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Diffstat (limited to 'libFDK/include/qmf.h')
| -rw-r--r-- | libFDK/include/qmf.h | 419 |
1 files changed, 236 insertions, 183 deletions
diff --git a/libFDK/include/qmf.h b/libFDK/include/qmf.h index be69477..609c6f1 100644 --- a/libFDK/include/qmf.h +++ b/libFDK/include/qmf.h @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- 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. +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. 2. COPYRIGHT LICENSE -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." 3. NO PATENT LICENSE -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION @@ -79,170 +90,212 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/******************* Library for basic calculation routines ******************** + + Author(s): + + Description: + +*******************************************************************************/ /*! \file qmf.h - \brief Complex qmf analysis/synthesis + \brief Complex qmf analysis/synthesis \author Markus Werner */ -#ifndef __QMF_H -#define __QMF_H - +#ifndef QMF_H +#define QMF_H #include "common_fix.h" #include "FDK_tools_rom.h" #include "dct.h" -/* - * Filter coefficient type definition - */ -#ifdef QMF_DATA_16BIT -#define FIXP_QMF FIXP_SGL -#define FX_DBL2FX_QMF FX_DBL2FX_SGL -#define FX_QMF2FX_DBL FX_SGL2FX_DBL -#define QFRACT_BITS FRACT_BITS -#else -#define FIXP_QMF FIXP_DBL -#define FX_DBL2FX_QMF -#define FX_QMF2FX_DBL -#define QFRACT_BITS DFRACT_BITS -#endif - -/* ARM neon optimized QMF analysis filter requires 32 bit input. - Implemented for RVCT only, currently disabled. See src/arm/qmf_arm.cpp:45 */ #define FIXP_QAS FIXP_PCM #define QAS_BITS SAMPLE_BITS -#ifdef QMFSYN_STATES_16BIT -#define FIXP_QSS FIXP_SGL -#define QSS_BITS FRACT_BITS -#else #define FIXP_QSS FIXP_DBL #define QSS_BITS DFRACT_BITS -#endif /* Flags for QMF intialization */ /* Low Power mode flag */ -#define QMF_FLAG_LP 1 -/* Filter is not symetric. This flag is set internally in the QMF initialization as required. */ +#define QMF_FLAG_LP 1 +/* Filter is not symmetric. This flag is set internally in the QMF + * initialization as required. */ +/* DO NOT PASS THIS FLAG TO qmfInitAnalysisFilterBank or + * qmfInitSynthesisFilterBank */ #define QMF_FLAG_NONSYMMETRIC 2 /* Complex Low Delay Filter Bank (or std symmetric filter bank) */ -#define QMF_FLAG_CLDFB 4 +#define QMF_FLAG_CLDFB 4 /* Flag indicating that the states should be kept. */ -#define QMF_FLAG_KEEP_STATES 8 +#define QMF_FLAG_KEEP_STATES 8 /* Complex Low Delay Filter Bank used in MPEG Surround Encoder */ -#define QMF_FLAG_MPSLDFB 16 -/* Complex Low Delay Filter Bank used in MPEG Surround Encoder allows a optimized calculation of the modulation in qmfForwardModulationHQ() */ -#define QMF_FLAG_MPSLDFB_OPTIMIZE_MODULATION 32 -/* Flag to indicate HE-AAC down-sampled SBR mode (decoder) -> adapt analysis post twiddling */ -#define QMF_FLAG_DOWNSAMPLED 64 - - -typedef struct -{ - int lb_scale; /*!< Scale of low band area */ - int ov_lb_scale; /*!< Scale of adjusted overlap low band area */ - int hb_scale; /*!< Scale of high band area */ - int ov_hb_scale; /*!< Scale of adjusted overlap high band area */ +#define QMF_FLAG_MPSLDFB 16 +/* Complex Low Delay Filter Bank used in MPEG Surround Encoder allows a + * optimized calculation of the modulation in qmfForwardModulationHQ() */ +#define QMF_FLAG_MPSLDFB_OPTIMIZE_MODULATION 32 +/* Flag to indicate HE-AAC down-sampled SBR mode (decoder) -> adapt analysis + * post twiddling */ +#define QMF_FLAG_DOWNSAMPLED 64 + +#define QMF_MAX_SYNTHESIS_BANDS (64) + +/*! + * \brief Algorithmic scaling in sbrForwardModulation() + * + * The scaling in sbrForwardModulation() is caused by: + * + * \li 1 R_SHIFT in sbrForwardModulation() + * \li 5/6 R_SHIFT in dct3() if using 32/64 Bands + * \li 1 omitted gain of 2.0 in qmfForwardModulation() + */ +#define ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK 7 + +/*! + * \brief Algorithmic scaling in cplxSynthesisQmfFiltering() + * + * The scaling in cplxSynthesisQmfFiltering() is caused by: + * + * \li 5/6 R_SHIFT in dct2() if using 32/64 Bands + * \li 1 omitted gain of 2.0 in qmfInverseModulation() + * \li -6 division by 64 in synthesis filterbank + * \li x bits external influence + */ +#define ALGORITHMIC_SCALING_IN_SYNTHESIS_FILTERBANK 1 + +typedef struct { + int lb_scale; /*!< Scale of low band area */ + int ov_lb_scale; /*!< Scale of adjusted overlap low band area */ + int hb_scale; /*!< Scale of high band area */ + int ov_hb_scale; /*!< Scale of adjusted overlap high band area */ } QMF_SCALE_FACTOR; -struct QMF_FILTER_BANK -{ - const FIXP_PFT *p_filter; /*!< Pointer to filter coefficients */ +struct QMF_FILTER_BANK { + const FIXP_PFT *p_filter; /*!< Pointer to filter coefficients */ - void *FilterStates; /*!< Pointer to buffer of filter states - FIXP_PCM in analyse and - FIXP_DBL in synthesis filter */ - int FilterSize; /*!< Size of prototype filter. */ - const FIXP_QTW *t_cos; /*!< Modulation tables. */ + void *FilterStates; /*!< Pointer to buffer of filter states + FIXP_PCM in analyse and + FIXP_DBL in synthesis filter */ + int FilterSize; /*!< Size of prototype filter. */ + const FIXP_QTW *t_cos; /*!< Modulation tables. */ const FIXP_QTW *t_sin; - int filterScale; /*!< filter scale */ - - int no_channels; /*!< Total number of channels (subbands) */ - int no_col; /*!< Number of time slots */ - int lsb; /*!< Top of low subbands */ - int usb; /*!< Top of high subbands */ + int filterScale; /*!< filter scale */ - int outScalefactor; /*!< Scale factor of output data (syn only) */ - FIXP_DBL outGain; /*!< Gain output data (syn only) (init with 0x80000000 to ignore) */ + int no_channels; /*!< Total number of channels (subbands) */ + int no_col; /*!< Number of time slots */ + int lsb; /*!< Top of low subbands */ + int usb; /*!< Top of high subbands */ - UINT flags; /*!< flags */ - UCHAR p_stride; /*!< Stride Factor of polyphase filters */ + int synScalefactor; /*!< Scale factor of synthesis qmf (syn only) */ + int outScalefactor; /*!< Scale factor of output data (syn only) */ + FIXP_DBL outGain_m; /*!< Mantissa of gain output data (syn only) (init with + 0x80000000 to ignore) */ + int outGain_e; /*!< Exponent of gain output data (syn only) */ + UINT flags; /*!< flags */ + UCHAR p_stride; /*!< Stride Factor of polyphase filters */ }; typedef struct QMF_FILTER_BANK *HANDLE_QMF_FILTER_BANK; -void -qmfAnalysisFiltering( HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Analysis Bank */ - FIXP_QMF **qmfReal, /*!< Pointer to real subband slots */ - FIXP_QMF **qmfImag, /*!< Pointer to imag subband slots */ - QMF_SCALE_FACTOR *scaleFactor, /*!< Scale factors of QMF data */ - const INT_PCM *timeIn, /*!< Time signal */ - const int stride, /*!< Stride factor of audio data */ - FIXP_QMF *pWorkBuffer /*!< pointer to temporal working buffer */ - ); - -void -qmfSynthesisFiltering( HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */ - FIXP_QMF **QmfBufferReal, /*!< Pointer to real subband slots */ - FIXP_QMF **QmfBufferImag, /*!< Pointer to imag subband slots */ - const QMF_SCALE_FACTOR *scaleFactor, /*!< Scale factors of QMF data */ - const int ov_len, /*!< Length of band overlap */ - INT_PCM *timeOut, /*!< Time signal */ - const int stride, /*!< Stride factor of audio data */ - FIXP_QMF *pWorkBuffer /*!< pointer to temporal working buffer */ - ); - -int -qmfInitAnalysisFilterBank( HANDLE_QMF_FILTER_BANK h_Qmf, /*!< QMF Handle */ - FIXP_QAS *pFilterStates, /*!< Pointer to filter state buffer */ - int noCols, /*!< Number of time slots */ - int lsb, /*!< Number of lower bands */ - int usb, /*!< Number of upper bands */ - int no_channels, /*!< Number of critically sampled bands */ - int flags); /*!< Flags */ - -void -qmfAnalysisFilteringSlot( HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Synthesis Bank */ - FIXP_QMF *qmfReal, /*!< Low and High band, real */ - FIXP_QMF *qmfImag, /*!< Low and High band, imag */ - const INT_PCM *timeIn, /*!< Pointer to input */ - const int stride, /*!< stride factor of input */ - FIXP_QMF *pWorkBuffer /*!< pointer to temporal working buffer */ - ); - -int -qmfInitSynthesisFilterBank( HANDLE_QMF_FILTER_BANK h_Qmf, /*!< QMF Handle */ - FIXP_QSS *pFilterStates, /*!< Pointer to filter state buffer */ - int noCols, /*!< Number of time slots */ - int lsb, /*!< Number of lower bands */ - int usb, /*!< Number of upper bands */ - int no_channels, /*!< Number of critically sampled bands */ - int flags); /*!< Flags */ - -void qmfSynthesisFilteringSlot( HANDLE_QMF_FILTER_BANK synQmf, - const FIXP_QMF *realSlot, - const FIXP_QMF *imagSlot, - const int scaleFactorLowBand, - const int scaleFactorHighBand, - INT_PCM *timeOut, - const int stride, - FIXP_QMF *pWorkBuffer); - -void -qmfChangeOutScalefactor (HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */ - int outScalefactor /*!< New scaling factor for output data */ - ); - -void -qmfChangeOutGain (HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */ - FIXP_DBL outputGain /*!< New gain for output data */ - ); - - - -#endif /* __QMF_H */ +void qmfAnalysisFiltering( + HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Analysis Bank */ + FIXP_DBL **qmfReal, /*!< Pointer to real subband slots */ + FIXP_DBL **qmfImag, /*!< Pointer to imag subband slots */ + QMF_SCALE_FACTOR *scaleFactor, /*!< Scale factors of QMF data */ + const LONG *timeIn, /*!< Time signal */ + const int timeIn_e, /*!< Exponent of audio data */ + const int stride, /*!< Stride factor of audio data */ + FIXP_DBL *pWorkBuffer /*!< pointer to temporary working buffer */ +); + +void qmfAnalysisFiltering( + HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Analysis Bank */ + FIXP_DBL **qmfReal, /*!< Pointer to real subband slots */ + FIXP_DBL **qmfImag, /*!< Pointer to imag subband slots */ + QMF_SCALE_FACTOR *scaleFactor, /*!< Scale factors of QMF data */ + const INT_PCM *timeIn, /*!< Time signal */ + const int timeIn_e, /*!< Exponent of audio data */ + const int stride, /*!< Stride factor of audio data */ + FIXP_DBL *pWorkBuffer /*!< pointer to temporal working buffer */ +); + +void qmfSynthesisFiltering( + HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */ + FIXP_DBL **QmfBufferReal, /*!< Pointer to real subband slots */ + FIXP_DBL **QmfBufferImag, /*!< Pointer to imag subband slots */ + const QMF_SCALE_FACTOR *scaleFactor, /*!< Scale factors of QMF data */ + const int ov_len, /*!< Length of band overlap */ + INT_PCM *timeOut, /*!< Time signal */ + const INT stride, /*!< Stride factor of audio data */ + FIXP_DBL *pWorkBuffer /*!< pointer to temporary working buffer. It must be + aligned */ +); + +int qmfInitAnalysisFilterBank( + HANDLE_QMF_FILTER_BANK h_Qmf, /*!< QMF Handle */ + FIXP_QAS *pFilterStates, /*!< Pointer to filter state buffer */ + int noCols, /*!< Number of time slots */ + int lsb, /*!< Number of lower bands */ + int usb, /*!< Number of upper bands */ + int no_channels, /*!< Number of critically sampled bands */ + int flags); /*!< Flags */ + +void qmfAnalysisFilteringSlot( + HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Synthesis Bank */ + FIXP_DBL *qmfReal, /*!< Low and High band, real */ + FIXP_DBL *qmfImag, /*!< Low and High band, imag */ + const LONG *timeIn, /*!< Pointer to input */ + const int stride, /*!< stride factor of input */ + FIXP_DBL *pWorkBuffer /*!< pointer to temporary working buffer */ +); + +void qmfAnalysisFilteringSlot( + HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Synthesis Bank */ + FIXP_DBL *qmfReal, /*!< Low and High band, real */ + FIXP_DBL *qmfImag, /*!< Low and High band, imag */ + const INT_PCM *timeIn, /*!< Pointer to input */ + const int stride, /*!< stride factor of input */ + FIXP_DBL *pWorkBuffer /*!< pointer to temporal working buffer */ +); +int qmfInitSynthesisFilterBank( + HANDLE_QMF_FILTER_BANK h_Qmf, /*!< QMF Handle */ + FIXP_QSS *pFilterStates, /*!< Pointer to filter state buffer */ + int noCols, /*!< Number of time slots */ + int lsb, /*!< Number of lower bands */ + int usb, /*!< Number of upper bands */ + int no_channels, /*!< Number of critically sampled bands */ + int flags); /*!< Flags */ + +void qmfSynthesisFilteringSlot(HANDLE_QMF_FILTER_BANK synQmf, + const FIXP_DBL *realSlot, + const FIXP_DBL *imagSlot, + const int scaleFactorLowBand, + const int scaleFactorHighBand, INT_PCM *timeOut, + const int timeOut_e, FIXP_DBL *pWorkBuffer); + +void qmfChangeOutScalefactor( + HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */ + int outScalefactor /*!< New scaling factor for output data */ +); + +int qmfGetOutScalefactor( + HANDLE_QMF_FILTER_BANK synQmf /*!< Handle of Qmf Synthesis Bank */ +); + +void qmfChangeOutGain( + HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */ + FIXP_DBL outputGain, /*!< New gain for output data (mantissa) */ + int outputGainScale /*!< New gain for output data (exponent) */ +); +void qmfSynPrototypeFirSlot( + HANDLE_QMF_FILTER_BANK qmf, + FIXP_DBL *RESTRICT realSlot, /*!< Input: Pointer to real Slot */ + FIXP_DBL *RESTRICT imagSlot, /*!< Input: Pointer to imag Slot */ + INT_PCM *RESTRICT timeOut, /*!< Time domain data */ + const int timeOut_e); + +#endif /*ifndef QMF_H */ |