| /* ----------------------------------------------------------------------------- |
| Software License for The Fraunhofer FDK AAC Codec Library for Android |
| |
| © 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. |
| |
| 2. COPYRIGHT LICENSE |
| |
| Redistribution and use in source and binary forms, with or without modification, |
| are permitted without payment of copyright license fees provided that you |
| satisfy the following conditions: |
| |
| You must retain the complete text of this software license in redistributions of |
| the FDK AAC Codec or your modifications thereto in source code form. |
| |
| You must retain the complete text of this software license in the documentation |
| and/or other materials provided with redistributions of the FDK AAC Codec or |
| your modifications thereto in binary form. You must make available free of |
| charge copies of the complete source code of the FDK AAC Codec and your |
| modifications thereto to recipients of copies in binary form. |
| |
| The name of Fraunhofer may not be used to endorse or promote products derived |
| from this library without prior written permission. |
| |
| You may not charge copyright license fees for anyone to use, copy or distribute |
| the FDK AAC Codec software or your modifications thereto. |
| |
| Your modified versions of the FDK AAC Codec must carry prominent notices stating |
| that you changed the software and the date of any change. For modified versions |
| of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" |
| must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK |
| AAC Codec Library for Android." |
| |
| 3. NO PATENT LICENSE |
| |
| NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without |
| limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. |
| Fraunhofer provides no warranty of patent non-infringement with respect to this |
| software. |
| |
| You may use this FDK AAC Codec software or modifications thereto only for |
| purposes that are authorized by appropriate patent licenses. |
| |
| 4. DISCLAIMER |
| |
| This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright |
| holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, |
| including but not limited to the implied warranties of merchantability and |
| fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR |
| CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, |
| or consequential damages, including but not limited to procurement of substitute |
| goods or services; loss of use, data, or profits, or business interruption, |
| however caused and on any theory of liability, whether in contract, strict |
| liability, or tort (including negligence), arising in any way out of the use of |
| this software, even if advised of the possibility of such damage. |
| |
| 5. CONTACT INFORMATION |
| |
| Fraunhofer Institute for Integrated Circuits IIS |
| Attention: Audio and Multimedia Departments - FDK AAC LL |
| Am Wolfsmantel 33 |
| 91058 Erlangen, Germany |
| |
| 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 |
| \author Markus Werner |
| |
| */ |
| |
| #ifndef QMF_H |
| #define QMF_H |
| |
| #include "common_fix.h" |
| #include "FDK_tools_rom.h" |
| #include "dct.h" |
| |
| #define FIXP_QAS FIXP_PCM |
| #define QAS_BITS SAMPLE_BITS |
| |
| #define FIXP_QSS FIXP_DBL |
| #define QSS_BITS DFRACT_BITS |
| |
| /* Flags for QMF intialization */ |
| /* Low Power mode flag */ |
| #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 |
| /* Flag indicating that the states should be kept. */ |
| #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 |
| |
| #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 */ |
| |
| 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 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_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 */ |
