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diff --git a/libFDK/include/qmf.h b/libFDK/include/qmf.h deleted file mode 100644 index be69477..0000000 --- a/libFDK/include/qmf.h +++ /dev/null @@ -1,248 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -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 */ |