diff options
| author | Wim Taymans <wtaymans@redhat.com> | 2017-10-09 16:54:45 +0200 |
|---|---|---|
| committer | Wim Taymans <wtaymans@redhat.com> | 2018-09-25 10:53:45 +0200 |
| commit | 8dcaebbdb5112d5930b6202a8880f89aeabb3722 (patch) | |
| tree | 7d399e758ffdb9659b57813b3321a91009a44fab | |
| parent | a30bfced6b6d6d976c728552d247cb30dd86e238 (diff) | |
Strip minimal decoder and encoder
141 files changed, 54 insertions, 53414 deletions
@@ -9,8 +9,6 @@ cc_library_static { "libSYS/src/*.cpp", "libMpegTPDec/src/*.cpp", "libMpegTPEnc/src/*.cpp", - "libSBRdec/src/*.cpp", - "libSBRenc/src/*.cpp", ], cflags: [ "-Werror", @@ -29,7 +27,5 @@ cc_library_static { "libSYS/include", "libMpegTPDec/include", "libMpegTPEnc/include", - "libSBRdec/include", - "libSBRenc/include", ], } diff --git a/Makefile.am b/Makefile.am index 1ee2dfc..fe998e5 100644 --- a/Makefile.am +++ b/Makefile.am @@ -4,8 +4,6 @@ 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 \ @@ -44,24 +42,17 @@ 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 = \ @@ -73,7 +64,6 @@ AACENC_SRC = \ 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 \ @@ -87,15 +77,12 @@ AACENC_SRC = \ 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 = \ @@ -133,45 +120,6 @@ 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 \ @@ -181,7 +129,6 @@ SYS_SRC = \ libfdk_aac_la_SOURCES = \ $(AACDEC_SRC) $(AACENC_SRC) \ $(MPEGTPDEC_SRC) $(MPEGTPENC_SRC) \ - $(SBRDEC_SRC) $(SBRENC_SRC) \ $(PCMUTILS_SRC) $(FDK_SRC) $(SYS_SRC) EXTRA_DIST = \ @@ -196,11 +143,6 @@ EXTRA_DIST = \ $(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 \ diff --git a/Makefile.vc b/Makefile.vc index 6fccb9c..42dc764 100644 --- a/Makefile.vc +++ b/Makefile.vc @@ -21,8 +21,6 @@ AM_CPPFLAGS = \ -Iwin32 \ -IlibAACdec/include \ -IlibAACenc/include \ - -IlibSBRdec/include \ - -IlibSBRenc/include \ -IlibMpegTPDec/include \ -IlibMpegTPEnc/include \ -IlibSYS/include \ @@ -31,24 +29,17 @@ AM_CPPFLAGS = \ 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 = \ @@ -120,45 +111,6 @@ 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 \ @@ -168,7 +120,6 @@ SYS_SRC = \ libfdk_aac_SOURCES = \ $(AACDEC_SRC) $(AACENC_SRC) \ $(MPEGTPDEC_SRC) $(MPEGTPENC_SRC) \ - $(SBRDEC_SRC) $(SBRENC_SRC) \ $(PCMUTILS_SRC) $(FDK_SRC) $(SYS_SRC) @@ -212,8 +163,6 @@ clean: 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) 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..5ba4160 100644 --- a/libAACdec/src/aac_rom.cpp +++ b/libAACdec/src/aac_rom.cpp @@ -1265,119 +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 */ @@ -1762,79 +1649,6 @@ WTC(0xfe99e090), WTC(0xfed6d644), WTC(0xff0e8f6e), WTC(0xff40f667), WTC(0xff6df5 WTC(0xffea32f4), WTC(0xfffae64c), WTC(0x0005aff3), WTC(0x000a7a44), WTC(0x00092f9c), WTC(0x0001ba54), WTC(0xfff404ca), WTC(0xffdff957) }; - - - -/* - * 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..b9e9899 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 */ @@ -139,8 +137,7 @@ typedef struct 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 +155,6 @@ 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..beb97ce 100644 --- a/libAACdec/src/aacdec_drc.cpp +++ b/libAACdec/src/aacdec_drc.cpp @@ -94,8 +94,6 @@ amm-info@iis.fraunhofer.de #include "channelinfo.h" #include "aac_rom.h" - #include "sbrdecoder.h" - /* * Dynamic Range Control */ @@ -844,13 +842,11 @@ 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 +877,6 @@ 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 +1037,6 @@ void aacDecoder_drcApply ( * short blocks must take care that bands fall on * block boundaries! */ - if (!bSbrPresent) { bottom = 0; @@ -1081,23 +1075,6 @@ 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 <MODULO_DIVISOR_HCR>-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 <input> is smaller, one bufferlength is added, - if <input> 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..74c04f7 100644 --- a/libAACdec/src/aacdec_pns.cpp +++ b/libAACdec/src/aacdec_pns.cpp @@ -103,115 +103,6 @@ amm-info@iis.fraunhofer.de #define NOISE_OFFSET 90 /* cf. ISO 14496-3 p. 175 */ /*! - \brief Reset InterChannel and PNS data - - The function resets the InterChannel and PNS data -*/ -void CPns_ResetData( - CPnsData *pPnsData, - CPnsInterChannelData *pPnsInterChannelData - ) -{ - /* Assign pointer always, since pPnsData is not persistent data */ - pPnsData->pPnsInterChannelData = pPnsInterChannelData; - pPnsData->PnsActive = 0; - pPnsData->CurrentEnergy = 0; - - FDKmemclear(pPnsData->pnsUsed,(8*16)*sizeof(UCHAR)); - FDKmemclear(pPnsInterChannelData->correlated,(8*16)*sizeof(UCHAR)); -} - -/*! - \brief Initialize PNS data - - The function initializes the PNS data -*/ -void CPns_InitPns( - CPnsData *pPnsData, - CPnsInterChannelData *pPnsInterChannelData, - INT* currentSeed, INT* randomSeed) -{ - /* save pointer to inter channel data */ - pPnsData->pPnsInterChannelData = pPnsInterChannelData; - - /* use pointer because seed has to be - same, left and right channel ! */ - pPnsData->currentSeed = currentSeed; - pPnsData->randomSeed = randomSeed; -} - -/*! - \brief Indicates if PNS is used - - The function returns a value indicating whether PNS is used or not - acordding to the noise energy - - \return PNS used -*/ -int CPns_IsPnsUsed (const CPnsData *pPnsData, - const int group, - const int band) -{ - unsigned pns_band = group*16+band; - - return pPnsData->pnsUsed[pns_band] & (UCHAR)1; -} - -/*! - \brief Set correlation - - The function activates the noise correlation between the channel pair -*/ -void CPns_SetCorrelation(CPnsData *pPnsData, - const int group, - const int band, - const int outofphase) -{ - CPnsInterChannelData *pInterChannelData = pPnsData->pPnsInterChannelData; - unsigned pns_band = group*16+band; - - pInterChannelData->correlated[pns_band] = (outofphase) ? 3 : 1; -} - -/*! - \brief Indicates if correlation is used - - The function indicates if the noise correlation between the channel pair - is activated - - \return PNS is correlated -*/ -static -int CPns_IsCorrelated(const CPnsData *pPnsData, - const int group, - const int band) -{ - CPnsInterChannelData *pInterChannelData = pPnsData->pPnsInterChannelData; - unsigned pns_band = group*16+band; - - return (pInterChannelData->correlated[pns_band] & 0x01) ? 1 : 0; -} - -/*! - \brief Indicates if correlated out of phase mode is used. - - The function indicates if the noise correlation between the channel pair - is activated in out-of-phase mode. - - \return PNS is out-of-phase -*/ -static -int CPns_IsOutOfPhase(const CPnsData *pPnsData, - const int group, - const int band) -{ - CPnsInterChannelData *pInterChannelData = pPnsData->pPnsInterChannelData; - unsigned pns_band = group*16+band; - - return (pInterChannelData->correlated[pns_band] & 0x02) ? 1 : 0; -} - -/*! \brief Read PNS information The function reads the PNS information from the bitstream @@ -244,139 +135,3 @@ void CPns_Read (CPnsData *pPnsData, pPnsData->pnsUsed[pns_band] = 1; } - - -/** - * \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<size; i++) - { - randomState = (1664525L * randomState) + 1013904223L; // Numerical Recipes - nrg_m = fPow2AddDiv2(nrg_m, (FIXP_DBL)randomState>>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 - - The function applies PNS (i.e. it generates noise) on the bands - flagged as noisy bands - -*/ -void CPns_Apply (const CPnsData *pPnsData, - const CIcsInfo *pIcsInfo, - SPECTRAL_PTR pSpectrum, - const SHORT *pSpecScale, - const SHORT *pScaleFactor, - const SamplingRateInfo *pSamplingRateInfo, - const INT granuleLength, - const int channel) -{ - 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) ; - } - } - } - } - } -} diff --git a/libAACdec/src/aacdec_pns.h b/libAACdec/src/aacdec_pns.h index fc9bdcb..953975f 100644 --- a/libAACdec/src/aacdec_pns.h +++ b/libAACdec/src/aacdec_pns.h @@ -93,7 +93,6 @@ amm-info@iis.fraunhofer.de #include "common_fix.h" - #define NO_OFBANDS ((8*16)) typedef struct { @@ -105,16 +104,6 @@ typedef struct { UCHAR pnsUsed[NO_OFBANDS]; int CurrentEnergy; UCHAR PnsActive; - INT *currentSeed; - INT *randomSeed; } CPnsData; -void CPns_InitPns ( CPnsData *pPnsData, - CPnsInterChannelData *pPnsInterChannelData, - INT* currentSeed, - INT* randomSeed ); - -void CPns_ResetData ( CPnsData *pPnsData, CPnsInterChannelData *pPnsInterChannelData ); - - #endif /* #ifndef PNS_H */ 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<size; i++) { - maxVal = fixMax(maxVal,fixp_abs(spec[i])); - } - - if ( maxVal > 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) + fMultDiv2 (*pCoeff++, pState [0]) ; - -#define INNER_FLTR_INLINE \ - x = fMultAddDiv2 (x, *pCoeff, pState [1]); \ - pState [0] = pState [1] - (fMultDiv2 (*pCoeff++, x) <<2) ; \ - pState++; - -#define LAST_PART_FLTR \ - if (s < 0) \ - *pSpec = x << 1; \ - else \ - *pSpec = x >> 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..b5f04ea 100644 --- a/libAACdec/src/aacdecoder.cpp +++ b/libAACdec/src/aacdecoder.cpp @@ -144,19 +144,9 @@ 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 +171,6 @@ 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 +503,7 @@ 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 +513,7 @@ 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 +699,6 @@ 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 +709,7 @@ LINKSPEC_CPP HANDLE_AACDECODER CAacDecoder_Open(TRANSPORT_TYPE bsFormat) /*!< aacDecoder_drcSetParam ( self->hDrcInfo, DRC_BS_DELAY, - CConcealment_GetDelay(&self->concealCommonData) + 0 ); @@ -835,18 +787,15 @@ 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 +901,6 @@ 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 +912,7 @@ 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; @@ -980,11 +924,6 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_Init(HANDLE_AACDECODER self, const CS self->flags |= AC_SCALABLE; } - - if (asc->m_sbrPresentFlag) { - self->sbrEnabled = 1; - self->sbrEnabledPrev = 1; - } if (asc->m_psPresentFlag) { self->flags |= AC_PS_PRESENT; } @@ -1051,7 +990,6 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_Init(HANDLE_AACDECODER self, const CS self->pAacDecoderChannelInfo[ch]->pSpectralCoefficient = (SPECTRAL_PTR) &self->aacCommonData.workBufferCore2[ch*1024]; } - CPns_InitPns(&self->pAacDecoderChannelInfo[ch]->data.aac.PnsData, &self->aacCommonData.pnsInterChannelData, &self->aacCommonData.pnsCurrentSeed, self->aacCommonData.pnsRandomSeed); } if (ascChannels > self->aacChannels) @@ -1059,9 +997,6 @@ 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 +1026,6 @@ 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 +1120,6 @@ 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 +1147,6 @@ 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; @@ -1249,14 +1171,6 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame( break; } - if ( !(self->flags & (AC_USAC|AC_RSVD50)) ) { - int ch; - for (ch=0; ch < el_channels; ch+=1) { - CPns_ResetData(&self->pAacDecoderChannelInfo[aacChannels+ch]->data.aac.PnsData, - &self->pAacDecoderChannelInfo[aacChannels+ch]->pComData->pnsInterChannelData); - } - } - if(self->frameOK) { ErrorStatus = CChannelElement_Read( bs, &self->pAacDecoderChannelInfo[aacChannels], @@ -1306,27 +1220,6 @@ 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]++; @@ -1358,9 +1251,6 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame( /* Assume AAC-LC */ tmpAacDecoderChannelInfo.granuleLength = self->streamInfo.aacSamplesPerFrame / 8; - /* Reset PNS data. */ - CPns_ResetData(&tmpAacDecoderChannelInfo.data.aac.PnsData, &tmpAacDecoderChannelInfo.pComData->pnsInterChannelData); - pTmpAacDecoderChannelInfo = &tmpAacDecoderChannelInfo; /* do CCE parsing */ ErrorStatus = CChannelElement_Read( bs, @@ -1497,42 +1387,12 @@ 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 +1490,11 @@ 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 +1578,6 @@ 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 +1587,11 @@ 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 +1640,6 @@ 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..37b7b4b 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" @@ -154,10 +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 { @@ -197,18 +190,12 @@ 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..ba35c8d 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; @@ -666,13 +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 ); @@ -692,7 +600,7 @@ 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 +644,6 @@ 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 +750,6 @@ 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; @@ -907,86 +810,11 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame( CAacDecoder_SyncQmfMode(self); -/* 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<<UPS_SCALE)); - - /* Update data in streaminfo structure. Assume that the SBR upsampling factor is either 1 or 2 */ - self->flags |= AC_SBR_PRESENT; - if (self->streamInfo.aacSampleRate != self->streamInfo.sampleRate) { - if (self->streamInfo.frameSize == 768) { - upsampleFactor = FL2FXCONST_DBL(8.0f/(3<<UPS_SCALE)); - } else { - upsampleFactor = FL2FXCONST_DBL(2.0f/(1<<UPS_SCALE)); - } - } - /* Apply upsampling factor to both the core frame length and the core delay */ - self->streamInfo.frameSize = (INT)fMult((FIXP_DBL)self->streamInfo.aacSamplesPerFrame<<UPS_SCALE, upsampleFactor); - self->streamInfo.outputDelay = (UINT)(INT)fMult((FIXP_DBL)self->streamInfo.outputDelay<<UPS_SCALE, upsampleFactor); - self->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 +912,6 @@ 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 +933,6 @@ 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 8bee2d4..94df2c4 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 */ @@ -586,39 +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)) ) { @@ -639,20 +596,6 @@ void ApplyTools ( CAacDecoderChannelInfo *pAacDecoderChannelInfo[], const UINT flags, const int channel ) { - - if ( !(flags & (AC_USAC|AC_RSVD50|AC_MPS_RES)) ) { - CPns_Apply( - &pAacDecoderChannelInfo[channel]->data.aac.PnsData, - &pAacDecoderChannelInfo[channel]->icsInfo, - pAacDecoderChannelInfo[channel]->pSpectralCoefficient, - pAacDecoderChannelInfo[channel]->specScale, - pAacDecoderChannelInfo[channel]->pDynData->aScaleFactor, - pSamplingRateInfo, - pAacDecoderChannelInfo[channel]->granuleLength, - channel - ); - } - CTns_Apply ( &pAacDecoderChannelInfo[channel]->pDynData->TnsData, &pAacDecoderChannelInfo[channel]->icsInfo, diff --git a/libAACdec/src/block.h b/libAACdec/src/block.h index f9394f6..741d6f2 100644 --- a/libAACdec/src/block.h +++ b/libAACdec/src/block.h @@ -98,25 +98,13 @@ amm-info@iis.fraunhofer.de /* PNS (of block) */ void CPns_Read (CPnsData *pPnsData, - HANDLE_FDK_BITSTREAM bs, + HANDLE_FDK_BITSTREAM bs, const CodeBookDescription *hcb, SHORT *pScaleFactor, UCHAR global_gain, int band, int group); - -void CPns_Apply (const CPnsData *pPnsData, - const CIcsInfo *pIcsInfo, - SPECTRAL_PTR pSpectrum, - const SHORT *pSpecScale, - const SHORT *pScaleFactor, - const SamplingRateInfo *pSamplingRateInfo, - const INT granuleLength, - const int channel); - - - /* TNS (of block) */ /*! \brief Read tns data-present flag from bitstream diff --git a/libAACdec/src/channel.cpp b/libAACdec/src/channel.cpp index 4b182e0..5debd32 100644 --- a/libAACdec/src/channel.cpp +++ b/libAACdec/src/channel.cpp @@ -91,36 +91,9 @@ 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 -void MapMidSideMaskToPnsCorrelation (CAacDecoderChannelInfo *pAacDecoderChannelInfo[2]) -{ - int group; - - for (group = 0 ; group < pAacDecoderChannelInfo[L]->icsInfo.WindowGroups; group++) { - UCHAR groupMask = 1 << group; - - for (UCHAR band = 0 ; band < pAacDecoderChannelInfo[L]->icsInfo.MaxSfBands; band++) { - if (pAacDecoderChannelInfo[L]->pComData->jointStereoData.MsUsed[band] & groupMask) { /* channels are correlated */ - CPns_SetCorrelation(&pAacDecoderChannelInfo[L]->data.aac.PnsData, group, band, 0); - - if (CPns_IsPnsUsed(&pAacDecoderChannelInfo[L]->data.aac.PnsData, group, band) && - CPns_IsPnsUsed(&pAacDecoderChannelInfo[R]->data.aac.PnsData, group, band)) - pAacDecoderChannelInfo[L]->pComData->jointStereoData.MsUsed[band] ^= groupMask; /* clear the groupMask-bit */ - } - } - } -} /*! \brief Decode channel pair element @@ -154,9 +127,6 @@ void CChannelElement_Decode( CAacDecoderChannelInfo *pAacDecoderChannelInfo[2], if (pAacDecoderChannelInfo[L]->pDynData->RawDataInfo.CommonWindow) { int maxSfBandsL = GetScaleFactorBandsTransmitted(&pAacDecoderChannelInfo[L]->icsInfo); int maxSfBandsR = GetScaleFactorBandsTransmitted(&pAacDecoderChannelInfo[R]->icsInfo); - if (pAacDecoderChannelInfo[L]->data.aac.PnsData.PnsActive || pAacDecoderChannelInfo[R]->data.aac.PnsData.PnsActive) { - MapMidSideMaskToPnsCorrelation(pAacDecoderChannelInfo); - } CJointStereo_ApplyMS(pAacDecoderChannelInfo, GetScaleFactorBandOffsets(&pAacDecoderChannelInfo[L]->icsInfo, pSamplingRateInfo), @@ -186,13 +156,6 @@ void CChannelElement_Decode( CAacDecoderChannelInfo *pAacDecoderChannelInfo[2], } } - - CRvlc_ElementCheck( - pAacDecoderChannelInfo, - pAacDecoderStaticChannelInfo, - flags, - el_channels - ); } void CChannel_CodebookTableInit(CAacDecoderChannelInfo *pAacDecoderChannelInfo) @@ -330,8 +293,8 @@ 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); + /* RVLC not supported */ + error = AAC_DEC_DECODE_FRAME_ERROR; } else { @@ -369,17 +332,13 @@ 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..f69223a 100644 --- a/libAACdec/src/channelinfo.h +++ b/libAACdec/src/channelinfo.h @@ -105,11 +105,6 @@ 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 +188,6 @@ typedef struct CDrcChannelData drcData; - CConcealmentInfo concealmentInfo; } CAacDecoderStaticChannelInfo; @@ -214,8 +208,6 @@ 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; @@ -241,23 +233,7 @@ typedef struct { CWorkBufferCore1 *workBufferCore1; FIXP_DBL* workBufferCore2; - CPnsInterChannelData pnsInterChannelData; - INT pnsCurrentSeed; - 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; @@ -361,16 +337,6 @@ void CJointStereo_ApplyIS(CAacDecoderChannelInfo *pAacDecoderChannelInfo[2], const UINT CommonWindow); -/* aacdec_pns.cpp */ -int CPns_IsPnsUsed (const CPnsData *pPnsData, - const int group, - const int band); - -void CPns_SetCorrelation(CPnsData *pPnsData, - const int group, - const int band, - const int outofphase); - /****************** inline functions ******************/ inline UCHAR IsValid(const CIcsInfo *pIcsInfo) @@ -440,11 +406,5 @@ 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. - - <h2>How concealment can be used with errors on the transport layer</h2> - - 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<<CONCEAL_PARAMETER_BITS)-1 ) -/*#define CONCEAL_MIN_ATTENUATION_FACTOR_025 ( FL2FXCONST_DBL(0.971627951577106174) )*/ /* -0.25 dB */ -#define CONCEAL_MIN_ATTENUATION_FACTOR_025_LD FL2FXCONST_DBL(-0.041524101186092029596853445212299) -/*#define CONCEAL_MIN_ATTENUATION_FACTOR_050 ( FL2FXCONST_DBL(0.944060876285923380) )*/ /* -0.50 dB */ -#define CONCEAL_MIN_ATTENUATION_FACTOR_050_LD FL2FXCONST_DBL(-0.083048202372184059253597008145293) - -typedef enum { - CConcealment_NoExpand, - CConcealment_Expand, - CConcealment_Compress -} -CConcealmentExpandType; - -static const FIXP_SGL facMod4Table[4] = { - FL2FXCONST_SGL(0.500000000f), /* FIXP_SGL(0x4000), 2^-(1-0,00) */ - FL2FXCONST_SGL(0.594603558f), /* FIXP_SGL(0x4c1b), 2^-(1-0,25) */ - FL2FXCONST_SGL(0.707106781f), /* FIXP_SGL(0x5a82), 2^-(1-0,50) */ - FL2FXCONST_SGL(0.840896415f) /* FIXP_SGL(0x6ba2) 2^-(1-0,75) */ -}; - - - - -static void - CConcealment_CalcBandEnergy ( - FIXP_DBL *spectrum, - const SamplingRateInfo *pSamplingRateInfo, - const int blockType, - CConcealmentExpandType ex, - int *sfbEnergy - ); - -static void - CConcealment_InterpolateBuffer ( - FIXP_DBL *spectrum, - SHORT *pSpecScalePrev, - SHORT *pSpecScaleAct, - SHORT *pSpecScaleOut, - int *enPrv, - int *enAct, - int sfbCnt, - const SHORT *pSfbOffset - ); - -static int - CConcealment_ApplyInter ( - CConcealmentInfo *pConcealmentInfo, - CAacDecoderChannelInfo *pAacDecoderChannelInfo, - const SamplingRateInfo *pSamplingRateInfo, - const int samplesPerFrame, - const int improveTonal, - const int frameOk - ); - - - -static int - CConcealment_ApplyNoise ( - CConcealmentInfo *pConcealmentInfo, - CAacDecoderChannelInfo *pAacDecoderChannelInfo, - CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, - const SamplingRateInfo *pSamplingRateInfo, - const int samplesPerFrame, - const UINT flags - ); - -static void - CConcealment_UpdateState ( - CConcealmentInfo *pConcealmentInfo, - int frameOk - ); - -static void - CConcealment_ApplyRandomSign ( - int iRandomPhase, - FIXP_DBL *spec, - int samplesPerFrame - ); - - -static int CConcealment_GetWinSeq(int prevWinSeq) -{ - int newWinSeq = OnlyLongSequence; - - /* Try to have only long blocks */ - if ( prevWinSeq == LongStartSequence - || prevWinSeq == EightShortSequence ) - { - newWinSeq = LongStopSequence; - } - - return (newWinSeq); -} - - -/*! - \brief Init common concealment information data - - \pConcealCommonData Pointer to the concealment common data structure. - - \return none -*/ -void - CConcealment_InitCommonData (CConcealParams *pConcealCommonData) -{ - if (pConcealCommonData != NULL) - { - int i; - - /* Set default error concealment technique */ - pConcealCommonData->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; band<pRvlc->maxSfbTransmitted; 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; band<pRvlc->maxSfbTransmitted; band++) { - bnds = 16*group+band; - pAacDecoderStaticChannelInfo->concealmentInfo.aRvlcPreviousCodebook[bnds] = pAacDecoderChannelInfo->pDynData->aCodeBook[bnds]; - } - for (; band <lastSfbIndex; band++) { - bnds = 16*group+band; - FDK_ASSERT(bnds >= 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; band<pRvlc->maxSfbTransmitted; 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; group<pRvlc->numWindowGroups; 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; group<pRvlc->numWindowGroups; 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; group<pRvlc->numWindowGroups; 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/src/aacenc.cpp b/libAACenc/src/aacenc.cpp index 5e8c08d..d018eb0 100644 --- a/libAACenc/src/aacenc.cpp +++ b/libAACenc/src/aacenc.cpp @@ -313,7 +313,6 @@ void FDKaacEnc_AacInitDefaultConfig(AACENC_CONFIG *config) 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 */ @@ -580,7 +579,6 @@ AAC_ENCODER_ERROR FDKaacEnc_Initialize(HANDLE_AAC_ENC hAacEnc, psyBitrate, tnsMask, hAacEnc->bandwidth90dB, - config->usePns, config->useIS, config->syntaxFlags, initFlags); diff --git a/libAACenc/src/aacenc.h b/libAACenc/src/aacenc.h index dd09ed9..75bbe8e 100644 --- a/libAACenc/src/aacenc.h +++ b/libAACenc/src/aacenc.h @@ -96,8 +96,6 @@ amm-info@iis.fraunhofer.de #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 */ @@ -215,7 +213,6 @@ struct AACENC_CONFIG { 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 */ diff --git a/libAACenc/src/aacenc_lib.cpp b/libAACenc/src/aacenc_lib.cpp index fc58d6d..eef7878 100644 --- a/libAACenc/src/aacenc_lib.cpp +++ b/libAACenc/src/aacenc_lib.cpp @@ -109,8 +109,6 @@ amm-info@iis.fraunhofer.de #endif -#include "sbr_encoder.h" -#include "../src/sbr_ram.h" #include "channel_map.h" #include "psy_const.h" @@ -156,7 +154,6 @@ typedef struct { UINT userPeakBitrate; UCHAR userTns; /*!< Use TNS coding. */ - UCHAR userPns; /*!< Use PNS coding. */ UCHAR userIntensity; /*!< Use Intensity coding. */ TRANSPORT_TYPE userTpType; /*!< Transport type */ @@ -172,9 +169,6 @@ typedef struct { UCHAR userMetaDataMode; /*!< Meta data library configuration. */ - UCHAR userSbrEnabled; /*!< Enable SBR for ELD. */ - UINT userSbrRatio; /*!< SBR sampling rate ratio. Dual- or single-rate. */ - } USER_PARAM; //////////////////////////////////////////////////////////////////////////////////// @@ -195,9 +189,6 @@ struct AACENCODER 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 @@ -221,6 +212,7 @@ struct AACENCODER AACENC_EXT_PAYLOAD extPayload [MAX_TOTAL_EXT_PAYLOADS]; /* Extension payload */ +#define MAX_PAYLOAD_SIZE 256 UCHAR extPayloadData [(1)][(8)][MAX_PAYLOAD_SIZE]; UINT extPayloadSize [(1)][(8)]; /* payload sizes in bits */ @@ -230,8 +222,6 @@ struct AACENCODER /* Memory allocation info. */ INT nMaxAacElements; INT nMaxAacChannels; - INT nMaxSbrElements; - INT nMaxSbrChannels; UINT nMaxSubFrames; UINT encoder_modis; @@ -246,99 +236,8 @@ 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; @@ -367,35 +266,6 @@ static SBR_PS_SIGNALING getSbrSignalingMode( } 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; } @@ -558,7 +428,6 @@ AAC_ENCODER_ERROR aacEncDefaultConfig(HANDLE_AACENC_CONFIG hAacConfig, 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; @@ -582,39 +451,10 @@ AAC_ENCODER_ERROR aacEncDefaultConfig(HANDLE_AACENC_CONFIG hAacConfig, 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; el<channelMapping->nElements; 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, @@ -623,8 +463,6 @@ INT aacEncoder_LimitBitrate( const CHANNEL_MODE channelMode, INT bitRate, const INT nSubFrames, - const INT sbrActive, - const INT sbrDownSampleRate, const AUDIO_OBJECT_TYPE aot ) { @@ -633,11 +471,7 @@ INT aacEncoder_LimitBitrate( FDKaacEnc_InitChannelMapping(channelMode, CH_ORDER_MPEG, &cm); - if (sbrActive) { - coreSamplingRate = samplingRate >> (sbrEncoder_IsSingleRatePossible(aot) ? (sbrDownSampleRate-1):1); - } else { - coreSamplingRate = samplingRate; - } + coreSamplingRate = samplingRate; /* Consider bandwidth channel bit rate limit (see bandwidth.cpp: GetBandwidthEntry()) */ if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD) { @@ -665,63 +499,6 @@ INT aacEncoder_LimitBitrate( 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<cm.nElements; e++) - { - INT sbrElementBitRateIn, sbrBitRateOut; - - if (cm.elInfo[e].elType != ID_SCE && cm.elInfo[e].elType != ID_CPE) { - continue; - } - sbrElementBitRateIn = sbrElInfo[e].bitRate; - sbrBitRateOut = sbrEncoder_LimitBitRate(sbrElementBitRateIn , cm.elInfo[e].nChannelsInEl, coreSamplingRate, aot); - if (sbrBitRateOut == 0) { - return 0; - } - - /* If bitrates don't match, distribution and limiting needs to be determined again. - Abort element loop and restart with adapted bitrate. */ - if (sbrElementBitRateIn != sbrBitRateOut) { - - if (sbrElementBitRateIn < sbrBitRateOut) { - adjustedBitrate = fMax(initialBitrate, (INT)fDivNorm((FIXP_DBL)(sbrBitRateOut+8), cm.elInfo[e].relativeBits)); - break; - } - - if (sbrElementBitRateIn > 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; @@ -750,7 +527,6 @@ AACENC_ERROR FDKaacEnc_AdjustEncSettings(HANDLE_AACENCODER hAacEncoder, /* 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; @@ -799,7 +575,6 @@ AACENC_ERROR FDKaacEnc_AdjustEncSettings(HANDLE_AACENCODER hAacEncoder, 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) { @@ -848,15 +623,6 @@ AACENC_ERROR FDKaacEnc_AdjustEncSettings(HANDLE_AACENCODER hAacEncoder, 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 */ @@ -875,39 +641,6 @@ AACENC_ERROR FDKaacEnc_AdjustEncSettings(HANDLE_AACENCODER hAacEncoder, } } - /* 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); @@ -936,18 +669,9 @@ AACENC_ERROR FDKaacEnc_AdjustEncSettings(HANDLE_AACENCODER hAacEncoder, 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)) { @@ -959,14 +683,6 @@ AACENC_ERROR FDKaacEnc_AdjustEncSettings(HANDLE_AACENCODER hAacEncoder, 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) { @@ -987,25 +703,6 @@ AACENC_ERROR FDKaacEnc_AdjustEncSettings(HANDLE_AACENCODER hAacEncoder, 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) @@ -1013,7 +710,6 @@ static AACENC_ERROR aacEncInit(HANDLE_AACENCODER hAacEncoder, 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 */ @@ -1029,11 +725,6 @@ static AACENC_ERROR aacEncInit(HANDLE_AACENCODER hAacEncoder, 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 */ @@ -1053,69 +744,6 @@ static AACENC_ERROR aacEncInit(HANDLE_AACENCODER hAacEncoder, 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. */ @@ -1169,10 +797,6 @@ static AACENC_ERROR aacEncInit(HANDLE_AACENCODER hAacEncoder, { 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, @@ -1246,15 +870,11 @@ AACENC_ERROR aacEncOpen( /* 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)) ) { + if ( (hAacEncoder->nMaxAacChannels>(8)) ) { err = AACENC_INVALID_CONFIG; goto bail; } @@ -1262,7 +882,6 @@ AACENC_ERROR aacEncOpen( /* Max number of elements could be tuned any more. */ hAacEncoder->nMaxAacElements = fixMin((8), hAacEncoder->nMaxAacChannels); - hAacEncoder->nMaxSbrElements = fixMin((8), hAacEncoder->nMaxSbrChannels); hAacEncoder->nMaxSubFrames = (1); @@ -1270,19 +889,6 @@ AACENC_ERROR aacEncOpen( 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, @@ -1330,10 +936,6 @@ AACENC_ERROR aacEncOpen( 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); @@ -1379,9 +981,6 @@ AACENC_ERROR aacEncClose(HANDLE_AACENCODER *phAacEncoder) FreeRam_bsOutbuffer(&hAacEncoder->outBuffer); } - if (hAacEncoder->hEnvEnc) { - sbrEncoder_Close (&hAacEncoder->hEnvEnc); - } if (hAacEncoder->hAacEnc) { FDKaacEnc_Close (&hAacEncoder->hAacEnc); } @@ -1569,44 +1168,6 @@ AACENC_ERROR aacEncEncode( } - 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; @@ -1639,13 +1200,6 @@ AACENC_ERROR aacEncEncode( 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) { @@ -1724,8 +1278,6 @@ AACENC_ERROR aacEncGetLibInfo(LIB_INFO *info) 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; @@ -1900,20 +1452,10 @@ AACENC_ERROR aacEncoder_SetParam( } 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; - } + err = AACENC_INVALID_CONFIG; break; case AACENC_SBR_MODE: - if (settings->userSbrEnabled != value) { - settings->userSbrEnabled = value; - hAacEncoder->InitFlags |= AACENC_INIT_CONFIG | AACENC_INIT_STATES | AACENC_INIT_TRANSPORT; - } + err = AACENC_INVALID_CONFIG; break; case AACENC_TRANSMUX: if (settings->userTpType != (TRANSPORT_TYPE)value) { @@ -2064,10 +1606,10 @@ UINT aacEncoder_GetParam( value = (UINT)hAacEncoder->aacConfig.framelength; break; case AACENC_SBR_RATIO: - value = isSbrActive(&hAacEncoder->aacConfig) ? hAacEncoder->aacConfig.sbrRatio : 0; + value = 0; break; case AACENC_SBR_MODE: - value = (UINT) (hAacEncoder->aacConfig.syntaxFlags & AC_SBR_PRESENT) ? 1 : 0; + value = 0; break; case AACENC_TRANSMUX: value = (UINT)settings->userTpType; 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; sfb<MAX_GROUPED_SFB; sfb++) { - noiseNrg[sfb] = NO_NOISE_PNS; /* clear nrg's of previous frame */ - } - return; - } - } - else { - if(!pnsConf->usePns) - 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; sfb<maxSfbPerGroup-1; sfb++) { - if((pnsData->noiseFuzzyMeasure[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; sfb<maxSfbPerGroup-1; sfb++) { - if((pnsData->pnsFlag[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;sfb<sfbActive;sfb++) { - /* - MS post processing - */ - if( msMask[sfb] ) { - if( (pnsDataLeft->pnsFlag[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/bitenc.cpp b/libAACenc/src/bitenc.cpp index 8e477aa..3ed3b2f 100644 --- a/libAACenc/src/bitenc.cpp +++ b/libAACenc/src/bitenc.cpp @@ -293,21 +293,17 @@ 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; i<sectionData->noOfSections; i++){ @@ -326,24 +322,6 @@ static INT FDKaacEnc_encodeScaleFactorData(UINT *maxValueInSfb, } } /* sfb */ } - else if(sectionData->huffsection[i].codeBook == CODE_BOOK_PNS_NO) { - INT sfbStart = sectionData->huffsection[i].sfbStart; - INT tmp = sfbStart + sectionData->huffsection[i].sfbCnt; - for(j=sfbStart; j<tmp; j++) { - deltaPns = noiseNrg[j]-lastValPns; - lastValPns = noiseNrg[j]; - - if(noisePCMFlag){ - FDKwriteBits(hBitStream,deltaPns+(1<<(PNS_PCM_BITS-1)),PNS_PCM_BITS); - noisePCMFlag = FALSE; - } - else { - if(FDKaacEnc_codeScalefactorDelta(deltaPns,hBitStream)) { - return(1); - } - } - } /* sfb */ - } else { INT tmp = sectionData->huffsection[i].sfbStart+sectionData->huffsection[i].sfbCnt; for(j=sectionData->huffsection[i].sfbStart; j<tmp; j++){ @@ -1150,11 +1128,10 @@ AAC_ENCODER_ERROR FDKaacEnc_ChannelElementWrite( HANDLE_TRANSPORTENC hTpEnc, pChSectionData, pChScf, hBitStream, - psyOutChannel[ch]->noiseNrg, psyOutChannel[ch]->isScale, chGlobalGain ); if ( (hBitStream != NULL) - && (sfDataBits != (qcOutChannel[ch]->sectionData.scalefacBits + qcOutChannel[ch]->sectionData.noiseNrgBits)) ) { + && (sfDataBits != (qcOutChannel[ch]->sectionData.scalefacBits)) ) { error = AAC_ENC_WRITE_SCAL_ERROR; } bitDemand += sfDataBits; diff --git a/libAACenc/src/dyn_bits.cpp b/libAACenc/src/dyn_bits.cpp index 0c07109..c153014 100644 --- a/libAACenc/src/dyn_bits.cpp +++ b/libAACenc/src/dyn_bits.cpp @@ -91,7 +91,6 @@ amm-info@iis.fraunhofer.de #include "dyn_bits.h" #include "bit_cnt.h" #include "psy_const.h" -#include "aacenc_pns.h" #include "aacEnc_ram.h" #include "aacEnc_rom.h" @@ -247,7 +246,6 @@ 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 ) { @@ -258,12 +256,8 @@ static void FDKaacEnc_gmStage0( /* 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] ) { + /* intensity codebooks are already allocated in bitcount.c */ + if( isBook[i] ) { huffsection[i].codeBook=isBook[i]; huffsection[i].sectionBits = 0; } @@ -394,7 +388,6 @@ static void FDKaacEnc_noiselessCounter( const UINT* const maxValueInSfb, const INT* const sfbOffset, const INT blockType, - const INT* const noiseNrg, const INT* const isBook, const INT useVCB11 ) @@ -438,7 +431,7 @@ static void FDKaacEnc_noiselessCounter( huffsection); /* 0.Stage :Find minimum Codebooks */ - FDKaacEnc_gmStage0(huffsection, bitLookUp, sectionData->maxSfbPerGroup, noiseNrg+grpNdx, isBook+grpNdx); + FDKaacEnc_gmStage0(huffsection, bitLookUp, sectionData->maxSfbPerGroup, isBook+grpNdx); /* 1.Stage :Merge all connected regions with the same code book */ FDKaacEnc_gmStage1(huffsection, bitLookUp, sectionData->maxSfbPerGroup, sideInfoTab, useVCB11); @@ -661,68 +654,6 @@ static void FDKaacEnc_scfCount( } /* for (i=0; i<sectionData->noOfSections; 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; j<sfbEnd; j++) { - - if (noisePCMFlag) { - sectionData->noiseNrgBits+=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, @@ -734,7 +665,6 @@ INT FDKaacEnc_dynBitCount( 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 @@ -754,7 +684,6 @@ INT FDKaacEnc_dynBitCount( maxValueInSfb, sfbOffset, blockType, - noiseNrg, isBook, (syntaxFlags & AC_ER_VCB11)?1:0); @@ -764,13 +693,9 @@ INT FDKaacEnc_dynBitCount( sectionData, isScale); - FDKaacEnc_noiseCount(sectionData, - noiseNrg); - return (sectionData->huffmanBits + sectionData->sideInfoBits + - sectionData->scalefacBits + - sectionData->noiseNrgBits); + sectionData->scalefacBits); } INT FDKaacEnc_BCNew(BITCNTR_STATE **phBC diff --git a/libAACenc/src/dyn_bits.h b/libAACenc/src/dyn_bits.h index ae78a4c..045c8c4 100644 --- a/libAACenc/src/dyn_bits.h +++ b/libAACenc/src/dyn_bits.h @@ -125,7 +125,6 @@ typedef struct 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; @@ -143,10 +142,6 @@ INT FDKaacEnc_BCNew(BITCNTR_STATE **phBC 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, @@ -158,7 +153,6 @@ INT FDKaacEnc_dynBitCount( 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 diff --git a/libAACenc/src/intensity.cpp b/libAACenc/src/intensity.cpp index b45b27b..d4d01c2 100644 --- a/libAACenc/src/intensity.cpp +++ b/libAACenc/src/intensity.cpp @@ -547,7 +547,6 @@ FDKaacEnc_finalizeIntensityDecision(const FIXP_DBL *hrrErr, output: isBook isScale - pnsData->pnsFlag msDigest zeroed from start to sfbCnt msMask zeroed from start to sfbCnt mdctSpectrumRight zeroed where isBook!=0 @@ -580,8 +579,7 @@ void FDKaacEnc_IntensityStereoProcessing( const INT *sfbOffset, const INT allowIS, INT *isBook, - INT *isScale, - PNS_DATA *RESTRICT pnsData[2] + INT *isScale ) { INT sfb,sfboffs, j; @@ -650,15 +648,6 @@ void FDKaacEnc_IntensityStereoProcessing( &&(fMult(FL2FXCONST_DBL(1.0f/1.5f),sfbEnergyRight[sfb+sfboffs]) > 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]); diff --git a/libAACenc/src/intensity.h b/libAACenc/src/intensity.h index 2acc292..4dc1d71 100644 --- a/libAACenc/src/intensity.h +++ b/libAACenc/src/intensity.h @@ -91,7 +91,7 @@ amm-info@iis.fraunhofer.de #ifndef _INTENSITY_H #define _INTENSITY_H -#include "aacenc_pns.h" +#include "common_fix.h" void FDKaacEnc_IntensityStereoProcessing( @@ -114,8 +114,7 @@ void FDKaacEnc_IntensityStereoProcessing( const INT *sfbOffset, const INT allowIS, INT *isBook, - INT *isScale, - PNS_DATA *RESTRICT pnsData[2] + INT *isScale ); #endif /* _INTENSITY_H */ diff --git a/libAACenc/src/interface.h b/libAACenc/src/interface.h index 51fb72a..3f9493b 100644 --- a/libAACenc/src/interface.h +++ b/libAACenc/src/interface.h @@ -128,7 +128,6 @@ typedef struct { INT groupLen[MAX_NO_OF_GROUPS]; TNS_INFO tnsInfo; - INT noiseNrg[MAX_GROUPED_SFB]; INT isBook[MAX_GROUPED_SFB]; INT isScale[MAX_GROUPED_SFB]; 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; sfb<sfbActive; sfb++) { - - fuzzyTotal = (FIXP_SGL)MAXVAL_SGL; - sfbWidth = sfbOffset[sfb+1] - sfbOffset[sfb]; - - /* Reset output for lower bands or too small bands */ - if (sfb < np->startSfb || 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]; i<sfbOffset[sfb]+k; i++) { - fhelp1 = fPow2AddDiv2(fhelp1, mdctSpectrum[i]<<leadingBits); - fhelp2 = fPow2AddDiv2(fhelp2, mdctSpectrum[i+k]<<leadingBits); - fhelp3 = fPow2AddDiv2(fhelp3, mdctSpectrum[i+2*k]<<leadingBits); - fhelp4 = fPow2AddDiv2(fhelp4, mdctSpectrum[i+3*k]<<leadingBits); - } - - /* get max into fhelp: */ - maxVal = fixMax(fhelp1, fhelp2); - maxVal = fixMax(maxVal, fhelp3); - maxVal = fixMax(maxVal, fhelp4); - - /* get min into fhelp1: */ - minVal = fixMin(fhelp1, fhelp2); - minVal = fixMin(minVal, fhelp3); - minVal = fixMin(minVal, fhelp4); - - /* Normalize min and max Val */ - leadingBits = CountLeadingBits(maxVal); - testVal = maxVal << leadingBits; - refVal = minVal << leadingBits; - - /* calculate fuzzy value for power distribution */ - testVal = fMultDiv2(testVal, np->powDistPSDcurve[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/psy_configuration.h b/libAACenc/src/psy_configuration.h index 3629246..15ecccb 100644 --- a/libAACenc/src/psy_configuration.h +++ b/libAACenc/src/psy_configuration.h @@ -97,7 +97,6 @@ amm-info@iis.fraunhofer.de #include "psy_const.h" #include "aacenc_tns.h" -#include "aacenc_pns.h" #define THR_SHIFTBITS 4 #define PCM_QUANT_THR_SCALE 16 @@ -131,7 +130,6 @@ typedef struct{ FIXP_DBL sfbMinSnrLdData[MAX_SFB]; /* minimum snr (formerly known as bmax) */ TNS_CONFIG tnsConf; - PNS_CONFIG pnsConf; INT granuleLength; INT allowIS; diff --git a/libAACenc/src/psy_main.cpp b/libAACenc/src/psy_main.cpp index 446c894..6dc810f 100644 --- a/libAACenc/src/psy_main.cpp +++ b/libAACenc/src/psy_main.cpp @@ -102,8 +102,6 @@ amm-info@iis.fraunhofer.de #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" @@ -304,7 +302,6 @@ AAC_ENCODER_ERROR FDKaacEnc_psyMainInit(PSY_INTERNAL *hPsy, INT bitRate, INT tnsMask, INT bandwidth, - INT usePns, INT useIS, UINT syntaxFlags, ULONG initFlags) @@ -391,25 +388,6 @@ AAC_ENCODER_ERROR FDKaacEnc_psyMainInit(PSY_INTERNAL *hPsy, } } - 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; } @@ -466,7 +444,6 @@ AAC_ENCODER_ERROR FDKaacEnc_psyMain(INT channels, PSY_DATA *RESTRICT psyData[(2)]; TNS_DATA *RESTRICT tnsData[(2)]; - PNS_DATA *RESTRICT pnsData[(2)]; INT zeroSpec = TRUE; /* means all spectral lines are zero */ @@ -498,7 +475,6 @@ AAC_ENCODER_ERROR FDKaacEnc_psyMain(INT channels, { psyData[ch] = &psyDynamic->psyData[ch]; tnsData[ch] = &psyDynamic->tnsData[ch]; - pnsData[ch] = &psyDynamic->pnsData[ch]; psyData[ch]->mdctSpectrum = psyOutChannel[ch]->mdctSpectrum; } @@ -771,19 +747,6 @@ AAC_ENCODER_ERROR FDKaacEnc_psyMain(INT channels, 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}; @@ -1141,29 +1104,6 @@ AAC_ENCODER_ERROR FDKaacEnc_psyMain(INT channels, FDKmemclear(psyOutChannel[ch]->isScale, MAX_GROUPED_SFB*sizeof(INT)); } - for(ch=0;ch<channels;ch++) { - INT win = (isShortWindow[ch]?1:0); - if (!psyStatic[ch]->isLFE) - { - /* 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 */ @@ -1177,18 +1117,6 @@ AAC_ENCODER_ERROR FDKaacEnc_psyMain(INT channels, 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, @@ -1209,8 +1137,7 @@ AAC_ENCODER_ERROR FDKaacEnc_psyMain(INT channels, psyConf[0].sfbOffset, psyConf[0].allowIS && commonWindow, psyOutChannel[1]->isBook, - psyOutChannel[1]->isScale, - pnsData); + psyOutChannel[1]->isScale); FDKaacEnc_MsStereoProcessing( psyData, @@ -1222,15 +1149,6 @@ AAC_ENCODER_ERROR FDKaacEnc_psyMain(INT channels, 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, @@ -1252,8 +1170,7 @@ AAC_ENCODER_ERROR FDKaacEnc_psyMain(INT channels, psyData[0]->groupedSfbOffset, psyConf[0].allowIS && commonWindow, psyOutChannel[1]->isBook, - psyOutChannel[1]->isScale, - pnsData); + psyOutChannel[1]->isScale); /* it's OK to pass the ".Long" arrays here. They contain grouped short data since FDKaacEnc_groupShortData() */ FDKaacEnc_MsStereoProcessing( psyData, @@ -1268,26 +1185,6 @@ AAC_ENCODER_ERROR FDKaacEnc_psyMain(INT channels, } } - /* - PNS Coding - */ - for(ch=0;ch<channels;ch++) { - if (psyStatic[ch]->isLFE) { - /* 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 */ diff --git a/libAACenc/src/psy_main.h b/libAACenc/src/psy_main.h index 7bdcc38..54c0bbb 100644 --- a/libAACenc/src/psy_main.h +++ b/libAACenc/src/psy_main.h @@ -94,7 +94,6 @@ amm-info@iis.fraunhofer.de #include "psy_configuration.h" #include "qc_data.h" -#include "aacenc_pns.h" /* psych internal @@ -109,7 +108,6 @@ typedef struct { PSY_DATA psyData[(2)]; TNS_DATA tnsData[(2)]; - PNS_DATA pnsData[(2)]; }PSY_DYNAMIC; @@ -153,7 +151,6 @@ AAC_ENCODER_ERROR FDKaacEnc_psyMainInit(PSY_INTERNAL *hPsy, INT bitRate, INT tnsMask, INT bandwidth, - INT usePns, INT useIS, UINT syntaxFlags, ULONG initFlags); diff --git a/libAACenc/src/qc_main.cpp b/libAACenc/src/qc_main.cpp index 9cd73f6..00da236 100644 --- a/libAACenc/src/qc_main.cpp +++ b/libAACenc/src/qc_main.cpp @@ -865,20 +865,6 @@ AAC_ENCODER_ERROR FDKaacEnc_QCMain(QC_STATE* RESTRICT hQC, 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; ch<cm->nChannels; 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++ ) { @@ -1063,7 +1049,6 @@ AAC_ENCODER_ERROR FDKaacEnc_QCMain(QC_STATE* RESTRICT hQC, psyOutCh->sfbPerGroup, psyOutCh->sfbOffsets, &qcOutCh->sectionData, - psyOutCh->noiseNrg, psyOutCh->isBook, psyOutCh->isScale, syntaxFlags) ; 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<numberOfLines;j++) { - FIXP_DBL tmp = fMultDiv2(alpha_1, chaosMeasurePerLine[j]); - chaosMeasurePerLine[j] = fMultAdd(tmp, alpha_0, chaosMeasurePerLine[j-1]); - } - - FDKaacEnc_CalcSfbTonality(spectrum, - sfbMaxScaleSpec, - chaosMeasurePerLine, - sfbTonality, - sfbCnt, - sfbOffset, - sfbEnergyLD64); - - C_ALLOC_SCRATCH_END(chaosMeasurePerLine, FIXP_DBL, (1024)); -} - - -/***************************************************************************** - - functionname: CalculateTonalityIndex - description: computes tonality values out of unpredictability values - limits range and computes log() - returns: - input: ptr to energies, ptr to chaos measure values, - number of sfb - output: sfb wise tonality values - -*****************************************************************************/ -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 ) -{ - INT i, j; - - for (i=0; i<sfbCnt; i++) { - FIXP_DBL chaosMeasureSfbLD64; - INT shiftBits = fixMax(0,sfbMaxScaleSpec[i] - 4); /* max sfbWidth = 96 ; 2^7=128 => 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++)<<shiftBits; - FIXP_DBL lineNrg = fMultDiv2(tmp, tmp); - chaosMeasureSfb = fMultAddDiv2(chaosMeasureSfb, lineNrg, *chaosMeasure++); - } - - /* calc tonalityPerSfb */ - if (chaosMeasureSfb != FL2FXCONST_DBL(0.0)) - { - /* add ld(convtone)/64 and 2/64 bec.fMultDiv2 */ - chaosMeasureSfbLD64 = CalcLdData((chaosMeasureSfb)) - sfbEnergyLD64[i]; - chaosMeasureSfbLD64 += FL2FXCONST_DBL(3.0f/64) - ((FIXP_DBL)(shiftBits)<<(DFRACT_BITS-6)); - - if (chaosMeasureSfbLD64 > 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/libMpegTPDec/include/tp_data.h b/libMpegTPDec/include/tp_data.h index c6e89b5..6ce4777 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 */ @@ -207,27 +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 +219,6 @@ 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 +257,12 @@ 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 @@ -353,15 +353,6 @@ int transportDec_RegisterSscCallback ( 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. * So it grabs whatever it can from pBuffer and returns information (bytesValid) so that at a 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; cnt<eldExtLen; cnt++) { - FDKreadBits(hBs, 8 ); - } - break; - /* add future eld extension configs here */ - } - } -bail: - return (ErrorStatus); -} -#endif /* TP_ELD_ENABLE */ - - static TRANSPORTDEC_ERROR AudioSpecificConfig_ExtensionParse(CSAudioSpecificConfig *self, HANDLE_FDK_BITSTREAM bs, CSTpCallBacks *cb) { @@ -1317,16 +1205,6 @@ TRANSPORTDEC_ERROR AudioSpecificConfig_Parse( return TRANSPORTDEC_UNSUPPORTED_FORMAT; } break; -#ifdef TP_ELD_ENABLE - case AOT_ER_AAC_ELD: - if ((ErrorStatus = EldSpecificConfig_Parse(self, bs, cb)) != TRANSPORTDEC_OK ) { - return (ErrorStatus); - } - frameLengthFlag = self->m_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/tpenc_lib.h b/libMpegTPEnc/include/tpenc_lib.h index 2833e82..2361f55 100644 --- a/libMpegTPEnc/include/tpenc_lib.h +++ b/libMpegTPEnc/include/tpenc_lib.h @@ -128,19 +128,6 @@ typedef struct TRANSPORTENC *HANDLE_TRANSPORTENC; 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. diff --git a/libMpegTPEnc/src/tpenc_lib.cpp b/libMpegTPEnc/src/tpenc_lib.cpp index 24fb32f..7cbad63 100644 --- a/libMpegTPEnc/src/tpenc_lib.cpp +++ b/libMpegTPEnc/src/tpenc_lib.cpp @@ -333,17 +333,6 @@ 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, 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 <em>envelope</em> 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 - <em>lowSubband</em> and <em>highSubband</em> 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 <em>_m</em> and <em>_e</em> 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$. Therefore, - the scale factor <em>hb_scale</em> passed into this function will be converted - to an 'input exponent' (#input_e), which fits the internal representation. - - Before the actual processing, an exponent #adj_e for resulting adjusted - samples is derived from the maximum reference energy. - - Then, for each envelope, the following steps are performed: - - \li Calculate energy in the signal to be adjusted. Depending on the the value of - #interpolFreq (interpolation mode), this is either done seperately - for each QMF-subband or for each SBR-band. - The resulting energies are stored in #nrgEst_m[#MAX_FREQ_COEFFS] (mantissas) - and #nrgEst_e[#MAX_FREQ_COEFFS] (exponents). - \li Calculate gain and noise level for each subband:<br> - \f$ gain = \sqrt{ \frac{nrgRef}{nrgEst} \cdot (1 - noiseRatio) } - \hspace{2cm} - noise = \sqrt{ nrgRef \cdot noiseRatio } - \f$<br> - where <em>noiseRatio</em> and <em>nrgRef</em> are extracted from the - bitstream and <em>nrgEst</em> 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 - - <br> -*/ -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<ui; k++) { - sinePresentFlag |= (i >= sineMapped[cc]); - cc++; - } - - for (k=li; k<ui; k++) { - if (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; k<noSubbands; k++) { - FDK_sqrt_MantExp(&pNrgs->nrgSine[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; k<noSubbands; k++) - h_sbr_cal_env->filtBufferNoise[k] <<= shift; - } - else { - int shift = fixMin(DFRACT_BITS-1,-(int)(h_sbr_cal_env->filtBufferNoise_e - noise_e)); - for (k=0; k<noSubbands; k++) - h_sbr_cal_env->filtBufferNoise[k] >>= shift; - } - - h_sbr_cal_env->filtBufferNoise_e = noise_e; - } - - /* find best scaling! */ - scale_change = -(DFRACT_BITS-1); - for(k=0;k<noSubbands;k++) { - scale_change = fixMax(scale_change,(int)pNrgs->nrgGain_e[k]); - } - sc_change = (start_pos<no_cols)? adj_e - input_e : final_e - input_e; - - if ((scale_change-sc_change+1)<0) - scale_change-=(scale_change-sc_change+1); - - scale_change = (scale_change-sc_change)+1; - - for(k=0;k<noSubbands;k++) { - int sc = scale_change-pNrgs->nrgGain_e[k] + (sc_change-1); - pNrgs->nrgGain[k] >>= sc; - pNrgs->nrgGain_e[k] += sc; - } - - if (!useLP) { - for(k=0;k<noSubbands;k++) { - int sc = scale_change-h_sbr_cal_env->filtBuffer_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_pos<no_cols) ) - { - int shift = (int) (noise_e - final_e); - if (!useLP) - filtBufferNoiseShift = shift; /* shifting of h_sbr_cal_env->filtBufferNoise[k] will be applied in function adjustTimeSlotHQ() */ - if (shift>=0) { - shift = fixMin(DFRACT_BITS-1,shift); - for (k=0; k<noSubbands; k++) { - pNrgs->nrgSine[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; k<noSubbands; k++) { - pNrgs->nrgSine[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;k<noSubbands;k++) { - pNrgs->nrgGain[k] >>= -sc_change; - pNrgs->nrgGain_e[k] += -sc_change; - } - if (!useLP) { - for(k=0;k<noSubbands;k++) { - h_sbr_cal_env->filtBuffer[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; band<subbands; band++){ - diff = (int) (nrgGain_e[band] - filtBuffer_e[band]); - if (diff>0) { - 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<next_pos; l++) { - scaleValues(&re[l][lowSubband], width, shift); - scaleValues(&im[l][lowSubband], width, shift); - } - } else - { - for (int l=start_pos; l<next_pos; l++) { - scaleValues(&re[l][lowSubband], width, shift); - } - } - } -} - - -/*! - \brief Determine headroom for shifting - - Determine by how much the spectrum can be shifted left - for better accuracy in later processing. - - \return Number of free bits in the biggest spectral value -*/ - -FIXP_DBL maxSubbandSample( FIXP_DBL ** re, /*!< Real part of input and output subband samples */ - FIXP_DBL ** im, /*!< Real part of input and output subband samples */ - int lowSubband, /*!< Begin of frequency range to process */ - int highSubband, /*!< Number of QMF bands to process */ - int start_pos, /*!< Begin of time rage (QMF-timeslot) */ - int next_pos /*!< End of time rage (QMF-timeslot) */ - ) -{ - FIXP_DBL maxVal = FL2FX_DBL(0.0f); - unsigned int width = highSubband - lowSubband; - - FDK_ASSERT(width <= (64)); - - if ( width > 0 ) { - if (im!=NULL) - { - for (int l=start_pos; l<next_pos; l++) - { -#ifdef FUNCTION_FDK_get_maxval - maxVal = FDK_get_maxval(maxVal, &re[l][lowSubband], &im[l][lowSubband], width); -#else - int k=width; - FIXP_DBL *reTmp = &re[l][lowSubband]; - FIXP_DBL *imTmp = &im[l][lowSubband]; - do{ - FIXP_DBL tmp1 = *(reTmp++); - FIXP_DBL tmp2 = *(imTmp++); - maxVal |= (FIXP_DBL)((LONG)(tmp1)^((LONG)tmp1>>(DFRACT_BITS-1))); - maxVal |= (FIXP_DBL)((LONG)(tmp2)^((LONG)tmp2>>(DFRACT_BITS-1))); - } while(--k!=0); -#endif - } - } else - { - for (int l=start_pos; l<next_pos; l++) { - int k=width; - FIXP_DBL *reTmp = &re[l][lowSubband]; - do{ - FIXP_DBL tmp = *(reTmp++); - maxVal |= (FIXP_DBL)((LONG)(tmp)^((LONG)tmp>>(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<highSubband; k++) { - FIXP_DBL bufferReal[(((1024)/(32))+(6))]; - FIXP_DBL bufferImag[(((1024)/(32))+(6))]; - FIXP_DBL maxVal = FL2FX_DBL(0.0f); - - if (analysBufferImag!=NULL) - { - for (l=start_pos;l<next_pos;l++) - { - bufferImag[l] = analysBufferImag[l][k]; - maxVal |= (FIXP_DBL)((LONG)(bufferImag[l])^((LONG)bufferImag[l]>>(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<next_pos;l++) - { - bufferReal[l] = analysBufferReal[l][k]; - maxVal |= (FIXP_DBL)((LONG)(bufferReal[l])^((LONG)bufferReal[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<next_pos; l++) { - FIXP_DBL temp1 = bufferReal[l] << (int)preShift; - FIXP_DBL temp2 = bufferImag[l] << (int)preShift; - accu = fPow2AddDiv2(accu, temp1); - accu = fPow2AddDiv2(accu, temp2); - } - } else - { - for (l=start_pos; l<next_pos; l++) { - FIXP_DBL temp = bufferReal[l] << (int)preShift; - accu = fPow2AddDiv2(accu, temp); - } - } - } - else { /* if negative shift value */ - int negpreShift = -preShift; - if (analysBufferImag!=NULL) { - for (l=start_pos; l<next_pos; l++) { - FIXP_DBL temp1 = bufferReal[l] >> (int)negpreShift; - FIXP_DBL temp2 = bufferImag[l] >> (int)negpreShift; - accu = fPow2AddDiv2(accu, temp1); - accu = fPow2AddDiv2(accu, temp2); - } - } else - { - for (l=start_pos; l<next_pos; l++) { - FIXP_DBL temp = bufferReal[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<nSfb; j++) { - li = freqBandTable[j]; - ui = freqBandTable[j+1]; - - FIXP_DBL maxVal = maxSubbandSample( analysBufferReal, - analysBufferImag, - li, - ui, - start_pos, - next_pos ); - - if (maxVal!=FL2FXCONST_DBL(0.f)) { - - preShift = CntLeadingZeros(maxVal)-1; - - /* 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! - */ - preShift -= SHIFT_BEFORE_SQUARE; - - sumAll = FL2FXCONST_DBL(0.0f); - - - for (k=li; k<ui; k++) { - - sumLine = FL2FXCONST_DBL(0.0f); - - if (analysBufferImag!=NULL) { - if (preShift>=0) { - for (l=start_pos; l<next_pos; l++) { - temp = analysBufferReal[l][k] << (int)preShift; - sumLine += fPow2Div2(temp); - temp = analysBufferImag[l][k] << (int)preShift; - sumLine += fPow2Div2(temp); - - } - } else { - for (l=start_pos; l<next_pos; l++) { - temp = analysBufferReal[l][k] >> -(int)preShift; - sumLine += fPow2Div2(temp); - temp = analysBufferImag[l][k] >> -(int)preShift; - sumLine += fPow2Div2(temp); - } - } - } else - { - if (preShift>=0) { - for (l=start_pos; l<next_pos; l++) { - temp = analysBufferReal[l][k] << (int)preShift; - sumLine += fPow2Div2(temp); - } - } else { - for (l=start_pos; l<next_pos; l++) { - temp = analysBufferReal[l][k] >> -(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; k<ui; k++) - { - *nrgEst++ = sum; - *nrgEst_e++ = sum_e; - } - } -} - - -/*! - \brief Calculate gain, noise, and additional sine level for one subband. - - The resulting energy gain is given by mantissa and exponent. -*/ -static void calcSubbandGain(FIXP_DBL nrgRef, /*!< Reference Energy according to envelope data */ - SCHAR nrgRef_e, /*!< Reference Energy according to envelope data (exponent) */ - ENV_CALC_NRGS* nrgs, - int i, - FIXP_DBL tmpNoise, /*!< Relative noise level */ - SCHAR tmpNoise_e, /*!< Relative noise level (exponent) */ - UCHAR sinePresentFlag, /*!< Indicates if sine is present on band */ - UCHAR sineMapped, /*!< Indicates if sine must be added */ - int noNoiseFlag) /*!< Flag to suppress noise addition */ -{ - FIXP_DBL nrgEst = nrgs->nrgEst[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; k<highSubband; k++){ - /* Add nrgRef[k] to sumRef: */ - FDK_add_MantExp( sumRef, sumRef_e, - nrgRef[k], nrgRef_e[k], - &sumRef, &sumRef_e ); - - /* Add nrgEst[k] to sumEst: */ - FDK_add_MantExp( sumEst, sumEst_e, - nrgEst[k], nrgEst_e[k], - &sumEst, &sumEst_e ); - } - - FDK_divide_MantExp(sumRef, sumRef_e, - sumEst, sumEst_e, - ptrAvgGain, ptrAvgGain_e); - - *ptrSumRef = sumRef; - *ptrSumRef_e = sumRef_e; -} - -static void adjustTimeSlot_EldGrid( - 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 */ - int scale_diff_low) /*!< */ -{ - int k; - FIXP_DBL signalReal, sbNoise; - int tone_count = 0; - - 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 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<noSubbands; k++) { - /* - Smoothing: The old envelope has been bufferd and a certain ratio - of the old gains and noise levels is used. - */ - - smoothedGain = fMult(smooth_ratio,filtBuffer[k]) + - fMult(direct_ratio,gain[k]); - - if (filtBufferNoiseShift<0) { - smoothedNoise = (fMultDiv2(smooth_ratio,filtBufferNoise[k])>>shift) + - fMult(direct_ratio,noiseLevel[k]); - } - else { - smoothedNoise = (fMultDiv2(smooth_ratio,filtBufferNoise[k])<<shift) + - fMult(direct_ratio,noiseLevel[k]); - } - - /* - The next 2 multiplications constitute the actual envelope adjustment - of the signal and should be carried out with full accuracy - (supplying #DFRACT_BITS valid bits). - */ - signalReal = fMultDiv2(*ptrReal,smoothedGain)<<((int)scale_change); - signalImag = fMultDiv2(*ptrImag,smoothedGain)<<((int)scale_change); - - index++; - - if (pSineLevel[k] != FL2FXCONST_DBL(0.0f)) { - sineLevel = pSineLevel[k]; - - switch(harmIndex) { - case 0: - *ptrReal++ = (signalReal + sineLevel); - *ptrImag++ = (signalImag); - break; - case 2: - *ptrReal++ = (signalReal - sineLevel); - *ptrImag++ = (signalImag); - break; - case 1: - *ptrReal++ = (signalReal); - if (freqInvFlag) - *ptrImag++ = (signalImag - sineLevel); - else - *ptrImag++ = (signalImag + sineLevel); - break; - case 3: - *ptrReal++ = signalReal; - if (freqInvFlag) - *ptrImag++ = (signalImag + sineLevel); - else - *ptrImag++ = (signalImag - sineLevel); - break; - } - } - else { - if (noNoiseFlag) { - /* Just the amplified signal is saved */ - *ptrReal++ = (signalReal); - *ptrImag++ = (signalImag); - } - else { - /* Add noisefloor to the amplified signal */ - index &= (SBR_NF_NO_RANDOM_VAL - 1); - noiseReal = fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], smoothedNoise)<<4; - noiseImag = fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][1], smoothedNoise)<<4; - *ptrReal++ = (signalReal + noiseReal); - *ptrImag++ = (signalImag + noiseImag); - } - } - freqInvFlag ^= 1; - } - - } - else - { - for (k=0; k<noSubbands; k++) - { - smoothedGain = gain[k]; - signalReal = fMultDiv2(*ptrReal, smoothedGain) << scale_change; - signalImag = fMultDiv2(*ptrImag, smoothedGain) << scale_change; - - index++; - - if ((sineLevel = pSineLevel[k]) != FL2FXCONST_DBL(0.0f)) - { - switch (harmIndex) - { - case 0: - signalReal += sineLevel; - break; - case 1: - if (freqInvFlag) - signalImag -= sineLevel; - else - signalImag += sineLevel; - break; - case 2: - signalReal -= sineLevel; - break; - case 3: - if (freqInvFlag) - signalImag += sineLevel; - else - signalImag -= sineLevel; - break; - } - } - else - { - if (noNoiseFlag == 0) - { - /* Add noisefloor to the amplified signal */ - smoothedNoise = noiseLevel[k]; - index &= (SBR_NF_NO_RANDOM_VAL - 1); - noiseReal = fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], smoothedNoise); - noiseImag = fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][1], smoothedNoise); - signalReal += noiseReal<<4; - signalImag += noiseImag<<4; - } - } - *ptrReal++ = signalReal; - *ptrImag++ = signalImag; - - freqInvFlag ^= 1; - } - } -} - - -/*! - \brief Reset limiter bands. - - Build frequency band table for the gain limiter dependent on - the previously generated transposer patch areas. - - \return SBRDEC_OK if ok, SBRDEC_UNSUPPORTED_CONFIG on error -*/ -SBR_ERROR -ResetLimiterBands ( UCHAR *limiterBandTable, /*!< Resulting band borders in QMF channels */ - UCHAR *noLimiterBands, /*!< Resulting number of limiter band */ - UCHAR *freqBandTable, /*!< Table with possible band borders */ - int noFreqBands, /*!< Number of bands in freqBandTable */ - const PATCH_PARAM *patchParam, /*!< Transposer patch parameters */ - int noPatches, /*!< Number of transposer patches */ - int limiterBands) /*!< Selected 'band density' from bitstream */ -{ - int i, k, isPatchBorder[2], loLimIndex, hiLimIndex, tempNoLim, nBands; - UCHAR workLimiterBandTable[MAX_FREQ_COEFFS / 2 + MAX_NUM_PATCHES + 1]; - int patchBorders[MAX_NUM_PATCHES + 1]; - int kx, k2; - - int lowSubband = freqBandTable[0]; - int highSubband = freqBandTable[noFreqBands]; - - /* 1 limiter band. */ - if(limiterBands == 0) { - limiterBandTable[0] = 0; - limiterBandTable[1] = highSubband - lowSubband; - nBands = 1; - } else { - for (i = 0; i < noPatches; i++) { - patchBorders[i] = patchParam[i].guardStartBand - lowSubband; - } - patchBorders[i] = highSubband - lowSubband; - - /* 1.2, 2, or 3 limiter bands/octave plus bandborders at patchborders. */ - for (k = 0; k <= noFreqBands; k++) { - workLimiterBandTable[k] = freqBandTable[k] - lowSubband; - } - for (k = 1; k < noPatches; k++) { - workLimiterBandTable[noFreqBands + k] = patchBorders[k]; - } - - tempNoLim = nBands = noFreqBands + noPatches - 1; - shellsort(workLimiterBandTable, tempNoLim + 1); - - loLimIndex = 0; - hiLimIndex = 1; - - - while (hiLimIndex <= tempNoLim) { - FIXP_DBL div_m, oct_m, temp; - INT div_e = 0, oct_e = 0, temp_e = 0; - - k2 = workLimiterBandTable[hiLimIndex] + lowSubband; - kx = workLimiterBandTable[loLimIndex] + lowSubband; - - div_m = fDivNorm(k2, kx, &div_e); - - /* calculate number of octaves */ - oct_m = fLog2(div_m, div_e, &oct_e); - - /* multiply with limiterbands per octave */ - /* values 1, 1.2, 2, 3 -> 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;i<MAX_INVF_BANDS;i++) - h_sbr_data->sbr_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 - - <h1>About the SBR data format:</h1> - - 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 <h2>The SBR_HEADER_ELEMENT</h2> - - 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 <h2>The SBR_STANDARD_ELEMENT</h2> - - 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;i<hHeaderData->freqBandData.nSfb[1];i++){ - hFrameData->addHarmonics[i] = FDKreadBits (hBs, 1 ); - bitsRead++; - } - } - else { - for(i=0; i<MAX_FREQ_COEFFS; i++) - hFrameData->addHarmonics[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<cnt; i++) - FDKreadBits(hBs, 8); - nBitsLeft -= cnt * 8; - } else { - nBitsLeft -= ReadPsData(hParametricStereoDec, hBs, nBitsLeft); - bPsRead = 1; - } - } - - /* parametric stereo detected, could set channelMode accordingly here */ - /* */ - /* "The usage of this parametric stereo extension to HE-AAC is */ - /* signalled implicitly in the bitstream. Hence, if an sbr_extension() */ - /* with bs_extension_id==EXTENSION_ID_PS is found in the SBR part of */ - /* the bitstream, a decoder supporting the combination of SBR and PS */ - /* shall operate the PS tool to generate a stereo output signal." */ - /* source: ISO/IEC 14496-3:2001/FDAM 2:2004(E) */ - - break; - - - default: - cnt = nBitsLeft >> 3; /* number of remaining bytes */ - for (i=0; i<cnt; i++) - FDKreadBits(hBs, 8); - nBitsLeft -= cnt * 8; - break; - } - } - - if (nBitsLeft < 0) { - frameOk = 0; - goto bail; - } - else { - /* Read fill bits for byte alignment */ - FDKreadBits(hBs, nBitsLeft); - } - } - -bail: - return (frameOk); -} - - -/*! - \brief Read bitstream elements of one channel - - \return SbrFrameOK: 1=ok, 0=error -*/ -int -sbrGetSingleChannelElement (HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ - HANDLE_SBR_FRAME_DATA hFrameData, /*!< Control data of current frame */ - HANDLE_FDK_BITSTREAM hBs, /*!< Handle to struct BIT_BUF */ - HANDLE_PS_DEC hParametricStereoDec, /*!< Handle to PS decoder */ - const UINT flags, - const int overlap - ) -{ - int i; - - - hFrameData->coupling = 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; i<hHeaderData->freqBandData.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; i<hHeaderData->freqBandData.nInvfBands; i++) { - hFrameDataLeft->sbr_invf_mode[i] = (INVF_MODE) FDKreadBits (hBs, 2); - } - - if (hFrameDataLeft->coupling) { - for (i=0; i<hHeaderData->freqBandData.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; i<hHeaderData->freqBandData.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; i<h_frame_data->frameInfo.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; i<nEnv; i++) - hSbrFrameInfo->borders[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;i<nEnvelopes;i++) { - if(pFrameInfo->borders[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; i<nNoiseEnvelopes; i++) { - if(pFrameInfo->bordersNoise[i] >= pFrameInfo->bordersNoise[i+1]) - return 0; - } - - /* Check that every noise border is the same as an envelope border*/ - for(i=0; i<nNoiseEnvelopes; i++) { - startPosNoise = pFrameInfo->bordersNoise[i]; - - for(j=0; j<nEnvelopes; j++) { - if(pFrameInfo->borders[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;i<LPC_ORDER;i++){ - *plowBandReal++ = hLppTrans->lpcFilterStatesReal[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;i<pSettings->nCols+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<FL2FX_DBL(0.0f)) ^ (ac.det<FL2FX_DBL(0.0f))) { - alphar[1] = -alphar[1]; - } - } - } - - if (!useLP) - { - tmp = ( fMultDiv2(ac.r01i,ac.r12r) >> (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<FL2FX_DBL(0.0f)) ^ (ac.det<FL2FX_DBL(0.0f))) { - alphai[1] = -alphai[1]; - } - } - } - } - } - - alphar[0] = FL2FXCONST_SGL(0.0f); - if (!useLP) - alphai[0] = FL2FXCONST_SGL(0.0f); - - if ( ac.r11r != FL2FXCONST_DBL(0.0f) ) { - - /* ac.r11r is always >=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<FL2FX_DBL(0.0f))) - alphar[0] = -alphar[0]; - } - - if (!useLP) - { - tmp = (ac.r01i>>(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<FL2FX_DBL(0.0f))) - alphai[0] = -alphai[0]; - } - } - } - - - if (!useLP) - { - /* Now check the quadratic criteria */ - if( (fMultDiv2(alphar[0],alphar[0]) + fMultDiv2(alphai[0],alphai[0])) >= 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<FL2FX_DBL(0.0f)) ^ (ac.r11r<FL2FX_DBL(0.0f)))) { - k1 = -k1; - } - } - } - if(loBand > 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 (i<min) - return min; - else if (i>max) - 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; env<pBsData->noEnv; 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; env<pBsData->noEnv; 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; env<pBsData->noEnv; 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]<noSubSample for VAR_BORDERS by - duplicating last PS parameters and incrementing noEnv */ - if (pBsData->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; env<pBsData->noEnv; 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; env<pBsData->noEnv; env++) { - UCHAR i; - for (i=0; i<NO_HI_RES_IID_BINS; i++) { - h_ps_d->specificTo.mpeg.coef.aaIidIndexMapped[env][i] = pBsData->aaIidIndex[env][i]; - } - for (i=0; i<NO_HI_RES_ICC_BINS; i++) { - h_ps_d->specificTo.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; env<pBsData->noEnv; 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; env<pBsData->noEnv+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; env<pBsData->noEnv; 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; env<pBsData->noEnv; 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<<PS_EXTENSION_SIZE_BITS)-1) { - cnt += FDKreadBits(hBitBuf, PS_EXTENSION_ESC_COUNT_BITS); - } - while (cnt--) - FDKreadBits(hBitBuf, 8); - } - - - /* new PS data was read from bitstream */ - h_ps_d->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 1729f90..0000000 --- a/libSBRdec/src/psdec.cpp +++ /dev/null @@ -1,1422 +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; i<NO_QMF_BANDS_HYBRID20; i++) { - scale = fMin(scale, getScalefactor(h_ps_d->specificTo.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; i<NO_SAMPLE_DELAY_ALLPASS; i++) { - scale = fMin(scale, getScalefactor(h_ps_d->specificTo.mpeg.aaRealDelayBufferQmf[i], FIRST_DELAY_SB)); - scale = fMin(scale, getScalefactor(h_ps_d->specificTo.mpeg.aaImagDelayBufferQmf[i], FIRST_DELAY_SB)); - } - - for (i=0; i<NO_SAMPLE_DELAY_ALLPASS; i++) { - scale = fMin(scale, getScalefactor(h_ps_d->specificTo.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; i<FIRST_DELAY_SB; i++) { - scale = fMin(scale, getScalefactor(h_ps_d->specificTo.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; i<NO_SUB_QMF_CHANNELS; i++) { - scale = fMin(scale, getScalefactor(h_ps_d->specificTo.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; i<MAX_DELAY_BUFFER_SIZE; i++) - { - INT len; - if (i==0) - len = NO_QMF_CHANNELS-FIRST_DELAY_SB; - else - len = NO_DELAY_BUFFER_BANDS-FIRST_DELAY_SB; - - scale = fMin(scale, getScalefactor(h_ps_d->specificTo.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; i<NO_QMF_BANDS_HYBRID20; i++) { - scaleValues( h_ps_d->specificTo.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; i<NO_SAMPLE_DELAY_ALLPASS; i++) { - scaleValues( h_ps_d->specificTo.mpeg.aaRealDelayBufferQmf[i], FIRST_DELAY_SB, scale ); - scaleValues( h_ps_d->specificTo.mpeg.aaImagDelayBufferQmf[i], FIRST_DELAY_SB, scale ); - } - - for (i=0; i<NO_SAMPLE_DELAY_ALLPASS; i++) { - scaleValues( h_ps_d->specificTo.mpeg.aaRealDelayBufferSubQmf[i], NO_SUB_QMF_CHANNELS, scale ); - scaleValues( h_ps_d->specificTo.mpeg.aaImagDelayBufferSubQmf[i], NO_SUB_QMF_CHANNELS, scale ); - } - - for (i=0; i<FIRST_DELAY_SB; i++) { - scaleValues( h_ps_d->specificTo.mpeg.aaaRealDelayRBufferSerQmf[i], NO_DELAY_LENGTH_VECTORS, scale ); - scaleValues( h_ps_d->specificTo.mpeg.aaaImagDelayRBufferSerQmf[i], NO_DELAY_LENGTH_VECTORS, scale ); - } - - for (i=0; i<NO_SUB_QMF_CHANNELS; i++) { - scaleValues( h_ps_d->specificTo.mpeg.aaaRealDelayRBufferSerSubQmf[i], NO_DELAY_LENGTH_VECTORS, scale ); - scaleValues( h_ps_d->specificTo.mpeg.aaaImagDelayRBufferSerSubQmf[i], NO_DELAY_LENGTH_VECTORS, scale ); - } - - for (i=0; i<MAX_DELAY_BUFFER_SIZE; i++) { - INT len; - if (i==0) - len = NO_QMF_CHANNELS-FIRST_DELAY_SB; - else - len = NO_DELAY_BUFFER_BANDS-FIRST_DELAY_SB; - - scaleValues( h_ps_d->specificTo.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; i<MAX_NUM_COL; i++ ) { */ - for ( i=0; i<noCols; i++ ) { - scaleValues( &fixpQmfReal[i][lsb], (64)-lsb, scale ); - scaleValues( &fixpQmfImag[i][lsb], (64)-lsb, scale ); - } - } - - if ( maxScal != h_ps_d->specificTo.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; i<noCols + (6); i++ ) { - scaleValues( QmfBufferReal[i], lsb, h_ps_d->rescal ); - 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); - -/*! -<pre> - 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) - ---------------------------------------------------------------------------- -</pre> -*/ - - #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<<scale_left, peakDiff<<scale_left, 16); - } - } /* bin */ - - - - - #define DELAY_GROUP_OFFSET 20 - #define NR_OF_DELAY_GROUPS 2 - - FIXP_DBL rTmp, iTmp, rTmp0, iTmp0, rR0, iR0; - - INT TempDelay = h_ps_d->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; m<NO_SERIAL_ALLPASS_LINKS ; m++) { - - INT tmpDelayRSer = h_ps_d->specificTo.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; m<NO_SERIAL_ALLPASS_LINKS ; m++) { - - INT tmpDelayRSer = h_ps_d->specificTo.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; m<NO_SERIAL_ALLPASS_LINKS ; m++) { - if (++h_ps_d->specificTo.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, noFactors; - -/* const UCHAR *pQuantizedIIDs;*/ - - FIXP_SGL invL; - FIXP_DBL ScaleL, ScaleR; - FIXP_DBL Alpha, Beta, AlphasValue; - 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 ; k<NO_SAMPLE_DELAY_ALLPASS; k++) { - FDKmemclear(h_ps_d->specificTo.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; - noFactors = NO_IID_LEVELS_FINE; - /*pQuantizedIIDs = quantizedIIDsFine;*/ - } - - else - { - PScaleFactors = ScaleFactors; /* values are shiftet right by one */ - noIidSteps = NO_IID_STEPS; - noFactors = NO_IID_LEVELS; - /*pQuantizedIIDs = quantizedIIDs;*/ - } - - - /* dequantize and decode */ - for ( group = 0; group < NO_IID_GROUPS; group++ ) { - - bin = bins2groupMap20[group]; - - /*! - <h3> type 'A' rotation </h3> - mixing procedure R_a, used in baseline version<br> - - 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 */ - - ScaleL = ScaleR = 0; - if (noIidSteps + h_ps_d->specificTo.mpeg.coef.aaIidIndexMapped[env][bin] >= 0 && noIidSteps + h_ps_d->specificTo.mpeg.coef.aaIidIndexMapped[env][bin] < noFactors) - ScaleR = PScaleFactors[noIidSteps + h_ps_d->specificTo.mpeg.coef.aaIidIndexMapped[env][bin]]; - if (noIidSteps - h_ps_d->specificTo.mpeg.coef.aaIidIndexMapped[env][bin] >= 0 && noIidSteps - h_ps_d->specificTo.mpeg.coef.aaIidIndexMapped[env][bin] < noFactors) - ScaleL = PScaleFactors[noIidSteps - h_ps_d->specificTo.mpeg.coef.aaIidIndexMapped[env][bin]]; - - AlphasValue = 0; - if (h_ps_d->specificTo.mpeg.coef.aaIccIndexMapped[env][bin] >= 0) - AlphasValue = Alphas[h_ps_d->specificTo.mpeg.coef.aaIccIndexMapped[env][bin]]; - Beta = fMult (fMult( AlphasValue, ( ScaleR - ScaleL )), FIXP_SQRT05); - Alpha = AlphasValue>>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; - - - /**********************************************************************************************/ - /*! - <h2> Mapping </h2> - - The number of stereo bands that is actually used depends on the number of availble - parameters for IID and ICC: - <pre> - 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 - </pre> - 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. - **********************************************************************************************/ - - /************************************************************************************************/ - /*! - <h2> Mixing </h2> - - 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: - <pre> - 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) - </pre> - 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. - ************************************************************************************************/ - - - /************************************************************************************************/ - /*! - <h2>Phase parameters </h2> - With disabled phase parameters (which is the case in baseline version), the H-matrices are - just calculated by: - - <pre> - H11(k,n[e+1] = h11(b(k)) - (...) - b(k): parameter index according to mapping table - </pre> - - <h2>Processing of the samples in the sub subbands </h2> - 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. - - <pre> - - ------------- ---------- ------------- - | 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 - - </pre> - */ - - /* 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 -<pre> - 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); -</pre> -*/ - -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 -<pre> - 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) | | -</pre> -*/ - -/* 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 - -*******************************************************************************/ - -/*! <pre> - 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) ---/ - - </pre> - 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; slot<hSbrDec->LppTrans.pSettings->overlap; slot++) { - QmfBufferReal[slot] = ptr; ptr += (64); - } - - /* Assign timeslots to Workbuffer1 */ - ptr = hSbrDec->WorkBuffer1; - for(i=0; i<noCols; i++) { - QmfBufferReal[slot] = ptr; ptr += (64); - slot++; - } -} - - -static void assignHqTimeSlots( HANDLE_SBR_DEC hSbrDec, /*!< handle to Decoder channel */ - FIXP_DBL **QmfBufferReal, - FIXP_DBL **QmfBufferImag, - int noCols ) -{ - FIXP_DBL *ptr; - int slot; - - /* Number of QMF timeslots in one half of a frame (size of Workbuffer1 or 2): */ - int halflen = (noCols >> 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; slot<hSbrDec->LppTrans.pSettings->overlap; slot++) { - QmfBufferReal[slot] = ptr; ptr += (64); - QmfBufferImag[slot] = ptr; ptr += (64); - } - - /* Assign first half of timeslots to Workbuffer1 */ - ptr = hSbrDec->WorkBuffer1; - for(; slot<halflen; slot++) { - QmfBufferReal[slot] = ptr; ptr += (64); - QmfBufferImag[slot] = ptr; ptr += (64); - } - - /* Assign second half of timeslots to Workbuffer2 */ - ptr = hSbrDec->WorkBuffer2; - for(; slot<totCols; slot++) { - QmfBufferReal[slot] = ptr; ptr += (64); - QmfBufferImag[slot] = ptr; ptr += (64); - } -} - - -static void assignTimeSlots( HANDLE_SBR_DEC hSbrDec, /*!< handle to Decoder channel */ - int noCols, - int useLP ) -{ - /* assign qmf time slots */ - hSbrDec->useLP = 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; i<hHeaderData->freqBandData.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; i<LPC_ORDER; i++){ - /* - Store the unmodified qmf Slots values (required for LPC filtering) - */ - if (! (flags & SBRDEC_LOW_POWER)) { - FDKmemcpy(hSbrDec->LppTrans.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; i<hSbrDec->LppTrans.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; i<hSbrDec->LppTrans.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; l<hSbrDec->LppTrans.pSettings->overlap; l++) { - FDKmemclear(&OverlapBufferReal[l][startBand], size*sizeof(FIXP_DBL)); - FDKmemclear(&OverlapBufferImag[l][startBand], size*sizeof(FIXP_DBL)); - } - } else - for (l=startSlot; l<hSbrDec->LppTrans.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; l<startSlot; l++) { - scaleValues( OverlapBufferReal[l] + startBand, stopBand-startBand, delta_scale ); - scaleValues( OverlapBufferImag[l] + startBand, stopBand-startBand, delta_scale ); - } - } else - for (l=0; l<startSlot; l++) { - scaleValues( OverlapBufferReal[l] + startBand, stopBand-startBand, delta_scale ); - } - - - /* - Initialize transposer and limiter - */ - sbrError = resetLppTransposer (&hSbrDec->LppTrans, - 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 <br> - Dimension: (8) */ -C_ALLOC_MEM2(Ram_SbrDecElement, SBR_DECODER_ELEMENT, 1, (8)) -/*! SBR Decoder individual channel data <br> - Dimension: (8) */ -C_ALLOC_MEM2(Ram_SbrDecChannel, SBR_CHANNEL, 1, (8)+1) - -/*! Filter states for QMF-synthesis. <br> - 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 - <br> - Depending on the mode set by DONT_USE_CORE_WORKBUFFER, workbuffers are - defined additionally to the CoreWorkbuffer. - <br> - The size of WorkBuffers is ((1024)/(32))*(64) = 2048. - <br> - 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. - <br> - WorkBuffer1 is reused in the AAC core (-> aacdecoder.cpp, aac_ram.cpp) - <br> - - Use of WorkBuffers: - <pre> - - ------------------------------------------------------------- - 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 - - ------------------------------------------------------------- - - </pre> - -*/ -/* @{ */ -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<INV_TABLE_SIZE; i++) { - d = 0.5f / ( 0.5f+(double)(i+1)/(INV_TABLE_SIZE*2) ) ; - invTable[i] = FL2FX_SGL(d); - } -*/ -const FIXP_SGL FDK_sbrDecoder_invTable[INV_TABLE_SIZE] = -{ - 0x7f80, 0x7f01, 0x7e83, 0x7e07, 0x7d8b, 0x7d11, 0x7c97, 0x7c1e, - 0x7ba6, 0x7b2f, 0x7ab9, 0x7a44, 0x79cf, 0x795c, 0x78e9, 0x7878, - 0x7807, 0x7796, 0x7727, 0x76b9, 0x764b, 0x75de, 0x7572, 0x7506, - 0x749c, 0x7432, 0x73c9, 0x7360, 0x72f9, 0x7292, 0x722c, 0x71c6, - 0x7161, 0x70fd, 0x709a, 0x7037, 0x6fd5, 0x6f74, 0x6f13, 0x6eb3, - 0x6e54, 0x6df5, 0x6d97, 0x6d39, 0x6cdc, 0x6c80, 0x6c24, 0x6bc9, - 0x6b6f, 0x6b15, 0x6abc, 0x6a63, 0x6a0b, 0x69b3, 0x695c, 0x6906, - 0x68b0, 0x685a, 0x6806, 0x67b1, 0x675e, 0x670a, 0x66b8, 0x6666, - 0x6614, 0x65c3, 0x6572, 0x6522, 0x64d2, 0x6483, 0x6434, 0x63e6, - 0x6399, 0x634b, 0x62fe, 0x62b2, 0x6266, 0x621b, 0x61d0, 0x6185, - 0x613b, 0x60f2, 0x60a8, 0x6060, 0x6017, 0x5fcf, 0x5f88, 0x5f41, - 0x5efa, 0x5eb4, 0x5e6e, 0x5e28, 0x5de3, 0x5d9f, 0x5d5a, 0x5d17, - 0x5cd3, 0x5c90, 0x5c4d, 0x5c0b, 0x5bc9, 0x5b87, 0x5b46, 0x5b05, - 0x5ac4, 0x5a84, 0x5a44, 0x5a05, 0x59c6, 0x5987, 0x5949, 0x590a, - 0x58cd, 0x588f, 0x5852, 0x5815, 0x57d9, 0x579d, 0x5761, 0x5725, - 0x56ea, 0x56af, 0x5675, 0x563b, 0x5601, 0x55c7, 0x558e, 0x5555, - 0x551c, 0x54e3, 0x54ab, 0x5473, 0x543c, 0x5405, 0x53ce, 0x5397, - 0x5360, 0x532a, 0x52f4, 0x52bf, 0x5289, 0x5254, 0x521f, 0x51eb, - 0x51b7, 0x5183, 0x514f, 0x511b, 0x50e8, 0x50b5, 0x5082, 0x5050, - 0x501d, 0x4feb, 0x4fba, 0x4f88, 0x4f57, 0x4f26, 0x4ef5, 0x4ec4, - 0x4e94, 0x4e64, 0x4e34, 0x4e04, 0x4dd5, 0x4da6, 0x4d77, 0x4d48, - 0x4d19, 0x4ceb, 0x4cbd, 0x4c8f, 0x4c61, 0x4c34, 0x4c07, 0x4bd9, - 0x4bad, 0x4b80, 0x4b54, 0x4b27, 0x4afb, 0x4acf, 0x4aa4, 0x4a78, - 0x4a4d, 0x4a22, 0x49f7, 0x49cd, 0x49a2, 0x4978, 0x494e, 0x4924, - 0x48fa, 0x48d1, 0x48a7, 0x487e, 0x4855, 0x482d, 0x4804, 0x47dc, - 0x47b3, 0x478b, 0x4763, 0x473c, 0x4714, 0x46ed, 0x46c5, 0x469e, - 0x4677, 0x4651, 0x462a, 0x4604, 0x45de, 0x45b8, 0x4592, 0x456c, - 0x4546, 0x4521, 0x44fc, 0x44d7, 0x44b2, 0x448d, 0x4468, 0x4444, - 0x441f, 0x43fb, 0x43d7, 0x43b3, 0x4390, 0x436c, 0x4349, 0x4325, - 0x4302, 0x42df, 0x42bc, 0x4299, 0x4277, 0x4254, 0x4232, 0x4210, - 0x41ee, 0x41cc, 0x41aa, 0x4189, 0x4167, 0x4146, 0x4125, 0x4104, - 0x40e3, 0x40c2, 0x40a1, 0x4081, 0x4060, 0x4040, 0x4020, 0x4000 -}; - diff --git a/libSBRdec/src/sbr_rom.h b/libSBRdec/src/sbr_rom.h deleted file mode 100644 index 1f800bc..0000000 --- a/libSBRdec/src/sbr_rom.h +++ /dev/null @@ -1,235 +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 Declaration of constant tables - -*/ -#ifndef __rom_H -#define __rom_H - -#include "sbrdecoder.h" -#include "env_extr.h" -#include "qmf.h" - -#define INV_INT_TABLE_SIZE 49 -#define SBR_NF_NO_RANDOM_VAL 512 /*!< Size of random number array for noise floor */ - -/* - Frequency scales -*/ -extern const UCHAR FDK_sbrDecoder_sbr_start_freq_16[16]; -extern const UCHAR FDK_sbrDecoder_sbr_start_freq_22[16]; -extern const UCHAR FDK_sbrDecoder_sbr_start_freq_24[16]; -extern const UCHAR FDK_sbrDecoder_sbr_start_freq_32[16]; -extern const UCHAR FDK_sbrDecoder_sbr_start_freq_40[16]; -extern const UCHAR FDK_sbrDecoder_sbr_start_freq_44[16]; -extern const UCHAR FDK_sbrDecoder_sbr_start_freq_48[16]; -extern const UCHAR FDK_sbrDecoder_sbr_start_freq_64[16]; -extern const UCHAR FDK_sbrDecoder_sbr_start_freq_88[16]; - -/* - Low-Power-Profile Transposer -*/ -#define NUM_WHFACTOR_TABLE_ENTRIES 9 -extern const USHORT FDK_sbrDecoder_sbr_whFactorsIndex[NUM_WHFACTOR_TABLE_ENTRIES]; -extern const FIXP_DBL FDK_sbrDecoder_sbr_whFactorsTable[NUM_WHFACTOR_TABLE_ENTRIES][6]; - - - -/* - Envelope Adjustor -*/ -extern const FIXP_SGL FDK_sbrDecoder_sbr_limGains_m[4]; -extern const UCHAR FDK_sbrDecoder_sbr_limGains_e[4]; -extern const FIXP_SGL FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4[4]; -extern const FIXP_DBL FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4_DBL[4]; -extern const FIXP_SGL FDK_sbrDecoder_sbr_smoothFilter[4]; -extern const FIXP_SGL FDK_sbrDecoder_sbr_randomPhase[SBR_NF_NO_RANDOM_VAL][2]; -extern const FIXP_SGL harmonicPhaseX [2][4]; - -/* - Envelope Extractor -*/ -extern const int FDK_sbrDecoder_envelopeTable_8 [8][5]; -extern const int FDK_sbrDecoder_envelopeTable_15 [15][6]; -extern const int FDK_sbrDecoder_envelopeTable_16 [16][6]; - -extern const FRAME_INFO FDK_sbrDecoder_sbr_frame_info1_15; -extern const FRAME_INFO FDK_sbrDecoder_sbr_frame_info2_15; -extern const FRAME_INFO FDK_sbrDecoder_sbr_frame_info4_15; -extern const FRAME_INFO FDK_sbrDecoder_sbr_frame_info8_15; - -extern const FRAME_INFO FDK_sbrDecoder_sbr_frame_info1_16; -extern const FRAME_INFO FDK_sbrDecoder_sbr_frame_info2_16; -extern const FRAME_INFO FDK_sbrDecoder_sbr_frame_info4_16; -extern const FRAME_INFO FDK_sbrDecoder_sbr_frame_info8_16; - -extern const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvLevel10T[120][2]; -extern const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvLevel10F[120][2]; -extern const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvBalance10T[48][2]; -extern const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvBalance10F[48][2]; -extern const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvLevel11T[62][2]; -extern const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvLevel11F[62][2]; -extern const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvBalance11T[24][2]; -extern const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvBalance11F[24][2]; -extern const SCHAR FDK_sbrDecoder_sbr_huffBook_NoiseLevel11T[62][2]; -extern const SCHAR FDK_sbrDecoder_sbr_huffBook_NoiseBalance11T[24][2]; - - -/* - Parametric stereo -*/ - - -extern const FIXP_DBL decayScaleFactTable[NO_QMF_CHANNELS]; - -/* FIX_BORDER can have 0, 1, 2, 4 envelops */ -extern const UCHAR FDK_sbrDecoder_aFixNoEnvDecode[4]; - -/* IID & ICC Huffman codebooks */ -extern const SCHAR aBookPsIidTimeDecode[28][2]; -extern const SCHAR aBookPsIidFreqDecode[28][2]; -extern const SCHAR aBookPsIccTimeDecode[14][2]; -extern const SCHAR aBookPsIccFreqDecode[14][2]; - -/* IID-fine Huffman codebooks */ - -extern const SCHAR aBookPsIidFineTimeDecode[60][2]; -extern const SCHAR aBookPsIidFineFreqDecode[60][2]; - -/* the values of the following 3 tables are shiftet right by 1 ! */ -extern const FIXP_DBL ScaleFactors[NO_IID_LEVELS]; -extern const FIXP_DBL ScaleFactorsFine[NO_IID_LEVELS_FINE]; -extern const FIXP_DBL Alphas[NO_ICC_LEVELS]; - -#if defined(ARCH_PREFER_MULT_32x16) -extern const FIXP_SGL aAllpassLinkDecaySer[NO_SERIAL_ALLPASS_LINKS]; -extern const FIXP_SGL aaFractDelayPhaseFactorReQmf[NO_QMF_CHANNELS]; -extern const FIXP_SGL aaFractDelayPhaseFactorImQmf[NO_QMF_CHANNELS]; -extern const FIXP_SGL aaFractDelayPhaseFactorReSubQmf20[NO_SUB_QMF_CHANNELS]; -extern const FIXP_SGL aaFractDelayPhaseFactorImSubQmf20[NO_SUB_QMF_CHANNELS]; - -extern const FIXP_SGL aaFractDelayPhaseFactorSerReQmf[NO_QMF_CHANNELS][NO_SERIAL_ALLPASS_LINKS]; -extern const FIXP_SGL aaFractDelayPhaseFactorSerImQmf[NO_QMF_CHANNELS][NO_SERIAL_ALLPASS_LINKS]; -extern const FIXP_SGL aaFractDelayPhaseFactorSerReSubQmf20[NO_SUB_QMF_CHANNELS][NO_SERIAL_ALLPASS_LINKS]; -extern const FIXP_SGL aaFractDelayPhaseFactorSerImSubQmf20[NO_SUB_QMF_CHANNELS][NO_SERIAL_ALLPASS_LINKS]; - -extern const FIXP_SGL p8_13_20[13]; -extern const FIXP_SGL p2_13_20[13]; - -#else -extern const FIXP_DBL aAllpassLinkDecaySer[NO_SERIAL_ALLPASS_LINKS]; -extern const FIXP_DBL aaFractDelayPhaseFactorReQmf[NO_QMF_CHANNELS]; -extern const FIXP_DBL aaFractDelayPhaseFactorImQmf[NO_QMF_CHANNELS]; -extern const FIXP_DBL aaFractDelayPhaseFactorReSubQmf20[NO_SUB_QMF_CHANNELS]; -extern const FIXP_DBL aaFractDelayPhaseFactorImSubQmf20[NO_SUB_QMF_CHANNELS]; - -extern const FIXP_DBL aaFractDelayPhaseFactorSerReQmf[NO_QMF_CHANNELS][NO_SERIAL_ALLPASS_LINKS]; -extern const FIXP_DBL aaFractDelayPhaseFactorSerImQmf[NO_QMF_CHANNELS][NO_SERIAL_ALLPASS_LINKS]; -extern const FIXP_DBL aaFractDelayPhaseFactorSerReSubQmf20[NO_SUB_QMF_CHANNELS][NO_SERIAL_ALLPASS_LINKS]; -extern const FIXP_DBL aaFractDelayPhaseFactorSerImSubQmf20[NO_SUB_QMF_CHANNELS][NO_SERIAL_ALLPASS_LINKS]; - -extern const FIXP_DBL p8_13_20[13]; -extern const FIXP_DBL p2_13_20[13]; -#endif - -extern const UCHAR aAllpassLinkDelaySer[3]; -extern const UCHAR delayIndexQmf[NO_QMF_CHANNELS]; -extern const UCHAR groupBorders20[NO_IID_GROUPS + 1]; -extern const UCHAR groupBorders34[NO_IID_GROUPS_HI_RES + 1]; -extern const UCHAR bins2groupMap20[NO_IID_GROUPS]; -extern const UCHAR quantizedIIDs[NO_IID_STEPS]; -extern const UCHAR quantizedIIDsFine[NO_IID_STEPS_FINE]; -extern const UCHAR FDK_sbrDecoder_aNoIidBins[3]; -extern const UCHAR FDK_sbrDecoder_aNoIccBins[3]; - - -/* Lookup tables for some arithmetic functions */ - -#define INV_TABLE_BITS 8 -#define INV_TABLE_SIZE (1<<INV_TABLE_BITS) -extern const FIXP_SGL FDK_sbrDecoder_invTable[INV_TABLE_SIZE]; - -#endif // __rom_H diff --git a/libSBRdec/src/sbr_scale.h b/libSBRdec/src/sbr_scale.h deleted file mode 100644 index 5fccd71..0000000 --- a/libSBRdec/src/sbr_scale.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 Sbr scaling factors, -To deal with the dynamic range in the different processing stages, a -fixed point specific code has to rely on scaling factors. A floating -point code carries a scaling factor -- the exponent -- for each value, -so scaling is not necessary there. - -The output of the core decoder (low band) is scaled up to cover as much -as possible bits for each value. As high band and low band are processed -in different algorithm sections, they require their own scaling -factors. In addition, any static buffers, e.g. filter states, require a -separate scaling factor as well. The code takes care to do the proper -adjustment, if scaling factors of a filter state and the time signal differ. - -\sa #QMF_SCALE_FACTOR, \ref documentationOverview -*/ - -#ifndef __SBR_SCALE_H -#define __SBR_SCALE_H - -/*! -\verbatim - scale: - 0 left aligned e.g. |max| >=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<num_bands0;i++) - diff_tot[i] = dk; - - /* If linear scale wasn't achieved */ - /* and we got too wide SBR area */ - if (k2_diff < 0) { - incr = 1; - i = 0; - } - - /* If linear scale wasn't achieved */ - /* and we got too small SBR area */ - if (k2_diff > 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<num_bands; j++) - //temp = fMult(temp,bandfactor); - temp = fMultDiv2(temp,bandfactor)<<2; - - if (temp<start) { /* Factor too strong, make it weaker */ - if (direction == 0) - /* Halfen step. Right shift is not done as fract because otherwise the - lowest bit cannot be cleared due to rounding */ - step = (FIXP_DBL)((LONG)step >> 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 a341746..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 - <a HREF="http://www.doxygen.org">doxygen</a>. As part of this documentation - you can find more extensive descriptions about key concepts and algorithms at the following locations: - - <h2>Programming</h2> - - \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() - - <h2>Algorithmic details</h2> - \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; ch<self->pSbrElement[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; ch<SBRDEC_MAX_CH_PER_ELEMENT; ch++) { - if (self->pSbrElement[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; ch<elChannels; ch++) - { - if (self->pSbrElement[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<<i)) >> 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 || ((*numOutChannels == 2) && 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; sbrElementNum<self->numSbrElements; 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; sbrElementNum<self->numSbrElements; 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. <br> - The base for exponents is 2. Example: \f$ a = a\_m * 2^{a\_e} \f$<br> -*/ -/************************************************************************/ -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. <br> - The base for exponents is 2. Example: \f$ a = a\_m * 2^{a\_e} \f$<br> - - 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. <br> - The base for the exponent is 2. Example: \f$ a = a\_m * 2^{a\_e} \f$<br> - 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<<SI_SBR_EXTENSION_SIZE_BITS) - 1; - INT writtenNoBits = 0; /* needed to byte align the extended data */ - - payloadBits += FDKwriteBits (hBitStream, 1, SI_SBR_EXTENDED_DATA_BITS); - FDK_ASSERT(extDataSize <= SBR_EXTENDED_DATA_MAX_CNT); - - if (extDataSize < maxExtSize) { - payloadBits += FDKwriteBits (hBitStream, extDataSize, SI_SBR_EXTENSION_SIZE_BITS); - } else { - payloadBits += FDKwriteBits (hBitStream, maxExtSize, SI_SBR_EXTENSION_SIZE_BITS); - payloadBits += FDKwriteBits (hBitStream, extDataSize - maxExtSize, SI_SBR_EXTENSION_ESC_COUNT_BITS); - } - - /* parametric coding signalled here? */ - if(hParametricStereo){ - writtenNoBits += FDKwriteBits (hBitStream, EXTENSION_ID_PS_CODING, SI_SBR_EXTENSION_ID_BITS); - writtenNoBits += FDKsbrEnc_PSEnc_WritePSData(hParametricStereo, hBitStream); - } - - payloadBits += writtenNoBits; - - /* byte alignment */ - writtenNoBits = writtenNoBits%8; - if(writtenNoBits) - payloadBits += FDKwriteBits(hBitStream, 0, (8 - writtenNoBits)); - } else { - payloadBits += FDKwriteBits (hBitStream, 0, SI_SBR_EXTENDED_DATA_BITS); - } - - return payloadBits; -} - - -/***************************************************************************** - - functionname: writeSyntheticCodingData - description: writes bits corresponding to the "synthetic-coding"-extension - returns: number of bits written - input: - output: - -*****************************************************************************/ -static INT writeSyntheticCodingData (HANDLE_SBR_ENV_DATA sbrEnvData, - HANDLE_FDK_BITSTREAM hBitStream) - -{ - INT i; - INT payloadBits = 0; - - payloadBits += FDKwriteBits (hBitStream, sbrEnvData->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<<i)) ? (1) : (0) ; - - (*crc)<<=1; - if(flag) (*crc) ^= crcPoly; - } -} - - -/* ***************************** FDKsbrEnc_InitSbrBitstream **********************************/ -/** - * @fn - * @brief Inittialisation of sbr bitstream, write of dummy header and CRC - * @return none - * - * - * - */ - -INT FDKsbrEnc_InitSbrBitstream(HANDLE_COMMON_DATA hCmonData, - UCHAR *memoryBase, /*!< Pointer to bitstream buffer */ - INT memorySize, /*!< Length of bitstream buffer in bytes */ - HANDLE_FDK_CRCINFO hCrcInfo, - UINT sbrSyntaxFlags) /*!< SBR syntax flags */ -{ - INT crcRegion = 0; - - /* reset bit buffer */ - FDKresetBitbuffer(&hCmonData->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;i<numCrcBits;i++){ - INT bit; - bit = FDKreadBits(&tmpCRCBuf,1); - crcAdvance(SBR_CRC_POLY,SBR_CRC_MASK,&crcReg,bit,1); - } - crcReg &= (SBR_CRC_RANGE); - - /* - * Write CRC data. - */ - FDKwriteBits (&hCmonData->tmpWriteBitbuf, 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<stopBand; b++) { - energyBand[b]=FL2FXCONST_DBL(0.0f); - } - } else { - for (b=startBand; b<stopBand; b++) { - energyBand[b]=Energies[15][b]>>4; - } - } - - for (k=0; k<15; k++) { - for (b=startBand; b<stopBand; b++) { - energyBand[b] += Energies[k][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<maxNEnergyValues; k++) { - energyMax[k] = energyBand[startBand+k]; - no_enMaxBand[k] = startBand+k; - if (energyMaxMin > energyMax[k]) { - energyMaxMin = energyMax[k]; - posEnergyMaxMin = k; - } - } - - for (b=startBand+maxNEnergyValues; b<stopBand; b++) { - if (energyBand[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<maxNEnergyValues; k++) { - if (energyMaxMin > energyMax[k]) { - energyMaxMin = energyMax[k]; - posEnergyMaxMin = k; - } - } - } - } - /*** End determine 5 highest band-energies ***/ - - /* Get tonality values for 5 highest energies */ - for (e=0; e<maxNEnergyValues; e++) { - tonalityBand[e]=FL2FXCONST_DBL(0.0f); - for (k=0; k<noEstPerFrame; k++) { - tonalityBand[e] += quotaMatrix[startIndex + k][no_enMaxBand[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<numberCols; k++) { - scale = fixMin(scale, fixMin(getScalefactor(realValues[k], numberBands), getScalefactor(imagValues[k], numberBands))); - } - - /* Tweak scaling stability for zero signal to non-zero signal transitions */ - if (scale >= 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<numberCols; k+=2) - { - /* Load band vector addresses of 2 consecutive timeslots */ - FIXP_DBL *RESTRICT r0 = realValues[k]; - FIXP_DBL *RESTRICT i0 = imagValues[k]; - FIXP_DBL *RESTRICT r1 = realValues[k+1]; - FIXP_DBL *RESTRICT i1 = imagValues[k+1]; - for (j=0; j<numberBands; j++) - { - FIXP_DBL energy; - FIXP_DBL tr0,tr1,ti0,ti1; - - /* Read QMF values of 2 timeslots */ - tr0 = r0[j]; tr1 = r1[j]; ti0 = i0[j]; ti1 = i1[j]; - - /* Scale QMF Values and Calc Energy of both timeslots */ - tr0 <<= scale; - ti0 <<= scale; - energy = fPow2AddDiv2(fPow2Div2(tr0), ti0) >> 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<numberCols>>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<numberCols; k++) { - scale = fixMin(scale, fixMin(getScalefactor(realValues[k], numberBands), getScalefactor(imagValues[k], numberBands))); - } - - /* Tweak scaling stability for zero signal to non-zero signal transitions */ - if (scale >= 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<numberCols; k++) - { - /* Load band vector addresses of 2 consecutive timeslots */ - FIXP_DBL *RESTRICT r0 = realValues[k]; - FIXP_DBL *RESTRICT i0 = imagValues[k]; - for (j=0; j<numberBands; j++) - { - FIXP_DBL energy; - FIXP_DBL tr0,ti0; - - /* Read QMF values of 2 timeslots */ - tr0 = r0[j]; ti0 = i0[j]; - - /* Scale QMF Values and Calc Energy of both timeslots */ - tr0 <<= scale; - ti0 <<= scale; - energy = fPow2AddDiv2(fPow2Div2(tr0), ti0); - *nrgValues++ = energy; - - max_val = fixMax(max_val, energy); - - /* Write back scaled QMF values */ - r0[j] = tr0; i0[j] = ti0; - } - } - } - /* 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<numberCols; k++) { - scaleValues(energyValues[k], nrgValues, numberBands, scale); - nrgValues += numberBands; - } - *energyScale += scale; - } - - /* Free Scratch buffer */ - C_ALLOC_SCRATCH_END(tmpNrg, FIXP_DBL, QMF_MAX_TIME_SLOTS*QMF_CHANNELS/2); -} - -/***************************************************************************/ -/*! - - \brief Quantisation of the panorama value (balance) - - \return the quantized pan value - -****************************************************************************/ -static INT -mapPanorama(INT nrgVal, /*! integer value of the energy */ - INT ampRes, /*! amplitude resolution [1.5/3dB] */ - INT *quantError /*! quantization error of energy val*/ - ) -{ - int i; - INT min_val, val; - UCHAR panIndex; - INT sign; - - sign = nrgVal > 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; i<stopBandHigh; i++){ - if(h_sbr->encEnvData.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<<YBufferSzShift); - - if(missingHarmonic[j]){ - - int k; - count[j] = stop_pos - start_pos; - nrgLeft = FL2FXCONST_DBL(0.0f); - - for (k = li; k < ui; k++) { - FIXP_DBL tmpNrg; - tmpNrg = getEnvSfbEnergy(k, - k+1, - start_pos, - stop_pos, - border_pos, - YBufferLeft, - YBufferSzShift, - scaleLeft0, - scaleLeft1); - - nrgLeft = fixMax(nrgLeft, tmpNrg); - } - - /* Energy lowering compensation */ - nrgLeft = mhLoweringEnergy(nrgLeft, ui-li); - - if (stereoMode == SBR_COUPLING) { - - nrgRight = FL2FXCONST_DBL(0.0f); - - for (k = li; k < ui; k++) { - FIXP_DBL tmpNrg; - tmpNrg = getEnvSfbEnergy(k, - k+1, - start_pos, - stop_pos, - border_pos, - YBufferRight, - YBufferSzShift, - scaleRight0, - scaleRight1); - - nrgRight = fixMax(nrgRight, tmpNrg); - } - - /* Energy lowering compensation */ - nrgRight = mhLoweringEnergy(nrgRight, ui-li); - } - } /* end missingHarmonic */ - else{ - count[j] = (stop_pos - start_pos) * (ui - li); - - nrgLeft = getEnvSfbEnergy(li, - ui, - start_pos, - stop_pos, - border_pos, - YBufferLeft, - YBufferSzShift, - scaleLeft0, - scaleLeft1); - - if (stereoMode == SBR_COUPLING) { - nrgRight = getEnvSfbEnergy(li, - ui, - start_pos, - stop_pos, - border_pos, - YBufferRight, - YBufferSzShift, - scaleRight0, - scaleRight1); - } - } /* !missingHarmonic */ - - /* save energies */ - pNrgLeft[j] = nrgLeft; - pNrgRight[j] = nrgRight; - envNrgLeft += (nrgLeft>>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<<sc0) - CalcLdData(nrgRight<<sc1) - scaleFract; - - /* ld64 to integer conversion */ - nrgRight = (FIXP_DBL)(LONG)(nrgRight) >> (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; i<MAX_NUM_NOISE_VALUES; i++) - hEnvChan->encEnvData.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;i<MAX_FREQ_COEFFS;i++){ - /* - swap sfb energies - */ - itmp = h_envChan[ch]->sbrCodeEnvelope.sfb_nrg_prev[i]; - h_envChan[ch]->sbrCodeEnvelope.sfb_nrg_prev[i]=sfbNrgPrevTemp[ch][i]; - sfbNrgPrevTemp[ch][i]=itmp; - } - for(i=0;i<MAX_NUM_NOISE_COEFFS;i++){ - /* - swap noise energies - */ - itmp = h_envChan[ch]->sbrCodeNoiseFloor.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; ch<nChannels; ch++) - { - int YBufferLength = h_envChan[ch]->sbrExtractEnvelope.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; i<YBufferLength; i++) { - FDKmemclear( hSbrCut->YBuffer[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; i<nEnv; i++) - hSbrFrameInfo->borders[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; i<nEnv; i++){ - k = hSbrFrameInfo->borders[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; j<stopIndex; j++) { - for(i=startChannel; i<stopChannel; i++) { - quotaVecOrig[i] += fMult(quotaMatrixOrig[j][i], invIndex); - - if(indexVector[i] != -1) - quotaVecSbr[i] += fMult(quotaMatrixOrig[j][indexVector[i]], invIndex); - } - detectorValues->avgNrg += 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; i<stopChannel; i++) { - origQuota += fMultDiv2(quotaVecOrig[i], invChannel); - sbrQuota += fMultDiv2(quotaVecSbr[i], invChannel); - } - - /* - Calculate the mean value for the x strongest components - */ - FDKsbrEnc_Shellsort_fract(quotaVecOrig+startChannel,stopChannel-startChannel); - FDKsbrEnc_Shellsort_fract(quotaVecSbr+startChannel,stopChannel-startChannel); - - origQuotaMeanStrongest = FL2FXCONST_DBL(0.0f); - sbrQuotaMeanStrongest = FL2FXCONST_DBL(0.0f); - - temp = min(stopChannel - startChannel, numberOfStrongest); - invTemp = GetInvInt(temp); - - for(i=0; i<temp; i++) { - origQuotaMeanStrongest += fMultDiv2(quotaVecOrig[i + stopChannel - temp], invTemp); - sbrQuotaMeanStrongest += fMultDiv2(quotaVecSbr[i + stopChannel - temp], invTemp); - } - - /* - The value for the strongest component - */ - detectorValues->origQuotaMax = 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;i<INVF_SMOOTHING_LENGTH+1;i++) { - detectorValues->origQuotaMeanFilt += 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;i<hInvFilt->noDetectorBandsMax;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<lu;k++){ - maxValOrig = fixMax(maxValOrig, pTonalityOrig[k]); - maxValSbr = fixMax(maxValSbr, pTonalityOrig[indexVector[k]]); - } - - if ((maxValSbr >= 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<nSfb;i++) - { - INT ll = pFreqBandTable[i]; - INT lu = pFreqBandTable[i+1]; - pSfmOrigVec[i] = (FIXP_DBL)(MAXVAL_DBL>>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<lu; j++) { - sfmOrig = pQuotaBuffer[j]; - sfmTransp = pQuotaBuffer[indexVector[j]]; - - amOrig += fMult(sfmOrig, invBands); - amTransp += fMult(sfmTransp, invBands); - - shiftFac0 = CountLeadingBits(sfmOrig); - shiftFac1 = CountLeadingBits(sfmTransp); - - gmOrig = fMult(gmOrig, sfmOrig<<shiftFac0); - gmTransp = fMult(gmTransp, sfmTransp<<shiftFac1); - - shiftFacSum0 += shiftFac0; - shiftFacSum1 += shiftFac1; - } - - if (gmOrig > 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<nSfb;i++){ - if(maxDerivPos >= 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<nSfb; i++) { - pAddHarmSfb[i] = pAddHarmSfb[i] || detectionVectors[est][i]; - } - } - - if(newDetectionAllowed == 1){ - /* - * Check for duplication of sines located - * on the border of two scf-bands. - *************************************************/ - for(i=0;i<nSfb-1;i++) { - li = pFreqBandTable[i]; - ui = pFreqBandTable[i+1]; - - /* detection in adjacent channels.*/ - if(pAddHarmSfb[i] && pAddHarmSfb[i+1]) { - FIXP_DBL maxVal1, maxVal2; - INT maxPos1, maxPos2, maxPosTime1, maxPosTime2; - - li = pFreqBandTable[i]; - ui = pFreqBandTable[i+1]; - - /* Find maximum tonality in the the two scf bands.*/ - maxPosTime1 = start; - maxPos1 = li; - maxVal1 = quotaBuffer[start][li]; - for(est = start; est < stop; est++){ - for(j = li; j<ui; j++){ - if(quotaBuffer[est][j] > 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<ui; j++){ - if(quotaBuffer[est][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<stop; est++){ - detectionVectors[est][i+1] = 0; - } - } - else{ - if(pPrevAddHarmSfb[i] == 0 && pPrevAddHarmSfb[i+1] == 1){ - /* Keep the upper, remove the lower.*/ - pAddHarmSfb[i] = 0; - for(est=start; est<stop; est++){ - detectionVectors[est][i] = 0; - } - } - else{ - /* If the maximum values are in adjacent QMF-channels, and if the signs indicate that it is the same sine, - we need to remove the lowest of the two.*/ - if(maxVal1 > maxVal2){ - if(signBuffer[maxPosTime1][maxPos2] < 0 && signBuffer[maxPosTime1][maxPos1] > 0){ - /* Keep the lower, remove the upper.*/ - pAddHarmSfb[i+1] = 0; - for(est=start; est<stop; est++){ - detectionVectors[est][i+1] = 0; - } - } - } - else{ - if(signBuffer[maxPosTime2][maxPos2] < 0 && signBuffer[maxPosTime2][maxPos1] > 0){ - /* Keep the upper, remove the lower.*/ - pAddHarmSfb[i] = 0; - for(est=start; est<stop; est++){ - detectionVectors[est][i] = 0; - } - } - } - } - } - } - } - } - - /* Make sure that the detection is not the cut-off of a low pass filter. */ - removeLowPassDetection(pAddHarmSfb, - detectionVectors, - start, - stop, - nSfb, - pFreqBandTable, - pNrgVector, - mhThresh); - } - else { - /* - * If a missing harmonic wasn't missing the previous frame - * the transient-flag needs to be set in order to be allowed to detect it. - *************************************************************************/ - for(i=0;i<nSfb;i++){ - if(pAddHarmSfb[i] - pPrevAddHarmSfb[i] > 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<nSfb;i++){ - - thresTemp = (guideVectors.guideVectorDiff[i] != FL2FXCONST_DBL(0.0f)) - ? fMax(fMult(mhThresh.decayGuideDiff,guideVectors.guideVectorDiff[i]), mhThresh.thresHoldDiffGuide) - : mhThresh.thresHoldDiff; - - thresTemp = fMin(thresTemp, mhThresh.thresHoldDiff); - - if(pDiffVecScfb[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<nSfb;i++){ - ll = pFreqBandTable[i]; - lu = pFreqBandTable[i+1]; - - thresOrig = fixMax(fMult(guideVectors.guideVectorOrig[i], mhThresh.decayGuideOrig), mhThresh.thresHoldToneGuide); - thresOrig = fixMin(thresOrig, mhThresh.thresHoldTone); - - if(guideVectors.guideVectorOrig[i] != FL2FXCONST_DBL(0.0f)){ - for(j= ll;j<lu;j++){ - if(quotaBuffer[j] > 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<nSfb;i++){ - ll = pFreqBandTable[i]; - lu = pFreqBandTable[i+1]; - - if(pHarmVec[i] == 0){ - if(lu -ll > 1){ - for(j= ll;j<lu;j++){ - if(quotaBuffer[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<nSfb; i++){ - if(pAddHarmSfb[i]){ - *pAddHarmFlag = 1; - break; - } - } - - FDKmemcpy(pPrevAddHarmSfb, pAddHarmSfb, nSfb*sizeof(UCHAR)); - FDKmemcpy(guideVectors[0].guideVectorDetected,pAddHarmSfb,nSfb*sizeof(INT)); - - for(i=0; i<nSfb ; i++){ - - guideVectors[0].guideVectorDiff[i] = FL2FXCONST_DBL(0.0f); - guideVectors[0].guideVectorOrig[i] = FL2FXCONST_DBL(0.0f); - - if(pAddHarmSfb[i] == 1){ - /* If we had a detection use the guide-value in the next frame from the last estimate were the detection - was done.*/ - for(est=start; est < totNoEst; est++){ - if(guideVectors[est].guideVectorDiff[i] != FL2FXCONST_DBL(0.0f)){ - guideVectors[0].guideVectorDiff[i] = guideVectors[est].guideVectorDiff[i]; - } - if(guideVectors[est].guideVectorOrig[i] != FL2FXCONST_DBL(0.0f)){ - guideVectors[0].guideVectorOrig[i] = guideVectors[est].guideVectorOrig[i]; - } - } - } - } - -} - - -/**************************************************************************/ -/*! - \brief Calculates a compensation vector for the energy data. - - This function calculates a compensation vector for the energy data (i.e. - envelope data) that is calculated elsewhere. This is since, one sine on - the border of two scalefactor bands, will be replace by one sine in the - middle of either scalefactor band. However, since the sine that is replaced - will influence the energy estimate in both scalefactor bands (in the envelops - calculation function) a compensation value is required in order to avoid - noise substitution in the decoder next to the synthetic sine. - - \return none. - -*/ -/**************************************************************************/ -static void calculateCompVector(UCHAR *pAddHarmSfb, - FIXP_DBL **pTonalityMatrix, - INT ** pSignMatrix, - UCHAR *pEnvComp, - INT nSfb, - const UCHAR *freqBandTable, - INT totNoEst, - INT maxComp, - UCHAR *pPrevEnvComp, - INT newDetectionAllowed) -{ - - INT scfBand,est,l,ll,lu,maxPosF,maxPosT; - FIXP_DBL maxVal; - INT compValue; - FIXP_DBL tmp; - - FDKmemclear(pEnvComp,nSfb*sizeof(UCHAR)); - - for(scfBand=0; scfBand < nSfb; scfBand++){ - - if(pAddHarmSfb[scfBand]){ /* A missing sine was detected */ - ll = freqBandTable[scfBand]; - lu = freqBandTable[scfBand+1]; - - maxPosF = 0; /* First find the maximum*/ - maxPosT = 0; - maxVal = FL2FXCONST_DBL(0.0f); - - for(est=0;est<totNoEst;est++){ - for(l=ll; l<lu; l++){ - if(pTonalityMatrix[est][l] > 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;scfBand<nSfb;scfBand++){ - if(pEnvComp[scfBand] != 0 && pPrevEnvComp[scfBand] == 0) - pEnvComp[scfBand] = 0; - } - } - - /* remember the value for the next frame.*/ - FDKmemcpy(pPrevEnvComp,pEnvComp,nSfb*sizeof(UCHAR)); -} - - -/**************************************************************************/ -/*! - \brief Detects where strong tonal components will be missing after - HFR in the decoder. - - - \return none. - -*/ -/**************************************************************************/ -void -FDKsbrEnc_SbrMissingHarmonicsDetectorQmf(HANDLE_SBR_MISSING_HARMONICS_DETECTOR h_sbrMHDet, - 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 transientFlag = pTranInfo[1]; - INT transientPos = pTranInfo[0]; - INT newDetectionAllowed; - INT transientDetStart = 0; - - UCHAR ** detectionVectors = h_sbrMHDet->detectionVectors; - 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; i<MAX_NO_OF_ESTIMATES; i++) { - hs->guideVectors[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; i<totNoEst; i++) { - FDKmemclear (hs->guideVectors[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; i<totNoEst/2; i++) { - for(i=0; i<MAX_NO_OF_ESTIMATES/2; i++) { - FDKmemclear (hs->tonalityDiff[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;i<h_sbrNoiseFloorEstimate->noNoiseBands;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;band<nBands;band++) { - INT delta = (val[band] - lastVal) + tableOffset; - lastVal = val[band]; - if( (delta>maxVal) || (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;band<nBands;band++) { - INT delta = (val[band] - valLast[band]) + tableOffset; - if( (delta>maxVal) || (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; env<psOut->nEnvelopes; 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; env<psOut->nEnvelopes; env++) { - bitCnt += FDKsbrEnc_WriteBits_ps( hBitBuf, psOut->frameBorder[env], 5); - } - } - - if(psOut->enableIID==1) { - INT *iidLast = psOut->iidLast; - for(env=0; env<psOut->nEnvelopes; 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; env<psOut->nEnvelopes; 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<n; i++) - Z[i] = (X[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; i<PS_MAX_BANDS; i++) { - hPsData->iidIdxLast[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; env<PS_MAX_ENVELOPES; env++) { - hPsData->iccDiffMode[env] = PS_DELTA_FREQ; - hPsData->iccDiffMode[env] = PS_DELTA_FREQ; - - for (i=0; i<PS_MAX_BANDS; i++) { - hPsData->iidIdx[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<nBands;band++) { - for(idx=0; idx<nQuantSteps-1; idx++){ - if( fixp_abs((input[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<psBands; b++) { - dIid = (iid[2*e][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;env<nEnvelopes; env++) { - errIID += quantizeCoef(iid[env], psBands, iidQuant_fx, 7, 15, iidIdxCoarse[env]); - errIIDFine += quantizeCoef(iid[env], psBands, iidQuantFine_fx, 15, 31, iidIdxFine[env]); - } - - /* normalize error to number of envelopes, ps bands - errIID /= psBands*nEnvelopes; - errIIDFine /= psBands*nEnvelopes; */ - - - /* Check if IID coefficients should be used in this frame */ - psData->iidEnable = 0; - for(env=0;env<nEnvelopes; env++) { - for(band=0;band<psBands;band++) { - loudnDiff += fixp_abs(iidIdxCoarse[env][band]); - iidTransmit ++; - } - } - - if(loudnDiff > 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;env<nEnvelopes; env++) { - psData->iidDiffMode[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;env<nEnvelopes; env++) { - bitsIidFreq = FDKsbrEnc_EncodeIid(NULL, iidIdxCoarse[env], NULL, psBands, PS_IID_RES_COARSE, PS_DELTA_FREQ, &error); - bitsIidTime = FDKsbrEnc_EncodeIid(NULL, iidIdxCoarse[env], iidIdxCoarse[env-1], psBands, PS_IID_RES_COARSE, PS_DELTA_TIME, &error); - - /* decision DELTA_FREQ vs DELTA_TIME */ - if(bitsIidTime>bitsIidFreq) { - 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;env<nEnvelopes; env++) { - bitsIidFreq = FDKsbrEnc_EncodeIid(NULL, iidIdxFine[env], NULL, psBands, PS_IID_RES_FINE, PS_DELTA_FREQ, &error); - bitsIidTime = FDKsbrEnc_EncodeIid(NULL, iidIdxFine[env], iidIdxFine[env-1], psBands, PS_IID_RES_FINE, PS_DELTA_TIME, &error); - - /* decision DELTA_FREQ vs DELTA_TIME */ - if(bitsIidTime>bitsIidFreq) { - 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(bitsFineTot<bitsCoarseTot) { - psData->iidQuantMode = PS_IID_RES_FINE; - for(env=0;env<nEnvelopes; env++) { - psData->iidDiffMode[env] = diffModeFine[env]; - FDKmemcpy(psData->iidIdx[env], iidIdxFine[env], psBands*sizeof(INT)); - } - } - else { - psData->iidQuantMode = PS_IID_RES_COARSE; - for(env=0;env<nEnvelopes; env++) { - psData->iidDiffMode[env] = diffMode[env]; - FDKmemcpy(psData->iidIdx[env], iidIdxCoarse[env], psBands*sizeof(INT)); - } - } - - /* Count DELTA_TIME encoding streaks */ - for(env=0;env<nEnvelopes; env++) { - if(psData->iidDiffMode[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; env<nEnvelopes; env++) { - sumDiff = 0; - b = 0; - do { - diff = fixp_abs(psData->iidIdx[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 ((b<psBands) && (similar>0)); - } - } /* 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; env<nEnvelopes; env++) { - sumDiff = 0; - b = 0; - do { - diff = fixp_abs(psData->iccIdx[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 ((b<psBands) && (similar>0)); - } - } /* 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;env<nEnvelopes; env++) { - errICC += quantizeCoef(icc[env], psBands, iccQuant, 0, 8, psData->iccIdx[env]); - } - - /* Check if ICC coefficients should be used */ - psData->iccEnable = 0; - for(env=0;env<nEnvelopes; env++) { - for(band=0;band<psBands;band++) { - inCoherence += psData->iccIdx[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;env<nEnvelopes; env++) { - psData->iccDiffMode[env] = PS_DELTA_FREQ; - FDKmemclear(psData->iccIdx[env], sizeof(INT)*psBands); - } - return; - } - - for(env=0;env<nEnvelopes; env++) { - bitsIccFreq = FDKsbrEnc_EncodeIcc(NULL, psData->iccIdx[env], NULL, psBands, PS_DELTA_FREQ, &error); - - if(psData->iccTimeCnt<MAX_TIME_DIFF_FRAMES) { - bitsIccTime = FDKsbrEnc_EncodeIcc(NULL, psData->iccIdx[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<nEnvelopes;env++) { - for (i=0; i<psBands; i++) { - - /* iid[env][i] = 10.0f*(float)log10(pwrL[env][i]/pwrR[env][i]); - */ - FIXP_DBL IID = fMultDiv2( FL2FXCONST_DBL(LOG10_2_10/IID_SCALE_FT), (ldPwrL[env][i]-ldPwrR[env][i]) ); - - IID = fixMin( IID, (FIXP_DBL)(MAXVAL_DBL>>(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<nEnvelopes;env++) { - for (i=0; i<border; i++) { - - /* icc[env][i] = min( pwrCr[env][i] / (float) sqrt(pwrL[env][i] * pwrR[env][i]) , 1.f); - */ - FIXP_DBL ICC, invNrg = CalcInvLdData ( -((ldPwrL[env][i]>>1) + (ldPwrR[env][i]>>1) + (FIXP_DBL)1) ); - INT scale, invScale = CountLeadingBits(invNrg); - - scale = (DFRACT_BITS-1) - invScale; - ICC = fMult(pwrCr[env][i], invNrg<<invScale) ; - icc[env][i] = SATURATE_LEFT_SHIFT(ICC, scale, DFRACT_BITS); - } - - for (; i<psBands; i++) { - INT sc1, sc2; - FIXP_DBL cNrgR, cNrgI, ICC; - - sc1 = CountLeadingBits( fixMax(fixp_abs(pwrCr[env][i]),fixp_abs(pwrCi[env][i])) ) ; - cNrgR = fPow2Div2((pwrCr[env][i]<<sc1)); /* squared nrg's expect explicit scaling */ - cNrgI = fPow2Div2((pwrCi[env][i]<<sc1)); - - ICC = CalcInvLdData( (CalcLdData((cNrgR + cNrgI)>>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<<sc2,invNrg); - - sc1 = ( (DFRACT_BITS-1) - sc1 - sc2 ); - if (sc1 < 0) { - 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; env<nEnvelopes+1;env++) { - envBorder[env] = fMultI(GetInvInt(nEnvelopes),frameSize*env); - } - - for(env=0; env<nEnvelopes;env++) { - - /* clear energy array */ - for (band=0; band<psBands; band++) { - pwrData->pwrL[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]; col<envBorder[env+1]; col++) { - for (subband = hPsEncode->iidGroupBorders[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; e<nEnvelopes; e++) { - FDKsbrEnc_addFIXP_DBL(pwrData->pwrL[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->headerCnt<MAX_PS_NOHEADER_CNT) - && ( (hPsData->iidQuantMode == 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; env<nEnvelopes; env++) { - hPsOut->frameBorder[env] = envBorder[env+1]; - } - - for(env=0; env<hPsOut->nEnvelopes; env++) { - hPsOut->deltaIID[env] = (PS_DELTA)hPsData->iidDiffMode[env]; - - for(band=0; band<psBands; band++) { - hPsOut->iid[env][band] = hPsData->iidIdx[env][band]; - } - } - - for(env=0; env<hPsOut->nEnvelopes; env++) { - hPsOut->deltaICC[env] = (PS_DELTA)hPsData->iccDiffMode[env]; - for(band=0; band<psBands; band++) { - hPsOut->icc[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; env<PS_MAX_ENVELOPES; env++) { - hPsOut->deltaIPD[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; band<PS_MAX_BANDS; band++) { - hPsOut->iidLast[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; i<psBands; i++) { - hPsData->iidIdxLast[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<<IID_SCALE) - -#define PS_QUANT_SCALE_FT (64.f) /* error smaller (64-25)/64 * 20 bands * 4 env -> 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; i<MAX_PS_CHANNELS; i++) { - if (FDKhybridAnalysisOpen( - &hParametricStereo->fdkHybAnaFilter[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; ch<MAX_PS_CHANNELS; ch++) { - FDKhybridAnalysisInit( - &hParametricStereo->fdkHybAnaFilter[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; ch<MAX_PS_CHANNELS; ch ++) { - FIXP_DBL *pDynReal = GetRam_Sbr_envRBuffer (ch, dynamic_RAM); - FIXP_DBL *pDynImag = GetRam_Sbr_envIBuffer (ch, dynamic_RAM); - - for (i=0; i<HYBRID_FRAMESIZE; i++) { - hParametricStereo->pHybridData[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; i<HYBRID_READ_OFFSET; i++) { - hParametricStereo->pHybridData[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<noQmfSlots; n++) { - - FIXP_DBL tmpHybrid[2][MAX_HYBRID_BANDS]; - - for(k = 0; k<71; k++){ - int dynScale, sc; /* scaling */ - FIXP_QMF tmpLeftReal, tmpRightReal, tmpLeftImag, tmpRightImag; - FIXP_DBL tmpScaleFactor, stereoScaleFactor; - - tmpLeftReal = hybridData[n][0][0][k]; - tmpLeftImag = hybridData[n][0][1][k]; - tmpRightReal = hybridData[n][1][0][k]; - tmpRightImag = hybridData[n][1][1][k]; - - sc = fixMax(0,CntLeadingZeros( fixMax(fixMax(fixp_abs(tmpLeftReal),fixp_abs(tmpLeftImag)),fixMax(fixp_abs(tmpRightReal),fixp_abs(tmpRightImag))) )-2); - - tmpLeftReal <<= sc; tmpLeftImag <<= sc; - tmpRightReal <<= sc; tmpRightImag <<= sc; - dynScale = fixMin(sc-dynQmfScale,DFRACT_BITS-1); - - /* calc stereo scale factor to avoid loss of energy in bands */ - /* stereo scale factor = min(2.0f, sqrt( (abs(l(k, n)^2 + abs(r(k, n)^2 )))/(0.5f*abs(l(k, n) + r(k, n))) )) */ - stereoScaleFactor = fPow2Div2(tmpLeftReal) + fPow2Div2(tmpLeftImag) - + fPow2Div2(tmpRightReal) + fPow2Div2(tmpRightImag) ; - - /* might be that tmpScaleFactor becomes negative, so fabs(.) */ - tmpScaleFactor = fixp_abs(stereoScaleFactor + fMult(tmpLeftReal,tmpRightReal) + fMult(tmpLeftImag,tmpRightImag)); - - /* min(2.0f, sqrt(stereoScaleFactor/(0.5f*tmpScaleFactor))) */ - if ( (stereoScaleFactor>>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<<sc_denum, - 16) ; - - /* prevent odd scaling for next sqrt calculation */ - if (sc&0x1) { - sc++; - 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; i<noQmfSlots2; i++) { - FDKmemcpy(tmp[0], hParametricStereo->qmfDelayLines[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<noQmfSlots2; i++) { - for (j=0; j<noQmfBands; j++) { - mixRealQmfData[i+slotOffset][j] >>= 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; psCh<MAX_PS_CHANNELS; psCh ++) { - - for (i = 0; i < hQmfAnalysis[psCh]->no_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; i<HYBRID_READ_OFFSET; i++) { - FDKmemcpy(hParametricStereo->pHybridData[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; col<frameSize-HYBRID_READ_OFFSET; col++) { - for (i = hPsEncode->iidGroupBorders[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; col<frameSize; col++) { - for (i = hPsEncode->iidGroupBorders[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<psBands; band++) { -#ifndef MULT_16x16 - dynBandScale[band] = CountLeadingBits(fixMax(maxVal[0][band],maxBandValue[band])); -#else - dynBandScale[band] = fixMax(0,CountLeadingBits(fixMax(maxVal[0][band],maxBandValue[band]))-FRACT_BITS); -#endif - maxValue = fixMax(maxValue,fixMax(maxVal[0][band],maxVal[1][band])); - maxBandValue[band] = fixMax(maxVal[0][band], maxVal[1][band]); - } - - /* calculate maximal scaling for QMF downmix */ -#ifndef MULT_16x16 - *dmxScale = fixMin(DFRACT_BITS, CountLeadingBits(maxValue)); -#else - *dmxScale = fixMax(0,fixMin(FRACT_BITS, CountLeadingBits(FX_QMF2FX_DBL(maxValue)))); -#endif - -} - diff --git a/libSBRenc/src/ps_main.h b/libSBRenc/src/ps_main.h deleted file mode 100644 index 21b32ff..0000000 --- a/libSBRenc/src/ps_main.h +++ /dev/null @@ -1,271 +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: Markus Multrus - Contents/Description: PS Wrapper, Downmix header file - -******************************************************************************/ - -#ifndef __INCLUDED_PS_MAIN_H -#define __INCLUDED_PS_MAIN_H - -/* Includes ******************************************************************/ -#include "sbr_def.h" -#include "qmf.h" -#include "ps_encode.h" -#include "FDK_bitstream.h" -#include "FDK_hybrid.h" - - -/* Data Types ****************************************************************/ -typedef enum { - PSENC_STEREO_BANDS_INVALID = 0, - PSENC_STEREO_BANDS_10 = 10, - PSENC_STEREO_BANDS_20 = 20 - -} PSENC_STEREO_BANDS_CONFIG; - -typedef enum { - PSENC_NENV_1 = 1, - PSENC_NENV_2 = 2, - PSENC_NENV_4 = 4, - PSENC_NENV_DEFAULT = PSENC_NENV_2, - PSENC_NENV_MAX = PSENC_NENV_4 - -} PSENC_NENV_CONFIG; - -typedef struct { - UINT bitrateFrom; /* inclusive */ - UINT bitrateTo; /* exclusive */ - PSENC_STEREO_BANDS_CONFIG nStereoBands; - PSENC_NENV_CONFIG nEnvelopes; - LONG iidQuantErrorThreshold; /* quantization threshold to switch between coarse and fine iid quantization */ - -} psTuningTable_t; - -/* Function / Class Declarations *********************************************/ - -typedef struct T_PARAMETRIC_STEREO { - HANDLE_PS_ENCODE hPsEncode; - PS_OUT psOut[2]; - - FIXP_DBL __staticHybridData[HYBRID_READ_OFFSET][MAX_PS_CHANNELS][2][MAX_HYBRID_BANDS]; - FIXP_DBL *pHybridData[HYBRID_READ_OFFSET+HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2]; - - FIXP_QMF qmfDelayLines[2][QMF_MAX_TIME_SLOTS>>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;i<sizeof(filter_paramSet)/sizeof(struct FILTER_PARAM *);i++){ - if (filter_paramSet[i]->Wc <= 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; n<downRatio; n++) - { - FIXP_BQS (*states)[2] = downFilter->states; - 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; i<downFilter->noCoeffs; 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; i<downFilter->noCoeffs; 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; i<numInSamples; i+=DownSampler->ratio) - { - *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; i<MAX_NUM_NOISE_VALUES; i++) - fData->res[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; i<nElements; i++) { - hSbrEncoder->sbrElement[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; i<nChannels; i++) { - hSbrEncoder->pSbrChannel[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; i<fixMax(nChannels,(supportPS)?2:0); i++) { - hSbrEncoder->QmfAnalysis[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; coreEl<noElements; coreEl++) - { - /* SBR only handles SCE and CPE's */ - if (elInfo[coreEl].elType == ID_SCE || elInfo[coreEl].elType == ID_CPE) { - el++; - } else { - if (elInfo[coreEl].elType == ID_LFE) { - hSbrEncoder->lfeChIdx = 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; el<hEnvEnc->noElements; 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<noElements; coreEl++) - { - /* SBR only handles SCE and CPE's */ - if (elInfo[coreEl].elType != ID_SCE && elInfo[coreEl].elType != ID_CPE) { - continue; - } - /* check if desired configuration is available */ - if ( !FDKsbrEnc_IsSbrSettingAvail (elInfo[coreEl].bitRate, 0, elInfo[coreEl].nChannelsInEl, inputSampleRate, *coreSampleRate, aot) ) - { - error = 1; - goto bail; - } - } - - /* Determine Delay balancing and new encoder delay */ - if (lowDelay) { - { - delayDiff = (*delay * *downSampleFactor) + DELAY_ELD2SBR(coreFrameLength,*downSampleFactor); - *delay = DELAY_ELDSBR(coreFrameLength,*downSampleFactor); - } - } - else if (usePs) { - delayDiff = (*delay * *downSampleFactor) + DELAY_AAC2PS(coreFrameLength,*downSampleFactor); - *delay = DELAY_PS(coreFrameLength,*downSampleFactor); - } - else { - delayDiff = DELAY_AAC2SBR(coreFrameLength,*downSampleFactor); - delayDiff += (*delay * *downSampleFactor); - *delay = DELAY_SBR(coreFrameLength,*downSampleFactor); - } - - if (!usePs) { - timeDomainDownsample = *downSampleFactor-1; /* activate time domain downsampler when downSampleFactor is != 1 */ - } - - - /* Take care about downsampled data bound to the SBR path */ - if (!timeDomainDownsample && delayDiff > 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; coreEl<noElements; coreEl++) - { - /* SBR only handles SCE and CPE's */ - if (elInfo[coreEl].elType == ID_SCE || elInfo[coreEl].elType == ID_CPE) { - el++; - } else { - continue; - } - - /* Set parametric Stereo Flag. */ - if (usePs) { - elInfo[coreEl].fParametricStereo = 1; - } else { - elInfo[coreEl].fParametricStereo = 0; - } - - /* - * Init sbrConfig structure - */ - if ( ! FDKsbrEnc_InitializeSbrDefaults ( &sbrConfig[el], - *downSampleFactor, - coreFrameLength, - IS_LOWDELAY(aot) - ) ) - { - error = 1; - goto bail; - } - - /* - * Modify sbrConfig structure according to Element parameters - */ - if ( ! FDKsbrEnc_AdjustSbrSettings (&sbrConfig[el], - elInfo[coreEl].bitRate, - elInfo[coreEl].nChannelsInEl, - *coreSampleRate, - inputSampleRate, - transformFactor, - 24000, - 0, - 0, /* useSpeechConfig */ - 0, /* lcsMode */ - usePs, /* bParametricStereo */ - aot) ) - { - error = 1; - goto bail; - } - - /* Find common frequency border for all SBR elements */ - highestSbrStartFreq = fixMax(highestSbrStartFreq, sbrConfig[el].startFreq); - highestSbrStopFreq = fixMax(highestSbrStopFreq, sbrConfig[el].stopFreq); - - } /* first element loop */ - - /* Set element count (can be less than core encoder element count) */ - hSbrEncoder->noElements = el+1; - - FDKsbrEnc_Reallocate(hSbrEncoder, - elInfo, - noElements); - - for (el=0; el<hSbrEncoder->noElements; 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; el<hSbrEncoder->noElements; 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; ch<hSbrEl->elInfo.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; el<hSbrEncoder->noElements; 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; el<hSbrEncoder->noElements; 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. <br> - 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) <br> - Dimension #MAX_FREQ_COEFFS/2+1 -*/ -C_ALLOC_MEM2 (Ram_Sbr_freqBandTableLO, UCHAR, (MAX_FREQ_COEFFS/2+1), (8)) - -/*! Frequency band table (high res) <br> - Dimension #MAX_FREQ_COEFFS +1 -*/ -C_ALLOC_MEM2 (Ram_Sbr_freqBandTableHI, UCHAR, (MAX_FREQ_COEFFS+1), (8)) - -/*! vk matser table <br> - Dimension #MAX_FREQ_COEFFS +1 -*/ -C_ALLOC_MEM2 (Ram_Sbr_v_k_master, UCHAR, (MAX_FREQ_COEFFS+1), (8)) - - -/* - Missing harmonics detection -*/ - -/*! sbr_detectionVectors <br> - 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[ <br> - Dimension #MAX_NUM_CHANNELS*#MAX_FREQ_COEFFS] -*/ -C_ALLOC_MEM2 (Ram_Sbr_prevEnvelopeCompensation, UCHAR, MAX_FREQ_COEFFS, (8)) -/*! sbr_guideScfb[ <br> - Dimension #MAX_NUM_CHANNELS*#MAX_FREQ_COEFFS] -*/ -C_ALLOC_MEM2 (Ram_Sbr_guideScfb, UCHAR, MAX_FREQ_COEFFS, (8)) - -/*! sbr_guideVectorDetected <br> - 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 <br> - 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<num_bands0;i++) - diff_tot[i] = dk; - - /* If linear scale wasn't achived */ - /* and we got wide SBR are */ - if (k2_diff < 0) { - incr = 1; - i = 0; - } - - /* If linear scale wasn't achived */ - /* and we got small SBR are */ - if (k2_diff > 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; v<NUM_V_COMBINE; v++) - { - ptr[0] = sourceBufferReal[i][r+v]; - ptr[0+BAND_V_SIZE*NUM_V_COMBINE] = sourceBufferImag[i][r+v]; - ptr -= BAND_V_SIZE; - } - } - } - - blockLength = pBlockLength[0]; - - while(k <= buffLen - blockLength) - { - autoCorrScaling = fixMin(getScalefactor(&realBuf[k-LPC_ORDER], LPC_ORDER+blockLength), getScalefactor(&imagBuf[k-LPC_ORDER], LPC_ORDER+blockLength)); - autoCorrScaling = fixMax(0, autoCorrScaling-1); - - scaleValues(&realBuf[k-LPC_ORDER], LPC_ORDER+blockLength, autoCorrScaling); - scaleValues(&imagBuf[k-LPC_ORDER], LPC_ORDER+blockLength, autoCorrScaling); - - autoCorrScaling <<= 1; /* consider qmf buffer scaling twice */ - autoCorrScaling += autoCorr2nd_cplx ( ac, realBuf+k, imagBuf+k, blockLength ); - - - if(ac->det == 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; i<MAX_NO_OF_ESTIMATES; i++) { - hTonCorr->quotaMatrix[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; i<hTonCorr->numberOfEstimates; 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<YBufferWriteOffset; i++) { - energies_e_diff[i] = prevEnergies_e_diff; - } - for (i=YBufferWriteOffset; i<stop; i++) { - energies_e_diff[i] = newEnergies_e_diff; - } - - /* Sum up energies of all QMF-timeslots for both halfs */ - FDK_ASSERT(len1<=8); /* otherwise an overflow is possible */ - FDK_ASSERT(len2<=8); /* otherwise an overflow is possible */ - /* init with some energy to prevent division by zero - and to prevent splitting for very low levels */ - accu1_init = scaleValue((FL2FXCONST_DBL((1.0e6*NORM_QMF_ENERGY))),-energies_e); - accu2_init = scaleValue((FL2FXCONST_DBL((1.0e6*NORM_QMF_ENERGY))),-energies_e); - accu1_init = fMult(accu1_init, (FIXP_DBL)len1<<((DFRACT_BITS-1)-NRG_SHIFT-1))<<1; - accu2_init = fMult(accu2_init, (FIXP_DBL)len2<<((DFRACT_BITS-1)-NRG_SHIFT-1))<<1; - - for (j=0; j<nSfb; j++) { - - accu1 = accu1_init; - accu2 = accu2_init; - accu_e = energies_e+3; - - /* Sum up energies in first half */ - for (i=start; i<border; i++) { - accu1 = fAddSaturate(accu1, scaleValue(Energies[i][j], -energies_e_diff[i])); - } - - /* Sum up energies in second half */ - for (i=border; i<stop; i++) { - accu2 = fAddSaturate(accu2, scaleValue(Energies[i][j], -energies_e_diff[i])); - } - - /* Energy change in current band */ - #define LN2 FL2FXCONST_DBL(0.6931471806f) /* ln(2) */ - tmp0 = fLog2(accu2, accu_e) - fLog2(accu1, accu_e); - tmp1 = fLog2((FIXP_DBL)len1, 31) - fLog2((FIXP_DBL)len2, 31); - delta = fMult(LN2, (tmp0 + tmp1)); - delta = (FIXP_DBL)FDKabs( delta ); - - /* Weighting with amplitude ratio of this band */ - accu_e++; - accu1>>=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<YBufferWriteOffset; ts++) { - for (k = 0; k < freqBandTable[0]; k++) { - accu1 += Energies[ts][k] >> 6; - } - } - for (; ts<tran_offdiv2+(slots>>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 and by the factor (2^19-scaleEnergies[1]) 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<sbrSlots; slotOut++) { - slotIn = timeStep*slotOut; - - for (j=0; j<nSfb; j++) { - accu = FL2FXCONST_DBL(0.0f); - - li = freqBandTable[j]; - ui = freqBandTable[j + 1]; - - for (k=li; k<ui; k++) { - for (i=0; i<timeStep; i++) { - accu += (Energies[(slotIn+i)>>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<YBufferWriteOffset; slotOut++) { - for (j=0; j<nSfb; j++) { - accu += (EnergiesM[slotOut][j] >> scale[0]); - } - } - nrgTotal = accu >> (scaleEnergies[0]-scale[0]); - - for (slotOut=YBufferWriteOffset; slotOut<sbrSlots; slotOut++) { - for (j=0; j<nSfb; j++) { - accu += (EnergiesM[slotOut][j] >> 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<noRows; i++) - { - int startEnergy = (tran_off>>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<YBufferWriteOffset; j++) - accu0 += fMult(Energies[j][i], i_noCols); - for (; j<endEnergy; j++) - accu1 += fMult(Energies[j][i], i_noCols); - - mean_val = (accu0 >> 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<YBufferWriteOffset; j++) { - temp = ((FIXP_DBL)mean_val - ((FIXP_DBL)Energies[j][i] >> scaleFactor0))<<shift; - temp = fPow2(temp); - temp = fMult(temp, i_noCols1); - accu += temp; - } - for (; j<endEnergy; j++) { - temp = ((FIXP_DBL)mean_val - ((FIXP_DBL)Energies[j][i] >> scaleFactor1))<<shift; - temp = fPow2(temp); - temp = fMult(temp, i_noCols1); - accu += temp; - } - - std_val = sqrtFixp(accu)>>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))-1)>>YBufferSzShift; /* YBufferSzShift shifts because of half energy values. */ - - /* Compute differential values with two different weightings in every subband */ - for (i=start_band; i<stop_band; i++) - { - FIXP_DBL thres = thresholds[i]; - - if((LONG)thresholds[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<YBufferWriteOffset; j++) { - FIXP_DBL tmp = Energies[j][i]; - EnergiesTemp[(j<<1)+1] = EnergiesTemp[j<<1] = tmp>>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<YBufferWriteOffset; j++) { - FIXP_DBL tmp = Energies[j][i]; - EnergiesTemp[j] = tmp>>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; i<qmfStartSample + no_cols; 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) { - 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 <frameShift> qmf-slots of second frame */ - for (i=qmfStartSample+no_cols; i<qmfStartSample + no_cols+h_sbrTran->frameShift; 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; i<QMF_CHANNELS; i++){ - /* Calculate dBf over all qmf bands: - dBf = (10^(0.002266f/10*bw(slot)))^(band) = - = 2^(log2(10)*0.002266f/10*bw(slot)*band) = - = 2^(0.00075275f*bw(slot)*band) */ - - FIXP_DBL dBf_m; /* dBf mantissa */ - INT dBf_e; /* dBf exponent */ - INT tmp; - - INT dBf_int; /* dBf integer part */ - FIXP_DBL dBf_fract; /* dBf fractional part */ - - /* myExp*(i+1) = myExp_int - myExp_fract - myExp*(i+1) is split up here for better accuracy of CalcInvLdData(), - for its result can be split up into an integer and a fractional part */ - - /* Round up to next integer */ - FIXP_DBL myExp_int = (myExpSlot & (FIXP_DBL)0xfe000000) + (FIXP_DBL)0x02000000; - - /* This is the fractional part that needs to be substracted */ - FIXP_DBL myExp_fract = myExp_int - myExpSlot; - - /* Calc integer part */ - dBf_int = CalcInvLdData(myExp_int); - /* The result needs to be re-scaled. The ld(myExp_int) had been scaled by EXP_E, - the CalcInvLdData expects the operand to be scaled by LD_DATA_SHIFT. - Therefore, the correctly scaled result is dBf_int^(2^(EXP_E-LD_DATA_SHIFT)), - which is dBf_int^2 */ - dBf_int *= dBf_int; - - /* Calc fractional part */ - dBf_fract = CalcInvLdData(-myExp_fract); - /* The result needs to be re-scaled. The ld(myExp_fract) had been scaled by EXP_E, - the CalcInvLdData expects the operand to be scaled by LD_DATA_SHIFT. - Therefore, the correctly scaled result is dBf_fract^(2^(EXP_E-LD_DATA_SHIFT)), - which is dBf_fract^2 */ - dBf_fract = fMultNorm(dBf_fract, dBf_fract, &tmp); - - /* Get worst case scaling of multiplication result */ - dBf_e = (DFRACT_BITS-1 - tmp) - CountLeadingBits(dBf_int); - - /* Now multiply integer with fractional part of the result, thus resulting - in the overall accurate fractional result */ - dBf_m = fMultNorm(dBf_int, dBf_fract, &e); - dBf_m = scaleValueSaturate(dBf_m, e+DFRACT_BITS-1+tmp-dBf_e); - myExpSlot += myExp; - - /* Keep the results */ - h_sbrFastTransientDetector->dBf_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]<<headroomEnSlot, h_sbrFastTransientDetector->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 |