mirror of https://github.com/mstorsjo/fdk-aac.git
302 lines
13 KiB
C
302 lines
13 KiB
C
/* -----------------------------------------------------------------------------
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Software License for The Fraunhofer FDK AAC Codec Library for Android
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© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
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Forschung e.V. 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
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that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding
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scheme for digital audio. This FDK AAC Codec software is intended to be used on
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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
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general perceptual audio codecs. AAC-ELD is considered the best-performing
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full-bandwidth communications codec by independent studies and is widely
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deployed. AAC has been standardized by ISO and IEC as part of the MPEG
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specifications.
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Patent licenses for necessary patent claims for the FDK AAC Codec (including
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those of Fraunhofer) may be obtained through Via Licensing
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(www.vialicensing.com) or through the respective patent owners individually for
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the purpose of encoding or decoding bit streams in products that are compliant
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with the ISO/IEC MPEG audio standards. Please note that most manufacturers of
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Android devices already license these patent claims through Via Licensing or
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directly from the patent owners, and therefore FDK AAC Codec software may
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already be covered under those patent licenses when it is used for those
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licensed purposes only.
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Commercially-licensed AAC software libraries, including floating-point versions
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with enhanced sound quality, are also available from Fraunhofer. Users are
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encouraged to check the Fraunhofer website for additional applications
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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,
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are permitted without payment of copyright license fees provided that you
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satisfy the following conditions:
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You must retain the complete text of this software license in redistributions of
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the FDK AAC Codec or your modifications thereto in source code form.
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You must retain the complete text of this software license in the documentation
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and/or other materials provided with redistributions of the FDK AAC Codec or
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your modifications thereto in binary form. You must make available free of
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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
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from this library without prior written permission.
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You may not charge copyright license fees for anyone to use, copy or distribute
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the FDK AAC Codec software or your modifications thereto.
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Your modified versions of the FDK AAC Codec must carry prominent notices stating
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that you changed the software and the date of any change. For modified versions
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of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android"
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must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK
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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
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limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE.
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Fraunhofer provides no warranty of patent non-infringement with respect to this
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software.
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You may use this FDK AAC Codec software or modifications thereto only for
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purposes that are authorized 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
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holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES,
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including but not limited to the implied warranties of merchantability and
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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,
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or consequential damages, including but not limited to procurement of substitute
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goods or services; loss of use, data, or profits, or business interruption,
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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
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this software, even if 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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/******************* Library for basic calculation routines ********************
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Author(s):
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Description:
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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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#define FIXP_QAS FIXP_PCM
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#define QAS_BITS SAMPLE_BITS
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#define FIXP_QSS FIXP_DBL
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#define QSS_BITS DFRACT_BITS
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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 symmetric. This flag is set internally in the QMF
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* initialization as required. */
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/* DO NOT PASS THIS FLAG TO qmfInitAnalysisFilterBank or
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* qmfInitSynthesisFilterBank */
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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
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* 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
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* post twiddling */
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#define QMF_FLAG_DOWNSAMPLED 64
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#define QMF_MAX_SYNTHESIS_BANDS (64)
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/*!
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* \brief Algorithmic scaling in sbrForwardModulation()
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*
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* The scaling in sbrForwardModulation() is caused by:
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*
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* \li 1 R_SHIFT in sbrForwardModulation()
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* \li 5/6 R_SHIFT in dct3() if using 32/64 Bands
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* \li 1 omitted gain of 2.0 in qmfForwardModulation()
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*/
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#define ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK 7
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/*!
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* \brief Algorithmic scaling in cplxSynthesisQmfFiltering()
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*
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* The scaling in cplxSynthesisQmfFiltering() is caused by:
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*
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* \li 5/6 R_SHIFT in dct2() if using 32/64 Bands
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* \li 1 omitted gain of 2.0 in qmfInverseModulation()
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* \li -6 division by 64 in synthesis filterbank
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* \li x bits external influence
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*/
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#define ALGORITHMIC_SCALING_IN_SYNTHESIS_FILTERBANK 1
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typedef struct {
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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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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 synScalefactor; /*!< Scale factor of synthesis qmf (syn only) */
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int outScalefactor; /*!< Scale factor of output data (syn only) */
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FIXP_DBL outGain_m; /*!< Mantissa of gain output data (syn only) (init with
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0x80000000 to ignore) */
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int outGain_e; /*!< Exponent of gain output data (syn only) */
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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 qmfAnalysisFiltering(
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HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Analysis Bank */
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FIXP_DBL **qmfReal, /*!< Pointer to real subband slots */
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FIXP_DBL **qmfImag, /*!< Pointer to imag subband slots */
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QMF_SCALE_FACTOR *scaleFactor, /*!< Scale factors of QMF data */
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const LONG *timeIn, /*!< Time signal */
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const int timeIn_e, /*!< Exponent of audio data */
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const int stride, /*!< Stride factor of audio data */
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FIXP_DBL *pWorkBuffer /*!< pointer to temporary working buffer */
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);
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void qmfAnalysisFiltering(
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HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Analysis Bank */
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FIXP_DBL **qmfReal, /*!< Pointer to real subband slots */
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FIXP_DBL **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 timeIn_e, /*!< Exponent of audio data */
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const int stride, /*!< Stride factor of audio data */
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FIXP_DBL *pWorkBuffer /*!< pointer to temporal working buffer */
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);
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void qmfSynthesisFiltering(
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HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */
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FIXP_DBL **QmfBufferReal, /*!< Pointer to real subband slots */
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FIXP_DBL **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_DBL *pWorkBuffer /*!< pointer to temporary working buffer. It must be
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aligned */
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);
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int qmfInitAnalysisFilterBank(
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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 qmfAnalysisFilteringSlot(
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HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Synthesis Bank */
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FIXP_DBL *qmfReal, /*!< Low and High band, real */
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FIXP_DBL *qmfImag, /*!< Low and High band, imag */
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const LONG *timeIn, /*!< Pointer to input */
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const int stride, /*!< stride factor of input */
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FIXP_DBL *pWorkBuffer /*!< pointer to temporary working buffer */
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);
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void qmfAnalysisFilteringSlot(
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HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Synthesis Bank */
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FIXP_DBL *qmfReal, /*!< Low and High band, real */
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FIXP_DBL *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_DBL *pWorkBuffer /*!< pointer to temporal working buffer */
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);
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int qmfInitSynthesisFilterBank(
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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_DBL *realSlot,
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const FIXP_DBL *imagSlot,
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const int scaleFactorLowBand,
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const int scaleFactorHighBand, INT_PCM *timeOut,
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const int timeOut_e, FIXP_DBL *pWorkBuffer);
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void qmfChangeOutScalefactor(
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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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int qmfGetOutScalefactor(
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HANDLE_QMF_FILTER_BANK synQmf /*!< Handle of Qmf Synthesis Bank */
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);
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void qmfChangeOutGain(
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HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */
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FIXP_DBL outputGain, /*!< New gain for output data (mantissa) */
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int outputGainScale /*!< New gain for output data (exponent) */
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);
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void qmfSynPrototypeFirSlot(
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HANDLE_QMF_FILTER_BANK qmf,
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FIXP_DBL *RESTRICT realSlot, /*!< Input: Pointer to real Slot */
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FIXP_DBL *RESTRICT imagSlot, /*!< Input: Pointer to imag Slot */
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INT_PCM *RESTRICT timeOut, /*!< Time domain data */
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const int timeOut_e);
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#endif /*ifndef QMF_H */
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