/* -----------------------------------------------------------------------------
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Software License for The Fraunhofer FDK AAC Codec Library for Android
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© Copyright 1995 - 2019 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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/************************* MPEG-D DRC decoder library **************************
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Author(s):
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Description:
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*******************************************************************************/
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#ifndef DRCDEC_GAINDECODER_H
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#define DRCDEC_GAINDECODER_H
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#include "drcDecoder.h"
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/* Definitions common to gainDecoder submodule */
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#define NUM_LNB_FRAMES \
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5 /* previous frame + this frame + one frame for DM_REGULAR_DELAY + (maximum \
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delaySamples)/frameSize */
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/* QMF64 */
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#define SUBBAND_NUM_BANDS_QMF64 64
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#define SUBBAND_DOWNSAMPLING_FACTOR_QMF64 64
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#define SUBBAND_ANALYSIS_DELAY_QMF64 320
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/* QMF71 (according to ISO/IEC 23003-1:2007) */
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#define SUBBAND_NUM_BANDS_QMF71 71
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#define SUBBAND_DOWNSAMPLING_FACTOR_QMF71 64
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#define SUBBAND_ANALYSIS_DELAY_QMF71 320 + 384
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/* STFT256 (according to ISO/IEC 23008-3:2015/AMD3) */
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#define SUBBAND_NUM_BANDS_STFT256 256
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#define SUBBAND_DOWNSAMPLING_FACTOR_STFT256 256
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#define SUBBAND_ANALYSIS_DELAY_STFT256 256
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typedef enum {
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GAIN_DEC_DRC1,
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GAIN_DEC_DRC1_DRC2,
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GAIN_DEC_DRC2,
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GAIN_DEC_DRC3,
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GAIN_DEC_DRC2_DRC3
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} GAIN_DEC_LOCATION;
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typedef enum { GAIN_DEC_FRAME_SIZE, GAIN_DEC_SAMPLE_RATE } GAIN_DEC_PARAM;
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typedef struct {
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FIXP_DBL gainLin; /* e = 7 */
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SHORT time;
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} NODE_LIN;
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typedef struct {
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GAIN_INTERPOLATION_TYPE gainInterpolationType;
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int nNodes[NUM_LNB_FRAMES]; /* number of nodes, saturated to 16 */
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NODE_LIN linearNode[NUM_LNB_FRAMES][16];
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} LINEAR_NODE_BUFFER;
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typedef struct {
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int lnbPointer;
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LINEAR_NODE_BUFFER linearNodeBuffer[12];
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LINEAR_NODE_BUFFER dummyLnb;
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FIXP_DBL channelGain[8][NUM_LNB_FRAMES]; /* e = 8 */
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} DRC_GAIN_BUFFERS;
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typedef struct {
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int activeDrcOffset;
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DRC_INSTRUCTIONS_UNI_DRC* pInst;
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DRC_COEFFICIENTS_UNI_DRC* pCoef;
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DUCKING_MODIFICATION duckingModificationForChannelGroup[8];
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SCHAR channelGroupForChannel[8];
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UCHAR bandCountForChannelGroup[8];
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UCHAR gainElementForGroup[8];
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UCHAR channelGroupIsParametricDrc[8];
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UCHAR gainElementCount; /* number of different DRC gains inluding all DRC
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bands */
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int lnbIndexForChannel[8][NUM_LNB_FRAMES];
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int subbandGainsReady;
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} ACTIVE_DRC;
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typedef struct {
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int deltaTminDefault;
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INT frameSize;
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FIXP_DBL loudnessNormalisationGainDb;
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DELAY_MODE delayMode;
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int nActiveDrcs;
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ACTIVE_DRC activeDrc[MAX_ACTIVE_DRCS];
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int multiBandActiveDrcIndex;
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int channelGainActiveDrcIndex;
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FIXP_DBL channelGain[8]; /* e = 8 */
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DRC_GAIN_BUFFERS drcGainBuffers;
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FIXP_DBL subbandGains[12][4 * 1024 / 256];
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FIXP_DBL dummySubbandGains[4 * 1024 / 256];
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int status;
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int timeDomainSupported;
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SUBBAND_DOMAIN_MODE subbandDomainSupported;
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} DRC_GAIN_DECODER, *HANDLE_DRC_GAIN_DECODER;
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/* init functions */
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DRC_ERROR
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drcDec_GainDecoder_Open(HANDLE_DRC_GAIN_DECODER* phGainDec);
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DRC_ERROR
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drcDec_GainDecoder_Init(HANDLE_DRC_GAIN_DECODER hGainDec);
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DRC_ERROR
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drcDec_GainDecoder_SetParam(HANDLE_DRC_GAIN_DECODER hGainDec,
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const GAIN_DEC_PARAM paramType,
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const int paramValue);
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DRC_ERROR
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drcDec_GainDecoder_SetCodecDependentParameters(
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HANDLE_DRC_GAIN_DECODER hGainDec, const DELAY_MODE delayMode,
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const int timeDomainSupported,
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const SUBBAND_DOMAIN_MODE subbandDomainSupported);
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DRC_ERROR
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drcDec_GainDecoder_Config(HANDLE_DRC_GAIN_DECODER hGainDec,
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HANDLE_UNI_DRC_CONFIG hUniDrcConfig,
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const UCHAR numSelectedDrcSets,
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const SCHAR* selectedDrcSetIds,
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const UCHAR* selectedDownmixIds);
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/* close functions */
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DRC_ERROR
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drcDec_GainDecoder_Close(HANDLE_DRC_GAIN_DECODER* phGainDec);
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/* process functions */
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/* call drcDec_GainDecoder_Preprocess first */
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DRC_ERROR
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drcDec_GainDecoder_Preprocess(HANDLE_DRC_GAIN_DECODER hGainDec,
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HANDLE_UNI_DRC_GAIN hUniDrcGain,
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const FIXP_DBL loudnessNormalizationGainDb,
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const FIXP_SGL boost, const FIXP_SGL compress);
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/* Then call one of drcDec_GainDecoder_ProcessTimeDomain or
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* drcDec_GainDecoder_ProcessSubbandDomain */
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DRC_ERROR
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drcDec_GainDecoder_ProcessTimeDomain(
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HANDLE_DRC_GAIN_DECODER hGainDec, const int delaySamples,
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const GAIN_DEC_LOCATION drcLocation, const int channelOffset,
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const int drcChannelOffset, const int numChannelsProcessed,
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const int timeDataChannelOffset, FIXP_DBL* audioIOBuffer);
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DRC_ERROR
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drcDec_GainDecoder_ProcessSubbandDomain(
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HANDLE_DRC_GAIN_DECODER hGainDec, const int delaySamples,
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GAIN_DEC_LOCATION drcLocation, const int channelOffset,
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const int drcChannelOffset, const int numChannelsProcessed,
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const int processSingleTimeslot, FIXP_DBL* audioIOBufferReal[],
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FIXP_DBL* audioIOBufferImag[]);
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DRC_ERROR
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drcDec_GainDecoder_Conceal(HANDLE_DRC_GAIN_DECODER hGainDec,
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HANDLE_UNI_DRC_CONFIG hUniDrcConfig,
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HANDLE_UNI_DRC_GAIN hUniDrcGain);
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DRC_ERROR
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drcDec_GainDecoder_SetLoudnessNormalizationGainDb(
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HANDLE_DRC_GAIN_DECODER hGainDec, FIXP_DBL loudnessNormalizationGainDb);
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int drcDec_GainDecoder_GetFrameSize(HANDLE_DRC_GAIN_DECODER hGainDec);
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int drcDec_GainDecoder_GetDeltaTminDefault(HANDLE_DRC_GAIN_DECODER hGainDec);
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void drcDec_GainDecoder_SetChannelGains(HANDLE_DRC_GAIN_DECODER hGainDec,
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const int numChannels,
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const int frameSize,
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const FIXP_DBL* channelGainDb,
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const int audioBufferChannelOffset,
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FIXP_DBL* audioBuffer);
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#endif
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