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author | Fraunhofer IIS FDK <audio-fdk@iis.fraunhofer.de> | 2019-12-19 17:27:25 +0100 |
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committer | Jean-Michel Trivi <jmtrivi@google.com> | 2020-02-14 08:51:24 -0800 |
commit | 548244319396da7ed6a5601e6cfed0fcd946413a (patch) | |
tree | d927aa872f2b967210584765ab4dc131dc7e278a /libFDK/include/qmf.h | |
parent | 443d38963e804ce1133e2338937e1cf389092b52 (diff) | |
download | fdk-aac-548244319396da7ed6a5601e6cfed0fcd946413a.tar.gz fdk-aac-548244319396da7ed6a5601e6cfed0fcd946413a.tar.bz2 fdk-aac-548244319396da7ed6a5601e6cfed0fcd946413a.zip |
Restructure QMF implementation and provide 16/32 bit implementation in parallel.
Bug: 149514474
Test: atest DecoderTestXheAac DecoderTestAacDrc
Change-Id: I2c6b332a0b5c4966a3f680d7a4d562d9bf18752c
Diffstat (limited to 'libFDK/include/qmf.h')
-rw-r--r-- | libFDK/include/qmf.h | 105 |
1 files changed, 69 insertions, 36 deletions
diff --git a/libFDK/include/qmf.h b/libFDK/include/qmf.h index 609c6f1..78770b3 100644 --- a/libFDK/include/qmf.h +++ b/libFDK/include/qmf.h @@ -1,7 +1,7 @@ /* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -116,6 +116,7 @@ amm-info@iis.fraunhofer.de #define FIXP_QAS FIXP_PCM #define QAS_BITS SAMPLE_BITS +#define INT_PCM_QMFIN INT_PCM #define FIXP_QSS FIXP_DBL #define QSS_BITS DFRACT_BITS @@ -201,66 +202,70 @@ struct QMF_FILTER_BANK { typedef struct QMF_FILTER_BANK *HANDLE_QMF_FILTER_BANK; +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 */ +#if SAMPLE_BITS == 16 + +int qmfInitAnalysisFilterBank( + HANDLE_QMF_FILTER_BANK h_Qmf, /*!< QMF Handle */ + FIXP_DBL *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 */ +#endif + 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_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 temporary working buffer */ + FIXP_DBL *pWorkBuffer /*!< pointer to temporal working buffer */ ); +#if SAMPLE_BITS == 16 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 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 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 */ + FIXP_DBL *pWorkBuffer /*!< pointer to temporary 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 */ +#endif 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_PCM *timeIn, /*!< Pointer to input */ const int stride, /*!< stride factor of input */ - FIXP_DBL *pWorkBuffer /*!< pointer to temporary working buffer */ + FIXP_DBL *pWorkBuffer /*!< pointer to temporal working buffer */ ); +#if SAMPLE_BITS == 16 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 LONG *timeIn, /*!< Pointer to input */ const int stride, /*!< stride factor of input */ - FIXP_DBL *pWorkBuffer /*!< pointer to temporal working buffer */ + FIXP_DBL *pWorkBuffer /*!< pointer to temporary working buffer */ ); +#endif + int qmfInitSynthesisFilterBank( HANDLE_QMF_FILTER_BANK h_Qmf, /*!< QMF Handle */ FIXP_QSS *pFilterStates, /*!< Pointer to filter state buffer */ @@ -270,12 +275,46 @@ int qmfInitSynthesisFilterBank( int no_channels, /*!< Number of critically sampled bands */ int flags); /*!< Flags */ +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 */ +); +#if SAMPLE_BITS == 16 + +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 */ + LONG *timeOut, /*!< Time signal */ + const int timeOut_e, /*!< Target exponent for timeOut */ + FIXP_DBL *pWorkBuffer /*!< pointer to temporary working buffer */ +); +#endif + 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); +#if SAMPLE_BITS == 16 + +void qmfSynthesisFilteringSlot(HANDLE_QMF_FILTER_BANK synQmf, + const FIXP_DBL *realSlot, + const FIXP_DBL *imagSlot, + const int scaleFactorLowBand, + const int scaleFactorHighBand, LONG *timeOut, + const int timeOut_e, FIXP_DBL *pWorkBuffer); +#endif void qmfChangeOutScalefactor( HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */ @@ -291,11 +330,5 @@ void qmfChangeOutGain( 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 */ |