diff options
| author | Wim Taymans <wtaymans@redhat.com> | 2017-09-08 15:00:19 +0200 |
|---|---|---|
| committer | Wim Taymans <wtaymans@redhat.com> | 2017-09-08 15:00:19 +0200 |
| commit | 4b07abf077727764ba16fe0fe28ce5561e66b809 (patch) | |
| tree | 9f4b80f0ecf0040b12e9f210b642e1edb98aed51 /libSBRdec | |
| parent | a3d11689433a046ad57add8ea22dedceb2fe722d (diff) | |
Strip out minimal decoderstripped
Diffstat (limited to 'libSBRdec')
36 files changed, 0 insertions, 18187 deletions
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 965917a..0000000 --- a/libSBRdec/src/psdec.cpp +++ /dev/null @@ -1,1414 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief parametric stereo decoder -*/ - -#include "psdec.h" - - - -#include "FDK_bitbuffer.h" -#include "psdec_hybrid.h" - -#include "sbr_rom.h" -#include "sbr_ram.h" - -#include "FDK_tools_rom.h" - -#include "genericStds.h" - -#include "FDK_trigFcts.h" - - -/********************************************************************/ -/* MLQUAL DEFINES */ -/********************************************************************/ - - #define FRACT_ZERO FRACT_BITS-1 -/********************************************************************/ - -SBR_ERROR ResetPsDec( HANDLE_PS_DEC h_ps_d ); - -void ResetPsDeCor( HANDLE_PS_DEC h_ps_d ); - - -/***** HELPERS *****/ - -static void assignTimeSlotsPS (FIXP_DBL *bufAdr, FIXP_DBL **bufPtr, const int numSlots, const int numChan); - - - -/*******************/ - -#define DIV3 FL2FXCONST_DBL(1.f/3.f) /* division 3.0 */ -#define DIV1_5 FL2FXCONST_DBL(2.f/3.f) /* division 1.5 */ - -/***************************************************************************/ -/*! - \brief Creates one instance of the PS_DEC struct - - \return Error info - -****************************************************************************/ -int -CreatePsDec( HANDLE_PS_DEC *h_PS_DEC, /*!< pointer to the module state */ - int aacSamplesPerFrame - ) -{ - SBR_ERROR errorInfo = SBRDEC_OK; - HANDLE_PS_DEC h_ps_d; - int i; - - if (*h_PS_DEC == NULL) { - /* Get ps dec ram */ - h_ps_d = GetRam_ps_dec(); - if (h_ps_d == NULL) { - errorInfo = SBRDEC_MEM_ALLOC_FAILED; - goto bail; - } - } else { - /* Reset an open instance */ - h_ps_d = *h_PS_DEC; - } - - /* initialisation */ - switch (aacSamplesPerFrame) { - case 960: - h_ps_d->noSubSamples = 30; /* col */ - break; - case 1024: - h_ps_d->noSubSamples = 32; /* col */ - break; - default: - h_ps_d->noSubSamples = -1; - break; - } - - if (h_ps_d->noSubSamples > MAX_NUM_COL - || h_ps_d->noSubSamples <= 0) - { - goto bail; - } - h_ps_d->noChannels = NO_QMF_CHANNELS; /* row */ - - h_ps_d->psDecodedPrv = 0; - h_ps_d->procFrameBased = -1; - for (i = 0; i < (1)+1; i++) { - h_ps_d->bPsDataAvail[i] = ppt_none; - } - - - for (i = 0; i < (1)+1; i++) { - FDKmemclear(&h_ps_d->bsData[i].mpeg, sizeof(MPEG_PS_BS_DATA)); - } - - errorInfo = ResetPsDec( h_ps_d ); - - if ( errorInfo != SBRDEC_OK ) - goto bail; - - ResetPsDeCor( h_ps_d ); - - *h_PS_DEC = h_ps_d; - - - - return 0; - -bail: - DeletePsDec(&h_ps_d); - - return -1; -} /*END CreatePsDec */ - -/***************************************************************************/ -/*! - \brief Delete one instance of the PS_DEC struct - - \return Error info - -****************************************************************************/ -int -DeletePsDec( HANDLE_PS_DEC *h_PS_DEC) /*!< pointer to the module state */ -{ - if (*h_PS_DEC == NULL) { - return -1; - } - - - FreeRam_ps_dec(h_PS_DEC); - - - return 0; -} /*END DeletePsDec */ - -/***************************************************************************/ -/*! - \brief resets some values of the PS handle to default states - - \return - -****************************************************************************/ -SBR_ERROR ResetPsDec( HANDLE_PS_DEC h_ps_d ) /*!< pointer to the module state */ -{ - SBR_ERROR errorInfo = SBRDEC_OK; - INT i; - - const UCHAR noQmfBandsInHybrid20 = 3; - /* const UCHAR noQmfBandsInHybrid34 = 5; */ - - const UCHAR aHybridResolution20[] = { HYBRID_8_CPLX, - HYBRID_2_REAL, - HYBRID_2_REAL }; - - h_ps_d->specificTo.mpeg.delayBufIndex = 0; - - /* explicitly init state variables to safe values (until first ps header arrives) */ - - h_ps_d->specificTo.mpeg.lastUsb = 0; - - h_ps_d->specificTo.mpeg.scaleFactorPsDelayBuffer = -(DFRACT_BITS-1); - - FDKmemclear(h_ps_d->specificTo.mpeg.aDelayBufIndexDelayQmf, (NO_QMF_CHANNELS-FIRST_DELAY_SB)*sizeof(UCHAR)); - h_ps_d->specificTo.mpeg.noSampleDelay = delayIndexQmf[0]; - - for (i=0 ; i < NO_SERIAL_ALLPASS_LINKS; i++) { - h_ps_d->specificTo.mpeg.aDelayRBufIndexSer[i] = 0; - } - - h_ps_d->specificTo.mpeg.pAaRealDelayBufferQmf[0] = h_ps_d->specificTo.mpeg.aaQmfDelayBufReal; - - assignTimeSlotsPS ( h_ps_d->specificTo.mpeg.pAaRealDelayBufferQmf[0] + (NO_QMF_CHANNELS-FIRST_DELAY_SB), - &h_ps_d->specificTo.mpeg.pAaRealDelayBufferQmf[1], - h_ps_d->specificTo.mpeg.noSampleDelay-1, - (NO_DELAY_BUFFER_BANDS-FIRST_DELAY_SB)); - - h_ps_d->specificTo.mpeg.pAaImagDelayBufferQmf[0] = h_ps_d->specificTo.mpeg.aaQmfDelayBufImag; - - assignTimeSlotsPS ( h_ps_d->specificTo.mpeg.pAaImagDelayBufferQmf[0] + (NO_QMF_CHANNELS-FIRST_DELAY_SB), - &h_ps_d->specificTo.mpeg.pAaImagDelayBufferQmf[1], - h_ps_d->specificTo.mpeg.noSampleDelay-1, - (NO_DELAY_BUFFER_BANDS-FIRST_DELAY_SB)); - - /* Hybrid Filter Bank 1 creation. */ - errorInfo = InitHybridFilterBank ( &h_ps_d->specificTo.mpeg.hybrid, - h_ps_d->noSubSamples, - noQmfBandsInHybrid20, - aHybridResolution20 ); - - for ( i = 0; i < NO_IID_GROUPS; i++ ) - { - h_ps_d->specificTo.mpeg.h11rPrev[i] = FL2FXCONST_DBL(0.5f); - h_ps_d->specificTo.mpeg.h12rPrev[i] = FL2FXCONST_DBL(0.5f); - } - - FDKmemclear( h_ps_d->specificTo.mpeg.h21rPrev, sizeof( h_ps_d->specificTo.mpeg.h21rPrev ) ); - FDKmemclear( h_ps_d->specificTo.mpeg.h22rPrev, sizeof( h_ps_d->specificTo.mpeg.h22rPrev ) ); - - return errorInfo; -} - -/***************************************************************************/ -/*! - \brief clear some buffers used in decorrelation process - - \return - -****************************************************************************/ -void ResetPsDeCor( HANDLE_PS_DEC h_ps_d ) /*!< pointer to the module state */ -{ - INT i; - - FDKmemclear(h_ps_d->specificTo.mpeg.aPeakDecayFastBin, NO_MID_RES_BINS*sizeof(FIXP_DBL)); - FDKmemclear(h_ps_d->specificTo.mpeg.aPrevNrgBin, NO_MID_RES_BINS*sizeof(FIXP_DBL)); - FDKmemclear(h_ps_d->specificTo.mpeg.aPrevPeakDiffBin, NO_MID_RES_BINS*sizeof(FIXP_DBL)); - FDKmemclear(h_ps_d->specificTo.mpeg.aPowerPrevScal, NO_MID_RES_BINS*sizeof(SCHAR)); - - for (i=0 ; i < FIRST_DELAY_SB ; i++) { - FDKmemclear(h_ps_d->specificTo.mpeg.aaaRealDelayRBufferSerQmf[i], NO_DELAY_LENGTH_VECTORS*sizeof(FIXP_DBL)); - FDKmemclear(h_ps_d->specificTo.mpeg.aaaImagDelayRBufferSerQmf[i], NO_DELAY_LENGTH_VECTORS*sizeof(FIXP_DBL)); - } - for (i=0 ; i < NO_SUB_QMF_CHANNELS ; i++) { - FDKmemclear(h_ps_d->specificTo.mpeg.aaaRealDelayRBufferSerSubQmf[i], NO_DELAY_LENGTH_VECTORS*sizeof(FIXP_DBL)); - FDKmemclear(h_ps_d->specificTo.mpeg.aaaImagDelayRBufferSerSubQmf[i], NO_DELAY_LENGTH_VECTORS*sizeof(FIXP_DBL)); - } - -} - -/*******************************************************************************/ - -/* slot based funcion prototypes */ - -static void deCorrelateSlotBased( HANDLE_PS_DEC h_ps_d, - - FIXP_DBL *mHybridRealLeft, - FIXP_DBL *mHybridImagLeft, - SCHAR sf_mHybridLeft, - - FIXP_DBL *rIntBufferLeft, - FIXP_DBL *iIntBufferLeft, - SCHAR sf_IntBuffer, - - FIXP_DBL *mHybridRealRight, - FIXP_DBL *mHybridImagRight, - - FIXP_DBL *rIntBufferRight, - FIXP_DBL *iIntBufferRight ); - -static void applySlotBasedRotation( HANDLE_PS_DEC h_ps_d, - - FIXP_DBL *mHybridRealLeft, - FIXP_DBL *mHybridImagLeft, - - FIXP_DBL *QmfLeftReal, - FIXP_DBL *QmfLeftImag, - - FIXP_DBL *mHybridRealRight, - FIXP_DBL *mHybridImagRight, - - FIXP_DBL *QmfRightReal, - FIXP_DBL *QmfRightImag - ); - - -/***************************************************************************/ -/*! - \brief Get scale factor for all ps delay buffer. - - \return - -****************************************************************************/ -static -int getScaleFactorPsStatesBuffer(HANDLE_PS_DEC h_ps_d) -{ - INT i; - int scale = DFRACT_BITS-1; - - for (i=0; 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; - -/* const UCHAR *pQuantizedIIDs;*/ - - FIXP_SGL invL; - FIXP_DBL ScaleL, ScaleR; - FIXP_DBL Alpha, Beta; - FIXP_DBL h11r, h12r, h21r, h22r; - - const FIXP_DBL *PScaleFactors; - - /* Overwrite old values in delay buffers when upper subband is higher than in last frame */ - if (env == 0) { - - if ((usb > h_ps_d->specificTo.mpeg.lastUsb) && h_ps_d->specificTo.mpeg.lastUsb) { - - INT i,k,length; - - for (i=h_ps_d->specificTo.mpeg.lastUsb ; i < FIRST_DELAY_SB; i++) { - FDKmemclear(h_ps_d->specificTo.mpeg.aaaRealDelayRBufferSerQmf[i], NO_DELAY_LENGTH_VECTORS*sizeof(FIXP_DBL)); - FDKmemclear(h_ps_d->specificTo.mpeg.aaaImagDelayRBufferSerQmf[i], NO_DELAY_LENGTH_VECTORS*sizeof(FIXP_DBL)); - } - - for (k=0 ; 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; - /*pQuantizedIIDs = quantizedIIDsFine;*/ - } - - else - { - PScaleFactors = ScaleFactors; /* values are shiftet right by one */ - noIidSteps = NO_IID_STEPS; - /*pQuantizedIIDs = quantizedIIDs;*/ - } - - - /* dequantize and decode */ - for ( group = 0; group < NO_IID_GROUPS; group++ ) { - - bin = bins2groupMap20[group]; - - /*! - <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 */ - - ScaleR = PScaleFactors[noIidSteps + h_ps_d->specificTo.mpeg.coef.aaIidIndexMapped[env][bin]]; - ScaleL = PScaleFactors[noIidSteps - h_ps_d->specificTo.mpeg.coef.aaIidIndexMapped[env][bin]]; - - Beta = fMult (fMult( Alphas[h_ps_d->specificTo.mpeg.coef.aaIccIndexMapped[env][bin]], ( ScaleR - ScaleL )), FIXP_SQRT05); - Alpha = Alphas[h_ps_d->specificTo.mpeg.coef.aaIccIndexMapped[env][bin]]>>1; - - /* Alpha and Beta are now both scaled by 2 shifts right */ - - /* calculate the coefficients h11... h22 from scale-factors and ICC parameters */ - - /* h values are scaled by 1 shift right */ - { - FIXP_DBL trigData[4]; - - inline_fixp_cos_sin(Beta + Alpha, Beta - Alpha, 2, trigData); - h11r = fMult( ScaleL, trigData[0]); - h12r = fMult( ScaleR, trigData[2]); - h21r = fMult( ScaleL, trigData[1]); - h22r = fMult( ScaleR, trigData[3]); - } - /*****************************************************************************************/ - /* Interpolation of the matrices H11... H22: */ - /* */ - /* H11(k,n) = H11(k,n[e]) + (n-n[e]) * (H11(k,n[e+1] - H11(k,n[e])) / (n[e+1] - n[e]) */ - /* ... */ - /*****************************************************************************************/ - - /* invL = 1/(length of envelope) */ - invL = FX_DBL2FX_SGL(GetInvInt(h_ps_d->bsData[h_ps_d->processSlot].mpeg.aEnvStartStop[env + 1] - h_ps_d->bsData[h_ps_d->processSlot].mpeg.aEnvStartStop[env])); - - h_ps_d->specificTo.mpeg.coef.H11r[group] = h_ps_d->specificTo.mpeg.h11rPrev[group]; - h_ps_d->specificTo.mpeg.coef.H12r[group] = h_ps_d->specificTo.mpeg.h12rPrev[group]; - h_ps_d->specificTo.mpeg.coef.H21r[group] = h_ps_d->specificTo.mpeg.h21rPrev[group]; - h_ps_d->specificTo.mpeg.coef.H22r[group] = h_ps_d->specificTo.mpeg.h22rPrev[group]; - - h_ps_d->specificTo.mpeg.coef.DeltaH11r[group] = fMult ( h11r - h_ps_d->specificTo.mpeg.coef.H11r[group], invL ); - h_ps_d->specificTo.mpeg.coef.DeltaH12r[group] = fMult ( h12r - h_ps_d->specificTo.mpeg.coef.H12r[group], invL ); - h_ps_d->specificTo.mpeg.coef.DeltaH21r[group] = fMult ( h21r - h_ps_d->specificTo.mpeg.coef.H21r[group], invL ); - h_ps_d->specificTo.mpeg.coef.DeltaH22r[group] = fMult ( h22r - h_ps_d->specificTo.mpeg.coef.H22r[group], invL ); - - /* update prev coefficients for interpolation in next envelope */ - - h_ps_d->specificTo.mpeg.h11rPrev[group] = h11r; - h_ps_d->specificTo.mpeg.h12rPrev[group] = h12r; - h_ps_d->specificTo.mpeg.h21rPrev[group] = h21r; - h_ps_d->specificTo.mpeg.h22rPrev[group] = h22r; - - } /* group loop */ -} - - -static void applySlotBasedRotation( HANDLE_PS_DEC h_ps_d, /*!< pointer to the module state */ - - FIXP_DBL *mHybridRealLeft, /*!< hybrid values real left */ - FIXP_DBL *mHybridImagLeft, /*!< hybrid values imag left */ - - FIXP_DBL *QmfLeftReal, /*!< real bands left qmf channel */ - FIXP_DBL *QmfLeftImag, /*!< imag bands left qmf channel */ - - FIXP_DBL *mHybridRealRight, /*!< hybrid values real right */ - FIXP_DBL *mHybridImagRight, /*!< hybrid values imag right */ - - FIXP_DBL *QmfRightReal, /*!< real bands right qmf channel */ - FIXP_DBL *QmfRightImag /*!< imag bands right qmf channel */ - ) -{ - INT group; - INT subband; - - FIXP_DBL *RESTRICT HybrLeftReal; - FIXP_DBL *RESTRICT HybrLeftImag; - FIXP_DBL *RESTRICT HybrRightReal; - FIXP_DBL *RESTRICT HybrRightImag; - - FIXP_DBL tmpLeft, tmpRight; - - - /**********************************************************************************************/ - /*! - <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 7d9468c..0000000 --- a/libSBRdec/src/sbrdecoder.cpp +++ /dev/null @@ -1,1764 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. - -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief SBR decoder frontend - This module provides a frontend to the SBR decoder. The function openSBR() is called for - initialization. The function sbrDecoder_Apply() is called for each frame. sbr_Apply() will call the - required functions to decode the raw SBR data (provided by env_extr.cpp), to decode the envelope data and noise floor levels [decodeSbrData()], - and to finally apply SBR to the current frame [sbr_dec()]. - - \sa sbrDecoder_Apply(), \ref documentationOverview -*/ - -/*! - \page documentationOverview Overview of important information resources and source code documentation - - The primary source code documentation is based on generated and cross-referenced HTML files using - <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 || 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 |