| |
| /* ----------------------------------------------------------------------------------------------------------- |
| 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 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" |
| |
| /* |
| * 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_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 |
| |
| |
| 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 outScalefactor; /*!< Scale factor of output data (syn only) */ |
| FIXP_DBL outGain; /*!< Gain output data (syn only) (init with 0x80000000 to ignore) */ |
| |
| 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 */ |
