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diff --git a/libFDK/include/qmf.h b/libFDK/include/qmf.h new file mode 100644 index 0000000..16d2b70 --- /dev/null +++ b/libFDK/include/qmf.h @@ -0,0 +1,246 @@ + +/* ----------------------------------------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2012 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 + + +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 */ |