mirror of https://github.com/mstorsjo/fdk-aac.git
249 lines
12 KiB
C
249 lines
12 KiB
C
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/* -----------------------------------------------------------------------------------------------------------
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Software License for The Fraunhofer FDK AAC Codec Library for Android
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© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
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All rights reserved.
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1. INTRODUCTION
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The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
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the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
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This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
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AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
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audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
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independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
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of the MPEG specifications.
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Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
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may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
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individually for the purpose of encoding or decoding bit streams in products that are compliant with
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the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
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these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
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software may already be covered under those patent licenses when it is used for those licensed purposes only.
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Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
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are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
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applications information and documentation.
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2. COPYRIGHT LICENSE
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Redistribution and use in source and binary forms, with or without modification, are permitted without
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payment of copyright license fees provided that you satisfy the following conditions:
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You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
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your modifications thereto in source code form.
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You must retain the complete text of this software license in the documentation and/or other materials
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provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
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You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
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modifications thereto to recipients of copies in binary form.
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The name of Fraunhofer may not be used to endorse or promote products derived from this library without
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prior written permission.
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You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
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software or your modifications thereto.
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Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
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and the date of any change. For modified versions of the FDK AAC Codec, the term
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"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
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"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
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3. NO PATENT LICENSE
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NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
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ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
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respect to this software.
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You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
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by appropriate patent licenses.
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4. DISCLAIMER
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This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
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"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
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of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
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CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
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including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
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or business interruption, however caused and on any theory of liability, whether in contract, strict
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liability, or tort (including negligence), arising in any way out of the use of this software, even if
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advised of the possibility of such damage.
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5. CONTACT INFORMATION
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Fraunhofer Institute for Integrated Circuits IIS
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Attention: Audio and Multimedia Departments - FDK AAC LL
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Am Wolfsmantel 33
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91058 Erlangen, Germany
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www.iis.fraunhofer.de/amm
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amm-info@iis.fraunhofer.de
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----------------------------------------------------------------------------------------------------------- */
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/*!
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\file qmf.h
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\brief Complex qmf analysis/synthesis
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\author Markus Werner
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*/
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#ifndef __QMF_H
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#define __QMF_H
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#include "common_fix.h"
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#include "FDK_tools_rom.h"
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#include "dct.h"
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/*
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* Filter coefficient type definition
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*/
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#ifdef QMF_DATA_16BIT
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#define FIXP_QMF FIXP_SGL
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#define FX_DBL2FX_QMF FX_DBL2FX_SGL
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#define FX_QMF2FX_DBL FX_SGL2FX_DBL
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#define QFRACT_BITS FRACT_BITS
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#else
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#define FIXP_QMF FIXP_DBL
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#define FX_DBL2FX_QMF
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#define FX_QMF2FX_DBL
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#define QFRACT_BITS DFRACT_BITS
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#endif
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/* ARM neon optimized QMF analysis filter requires 32 bit input.
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Implemented for RVCT only, currently disabled. See src/arm/qmf_arm.cpp:45 */
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#define FIXP_QAS FIXP_PCM
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#define QAS_BITS SAMPLE_BITS
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#ifdef QMFSYN_STATES_16BIT
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#define FIXP_QSS FIXP_SGL
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#define QSS_BITS FRACT_BITS
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#else
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#define FIXP_QSS FIXP_DBL
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#define QSS_BITS DFRACT_BITS
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#endif
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/* Flags for QMF intialization */
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/* Low Power mode flag */
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#define QMF_FLAG_LP 1
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/* Filter is not symetric. This flag is set internally in the QMF initialization as required. */
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#define QMF_FLAG_NONSYMMETRIC 2
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/* Complex Low Delay Filter Bank (or std symmetric filter bank) */
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#define QMF_FLAG_CLDFB 4
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/* Flag indicating that the states should be kept. */
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#define QMF_FLAG_KEEP_STATES 8
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/* Complex Low Delay Filter Bank used in MPEG Surround Encoder */
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#define QMF_FLAG_MPSLDFB 16
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/* Complex Low Delay Filter Bank used in MPEG Surround Encoder allows a optimized calculation of the modulation in qmfForwardModulationHQ() */
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#define QMF_FLAG_MPSLDFB_OPTIMIZE_MODULATION 32
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/* Flag to indicate HE-AAC down-sampled SBR mode (decoder) -> adapt analysis post twiddling */
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#define QMF_FLAG_DOWNSAMPLED 64
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typedef struct
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{
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int lb_scale; /*!< Scale of low band area */
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int ov_lb_scale; /*!< Scale of adjusted overlap low band area */
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int hb_scale; /*!< Scale of high band area */
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int ov_hb_scale; /*!< Scale of adjusted overlap high band area */
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} QMF_SCALE_FACTOR;
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struct QMF_FILTER_BANK
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{
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const FIXP_PFT *p_filter; /*!< Pointer to filter coefficients */
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void *FilterStates; /*!< Pointer to buffer of filter states
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FIXP_PCM in analyse and
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FIXP_DBL in synthesis filter */
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int FilterSize; /*!< Size of prototype filter. */
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const FIXP_QTW *t_cos; /*!< Modulation tables. */
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const FIXP_QTW *t_sin;
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int filterScale; /*!< filter scale */
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int no_channels; /*!< Total number of channels (subbands) */
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int no_col; /*!< Number of time slots */
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int lsb; /*!< Top of low subbands */
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int usb; /*!< Top of high subbands */
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int outScalefactor; /*!< Scale factor of output data (syn only) */
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FIXP_DBL outGain; /*!< Gain output data (syn only) (init with 0x80000000 to ignore) */
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UINT flags; /*!< flags */
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UCHAR p_stride; /*!< Stride Factor of polyphase filters */
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};
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typedef struct QMF_FILTER_BANK *HANDLE_QMF_FILTER_BANK;
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void
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qmfAnalysisFiltering( HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Analysis Bank */
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FIXP_QMF **qmfReal, /*!< Pointer to real subband slots */
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FIXP_QMF **qmfImag, /*!< Pointer to imag subband slots */
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QMF_SCALE_FACTOR *scaleFactor, /*!< Scale factors of QMF data */
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const INT_PCM *timeIn, /*!< Time signal */
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const int stride, /*!< Stride factor of audio data */
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FIXP_QMF *pWorkBuffer /*!< pointer to temporal working buffer */
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);
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void
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qmfSynthesisFiltering( HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */
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FIXP_QMF **QmfBufferReal, /*!< Pointer to real subband slots */
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FIXP_QMF **QmfBufferImag, /*!< Pointer to imag subband slots */
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const QMF_SCALE_FACTOR *scaleFactor, /*!< Scale factors of QMF data */
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const int ov_len, /*!< Length of band overlap */
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INT_PCM *timeOut, /*!< Time signal */
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const int stride, /*!< Stride factor of audio data */
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FIXP_QMF *pWorkBuffer /*!< pointer to temporal working buffer */
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);
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int
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qmfInitAnalysisFilterBank( HANDLE_QMF_FILTER_BANK h_Qmf, /*!< QMF Handle */
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FIXP_QAS *pFilterStates, /*!< Pointer to filter state buffer */
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int noCols, /*!< Number of time slots */
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int lsb, /*!< Number of lower bands */
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int usb, /*!< Number of upper bands */
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int no_channels, /*!< Number of critically sampled bands */
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int flags); /*!< Flags */
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void
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qmfAnalysisFilteringSlot( HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Synthesis Bank */
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FIXP_QMF *qmfReal, /*!< Low and High band, real */
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FIXP_QMF *qmfImag, /*!< Low and High band, imag */
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const INT_PCM *timeIn, /*!< Pointer to input */
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const int stride, /*!< stride factor of input */
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FIXP_QMF *pWorkBuffer /*!< pointer to temporal working buffer */
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);
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int
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qmfInitSynthesisFilterBank( HANDLE_QMF_FILTER_BANK h_Qmf, /*!< QMF Handle */
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FIXP_QSS *pFilterStates, /*!< Pointer to filter state buffer */
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int noCols, /*!< Number of time slots */
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int lsb, /*!< Number of lower bands */
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int usb, /*!< Number of upper bands */
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int no_channels, /*!< Number of critically sampled bands */
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int flags); /*!< Flags */
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void qmfSynthesisFilteringSlot( HANDLE_QMF_FILTER_BANK synQmf,
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const FIXP_QMF *realSlot,
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const FIXP_QMF *imagSlot,
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const int scaleFactorLowBand,
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const int scaleFactorHighBand,
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INT_PCM *timeOut,
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const int stride,
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FIXP_QMF *pWorkBuffer);
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void
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qmfChangeOutScalefactor (HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */
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int outScalefactor /*!< New scaling factor for output data */
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);
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void
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qmfChangeOutGain (HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */
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FIXP_DBL outputGain /*!< New gain for output data */
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);
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#endif /* __QMF_H */
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