if (freqScale==3) b_p_o = 8; - - - if(freqScale > 0) /*Bark*/ - { - if(alterScale==0) - warp = FL2FXCONST_DBL(0.5f); /* 1.0/(1.0*2.0) */ - else - warp = FL2FXCONST_DBL(1.0f/2.6f); /* 1.0/(1.3*2.0); */ - - - if(4*k2 >= 9*k0) /*two or more regions (how many times the basis band is copied)*/ - { - k1=2*k0; - - num_bands0=numberOfBands(b_p_o, k0, k1, FL2FXCONST_DBL(0.5f)); - num_bands1=numberOfBands(b_p_o, k1, k2, warp); - - CalcBands(diff0, k0, k1, num_bands0);/*CalcBands1 => diff0 */ - FDKsbrEnc_Shellsort_int( diff0, num_bands0);/*SortBands sort diff0 */ - - if (diff0[0] == 0) /* too wide FB bands for target tuning */ - { - return (1);/* raise the cross-over frequency and/or lower the number - of target bands per octave (or lower the sampling frequency */ - } - - cumSum(k0, diff0, num_bands0, v_k_master); /* cumsum */ - - CalcBands(diff1, k1, k2, num_bands1); /* CalcBands2 => diff1 */ - FDKsbrEnc_Shellsort_int( diff1, num_bands1); /* SortBands sort diff1 */ - if(diff0[num_bands0-1] > diff1[0]) /* max(1) > min(2) */ - { - if(modifyBands(diff0[num_bands0-1],diff1, num_bands1)) - return(1); - } - - /* Add 2'nd region */ - cumSum(k1, diff1, num_bands1, &v_k_master[num_bands0]); - *h_num_bands=num_bands0+num_bands1; /* Output nr of bands */ - - } - else /* one region */ - { - k1=k2; - - num_bands0=numberOfBands(b_p_o, k0, k1, FL2FXCONST_DBL(0.5f)); - CalcBands(diff0, k0, k1, num_bands0);/* CalcBands1 => diff0 */ - FDKsbrEnc_Shellsort_int( diff0, num_bands0); /* SortBands sort diff0 */ - - if (diff0[0] == 0) /* too wide FB bands for target tuning */ - { - return (1); /* raise the cross-over frequency and/or lower the number - of target bands per octave (or lower the sampling frequency */ - } - - cumSum(k0, diff0, num_bands0, v_k_master);/* cumsum */ - *h_num_bands=num_bands0; /* Output nr of bands */ - - } - } - else /* Linear mode */ - { - if (alterScale==0) { - dk = 1; - num_bands0 = 2 * ((k2 - k0)/2); /* FLOOR to get to few number of bands*/ - } else { - dk = 2; - num_bands0 = 2 * (((k2 - k0)/dk +1)/2); /* ROUND to get closest fit */ - } - - k2_achived = k0 + num_bands0*dk; - k2_diff = k2 - k2_achived; - - for(i=0;i 0) { - incr = -1; - i = num_bands0-1; - } - - /* Adjust diff vector to get sepc. SBR range */ - while (k2_diff != 0) { - diff_tot[i] = diff_tot[i] - incr; - i = i + incr; - k2_diff = k2_diff + incr; - } - - cumSum(k0, diff_tot, num_bands0, v_k_master);/* cumsum */ - *h_num_bands=num_bands0; /* Output nr of bands */ - - } - - if (*h_num_bands < 1) - return(1); /*To small sbr area */ - - return (0); -}/* End FDKsbrEnc_UpdateFreqScale */ - -static INT -numberOfBands(INT b_p_o, INT start, INT stop, FIXP_DBL warp_factor) -{ - INT result=0; - /* result = 2* (INT) ( (double)b_p_o * (double)(FDKlog((double)stop/(double)start)/FDKlog((double)2)) * (double)FX_DBL2FL(warp_factor) + 0.5); */ - result = ( ( b_p_o * fMult( (CalcLdInt(stop) - CalcLdInt(start)), warp_factor) + (FL2FX_DBL(0.5f)>>LD_DATA_SHIFT) - ) >> ((DFRACT_BITS-1)-LD_DATA_SHIFT) ) << 1; /* do not optimize anymore (rounding!!) */ - - return(result); -} - - -static void -CalcBands(INT * diff, INT start , INT stop , INT num_bands) -{ - INT i, qb, qe, qtmp; - INT previous; - INT current; - FIXP_DBL base, exp, tmp; - - previous=start; - for(i=1; i<= num_bands; i++) - { - base = fDivNorm((FIXP_DBL)stop, (FIXP_DBL)start, &qb); - exp = fDivNorm((FIXP_DBL)i, (FIXP_DBL)num_bands, &qe); - tmp = fPow(base, qb, exp, qe, &qtmp); - tmp = fMult(tmp, (FIXP_DBL)(start<<24)); - current = (INT)scaleValue(tmp, qtmp-23); - current = (current+1) >> 1; /* rounding*/ - diff[i-1] = current-previous; - previous = current; - } - -}/* End CalcBands */ - - -static void -cumSum(INT start_value, INT* diff, INT length, UCHAR *start_adress) -{ - INT i; - start_adress[0]=start_value; - for(i=1;i<=length;i++) - start_adress[i]=start_adress[i-1]+diff[i-1]; -} /* End cumSum */ - - -static INT -modifyBands(INT max_band_previous, INT * diff, INT length) -{ - INT change=max_band_previous-diff[0]; - - /* Limit the change so that the last band cannot get narrower than the first one */ - if ( change > (diff[length-1] - diff[0]) / 2 ) - change = (diff[length-1] - diff[0]) / 2; - - diff[0] += change; - diff[length-1] -= change; - FDKsbrEnc_Shellsort_int(diff, length); - - return(0); -}/* End modifyBands */ - - -/******************************************************************************* - Functionname: FDKsbrEnc_UpdateHiRes - ******************************************************************************* - Description: - - - Arguments: - - Return: - *******************************************************************************/ -INT -FDKsbrEnc_UpdateHiRes( - UCHAR *h_hires, - INT *num_hires, - UCHAR *v_k_master, - INT num_master, - INT *xover_band - ) -{ - INT i; - INT max1,max2; - - if( (v_k_master[*xover_band] > 32 ) || /* v_k_master[*xover_band] > noQMFChannels(dualRate)/divider */ - ( *xover_band > num_master ) ) { - /* xover_band error, too big for this startFreq. Will be clipped */ - - /* Calculate maximum value for xover_band */ - max1=0; - max2=num_master; - while( (v_k_master[max1+1] < 32 ) && /* noQMFChannels(dualRate)/divider */ - ( (max1+1) < max2) ) - { - max1++; - } - - *xover_band=max1; - } - - *num_hires = num_master - *xover_band; - for(i = *xover_band; i <= num_master; i++) - { - h_hires[i - *xover_band] = v_k_master[i]; - } - - return (0); -}/* End FDKsbrEnc_UpdateHiRes */ - - -/******************************************************************************* - Functionname: FDKsbrEnc_UpdateLoRes - ******************************************************************************* - Description: - - Arguments: - - Return: - *******************************************************************************/ -void -FDKsbrEnc_UpdateLoRes(UCHAR * h_lores, INT *num_lores, UCHAR * h_hires, INT num_hires) -{ - INT i; - - if(num_hires%2 == 0) /* if even number of hires bands */ - { - *num_lores=num_hires/2; - /* Use every second lores=hires[0,2,4...] */ - for(i=0;i<=*num_lores;i++) - h_lores[i]=h_hires[i*2]; - - } - else /* odd number of hires which means xover is odd */ - { - *num_lores=(num_hires+1)/2; - - /* Use lores=hires[0,1,3,5 ...] */ - h_lores[0]=h_hires[0]; - for(i=1;i<=*num_lores;i++) - { - h_lores[i]=h_hires[i*2-1]; - } - } - -}/* End FDKsbrEnc_UpdateLoRes */ diff --git a/libSBRenc/src/sbrenc_freq_sca.h b/libSBRenc/src/sbrenc_freq_sca.h deleted file mode 100644 index 6f2bb84..0000000 --- a/libSBRenc/src/sbrenc_freq_sca.h +++ /dev/null @@ -1,137 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief frequency scale prototypes -*/ -#ifndef __FREQ_SCA2_H -#define __FREQ_SCA2_H - -#include "sbr_encoder.h" -#include "sbr_def.h" - -#define MAX_OCTAVE 29 -#define MAX_SECOND_REGION 50 - - -INT -FDKsbrEnc_UpdateFreqScale( - UCHAR *v_k_master, - INT *h_num_bands, - const INT k0, - const INT k2, - const INT freq_scale, - const INT alter_scale - ); - -INT -FDKsbrEnc_UpdateHiRes( - UCHAR *h_hires, - INT *num_hires, - UCHAR *v_k_master, - INT num_master, - INT *xover_band - ); - -void FDKsbrEnc_UpdateLoRes( - UCHAR *v_lores, - INT *num_lores, - UCHAR *v_hires, - INT num_hires - ); - -INT -FDKsbrEnc_FindStartAndStopBand( - const INT srSbr, - const INT srCore, - const INT noChannels, - const INT startFreq, - const INT stop_freq, - INT *k0, - INT *k2 - ); - -INT FDKsbrEnc_getSbrStartFreqRAW (INT startFreq, INT fsCore); -INT FDKsbrEnc_getSbrStopFreqRAW (INT stopFreq, INT fsCore); -#endif diff --git a/libSBRenc/src/ton_corr.cpp b/libSBRenc/src/ton_corr.cpp deleted file mode 100644 index af5afba..0000000 --- a/libSBRenc/src/ton_corr.cpp +++ /dev/null @@ -1,881 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -#include "ton_corr.h" - -#include "sbr_ram.h" -#include "sbr_misc.h" -#include "genericStds.h" -#include "autocorr2nd.h" - - - -/*************************************************************************** - - Send autoCorrSecondOrder to mlfile - -****************************************************************************/ - -/**************************************************************************/ -/*! - \brief Calculates the tonal to noise ration for different frequency bands - and time segments. - - The ratio between the predicted energy (tonal energy A) and the total - energy (A + B) is calculated. This is converted to the ratio between - the predicted energy (tonal energy A) and the non-predictable energy - (noise energy B). Hence the quota-matrix contains A/B = q/(1-q). - - The samples in nrgVector are scaled by 1.0/16.0 - The samples in pNrgVectorFreq are scaled by 1.0/2.0 - The samples in quotaMatrix are scaled by RELAXATION - - \return none. - -*/ -/**************************************************************************/ - -void -FDKsbrEnc_CalculateTonalityQuotas( HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Handle to SBR_TON_CORR struct. */ - FIXP_DBL **RESTRICT sourceBufferReal, /*!< The real part of the QMF-matrix. */ - FIXP_DBL **RESTRICT sourceBufferImag, /*!< The imaginary part of the QMF-matrix. */ - INT usb, /*!< upper side band, highest + 1 QMF band in the SBR range. */ - INT qmfScale /*!< sclefactor of QMF subsamples */ - ) -{ - INT i, k, r, r2, timeIndex, autoCorrScaling; - - INT startIndexMatrix = hTonCorr->startIndexMatrix; - INT totNoEst = hTonCorr->numberOfEstimates; - INT noEstPerFrame = hTonCorr->numberOfEstimatesPerFrame; - INT move = hTonCorr->move; - INT noQmfChannels = hTonCorr->noQmfChannels; /* Numer of Bands */ - INT buffLen = hTonCorr->bufferLength; /* Numer of Slots */ - INT stepSize = hTonCorr->stepSize; - INT *pBlockLength = hTonCorr->lpcLength; - INT** RESTRICT signMatrix = hTonCorr->signMatrix; - FIXP_DBL* RESTRICT nrgVector = hTonCorr->nrgVector; - FIXP_DBL** RESTRICT quotaMatrix = hTonCorr->quotaMatrix; - FIXP_DBL* RESTRICT pNrgVectorFreq = hTonCorr->nrgVectorFreq; - -#define BAND_V_SIZE QMF_MAX_TIME_SLOTS -#define NUM_V_COMBINE 8 /* Must be a divisor of 64 and fulfill the ASSERTs below */ - - FIXP_DBL *realBuf; - FIXP_DBL *imagBuf; - - FIXP_DBL alphar[2],alphai[2],fac; - - C_ALLOC_SCRATCH_START(ac, ACORR_COEFS, 1); - C_ALLOC_SCRATCH_START(realBufRef, FIXP_DBL, 2*BAND_V_SIZE*NUM_V_COMBINE); - - realBuf = realBufRef; - imagBuf = realBuf + BAND_V_SIZE*NUM_V_COMBINE; - - - FDK_ASSERT(buffLen <= BAND_V_SIZE); - FDK_ASSERT(sizeof(FIXP_DBL)*NUM_V_COMBINE*BAND_V_SIZE*2 < (1024*sizeof(FIXP_DBL)-sizeof(ACORR_COEFS)) ); - - /* - * Buffering of the quotaMatrix and the quotaMatrixTransp. - *********************************************************/ - for(i = 0 ; i < move; i++){ - FDKmemcpy(quotaMatrix[i],quotaMatrix[i + noEstPerFrame],noQmfChannels * sizeof(FIXP_DBL)); - FDKmemcpy(signMatrix[i],signMatrix[i + noEstPerFrame],noQmfChannels * sizeof(INT)); - } - - FDKmemmove(nrgVector,nrgVector+noEstPerFrame,move*sizeof(FIXP_DBL)); - FDKmemclear(nrgVector+startIndexMatrix,(totNoEst-startIndexMatrix)*sizeof(FIXP_DBL)); - FDKmemclear(pNrgVectorFreq,noQmfChannels * sizeof(FIXP_DBL)); - - /* - * Calculate the quotas for the current time steps. - **************************************************/ - - for (r = 0; r < usb; r++) - { - int blockLength; - - k = hTonCorr->nextSample; /* startSample */ - timeIndex = startIndexMatrix; - /* Copy as many as possible Band accross all Slots at once */ - if (realBuf != realBufRef) { - realBuf -= BAND_V_SIZE; - imagBuf -= BAND_V_SIZE; - } else { - realBuf += BAND_V_SIZE*(NUM_V_COMBINE-1); - imagBuf += BAND_V_SIZE*(NUM_V_COMBINE-1); - for (i = 0; i < buffLen; i++) { - int v; - FIXP_DBL *ptr; - ptr = realBuf+i; - for (v=0; vdet == FL2FXCONST_DBL(0.0f)){ - alphar[1] = alphai[1] = FL2FXCONST_DBL(0.0f); - - alphar[0] = (ac->r01r)>>2; - alphai[0] = (ac->r01i)>>2; - - fac = fMultDiv2(ac->r00r, ac->r11r)>>1; - } - else{ - alphar[1] = (fMultDiv2(ac->r01r, ac->r12r)>>1) - (fMultDiv2(ac->r01i, ac->r12i)>>1) - (fMultDiv2(ac->r02r, ac->r11r)>>1); - alphai[1] = (fMultDiv2(ac->r01i, ac->r12r)>>1) + (fMultDiv2(ac->r01r, ac->r12i)>>1) - (fMultDiv2(ac->r02i, ac->r11r)>>1); - - alphar[0] = (fMultDiv2(ac->r01r, ac->det)>>(ac->det_scale+1)) + fMult(alphar[1], ac->r12r) + fMult(alphai[1], ac->r12i); - alphai[0] = (fMultDiv2(ac->r01i, ac->det)>>(ac->det_scale+1)) + fMult(alphai[1], ac->r12r) - fMult(alphar[1], ac->r12i); - - fac = fMultDiv2(ac->r00r, fMult(ac->det, ac->r11r))>>(ac->det_scale+1); - } - - if(fac == FL2FXCONST_DBL(0.0f)){ - quotaMatrix[timeIndex][r] = FL2FXCONST_DBL(0.0f); - signMatrix[timeIndex][r] = 0; - } - else { - /* quotaMatrix is scaled with the factor RELAXATION - parse RELAXATION in fractional part and shift factor: 1/(1/0.524288 * 2^RELAXATION_SHIFT) */ - FIXP_DBL tmp,num,denom; - INT numShift,denomShift,commonShift; - INT sign; - - num = fMultDiv2(alphar[0], ac->r01r) + fMultDiv2(alphai[0], ac->r01i) - fMultDiv2(alphar[1], fMult(ac->r02r, ac->r11r)) - fMultDiv2(alphai[1], fMult(ac->r02i, ac->r11r)); - num = fixp_abs(num); - - denom = (fac>>1) + (fMultDiv2(fac,RELAXATION_FRACT)>>RELAXATION_SHIFT) - num; - denom = fixp_abs(denom); - - num = fMult(num,RELAXATION_FRACT); - - numShift = CountLeadingBits(num) - 2; - num = scaleValue(num, numShift); - - denomShift = CountLeadingBits(denom); - denom = (FIXP_DBL)denom << denomShift; - - if ((num > FL2FXCONST_DBL(0.0f)) && (denom != FL2FXCONST_DBL(0.0f))) { - commonShift = fixMin(numShift - denomShift + RELAXATION_SHIFT, DFRACT_BITS-1); - if (commonShift < 0) { - commonShift = -commonShift; - tmp = schur_div(num,denom,16); - commonShift = fixMin(commonShift,CountLeadingBits(tmp)); - quotaMatrix[timeIndex][r] = tmp << commonShift; - } - else { - quotaMatrix[timeIndex][r] = schur_div(num,denom,16) >> commonShift; - } - } - else { - quotaMatrix[timeIndex][r] = FL2FXCONST_DBL(0.0f); - } - - if (ac->r11r != FL2FXCONST_DBL(0.0f)) { - if ( ( (ac->r01r >= FL2FXCONST_DBL(0.0f) ) && ( ac->r11r >= FL2FXCONST_DBL(0.0f) ) ) - ||( (ac->r01r < FL2FXCONST_DBL(0.0f) ) && ( ac->r11r < FL2FXCONST_DBL(0.0f) ) ) ) { - sign = 1; - } - else { - sign = -1; - } - } - else { - sign = 1; - } - - if(sign < 0) { - r2 = r; /* (INT) pow(-1, band); */ - } - else { - r2 = r + 1; /* (INT) pow(-1, band+1); */ - } - signMatrix[timeIndex][r] = 1 - 2*(r2 & 0x1); - } - - nrgVector[timeIndex] += ((ac->r00r) >> fixMin(DFRACT_BITS-1,(2*qmfScale+autoCorrScaling + SCALE_NRGVEC))); - /* pNrgVectorFreq[r] finally has to be divided by noEstPerFrame, replaced division by shifting with one */ - pNrgVectorFreq[r] = pNrgVectorFreq[r] + ((ac->r00r) >> fixMin(DFRACT_BITS-1,(2*qmfScale+autoCorrScaling + SCALE_NRGVEC))); - - blockLength = pBlockLength[1]; - k += stepSize; - timeIndex++; - } - } - - - C_ALLOC_SCRATCH_END(realBuf, FIXP_DBL, 2*BAND_V_SIZE*NUM_V_COMBINE); - C_ALLOC_SCRATCH_END(ac, ACORR_COEFS, 1); -} - -/**************************************************************************/ -/*! - \brief Extracts the parameters required in the decoder to obtain the - correct tonal to noise ratio after SBR. - - Estimates the tonal to noise ratio of the original signal (using LPC). - Predicts the tonal to noise ration of the SBR signal (in the decoder) by - patching the tonal to noise ratio values similar to the patching of the - lowband in the decoder. Given the tonal to noise ratio of the original - and the SBR signal, it estimates the required amount of inverse filtering, - additional noise as well as any additional sines. - - \return none. - -*/ -/**************************************************************************/ -void -FDKsbrEnc_TonCorrParamExtr(HANDLE_SBR_TON_CORR_EST hTonCorr,/*!< Handle to SBR_TON_CORR struct. */ - INVF_MODE* infVec, /*!< Vector where the inverse filtering levels will be stored. */ - FIXP_DBL * noiseLevels, /*!< Vector where the noise levels will be stored. */ - INT* missingHarmonicFlag, /*!< Flag set to one or zero, dependent on if any strong sines are missing.*/ - UCHAR * missingHarmonicsIndex, /*!< Vector indicating where sines are missing. */ - UCHAR * envelopeCompensation, /*!< Vector to store compensation values for the energies in. */ - const SBR_FRAME_INFO *frameInfo, /*!< Frame info struct, contains the time and frequency grid of the current frame.*/ - UCHAR* transientInfo, /*!< Transient info.*/ - UCHAR* freqBandTable, /*!< Frequency band tables for high-res.*/ - INT nSfb, /*!< Number of scalefactor bands for high-res. */ - XPOS_MODE xposType, /*!< Type of transposer used in the decoder.*/ - UINT sbrSyntaxFlags - ) -{ - INT band; - INT transientFlag = transientInfo[1] ; /*!< Flag indicating if a transient is present in the current frame. */ - INT transientPos = transientInfo[0]; /*!< Position of the transient.*/ - INT transientFrame, transientFrameInvfEst; - INVF_MODE* infVecPtr; - - - /* Determine if this is a frame where a transient starts... - - The detection of noise-floor, missing harmonics and invf_est, is not in sync for the - non-buf-opt decoder such as AAC. Hence we need to keep track on the transient in the - present frame as well as in the next. - */ - transientFrame = 0; - if(hTonCorr->transientNextFrame){ /* The transient was detected in the previous frame, but is actually */ - transientFrame = 1; - hTonCorr->transientNextFrame = 0; - - if(transientFlag){ - if(transientPos + hTonCorr->transientPosOffset >= frameInfo->borders[frameInfo->nEnvelopes]){ - hTonCorr->transientNextFrame = 1; - } - } - } - else{ - if(transientFlag){ - if(transientPos + hTonCorr->transientPosOffset < frameInfo->borders[frameInfo->nEnvelopes]){ - transientFrame = 1; - hTonCorr->transientNextFrame = 0; - } - else{ - hTonCorr->transientNextFrame = 1; - } - } - } - transientFrameInvfEst = transientFrame; - - - /* - Estimate the required invese filtereing level. - */ - if (hTonCorr->switchInverseFilt) - FDKsbrEnc_qmfInverseFilteringDetector(&hTonCorr->sbrInvFilt, - hTonCorr->quotaMatrix, - hTonCorr->nrgVector, - hTonCorr->indexVector, - hTonCorr->frameStartIndexInvfEst, - hTonCorr->numberOfEstimatesPerFrame + hTonCorr->frameStartIndexInvfEst, - transientFrameInvfEst, - infVec); - - /* - Detect what tones will be missing. - */ - if (xposType == XPOS_LC ){ - FDKsbrEnc_SbrMissingHarmonicsDetectorQmf(&hTonCorr->sbrMissingHarmonicsDetector, - hTonCorr->quotaMatrix, - hTonCorr->signMatrix, - hTonCorr->indexVector, - frameInfo, - transientInfo, - missingHarmonicFlag, - missingHarmonicsIndex, - freqBandTable, - nSfb, - envelopeCompensation, - hTonCorr->nrgVectorFreq); - } - else{ - *missingHarmonicFlag = 0; - FDKmemclear(missingHarmonicsIndex,nSfb*sizeof(UCHAR)); - } - - - - /* - Noise floor estimation - */ - - infVecPtr = hTonCorr->sbrInvFilt.prevInvfMode; - - FDKsbrEnc_sbrNoiseFloorEstimateQmf(&hTonCorr->sbrNoiseFloorEstimate, - frameInfo, - noiseLevels, - hTonCorr->quotaMatrix, - hTonCorr->indexVector, - *missingHarmonicFlag, - hTonCorr->frameStartIndex, - hTonCorr->numberOfEstimatesPerFrame, - transientFrame, - infVecPtr, - sbrSyntaxFlags); - - - /* Store the invfVec data for the next frame...*/ - for(band = 0 ; band < hTonCorr->sbrInvFilt.noDetectorBands; band++){ - hTonCorr->sbrInvFilt.prevInvfMode[band] = infVec[band]; - } -} - -/**************************************************************************/ -/*! - \brief Searches for the closest match in the frequency master table. - - - - \return closest entry. - -*/ -/**************************************************************************/ -static INT -findClosestEntry(INT goalSb, - UCHAR *v_k_master, - INT numMaster, - INT direction) -{ - INT index; - - if( goalSb <= v_k_master[0] ) - return v_k_master[0]; - - if( goalSb >= v_k_master[numMaster] ) - return v_k_master[numMaster]; - - if(direction) { - index = 0; - while( v_k_master[index] < goalSb ) { - index++; - } - } else { - index = numMaster; - while( v_k_master[index] > goalSb ) { - index--; - } - } - - return v_k_master[index]; -} - - -/**************************************************************************/ -/*! - \brief resets the patch - - - - \return errorCode, noError if successful. - -*/ -/**************************************************************************/ -static INT -resetPatch(HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Handle to SBR_TON_CORR struct. */ - INT xposctrl, /*!< Different patch modes. */ - INT highBandStartSb, /*!< Start band of the SBR range. */ - UCHAR *v_k_master, /*!< Master frequency table from which all other table are derived.*/ - INT numMaster, /*!< Number of elements in the master table. */ - INT fs, /*!< Sampling frequency. */ - INT noChannels) /*!< Number of QMF-channels. */ -{ - INT patch,k,i; - INT targetStopBand; - - PATCH_PARAM *patchParam = hTonCorr->patchParam; - - INT sbGuard = hTonCorr->guard; - INT sourceStartBand; - INT patchDistance; - INT numBandsInPatch; - - INT lsb = v_k_master[0]; /* Lowest subband related to the synthesis filterbank */ - INT usb = v_k_master[numMaster]; /* Stop subband related to the synthesis filterbank */ - INT xoverOffset = highBandStartSb - v_k_master[0]; /* Calculate distance in subbands between k0 and kx */ - - INT goalSb; - - - /* - * Initialize the patching parameter - */ - - if (xposctrl == 1) { - lsb += xoverOffset; - xoverOffset = 0; - } - - goalSb = (INT)( (2 * noChannels * 16000 + (fs>>1)) / fs ); /* 16 kHz band */ - goalSb = findClosestEntry(goalSb, v_k_master, numMaster, 1); /* Adapt region to master-table */ - - /* First patch */ - sourceStartBand = hTonCorr->shiftStartSb + xoverOffset; - targetStopBand = lsb + xoverOffset; - - /* even (odd) numbered channel must be patched to even (odd) numbered channel */ - patch = 0; - while(targetStopBand < usb) { - - /* To many patches */ - if (patch >= MAX_NUM_PATCHES) - return(1); /*Number of patches to high */ - - patchParam[patch].guardStartBand = targetStopBand; - targetStopBand += sbGuard; - patchParam[patch].targetStartBand = targetStopBand; - - numBandsInPatch = goalSb - targetStopBand; /* get the desired range of the patch */ - - if ( numBandsInPatch >= lsb - sourceStartBand ) { - /* desired number bands are not available -> patch whole source range */ - patchDistance = targetStopBand - sourceStartBand; /* get the targetOffset */ - patchDistance = patchDistance & ~1; /* rounding off odd numbers and make all even */ - numBandsInPatch = lsb - (targetStopBand - patchDistance); - numBandsInPatch = findClosestEntry(targetStopBand + numBandsInPatch, v_k_master, numMaster, 0) - - targetStopBand; /* Adapt region to master-table */ - } - - /* desired number bands are available -> get the minimal even patching distance */ - patchDistance = numBandsInPatch + targetStopBand - lsb; /* get minimal distance */ - patchDistance = (patchDistance + 1) & ~1; /* rounding up odd numbers and make all even */ - - if (numBandsInPatch <= 0) { - patch--; - } else { - patchParam[patch].sourceStartBand = targetStopBand - patchDistance; - patchParam[patch].targetBandOffs = patchDistance; - patchParam[patch].numBandsInPatch = numBandsInPatch; - patchParam[patch].sourceStopBand = patchParam[patch].sourceStartBand + numBandsInPatch; - - targetStopBand += patchParam[patch].numBandsInPatch; - } - - /* All patches but first */ - sourceStartBand = hTonCorr->shiftStartSb; - - /* Check if we are close to goalSb */ - if( fixp_abs(targetStopBand - goalSb) < 3) { - goalSb = usb; - } - - patch++; - - } - - patch--; - - /* if highest patch contains less than three subband: skip it */ - if ( patchParam[patch].numBandsInPatch < 3 && patch > 0 ) { - patch--; - targetStopBand = patchParam[patch].targetStartBand + patchParam[patch].numBandsInPatch; - } - - hTonCorr->noOfPatches = patch + 1; - - - /* Assign the index-vector, so we know where to look for the high-band. - -1 represents a guard-band. */ - for(k = 0; k < hTonCorr->patchParam[0].guardStartBand; k++) - hTonCorr->indexVector[k] = k; - - for(i = 0; i < hTonCorr->noOfPatches; i++) - { - INT sourceStart = hTonCorr->patchParam[i].sourceStartBand; - INT targetStart = hTonCorr->patchParam[i].targetStartBand; - INT numberOfBands = hTonCorr->patchParam[i].numBandsInPatch; - INT startGuardBand = hTonCorr->patchParam[i].guardStartBand; - - for(k = 0; k < (targetStart- startGuardBand); k++) - hTonCorr->indexVector[startGuardBand+k] = -1; - - for(k = 0; k < numberOfBands; k++) - hTonCorr->indexVector[targetStart+k] = sourceStart+k; - } - - return (0); -} - -/**************************************************************************/ -/*! - \brief Creates an instance of the tonality correction parameter module. - - The module includes modules for inverse filtering level estimation, - missing harmonics detection and noise floor level estimation. - - \return errorCode, noError if successful. -*/ -/**************************************************************************/ -INT -FDKsbrEnc_CreateTonCorrParamExtr(HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Pointer to handle to SBR_TON_CORR struct. */ - INT chan) /*!< Channel index, needed for mem allocation */ -{ - INT i; - FIXP_DBL* quotaMatrix = GetRam_Sbr_quotaMatrix(chan); - INT* signMatrix = GetRam_Sbr_signMatrix(chan); - - FDKmemclear(hTonCorr, sizeof(SBR_TON_CORR_EST)); - - for (i=0; iquotaMatrix[i] = quotaMatrix + (i*QMF_CHANNELS); - hTonCorr->signMatrix[i] = signMatrix + (i*QMF_CHANNELS); - } - - FDKsbrEnc_CreateSbrMissingHarmonicsDetector (&hTonCorr->sbrMissingHarmonicsDetector, chan); - - return 0; -} - - - -/**************************************************************************/ -/*! - \brief Initialize an instance of the tonality correction parameter module. - - The module includes modules for inverse filtering level estimation, - missing harmonics detection and noise floor level estimation. - - \return errorCode, noError if successful. -*/ -/**************************************************************************/ -INT -FDKsbrEnc_InitTonCorrParamExtr (INT frameSize, /*!< Current SBR frame size. */ - HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Pointer to handle to SBR_TON_CORR struct. */ - HANDLE_SBR_CONFIG_DATA sbrCfg, /*!< Pointer to SBR configuration parameters. */ - INT timeSlots, /*!< Number of time-slots per frame */ - INT xposCtrl, /*!< Different patch modes. */ - INT ana_max_level, /*!< Maximum level of the adaptive noise. */ - INT noiseBands, /*!< Number of noise bands per octave. */ - INT noiseFloorOffset, /*!< Noise floor offset. */ - UINT useSpeechConfig) /*!< Speech or music tuning. */ -{ - INT nCols = sbrCfg->noQmfSlots; - INT fs = sbrCfg->sampleFreq; - INT noQmfChannels = sbrCfg->noQmfBands; - - INT highBandStartSb = sbrCfg->freqBandTable[LOW_RES][0]; - UCHAR *v_k_master = sbrCfg->v_k_master; - INT numMaster = sbrCfg->num_Master; - - UCHAR **freqBandTable = sbrCfg->freqBandTable; - INT *nSfb = sbrCfg->nSfb; - - INT i; - - /* - Reset the patching and allocate memory for the quota matrix. - Assuming parameters for the LPC analysis. - */ - if (sbrCfg->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) { - switch (timeSlots) { - case NUMBER_TIME_SLOTS_1920: - hTonCorr->lpcLength[0] = 8 - LPC_ORDER; - hTonCorr->lpcLength[1] = 7 - LPC_ORDER; - hTonCorr->numberOfEstimates = NO_OF_ESTIMATES_LD; - hTonCorr->numberOfEstimatesPerFrame = 2; /* sbrCfg->noQmfSlots / 7 */ - hTonCorr->frameStartIndexInvfEst = 0; - hTonCorr->transientPosOffset = FRAME_MIDDLE_SLOT_512LD; - break; - case NUMBER_TIME_SLOTS_2048: - hTonCorr->lpcLength[0] = 8 - LPC_ORDER; - hTonCorr->lpcLength[1] = 8 - LPC_ORDER; - hTonCorr->numberOfEstimates = NO_OF_ESTIMATES_LD; - hTonCorr->numberOfEstimatesPerFrame = 2; /* sbrCfg->noQmfSlots / 8 */ - hTonCorr->frameStartIndexInvfEst = 0; - hTonCorr->transientPosOffset = FRAME_MIDDLE_SLOT_512LD; - break; - } - } else - switch (timeSlots) { - case NUMBER_TIME_SLOTS_2048: - hTonCorr->lpcLength[0] = 16 - LPC_ORDER; /* blockLength[0] */ - hTonCorr->lpcLength[1] = 16 - LPC_ORDER; /* blockLength[0] */ - hTonCorr->numberOfEstimates = NO_OF_ESTIMATES_LC; - hTonCorr->numberOfEstimatesPerFrame = sbrCfg->noQmfSlots / 16; - hTonCorr->frameStartIndexInvfEst = 0; - hTonCorr->transientPosOffset = FRAME_MIDDLE_SLOT_2048; - break; - case NUMBER_TIME_SLOTS_1920: - hTonCorr->lpcLength[0] = 15 - LPC_ORDER; /* blockLength[0] */ - hTonCorr->lpcLength[1] = 15 - LPC_ORDER; /* blockLength[0] */ - hTonCorr->numberOfEstimates = NO_OF_ESTIMATES_LC; - hTonCorr->numberOfEstimatesPerFrame = sbrCfg->noQmfSlots / 15; - hTonCorr->frameStartIndexInvfEst = 0; - hTonCorr->transientPosOffset = FRAME_MIDDLE_SLOT_1920; - break; - default: - return -1; - } - - hTonCorr->bufferLength = nCols; - hTonCorr->stepSize = hTonCorr->lpcLength[0] + LPC_ORDER; /* stepSize[0] implicitly 0. */ - - hTonCorr->nextSample = LPC_ORDER; /* firstSample */ - hTonCorr->move = hTonCorr->numberOfEstimates - hTonCorr->numberOfEstimatesPerFrame; /* Number of estimates to move when buffering.*/ - hTonCorr->startIndexMatrix = hTonCorr->numberOfEstimates - hTonCorr->numberOfEstimatesPerFrame; /* Where to store the latest estimations in the tonality Matrix.*/ - hTonCorr->frameStartIndex = 0; /* Where in the tonality matrix the current frame (to be sent to the decoder) starts. */ - hTonCorr->prevTransientFlag = 0; - hTonCorr->transientNextFrame = 0; - - hTonCorr->noQmfChannels = noQmfChannels; - - for (i=0; inumberOfEstimates; i++) { - FDKmemclear (hTonCorr->quotaMatrix[i] , sizeof(FIXP_DBL)*noQmfChannels); - FDKmemclear (hTonCorr->signMatrix[i] , sizeof(INT)*noQmfChannels); - } - - /* Reset the patch.*/ - hTonCorr->guard = 0; - hTonCorr->shiftStartSb = 1; - - if(resetPatch(hTonCorr, - xposCtrl, - highBandStartSb, - v_k_master, - numMaster, - fs, - noQmfChannels)) - return(1); - - if(FDKsbrEnc_InitSbrNoiseFloorEstimate (&hTonCorr->sbrNoiseFloorEstimate, - ana_max_level, - freqBandTable[LO], - nSfb[LO], - noiseBands, - noiseFloorOffset, - timeSlots, - useSpeechConfig)) - return(1); - - - if(FDKsbrEnc_initInvFiltDetector(&hTonCorr->sbrInvFilt, - hTonCorr->sbrNoiseFloorEstimate.freqBandTableQmf, - hTonCorr->sbrNoiseFloorEstimate.noNoiseBands, - useSpeechConfig)) - return(1); - - - - if(FDKsbrEnc_InitSbrMissingHarmonicsDetector( - &hTonCorr->sbrMissingHarmonicsDetector, - fs, - frameSize, - nSfb[HI], - noQmfChannels, - hTonCorr->numberOfEstimates, - hTonCorr->move, - hTonCorr->numberOfEstimatesPerFrame, - sbrCfg->sbrSyntaxFlags)) - return(1); - - - - return (0); -} - - - -/**************************************************************************/ -/*! - \brief resets tonality correction parameter module. - - - - \return errorCode, noError if successful. - -*/ -/**************************************************************************/ -INT -FDKsbrEnc_ResetTonCorrParamExtr(HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Handle to SBR_TON_CORR struct. */ - INT xposctrl, /*!< Different patch modes. */ - INT highBandStartSb, /*!< Start band of the SBR range. */ - UCHAR *v_k_master, /*!< Master frequency table from which all other table are derived.*/ - INT numMaster, /*!< Number of elements in the master table. */ - INT fs, /*!< Sampling frequency (of the SBR part). */ - UCHAR ** freqBandTable, /*!< Frequency band table for low-res and high-res. */ - INT* nSfb, /*!< Number of frequency bands (hig-res and low-res). */ - INT noQmfChannels /*!< Number of QMF channels. */ - ) -{ - - /* Reset the patch.*/ - hTonCorr->guard = 0; - hTonCorr->shiftStartSb = 1; - - if(resetPatch(hTonCorr, - xposctrl, - highBandStartSb, - v_k_master, - numMaster, - fs, - noQmfChannels)) - return(1); - - - - /* Reset the noise floor estimate.*/ - if(FDKsbrEnc_resetSbrNoiseFloorEstimate (&hTonCorr->sbrNoiseFloorEstimate, - freqBandTable[LO], - nSfb[LO])) - return(1); - - /* - Reset the inveerse filtereing detector. - */ - if(FDKsbrEnc_resetInvFiltDetector(&hTonCorr->sbrInvFilt, - hTonCorr->sbrNoiseFloorEstimate.freqBandTableQmf, - hTonCorr->sbrNoiseFloorEstimate.noNoiseBands)) - return(1); -/* Reset the missing harmonics detector. */ - if(FDKsbrEnc_ResetSbrMissingHarmonicsDetector (&hTonCorr->sbrMissingHarmonicsDetector, - nSfb[HI])) - return(1); - - return (0); -} - - - - - -/**************************************************************************/ -/*! - \brief Deletes the tonality correction paramtere module. - - - - \return none - -*/ -/**************************************************************************/ -void -FDKsbrEnc_DeleteTonCorrParamExtr (HANDLE_SBR_TON_CORR_EST hTonCorr) /*!< Handle to SBR_TON_CORR struct. */ -{ - - if (hTonCorr) { - - FreeRam_Sbr_quotaMatrix(hTonCorr->quotaMatrix); - - FreeRam_Sbr_signMatrix(hTonCorr->signMatrix); - - FDKsbrEnc_DeleteSbrMissingHarmonicsDetector (&hTonCorr->sbrMissingHarmonicsDetector); - } -} diff --git a/libSBRenc/src/ton_corr.h b/libSBRenc/src/ton_corr.h deleted file mode 100644 index 504ab03..0000000 --- a/libSBRenc/src/ton_corr.h +++ /dev/null @@ -1,212 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief General tonality correction detector module. -*/ -#ifndef _TON_CORR_EST_H -#define _TON_CORR_EST_H - -#include "sbr_encoder.h" -#include "mh_det.h" -#include "nf_est.h" -#include "invf_est.h" - - -#define MAX_NUM_PATCHES 6 -#define SCALE_NRGVEC 4 - -/** parameter set for one single patch */ -typedef struct { - INT sourceStartBand; /*!< first band in lowbands where to take the samples from */ - INT sourceStopBand; /*!< first band in lowbands which is not included in the patch anymore */ - INT guardStartBand; /*!< first band in highbands to be filled with zeros in order to - reduce interferences between patches */ - INT targetStartBand; /*!< first band in highbands to be filled with whitened lowband signal */ - INT targetBandOffs; /*!< difference between 'startTargetBand' and 'startSourceBand' */ - INT numBandsInPatch; /*!< number of consecutive bands in this one patch */ -} PATCH_PARAM; - - - - -typedef struct -{ - INT switchInverseFilt; /*!< Flag to enable dynamic adaption of invf. detection */ - INT noQmfChannels; - INT bufferLength; /*!< Length of the r and i buffers. */ - INT stepSize; /*!< Stride for the lpc estimate. */ - INT numberOfEstimates; /*!< The total number of estiamtes, available in the quotaMatrix.*/ - UINT numberOfEstimatesPerFrame; /*!< The number of estimates per frame available in the quotaMatrix.*/ - INT lpcLength[2]; /*!< Segment length used for second order LPC analysis.*/ - INT nextSample; /*!< Where to start the LPC analysis of the current frame.*/ - INT move; /*!< How many estimates to move in the quotaMatrix, when buffering. */ - INT frameStartIndex; /*!< The start index for the current frame in the r and i buffers. */ - INT startIndexMatrix; /*!< The start index for the current frame in the quotaMatrix. */ - INT frameStartIndexInvfEst; /*!< The start index of the inverse filtering, not the same as the others, - dependent on what decoder is used (buffer opt, or no buffer opt). */ - INT prevTransientFlag; /*!< The transisent flag (from the transient detector) for the previous frame. */ - INT transientNextFrame; /*!< Flag to indicate that the transient will show up in the next frame. */ - INT transientPosOffset; /*!< An offset value to match the transient pos as calculated by the transient detector - with the actual position in the frame.*/ - - INT *signMatrix[MAX_NO_OF_ESTIMATES]; /*!< Matrix holding the sign of each channe, i.e. indicating in what - part of a QMF channel a possible sine is. */ - - FIXP_DBL *quotaMatrix[MAX_NO_OF_ESTIMATES];/*!< Matrix holding the quota values for all estimates, all channels. */ - - FIXP_DBL nrgVector[MAX_NO_OF_ESTIMATES]; /*!< Vector holding the averaged energies for every QMF band. */ - FIXP_DBL nrgVectorFreq[QMF_CHANNELS]; /*!< Vector holding the averaged energies for every QMF channel */ - - SCHAR indexVector[QMF_CHANNELS]; /*!< Index vector poINTing to the correct lowband channel, - when indexing a highband channel, -1 represents a guard band */ - PATCH_PARAM patchParam[MAX_NUM_PATCHES]; /*!< new parameter set for patching */ - INT guard; /*!< number of guardbands between every patch */ - INT shiftStartSb; /*!< lowest subband of source range to be included in the patches */ - INT noOfPatches; /*!< number of patches */ - - SBR_MISSING_HARMONICS_DETECTOR sbrMissingHarmonicsDetector; /*!< SBR_MISSING_HARMONICS_DETECTOR struct. */ - SBR_NOISE_FLOOR_ESTIMATE sbrNoiseFloorEstimate; /*!< SBR_NOISE_FLOOR_ESTIMATE struct. */ - SBR_INV_FILT_EST sbrInvFilt; /*!< SBR_INV_FILT_EST struct. */ -} -SBR_TON_CORR_EST; - -typedef SBR_TON_CORR_EST *HANDLE_SBR_TON_CORR_EST; - -void -FDKsbrEnc_TonCorrParamExtr(HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Handle to SBR_TON_CORR struct. */ - INVF_MODE* infVec, /*!< Vector where the inverse filtering levels will be stored. */ - FIXP_DBL * noiseLevels, /*!< Vector where the noise levels will be stored. */ - INT* missingHarmonicFlag, /*!< Flag set to one or zero, dependent on if any strong sines are missing.*/ - UCHAR* missingHarmonicsIndex, /*!< Vector indicating where sines are missing. */ - UCHAR* envelopeCompensation, /*!< Vector to store compensation values for the energies in. */ - const SBR_FRAME_INFO *frameInfo, /*!< Frame info struct, contains the time and frequency grid of the current frame.*/ - UCHAR* transientInfo, /*!< Transient info.*/ - UCHAR * freqBandTable, /*!< Frequency band tables for high-res.*/ - INT nSfb, /*!< Number of scalefactor bands for high-res. */ - XPOS_MODE xposType, /*!< Type of transposer used in the decoder.*/ - UINT sbrSyntaxFlags - ); - -INT -FDKsbrEnc_CreateTonCorrParamExtr(HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Pointer to handle to SBR_TON_CORR struct. */ - INT chan); /*!< Channel index, needed for mem allocation */ - -INT -FDKsbrEnc_InitTonCorrParamExtr(INT frameSize, /*!< Current SBR frame size. */ - HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Pointer to handle to SBR_TON_CORR struct. */ - HANDLE_SBR_CONFIG_DATA sbrCfg, /*!< Pointer to SBR configuration parameters. */ - INT timeSlots, /*!< Number of time-slots per frame */ - INT xposCtrl, /*!< Different patch modes. */ - INT ana_max_level, /*!< Maximum level of the adaptive noise. */ - INT noiseBands, /*!< Number of noise bands per octave. */ - INT noiseFloorOffset, /*!< Noise floor offset. */ - UINT useSpeechConfig /*!< Speech or music tuning. */ - ); - -void -FDKsbrEnc_DeleteTonCorrParamExtr(HANDLE_SBR_TON_CORR_EST hTonCorr); /*!< Handle to SBR_TON_CORR struct. */ - - -void -FDKsbrEnc_CalculateTonalityQuotas(HANDLE_SBR_TON_CORR_EST hTonCorr, - FIXP_DBL **sourceBufferReal, - FIXP_DBL **sourceBufferImag, - INT usb, - INT qmfScale /*!< sclefactor of QMF subsamples */ - ); - -INT -FDKsbrEnc_ResetTonCorrParamExtr(HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Handle to SBR_TON_CORR struct. */ - INT xposctrl, /*!< Different patch modes. */ - INT highBandStartSb, /*!< Start band of the SBR range. */ - UCHAR *v_k_master, /*!< Master frequency table from which all other table are derived.*/ - INT numMaster, /*!< Number of elements in the master table. */ - INT fs, /*!< Sampling frequency (of the SBR part). */ - UCHAR** freqBandTable, /*!< Frequency band table for low-res and high-res. */ - INT* nSfb, /*!< Number of frequency bands (hig-res and low-res). */ - INT noQmfChannels /*!< Number of QMF channels. */ - ); -#endif - diff --git a/libSBRenc/src/tran_det.cpp b/libSBRenc/src/tran_det.cpp deleted file mode 100644 index 0e35ec3..0000000 --- a/libSBRenc/src/tran_det.cpp +++ /dev/null @@ -1,1069 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -#include "tran_det.h" - -#include "fram_gen.h" -#include "sbr_ram.h" -#include "sbr_misc.h" - -#include "genericStds.h" - -#define NORM_QMF_ENERGY 9.31322574615479E-10 /* 2^-30 */ - -/* static FIXP_DBL ABS_THRES = fixMax( FL2FXCONST_DBL(1.28e5 * NORM_QMF_ENERGY), (FIXP_DBL)1) Minimum threshold for detecting changes */ -#define ABS_THRES ((FIXP_DBL)16) - -/******************************************************************************* - Functionname: spectralChange - ******************************************************************************* - \brief Calculates a measure for the spectral change within the frame - - The function says how good it would be to split the frame at the given border - position into 2 envelopes. - - The return value delta_sum is scaled with the factor 1/64 - - \return calculated value -*******************************************************************************/ -#define NRG_SHIFT 3 /* for energy summation */ - -static FIXP_DBL spectralChange(FIXP_DBL Energies[NUMBER_TIME_SLOTS_2304][MAX_FREQ_COEFFS], - INT *scaleEnergies, - FIXP_DBL EnergyTotal, - INT nSfb, - INT start, - INT border, - INT YBufferWriteOffset, - INT stop, - INT *result_e) -{ - INT i,j; - INT len1,len2; - SCHAR energies_e_diff[NUMBER_TIME_SLOTS_2304], energies_e, energyTotal_e=19, energies_e_add; - SCHAR prevEnergies_e_diff, newEnergies_e_diff; - FIXP_DBL tmp0,tmp1; - FIXP_DBL accu1,accu2,accu1_init,accu2_init; - FIXP_DBL delta, delta_sum; - INT accu_e, tmp_e; - - delta_sum = FL2FXCONST_DBL(0.0f); - *result_e = 0; - - len1 = border-start; - len2 = stop-border; - - /* prefer borders near the middle of the frame */ - FIXP_DBL pos_weight; - pos_weight = FL2FXCONST_DBL(0.5f) - (len1*GetInvInt(len1+len2)); - pos_weight = /*FL2FXCONST_DBL(1.0)*/ (FIXP_DBL)MAXVAL_DBL - (fMult(pos_weight, pos_weight)<<2); - - /*** Calc scaling for energies ***/ - FDK_ASSERT(scaleEnergies[0] >= 0); - FDK_ASSERT(scaleEnergies[1] >= 0); - - energies_e = 19 - FDKmin(scaleEnergies[0], scaleEnergies[1]); - - /* limit shift for energy accumulation, energies_e can be -10 min. */ - if (energies_e < -10) { - energies_e_add = -10 - energies_e; - energies_e = -10; - } else if (energies_e > 17) { - energies_e_add = energies_e - 17; - energies_e = 17; - } else { - energies_e_add = 0; - } - - /* compensate scaling differences between scaleEnergies[0] and scaleEnergies[1] */ - prevEnergies_e_diff = scaleEnergies[0] - FDKmin(scaleEnergies[0], scaleEnergies[1]) + energies_e_add + NRG_SHIFT; - newEnergies_e_diff = scaleEnergies[1] - FDKmin(scaleEnergies[0], scaleEnergies[1]) + energies_e_add + NRG_SHIFT; - - prevEnergies_e_diff = fMin(prevEnergies_e_diff, DFRACT_BITS-1); - newEnergies_e_diff = fMin(newEnergies_e_diff, DFRACT_BITS-1); - - for (i=start; i>=1; - accu2>>=1; - if (accu_e & 1) { - accu_e++; - accu1>>=1; - accu2>>=1; - } - - delta_sum += fMult(sqrtFixp(accu1+accu2), delta); - *result_e = ((accu_e>>1) + LD_DATA_SHIFT); - } - - energyTotal_e+=1; /* for a defined square result exponent, the exponent has to be even */ - EnergyTotal<<=1; - delta_sum = fMult(delta_sum, invSqrtNorm2(EnergyTotal, &tmp_e)); - *result_e = *result_e + (tmp_e-(energyTotal_e>>1)); - - return fMult(delta_sum, pos_weight); - -} - - -/******************************************************************************* - Functionname: addLowbandEnergies - ******************************************************************************* - \brief Calculates total lowband energy - - The input values Energies[0] (low-band) are scaled by the factor - 2^(14-*scaleEnergies[0]) - The input values Energies[1] (high-band) are scaled by the factor - 2^(14-*scaleEnergies[1]) - - \return total energy in the lowband, scaled by the factor 2^19 -*******************************************************************************/ -static FIXP_DBL addLowbandEnergies(FIXP_DBL **Energies, - int *scaleEnergies, - int YBufferWriteOffset, - int nrgSzShift, - int tran_off, - UCHAR *freqBandTable, - int slots) -{ - FIXP_DBL nrgTotal; - FIXP_DBL accu1 = FL2FXCONST_DBL(0.0f); - FIXP_DBL accu2 = FL2FXCONST_DBL(0.0f); - int tran_offdiv2 = tran_off>>nrgSzShift; - int ts,k; - - /* Sum up lowband energy from one frame at offset tran_off */ - /* freqBandTable[LORES] has MAX_FREQ_COEFFS/2 +1 coeefs max. */ - for (ts=tran_offdiv2; ts> 6; - } - } - for (; ts>nrgSzShift); ts++) { - for (k = 0; k < freqBandTable[0]; k++) { - accu2 += Energies[ts][k] >> 9; - } - } - - nrgTotal = ( scaleValueSaturate(accu1, 1-scaleEnergies[0]) ) - + ( scaleValueSaturate(accu2, 4-scaleEnergies[1]) ); - - return(nrgTotal); -} - - -/******************************************************************************* - Functionname: addHighbandEnergies - ******************************************************************************* - \brief Add highband energies - - Highband energies are mapped to an array with smaller dimension: - Its time resolution is only 1 SBR-timeslot and its frequency resolution - is 1 SBR-band. Therefore the data to be fed into the spectralChange - function is reduced. - - The values EnergiesM are scaled by the factor (2^19-scaleEnergies[0]) for - slots=YBufferWriteOffset. - - \return total energy in the highband, scaled by factor 2^19 -*******************************************************************************/ - -static FIXP_DBL addHighbandEnergies(FIXP_DBL **RESTRICT Energies, /*!< input */ - INT *scaleEnergies, - INT YBufferWriteOffset, - FIXP_DBL EnergiesM[NUMBER_TIME_SLOTS_2304][MAX_FREQ_COEFFS], /*!< Combined output */ - UCHAR *RESTRICT freqBandTable, - INT nSfb, - INT sbrSlots, - INT timeStep) -{ - INT i,j,k,slotIn,slotOut,scale[2]; - INT li,ui; - FIXP_DBL nrgTotal; - FIXP_DBL accu = FL2FXCONST_DBL(0.0f); - - /* Combine QMF-timeslots to SBR-timeslots, - combine QMF-bands to SBR-bands, - combine Left and Right channel */ - for (slotOut=0; slotOut>1][k] >> 5); - } - } - EnergiesM[slotOut][j] = accu; - } - } - - /* scale energies down before add up */ - scale[0] = fixMin(8,scaleEnergies[0]); - scale[1] = fixMin(8,scaleEnergies[1]); - - if ((scaleEnergies[0]-scale[0]) > (DFRACT_BITS-1) || (scaleEnergies[1]-scale[0]) > (DFRACT_BITS-1)) - nrgTotal = FL2FXCONST_DBL(0.0f); - else { - /* Now add all energies */ - accu = FL2FXCONST_DBL(0.0f); - - for (slotOut=0; slotOut> scale[0]); - } - } - nrgTotal = accu >> (scaleEnergies[0]-scale[0]); - - for (slotOut=YBufferWriteOffset; slotOut> scale[0]); - } - } - nrgTotal = accu >> (scaleEnergies[1]-scale[1]); - } - - return(nrgTotal); -} - - -/******************************************************************************* - Functionname: FDKsbrEnc_frameSplitter - ******************************************************************************* - \brief Decides if a FIXFIX-frame shall be splitted into 2 envelopes - - If no transient has been detected before, the frame can still be splitted - into 2 envelopes. -*******************************************************************************/ -void -FDKsbrEnc_frameSplitter(FIXP_DBL **Energies, - INT *scaleEnergies, - HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTransientDetector, - UCHAR *freqBandTable, - UCHAR *tran_vector, - int YBufferWriteOffset, - int YBufferSzShift, - int nSfb, - int timeStep, - int no_cols, - FIXP_DBL* tonality) -{ - if (tran_vector[1]==0) /* no transient was detected */ - { - FIXP_DBL delta; - INT delta_e; - FIXP_DBL (*EnergiesM)[MAX_FREQ_COEFFS]; - FIXP_DBL EnergyTotal,newLowbandEnergy,newHighbandEnergy; - INT border; - INT sbrSlots = fMultI(GetInvInt(timeStep),no_cols); - C_ALLOC_SCRATCH_START(_EnergiesM, FIXP_DBL, NUMBER_TIME_SLOTS_2304*MAX_FREQ_COEFFS) - - FDK_ASSERT( sbrSlots * timeStep == no_cols ); - - EnergiesM = (FIXP_DBL(*)[MAX_FREQ_COEFFS])_EnergiesM; - - /* - Get Lowband-energy over a range of 2 frames (Look half a frame back and ahead). - */ - newLowbandEnergy = addLowbandEnergies(Energies, - scaleEnergies, - YBufferWriteOffset, - YBufferSzShift, - h_sbrTransientDetector->tran_off, - freqBandTable, - no_cols); - - newHighbandEnergy = addHighbandEnergies(Energies, - scaleEnergies, - YBufferWriteOffset, - EnergiesM, - freqBandTable, - nSfb, - sbrSlots, - timeStep); - - { - /* prevLowBandEnergy: Corresponds to 1 frame, starting with half a frame look-behind - newLowbandEnergy: Corresponds to 1 frame, starting in the middle of the current frame */ - EnergyTotal = (newLowbandEnergy + h_sbrTransientDetector->prevLowBandEnergy) >> 1; - EnergyTotal += newHighbandEnergy; - /* The below border should specify the same position as the middle border - of a FIXFIX-frame with 2 envelopes. */ - border = (sbrSlots+1) >> 1; - - if ( (INT)EnergyTotal&0xffffffe0 && (scaleEnergies[0]<32 || scaleEnergies[1]<32) ) /* i.e. > 31 */ { - delta = spectralChange(EnergiesM, - scaleEnergies, - EnergyTotal, - nSfb, - 0, - border, - YBufferWriteOffset, - sbrSlots, - &delta_e - ); - } else { - delta = FL2FXCONST_DBL(0.0f); - delta_e = 0; - - /* set tonality to 0 when energy is very low, since the amplitude - resolution should then be low as well */ - *tonality = FL2FXCONST_DBL(0.0f); - } - - - if ( fIsLessThan(h_sbrTransientDetector->split_thr_m, h_sbrTransientDetector->split_thr_e, delta, delta_e) ) { - tran_vector[0] = 1; /* Set flag for splitting */ - } else { - tran_vector[0] = 0; - } - - } - - /* Update prevLowBandEnergy */ - h_sbrTransientDetector->prevLowBandEnergy = newLowbandEnergy; - h_sbrTransientDetector->prevHighBandEnergy = newHighbandEnergy; - C_ALLOC_SCRATCH_END(_EnergiesM, FIXP_DBL, NUMBER_TIME_SLOTS_2304*MAX_FREQ_COEFFS) - } -} - -/* - * Calculate transient energy threshold for each QMF band - */ -static void -calculateThresholds(FIXP_DBL **RESTRICT Energies, - INT *RESTRICT scaleEnergies, - FIXP_DBL *RESTRICT thresholds, - int YBufferWriteOffset, - int YBufferSzShift, - int noCols, - int noRows, - int tran_off) -{ - FIXP_DBL mean_val,std_val,temp; - FIXP_DBL i_noCols; - FIXP_DBL i_noCols1; - FIXP_DBL accu,accu0,accu1; - int scaleFactor0,scaleFactor1,commonScale; - int i,j; - - i_noCols = GetInvInt(noCols + tran_off ) << YBufferSzShift; - i_noCols1 = GetInvInt(noCols + tran_off - 1) << YBufferSzShift; - - /* calc minimum scale of energies of previous and current frame */ - commonScale = fixMin(scaleEnergies[0],scaleEnergies[1]); - - /* calc scalefactors to adapt energies to common scale */ - scaleFactor0 = fixMin((scaleEnergies[0]-commonScale), (DFRACT_BITS-1)); - scaleFactor1 = fixMin((scaleEnergies[1]-commonScale), (DFRACT_BITS-1)); - - FDK_ASSERT((scaleFactor0 >= 0) && (scaleFactor1 >= 0)); - - /* calculate standard deviation in every subband */ - for (i=0; i>YBufferSzShift); - int endEnergy = ((noCols>>YBufferSzShift)+tran_off); - int shift; - - /* calculate mean value over decimated energy values (downsampled by 2). */ - accu0 = accu1 = FL2FXCONST_DBL(0.0f); - - for (j=startEnergy; j> scaleFactor0) + (accu1 >> scaleFactor1); /* average */ - shift = fixMax(0,CountLeadingBits(mean_val)-6); /* -6 to keep room for accumulating upto N = 24 values */ - - /* calculate standard deviation */ - accu = FL2FXCONST_DBL(0.0f); - - /* summe { ((mean_val-nrg)^2) * i_noCols1 } */ - for (j=startEnergy; j> scaleFactor0))<> scaleFactor1))<>shift; /* standard deviation */ - - /* - Take new threshold as average of calculated standard deviation ratio - and old threshold if greater than absolute threshold - */ - temp = ( commonScale<=(DFRACT_BITS-1) ) - ? fMult(FL2FXCONST_DBL(0.66f), thresholds[i]) + (fMult(FL2FXCONST_DBL(0.34f), std_val) >> commonScale) - : (FIXP_DBL) 0; - - thresholds[i] = fixMax(ABS_THRES,temp); - - FDK_ASSERT(commonScale >= 0); - } -} - -/* - * Calculate transient levels for each QMF time slot. - */ -static void -extractTransientCandidates(FIXP_DBL **RESTRICT Energies, - INT *RESTRICT scaleEnergies, - FIXP_DBL *RESTRICT thresholds, - FIXP_DBL *RESTRICT transients, - int YBufferWriteOffset, - int YBufferSzShift, - int noCols, - int start_band, - int stop_band, - int tran_off, - int addPrevSamples) -{ - FIXP_DBL i_thres; - C_ALLOC_SCRATCH_START(EnergiesTemp, FIXP_DBL, 2*QMF_MAX_TIME_SLOTS); - FIXP_DBL *RESTRICT pEnergiesTemp = EnergiesTemp; - int tmpScaleEnergies0, tmpScaleEnergies1; - int endCond; - int startEnerg,endEnerg; - int i,j,jIndex,jpBM; - - tmpScaleEnergies0 = scaleEnergies[0]; - tmpScaleEnergies1 = scaleEnergies[1]; - - /* Scale value for first energies, upto YBufferWriteOffset */ - tmpScaleEnergies0 = fixMin(tmpScaleEnergies0, MAX_SHIFT_DBL); - /* Scale value for first energies, from YBufferWriteOffset upwards */ - tmpScaleEnergies1 = fixMin(tmpScaleEnergies1, MAX_SHIFT_DBL); - - FDK_ASSERT((tmpScaleEnergies0 >= 0) && (tmpScaleEnergies1 >= 0)); - - /* Keep addPrevSamples extra previous transient candidates. */ - FDKmemmove(transients, transients + noCols - addPrevSamples, (tran_off+addPrevSamples) * sizeof (FIXP_DBL)); - FDKmemclear(transients + tran_off + addPrevSamples, noCols * sizeof (FIXP_DBL)); - - endCond = noCols; /* Amount of new transient values to be calculated. */ - startEnerg = (tran_off-3)>>YBufferSzShift; /* >>YBufferSzShift because of amount of energy values. -3 because of neighbors being watched. */ - endEnerg = ((noCols+ (YBufferWriteOffset<>YBufferSzShift; /* YBufferSzShift shifts because of half energy values. */ - - /* Compute differential values with two different weightings in every subband */ - for (i=start_band; i=256) - i_thres = (LONG)( (LONG)MAXVAL_DBL / ((((LONG)thresholds[i]))+1) )<<(32-24); - else - i_thres = (LONG)MAXVAL_DBL; - - /* Copy one timeslot and de-scale and de-squish */ - if (YBufferSzShift == 1) { - for(j=startEnerg; j>tmpScaleEnergies0; - } - for(; j<=endEnerg; j++) { - FIXP_DBL tmp = Energies[j][i]; - EnergiesTemp[(j<<1)+1] = EnergiesTemp[j<<1] = tmp>>tmpScaleEnergies1; - } - } else { - for(j=startEnerg; j>tmpScaleEnergies0; - } - for(; j<=endEnerg; j++) { - FIXP_DBL tmp = Energies[j][i]; - EnergiesTemp[j] = tmp>>tmpScaleEnergies1; - } - } - - /* Detect peaks in energy values. */ - - jIndex = tran_off; - jpBM = jIndex+addPrevSamples; - - for (j=endCond; j--; jIndex++, jpBM++) - { - - FIXP_DBL delta, tran; - int d; - - delta = (FIXP_DBL)0; - tran = (FIXP_DBL)0; - - for (d=1; d<4; d++) { - delta += pEnergiesTemp[jIndex+d]; /* R */ - delta -= pEnergiesTemp[jIndex-d]; /* L */ - delta -= thres; - - if ( delta > (FIXP_DBL)0 ) { - tran += fMult(i_thres, delta); - } - } - transients[jpBM] += tran; - } - } - C_ALLOC_SCRATCH_END(EnergiesTemp, FIXP_DBL, 2*QMF_MAX_TIME_SLOTS); -} - -void -FDKsbrEnc_transientDetect(HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTran, - FIXP_DBL **Energies, - INT *scaleEnergies, - UCHAR *transient_info, - int YBufferWriteOffset, - int YBufferSzShift, - int timeStep, - int frameMiddleBorder) -{ - int no_cols = h_sbrTran->no_cols; - int qmfStartSample; - int addPrevSamples; - int timeStepShift=0; - int i, cond; - - /* Where to start looking for transients in the transient candidate buffer */ - qmfStartSample = timeStep * frameMiddleBorder; - /* We need to look one value backwards in the transients, so we might need one more previous value. */ - addPrevSamples = (qmfStartSample > 0) ? 0: 1; - - switch (timeStep) { - case 1: timeStepShift = 0; break; - case 2: timeStepShift = 1; break; - case 4: timeStepShift = 2; break; - } - - calculateThresholds(Energies, - scaleEnergies, - h_sbrTran->thresholds, - YBufferWriteOffset, - YBufferSzShift, - h_sbrTran->no_cols, - h_sbrTran->no_rows, - h_sbrTran->tran_off); - - extractTransientCandidates(Energies, - scaleEnergies, - h_sbrTran->thresholds, - h_sbrTran->transients, - YBufferWriteOffset, - YBufferSzShift, - h_sbrTran->no_cols, - 0, - h_sbrTran->no_rows, - h_sbrTran->tran_off, - addPrevSamples ); - - transient_info[0] = 0; - transient_info[1] = 0; - transient_info[2] = 0; - - /* Offset by the amount of additional previous transient candidates being kept. */ - qmfStartSample += addPrevSamples; - - /* Check for transients in second granule (pick the last value of subsequent values) */ - for (i=qmfStartSample; itransients[i] < fMult(FL2FXCONST_DBL(0.9f), h_sbrTran->transients[i - 1]) ) - && (h_sbrTran->transients[i - 1] > h_sbrTran->tran_thr); - - if (cond) { - transient_info[0] = (i - qmfStartSample)>>timeStepShift; - transient_info[1] = 1; - break; - } - } - - if ( h_sbrTran->frameShift != 0) { - /* transient prediction for LDSBR */ - /* Check for transients in first qmf-slots of second frame */ - for (i=qmfStartSample+no_cols; iframeShift; i++) { - - cond = (h_sbrTran->transients[i] < fMult(FL2FXCONST_DBL(0.9f), h_sbrTran->transients[i - 1]) ) - && (h_sbrTran->transients[i - 1] > h_sbrTran->tran_thr); - - if (cond) { - int pos = (int) ( (i - qmfStartSample-no_cols) >> timeStepShift ); - if ((pos < 3) && (transient_info[1]==0)) { - transient_info[2] = 1; - } - break; - } - } - } -} - -int -FDKsbrEnc_InitSbrTransientDetector(HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTransientDetector, - UINT sbrSyntaxFlags, /* SBR syntax flags derived from AOT. */ - INT frameSize, - INT sampleFreq, - sbrConfigurationPtr params, - int tran_fc, - int no_cols, - int no_rows, - int YBufferWriteOffset, - int YBufferSzShift, - int frameShift, - int tran_off) -{ - INT totalBitrate = params->codecSettings.standardBitrate * params->codecSettings.nChannels; - INT codecBitrate = params->codecSettings.bitRate; - FIXP_DBL bitrateFactor_m, framedur_fix; - INT bitrateFactor_e, tmp_e; - - FDKmemclear(h_sbrTransientDetector,sizeof(SBR_TRANSIENT_DETECTOR)); - - h_sbrTransientDetector->frameShift = frameShift; - h_sbrTransientDetector->tran_off = tran_off; - - if(codecBitrate) { - bitrateFactor_m = fDivNorm((FIXP_DBL)totalBitrate, (FIXP_DBL)(codecBitrate<<2),&bitrateFactor_e); - bitrateFactor_e += 2; - } - else { - bitrateFactor_m = FL2FXCONST_DBL(1.0/4.0); - bitrateFactor_e = 2; - } - - framedur_fix = fDivNorm(frameSize, sampleFreq); - - /* The longer the frames, the more often should the FIXFIX- - case transmit 2 envelopes instead of 1. - Frame durations below 10 ms produce the highest threshold - so that practically always only 1 env is transmitted. */ - FIXP_DBL tmp = framedur_fix - FL2FXCONST_DBL(0.010); - - tmp = fixMax(tmp, FL2FXCONST_DBL(0.0001)); - tmp = fDivNorm(FL2FXCONST_DBL(0.000075), fPow2(tmp), &tmp_e); - - bitrateFactor_e = (tmp_e + bitrateFactor_e); - - if(sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) { - bitrateFactor_e--; /* divide by 2 */ - } - - FDK_ASSERT(no_cols <= QMF_MAX_TIME_SLOTS); - FDK_ASSERT(no_rows <= QMF_CHANNELS); - - h_sbrTransientDetector->no_cols = no_cols; - h_sbrTransientDetector->tran_thr = (FIXP_DBL)((params->tran_thr << (32-24-1)) / no_rows); - h_sbrTransientDetector->tran_fc = tran_fc; - h_sbrTransientDetector->split_thr_m = fMult(tmp, bitrateFactor_m); - h_sbrTransientDetector->split_thr_e = bitrateFactor_e; - h_sbrTransientDetector->no_rows = no_rows; - h_sbrTransientDetector->mode = params->tran_det_mode; - h_sbrTransientDetector->prevLowBandEnergy = FL2FXCONST_DBL(0.0f); - - return (0); -} - - -#define ENERGY_SCALING_SIZE 32 - -INT FDKsbrEnc_InitSbrFastTransientDetector( - HANDLE_FAST_TRAN_DET h_sbrFastTransientDetector, - const INT time_slots_per_frame, - const INT bandwidth_qmf_slot, - const INT no_qmf_channels, - const INT sbr_qmf_1st_band - ) -{ - - int i, e; - int buff_size; - FIXP_DBL myExp; - FIXP_DBL myExpSlot; - - h_sbrFastTransientDetector->lookahead = TRAN_DET_LOOKAHEAD; - h_sbrFastTransientDetector->nTimeSlots = time_slots_per_frame; - - buff_size = h_sbrFastTransientDetector->nTimeSlots + h_sbrFastTransientDetector->lookahead; - - for(i=0; i< buff_size; i++) { - h_sbrFastTransientDetector->delta_energy[i] = FL2FXCONST_DBL(0.0f); - h_sbrFastTransientDetector->energy_timeSlots[i] = FL2FXCONST_DBL(0.0f); - h_sbrFastTransientDetector->lowpass_energy[i] = FL2FXCONST_DBL(0.0f); - h_sbrFastTransientDetector->transientCandidates[i] = 0; - } - - FDK_ASSERT(bandwidth_qmf_slot > 0.f); - h_sbrFastTransientDetector->stopBand = fMin(TRAN_DET_STOP_FREQ/bandwidth_qmf_slot, no_qmf_channels); - h_sbrFastTransientDetector->startBand = fMin(sbr_qmf_1st_band, h_sbrFastTransientDetector->stopBand - TRAN_DET_MIN_QMFBANDS); - - FDK_ASSERT(h_sbrFastTransientDetector->startBand < no_qmf_channels); - FDK_ASSERT(h_sbrFastTransientDetector->startBand < h_sbrFastTransientDetector->stopBand); - FDK_ASSERT(h_sbrFastTransientDetector->startBand > 1); - FDK_ASSERT(h_sbrFastTransientDetector->stopBand > 1); - - /* the energy weighting and adding up has a headroom of 6 Bits, - so up to 64 bands can be added without potential overflow. */ - FDK_ASSERT(h_sbrFastTransientDetector->stopBand - h_sbrFastTransientDetector->startBand <= 64); - - /* QMF_HP_dB_SLOPE_FIX says that we want a 20 dB per 16 kHz HP filter. - The following lines map this to the QMF bandwidth. */ - #define EXP_E 7 /* QMF_CHANNELS (=64) multiplications max, max. allowed sum is 0.5 */ - myExp = fMultNorm(QMF_HP_dBd_SLOPE_FIX, (FIXP_DBL)bandwidth_qmf_slot, &e); - myExp = scaleValueSaturate(myExp, e+0+DFRACT_BITS-1-EXP_E); - myExpSlot = myExp; - - for(i=0; idBf_m[i] = dBf_m; - h_sbrFastTransientDetector->dBf_e[i] = dBf_e; - - } - - /* Make sure that dBf is greater than 1.0 (because it should be a highpass) */ - /* ... */ - - return 0; -} - -void FDKsbrEnc_fastTransientDetect( - const HANDLE_FAST_TRAN_DET h_sbrFastTransientDetector, - const FIXP_DBL *const *Energies, - const int *const scaleEnergies, - const INT YBufferWriteOffset, - UCHAR *const tran_vector - ) -{ - int timeSlot, band; - - FIXP_DBL max_delta_energy; /* helper to store maximum energy ratio */ - int max_delta_energy_scale; /* helper to store scale of maximum energy ratio */ - int ind_max = 0; /* helper to store index of maximum energy ratio */ - int isTransientInFrame = 0; - - const int nTimeSlots = h_sbrFastTransientDetector->nTimeSlots; - const int lookahead = h_sbrFastTransientDetector->lookahead; - const int startBand = h_sbrFastTransientDetector->startBand; - const int stopBand = h_sbrFastTransientDetector->stopBand; - - int * transientCandidates = h_sbrFastTransientDetector->transientCandidates; - - FIXP_DBL * energy_timeSlots = h_sbrFastTransientDetector->energy_timeSlots; - int * energy_timeSlots_scale = h_sbrFastTransientDetector->energy_timeSlots_scale; - - FIXP_DBL * delta_energy = h_sbrFastTransientDetector->delta_energy; - int * delta_energy_scale = h_sbrFastTransientDetector->delta_energy_scale; - - const FIXP_DBL thr = TRAN_DET_THRSHLD; - const INT thr_scale = TRAN_DET_THRSHLD_SCALE; - - /*reset transient info*/ - tran_vector[2] = 0; - - /* reset transient candidates */ - FDKmemclear(transientCandidates+lookahead, nTimeSlots*sizeof(int)); - - for(timeSlot = lookahead; timeSlot < nTimeSlots + lookahead; timeSlot++) { - int i, norm; - FIXP_DBL tmpE = FL2FXCONST_DBL(0.0f); - int headroomEnSlot = DFRACT_BITS-1; - - FIXP_DBL smallNRG = FL2FXCONST_DBL(1e-2f); - FIXP_DBL denominator; - INT denominator_scale; - - /* determine minimum headroom of energy values for this timeslot */ - for(band = startBand; band < stopBand; band++) { - int tmp_headroom = fNormz(Energies[timeSlot][band])-1; - if(tmp_headroom < headroomEnSlot){ - headroomEnSlot = tmp_headroom; - } - } - - for(i = 0, band = startBand; band < stopBand; band++, i++) { - /* energy is weighted by weightingfactor stored in dBf_m array */ - /* dBf_m index runs from 0 to stopBand-startband */ - /* energy shifted by calculated headroom for maximum precision */ - FIXP_DBL weightedEnergy = fMult(Energies[timeSlot][band]<dBf_m[i]); - - /* energy is added up */ - /* shift by 6 to have a headroom for maximum 64 additions */ - /* shift by dBf_e to handle weighting factor dependent scale factors */ - tmpE += weightedEnergy >> (6 + (10 - h_sbrFastTransientDetector->dBf_e[i])); - } - - /* store calculated energy for timeslot */ - energy_timeSlots[timeSlot] = tmpE; - - /* calculate overall scale factor for energy of this timeslot */ - /* = original scale factor of energies (-scaleEnergies[0]+2*QMF_SCALE_OFFSET or -scaleEnergies[1]+2*QMF_SCALE_OFFSET */ - /* depending on YBufferWriteOffset) */ - /* + weighting factor scale (10) */ - /* + adding up scale factor ( 6) */ - /* - headroom of energy value (headroomEnSlot) */ - if(timeSlot < YBufferWriteOffset){ - energy_timeSlots_scale[timeSlot] = (-scaleEnergies[0]+2*QMF_SCALE_OFFSET) + (10+6) - headroomEnSlot; - } else { - energy_timeSlots_scale[timeSlot] = (-scaleEnergies[1]+2*QMF_SCALE_OFFSET) + (10+6) - headroomEnSlot; - } - - /* Add a small energy to the denominator, thus making the transient - detection energy-dependent. Loud transients are being detected, - silent ones not. */ - - /* make sure that smallNRG does not overflow */ - if ( -energy_timeSlots_scale[timeSlot-1] + 1 > 5 ) - { - denominator = smallNRG; - denominator_scale = 0; - } else { - /* Leave an additional headroom of 1 bit for this addition. */ - smallNRG = scaleValue(smallNRG, -(energy_timeSlots_scale[timeSlot-1] + 1)); - denominator = (energy_timeSlots[timeSlot-1]>>1) + smallNRG; - denominator_scale = energy_timeSlots_scale[timeSlot-1]+1; - } - - delta_energy[timeSlot] = fDivNorm(energy_timeSlots[timeSlot], denominator, &norm); - delta_energy_scale[timeSlot] = energy_timeSlots_scale[timeSlot] - denominator_scale + norm; - } - - /*get transient candidates*/ - /* For every timeslot, check if delta(E) exceeds the threshold. If it did, - it could potentially be marked as a transient candidate. However, the 2 - slots before the current one must not be transients with an energy higher - than 1.4*E(current). If both aren't transients or if the energy of the - current timesolot is more than 1.4 times higher than the energy in the - last or the one before the last slot, it is marked as a transient.*/ - - FDK_ASSERT(lookahead >= 2); - for(timeSlot = lookahead; timeSlot < nTimeSlots + lookahead; timeSlot++) { - FIXP_DBL energy_cur_slot_weighted = fMult(energy_timeSlots[timeSlot],FL2FXCONST_DBL(1.0f/1.4f)); - if( !fIsLessThan(delta_energy[timeSlot], delta_energy_scale[timeSlot], thr, thr_scale) && - ( ((transientCandidates[timeSlot-2]==0) && (transientCandidates[timeSlot-1]==0)) || - !fIsLessThan(energy_cur_slot_weighted, energy_timeSlots_scale[timeSlot], energy_timeSlots[timeSlot-1], energy_timeSlots_scale[timeSlot-1] ) || - !fIsLessThan(energy_cur_slot_weighted, energy_timeSlots_scale[timeSlot], energy_timeSlots[timeSlot-2], energy_timeSlots_scale[timeSlot-2] ) - ) - ) -{ - /* in case of strong transients, subsequent - * qmf slots might be recognized as transients. */ - transientCandidates[timeSlot] = 1; - } - } - - /*get transient with max energy*/ - max_delta_energy = FL2FXCONST_DBL(0.0f); - max_delta_energy_scale = 0; - ind_max = 0; - isTransientInFrame = 0; - for(timeSlot = 0; timeSlot < nTimeSlots; timeSlot++) { - int scale = fMax(delta_energy_scale[timeSlot], max_delta_energy_scale); - if(transientCandidates[timeSlot] && ( (delta_energy[timeSlot] >> (scale - delta_energy_scale[timeSlot])) > (max_delta_energy >> (scale - max_delta_energy_scale)) ) ) { - max_delta_energy = delta_energy[timeSlot]; - max_delta_energy_scale = scale; - ind_max = timeSlot; - isTransientInFrame = 1; - } - } - - /*from all transient candidates take the one with the biggest energy*/ - if(isTransientInFrame) { - tran_vector[0] = ind_max; - tran_vector[1] = 1; - } else { - /*reset transient info*/ - tran_vector[0] = tran_vector[1] = 0; - } - - /*check for transients in lookahead*/ - for(timeSlot = nTimeSlots; timeSlot < nTimeSlots + lookahead; timeSlot++) { - if(transientCandidates[timeSlot]) { - tran_vector[2] = 1; - } - } - - /*update buffers*/ - for(timeSlot = 0; timeSlot < lookahead; timeSlot++) { - transientCandidates[timeSlot] = transientCandidates[nTimeSlots + timeSlot]; - - /* fixpoint stuff */ - energy_timeSlots[timeSlot] = energy_timeSlots[nTimeSlots + timeSlot]; - energy_timeSlots_scale[timeSlot] = energy_timeSlots_scale[nTimeSlots + timeSlot]; - - delta_energy[timeSlot] = delta_energy[nTimeSlots + timeSlot]; - delta_energy_scale[timeSlot] = delta_energy_scale[nTimeSlots + timeSlot]; - } -} - diff --git a/libSBRenc/src/tran_det.h b/libSBRenc/src/tran_det.h deleted file mode 100644 index 6fe1023..0000000 --- a/libSBRenc/src/tran_det.h +++ /dev/null @@ -1,203 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief Transient detector prototypes -*/ -#ifndef __TRAN_DET_H -#define __TRAN_DET_H - -#include "sbr_encoder.h" -#include "sbr_def.h" - -typedef struct -{ - FIXP_DBL transients[QMF_MAX_TIME_SLOTS+(QMF_MAX_TIME_SLOTS/2)]; - FIXP_DBL thresholds[QMF_CHANNELS]; - FIXP_DBL tran_thr; /* Master threshold for transient signals */ - FIXP_DBL split_thr_m; /* Threshold for splitting FIXFIX-frames into 2 env */ - INT split_thr_e; /* Scale for splitting threshold */ - FIXP_DBL prevLowBandEnergy; /* Energy of low band */ - FIXP_DBL prevHighBandEnergy; /* Energy of high band */ - INT tran_fc; /* Number of lowband subbands to discard */ - INT no_cols; - INT no_rows; - INT mode; - - int frameShift; - int tran_off; /* Offset for reading energy values. */ -} -SBR_TRANSIENT_DETECTOR; - - -typedef SBR_TRANSIENT_DETECTOR *HANDLE_SBR_TRANSIENT_DETECTOR; - -#define TRAN_DET_LOOKAHEAD 2 -#define TRAN_DET_START_FREQ 4500 /*start frequency for transient detection*/ -#define TRAN_DET_STOP_FREQ 13500 /*stop frequency for transient detection*/ -#define TRAN_DET_MIN_QMFBANDS 4 /* minimum qmf bands for transient detection */ -#define QMF_HP_dBd_SLOPE_FIX FL2FXCONST_DBL(0.00075275f) /* 0.002266f/10 * log2(10) */ -#define TRAN_DET_THRSHLD FL2FXCONST_DBL(3.2f/4.f) -#define TRAN_DET_THRSHLD_SCALE (2) - -typedef struct -{ - INT transientCandidates[QMF_MAX_TIME_SLOTS + TRAN_DET_LOOKAHEAD]; - INT nTimeSlots; - INT lookahead; - INT startBand; - INT stopBand; - - FIXP_DBL dBf_m[QMF_CHANNELS]; - INT dBf_e[QMF_CHANNELS]; - - FIXP_DBL energy_timeSlots[QMF_MAX_TIME_SLOTS + TRAN_DET_LOOKAHEAD]; - INT energy_timeSlots_scale[QMF_MAX_TIME_SLOTS + TRAN_DET_LOOKAHEAD]; - - FIXP_DBL delta_energy[QMF_MAX_TIME_SLOTS + TRAN_DET_LOOKAHEAD]; - INT delta_energy_scale[QMF_MAX_TIME_SLOTS + TRAN_DET_LOOKAHEAD]; - - FIXP_DBL lowpass_energy[QMF_MAX_TIME_SLOTS + TRAN_DET_LOOKAHEAD]; - INT lowpass_energy_scale[QMF_MAX_TIME_SLOTS + TRAN_DET_LOOKAHEAD]; -#if defined (FTD_LOG) - FDKFILE *ftd_log; -#endif -} -FAST_TRAN_DETECTOR; -typedef FAST_TRAN_DETECTOR *HANDLE_FAST_TRAN_DET; - - -INT FDKsbrEnc_InitSbrFastTransientDetector( - HANDLE_FAST_TRAN_DET h_sbrFastTransientDetector, - const INT time_slots_per_frame, - const INT bandwidth_qmf_slot, - const INT no_qmf_channels, - const INT sbr_qmf_1st_band - ); - -void FDKsbrEnc_fastTransientDetect( - const HANDLE_FAST_TRAN_DET h_sbrFastTransientDetector, - const FIXP_DBL *const *Energies, - const int *const scaleEnergies, - const INT YBufferWriteOffset, - UCHAR *const tran_vector - ); - -void -FDKsbrEnc_transientDetect(HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTransientDetector, - FIXP_DBL **Energies, - INT *scaleEnergies, - UCHAR *tran_vector, - int YBufferWriteOffset, - int YBufferSzShift, - int timeStep, - int frameMiddleBorder); - -int -FDKsbrEnc_InitSbrTransientDetector (HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTransientDetector, - UINT sbrSyntaxFlags, /* SBR syntax flags derived from AOT. */ - INT frameSize, - INT sampleFreq, - sbrConfigurationPtr params, - int tran_fc, - int no_cols, - int no_rows, - int YBufferWriteOffset, - int YBufferSzShift, - int frameShift, - int tran_off); - -void -FDKsbrEnc_frameSplitter(FIXP_DBL **Energies, - INT *scaleEnergies, - HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTransientDetector, - UCHAR *freqBandTable, - UCHAR *tran_vector, - int YBufferWriteOffset, - int YBufferSzShift, - int nSfb, - int timeStep, - int no_cols, - FIXP_DBL* tonality); -#endif diff --git a/wavreader.c b/wavreader.c deleted file mode 100644 index 898eb9c..0000000 --- a/wavreader.c +++ /dev/null @@ -1,193 +0,0 @@ -/* ------------------------------------------------------------------ - * Copyright (C) 2009 Martin Storsjo - * - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either - * express or implied. - * See the License for the specific language governing permissions - * and limitations under the License. - * ------------------------------------------------------------------- - */ - -#include "wavreader.h" -#include -#include -#include -#include - -#define TAG(a, b, c, d) (((a) << 24) | ((b) << 16) | ((c) << 8) | (d)) - -struct wav_reader { - FILE *wav; - uint32_t data_length; - - int format; - int sample_rate; - int bits_per_sample; - int channels; - int byte_rate; - int block_align; - - int streamed; -}; - -static uint32_t read_tag(struct wav_reader* wr) { - uint32_t tag = 0; - tag = (tag << 8) | fgetc(wr->wav); - tag = (tag << 8) | fgetc(wr->wav); - tag = (tag << 8) | fgetc(wr->wav); - tag = (tag << 8) | fgetc(wr->wav); - return tag; -} - -static uint32_t read_int32(struct wav_reader* wr) { - uint32_t value = 0; - value |= fgetc(wr->wav) << 0; - value |= fgetc(wr->wav) << 8; - value |= fgetc(wr->wav) << 16; - value |= fgetc(wr->wav) << 24; - return value; -} - -static uint16_t read_int16(struct wav_reader* wr) { - uint16_t value = 0; - value |= fgetc(wr->wav) << 0; - value |= fgetc(wr->wav) << 8; - return value; -} - -static void skip(FILE *f, int n) { - int i; - for (i = 0; i < n; i++) - fgetc(f); -} - -void* wav_read_open(const char *filename) { - struct wav_reader* wr = (struct wav_reader*) malloc(sizeof(*wr)); - long data_pos = 0; - memset(wr, 0, sizeof(*wr)); - - if (!strcmp(filename, "-")) - wr->wav = stdin; - else - wr->wav = fopen(filename, "rb"); - if (wr->wav == NULL) { - free(wr); - return NULL; - } - - while (1) { - uint32_t tag, tag2, length; - tag = read_tag(wr); - if (feof(wr->wav)) - break; - length = read_int32(wr); - if (!length || length >= 0x7fff0000) { - wr->streamed = 1; - length = ~0; - } - if (tag != TAG('R', 'I', 'F', 'F') || length < 4) { - fseek(wr->wav, length, SEEK_CUR); - continue; - } - tag2 = read_tag(wr); - length -= 4; - if (tag2 != TAG('W', 'A', 'V', 'E')) { - fseek(wr->wav, length, SEEK_CUR); - continue; - } - // RIFF chunk found, iterate through it - while (length >= 8) { - uint32_t subtag, sublength; - subtag = read_tag(wr); - if (feof(wr->wav)) - break; - sublength = read_int32(wr); - length -= 8; - if (length < sublength) - break; - if (subtag == TAG('f', 'm', 't', ' ')) { - if (sublength < 16) { - // Insufficient data for 'fmt ' - break; - } - wr->format = read_int16(wr); - wr->channels = read_int16(wr); - wr->sample_rate = read_int32(wr); - wr->byte_rate = read_int32(wr); - wr->block_align = read_int16(wr); - wr->bits_per_sample = read_int16(wr); - if (wr->format == 0xfffe) { - if (sublength < 28) { - // Insufficient data for waveformatex - break; - } - skip(wr->wav, 8); - wr->format = read_int32(wr); - skip(wr->wav, sublength - 28); - } else { - skip(wr->wav, sublength - 16); - } - } else if (subtag == TAG('d', 'a', 't', 'a')) { - data_pos = ftell(wr->wav); - wr->data_length = sublength; - if (!wr->data_length || wr->streamed) { - wr->streamed = 1; - return wr; - } - fseek(wr->wav, sublength, SEEK_CUR); - } else { - skip(wr->wav, sublength); - } - length -= sublength; - } - if (length > 0) { - // Bad chunk? - fseek(wr->wav, length, SEEK_CUR); - } - } - fseek(wr->wav, data_pos, SEEK_SET); - return wr; -} - -void wav_read_close(void* obj) { - struct wav_reader* wr = (struct wav_reader*) obj; - if (wr->wav != stdin) - fclose(wr->wav); - free(wr); -} - -int wav_get_header(void* obj, int* format, int* channels, int* sample_rate, int* bits_per_sample, unsigned int* data_length) { - struct wav_reader* wr = (struct wav_reader*) obj; - if (format) - *format = wr->format; - if (channels) - *channels = wr->channels; - if (sample_rate) - *sample_rate = wr->sample_rate; - if (bits_per_sample) - *bits_per_sample = wr->bits_per_sample; - if (data_length) - *data_length = wr->data_length; - return wr->format && wr->sample_rate; -} - -int wav_read_data(void* obj, unsigned char* data, unsigned int length) { - struct wav_reader* wr = (struct wav_reader*) obj; - int n; - if (wr->wav == NULL) - return -1; - if (length > wr->data_length && !wr->streamed) - length = wr->data_length; - n = fread(data, 1, length, wr->wav); - wr->data_length -= length; - return n; -} - diff --git a/wavreader.h b/wavreader.h deleted file mode 100644 index 57a13ff..0000000 --- a/wavreader.h +++ /dev/null @@ -1,37 +0,0 @@ -/* ------------------------------------------------------------------ - * Copyright (C) 2009 Martin Storsjo - * - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either - * express or implied. - * See the License for the specific language governing permissions - * and limitations under the License. - * ------------------------------------------------------------------- - */ - -#ifndef WAVREADER_H -#define WAVREADER_H - -#ifdef __cplusplus -extern "C" { -#endif - -void* wav_read_open(const char *filename); -void wav_read_close(void* obj); - -int wav_get_header(void* obj, int* format, int* channels, int* sample_rate, int* bits_per_sample, unsigned int* data_length); -int wav_read_data(void* obj, unsigned char* data, unsigned int length); - -#ifdef __cplusplus -} -#endif - -#endif - -- cgit v1.2.3