| /* ----------------------------------------------------------------------------- |
| Software License for The Fraunhofer FDK AAC Codec Library for Android |
| |
| © Copyright 1995 - 2018 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 |
| ----------------------------------------------------------------------------- */ |
| |
| /************************* MPEG-D DRC decoder library ************************** |
| |
| Author(s): |
| |
| Description: |
| |
| *******************************************************************************/ |
| |
| #ifndef DRCDEC_GAINDECODER_H |
| #define DRCDEC_GAINDECODER_H |
| |
| #include "drcDecoder.h" |
| |
| /* Definitions common to gainDecoder submodule */ |
| |
| #define NUM_LNB_FRAMES \ |
| 5 /* previous frame + this frame + one frame for DM_REGULAR_DELAY + (maximum \ |
| delaySamples)/frameSize */ |
| |
| /* QMF64 */ |
| #define SUBBAND_NUM_BANDS_QMF64 64 |
| #define SUBBAND_DOWNSAMPLING_FACTOR_QMF64 64 |
| #define SUBBAND_ANALYSIS_DELAY_QMF64 320 |
| |
| /* QMF71 (according to ISO/IEC 23003-1:2007) */ |
| #define SUBBAND_NUM_BANDS_QMF71 71 |
| #define SUBBAND_DOWNSAMPLING_FACTOR_QMF71 64 |
| #define SUBBAND_ANALYSIS_DELAY_QMF71 320 + 384 |
| |
| /* STFT256 (according to ISO/IEC 23008-3:2015/AMD3) */ |
| #define SUBBAND_NUM_BANDS_STFT256 256 |
| #define SUBBAND_DOWNSAMPLING_FACTOR_STFT256 256 |
| #define SUBBAND_ANALYSIS_DELAY_STFT256 256 |
| |
| typedef enum { |
| GAIN_DEC_DRC1, |
| GAIN_DEC_DRC1_DRC2, |
| GAIN_DEC_DRC2, |
| GAIN_DEC_DRC3, |
| GAIN_DEC_DRC2_DRC3 |
| } GAIN_DEC_LOCATION; |
| |
| typedef struct { |
| FIXP_DBL gainLin; /* e = 7 */ |
| SHORT time; |
| } NODE_LIN; |
| |
| typedef struct { |
| GAIN_INTERPOLATION_TYPE gainInterpolationType; |
| int nNodes[NUM_LNB_FRAMES]; /* number of nodes, saturated to 16 */ |
| NODE_LIN linearNode[NUM_LNB_FRAMES][16]; |
| } LINEAR_NODE_BUFFER; |
| |
| typedef struct { |
| int lnbPointer; |
| LINEAR_NODE_BUFFER linearNodeBuffer[12]; |
| LINEAR_NODE_BUFFER dummyLnb; |
| FIXP_DBL channelGain[8][NUM_LNB_FRAMES]; /* e = 8 */ |
| } DRC_GAIN_BUFFERS; |
| |
| typedef struct { |
| int activeDrcOffset; |
| DRC_INSTRUCTIONS_UNI_DRC* pInst; |
| DRC_COEFFICIENTS_UNI_DRC* pCoef; |
| |
| DUCKING_MODIFICATION duckingModificationForChannelGroup[8]; |
| SCHAR channelGroupForChannel[8]; |
| |
| UCHAR bandCountForChannelGroup[8]; |
| UCHAR gainElementForGroup[8]; |
| UCHAR channelGroupIsParametricDrc[8]; |
| UCHAR gainElementCount; /* number of different DRC gains inluding all DRC |
| bands */ |
| int lnbIndexForChannel[8][NUM_LNB_FRAMES]; |
| int subbandGainsReady; |
| } ACTIVE_DRC; |
| |
| typedef struct { |
| int deltaTminDefault; |
| INT frameSize; |
| FIXP_DBL loudnessNormalisationGainDb; |
| DELAY_MODE delayMode; |
| |
| int nActiveDrcs; |
| ACTIVE_DRC activeDrc[MAX_ACTIVE_DRCS]; |
| int multiBandActiveDrcIndex; |
| int channelGainActiveDrcIndex; |
| FIXP_DBL channelGain[8]; /* e = 8 */ |
| |
| DRC_GAIN_BUFFERS drcGainBuffers; |
| FIXP_DBL subbandGains[12][4 * 1024 / 256]; |
| FIXP_DBL dummySubbandGains[4 * 1024 / 256]; |
| |
| int status; |
| int timeDomainSupported; |
| SUBBAND_DOMAIN_MODE subbandDomainSupported; |
| } DRC_GAIN_DECODER, *HANDLE_DRC_GAIN_DECODER; |
| |
| /* init functions */ |
| DRC_ERROR |
| drcDec_GainDecoder_Open(HANDLE_DRC_GAIN_DECODER* phGainDec); |
| |
| DRC_ERROR |
| drcDec_GainDecoder_Init(HANDLE_DRC_GAIN_DECODER hGainDec, const int frameSize, |
| const int sampleRate); |
| |
| DRC_ERROR |
| drcDec_GainDecoder_SetCodecDependentParameters( |
| HANDLE_DRC_GAIN_DECODER hGainDec, const DELAY_MODE delayMode, |
| const int timeDomainSupported, |
| const SUBBAND_DOMAIN_MODE subbandDomainSupported); |
| |
| DRC_ERROR |
| drcDec_GainDecoder_Config(HANDLE_DRC_GAIN_DECODER hGainDec, |
| HANDLE_UNI_DRC_CONFIG hUniDrcConfig, |
| const UCHAR numSelectedDrcSets, |
| const SCHAR* selectedDrcSetIds, |
| const UCHAR* selectedDownmixIds); |
| |
| /* close functions */ |
| DRC_ERROR |
| drcDec_GainDecoder_Close(HANDLE_DRC_GAIN_DECODER* phGainDec); |
| |
| /* process functions */ |
| |
| /* call drcDec_GainDecoder_Preprocess first */ |
| DRC_ERROR |
| drcDec_GainDecoder_Preprocess(HANDLE_DRC_GAIN_DECODER hGainDec, |
| HANDLE_UNI_DRC_GAIN hUniDrcGain, |
| const FIXP_DBL loudnessNormalizationGainDb, |
| const FIXP_SGL boost, const FIXP_SGL compress); |
| |
| /* Then call one of drcDec_GainDecoder_ProcessTimeDomain or |
| * drcDec_GainDecoder_ProcessSubbandDomain */ |
| DRC_ERROR |
| drcDec_GainDecoder_ProcessTimeDomain( |
| HANDLE_DRC_GAIN_DECODER hGainDec, const int delaySamples, |
| const GAIN_DEC_LOCATION drcLocation, const int channelOffset, |
| const int drcChannelOffset, const int numChannelsProcessed, |
| const int timeDataChannelOffset, FIXP_DBL* audioIOBuffer); |
| |
| DRC_ERROR |
| drcDec_GainDecoder_ProcessSubbandDomain( |
| HANDLE_DRC_GAIN_DECODER hGainDec, const int delaySamples, |
| GAIN_DEC_LOCATION drcLocation, const int channelOffset, |
| const int drcChannelOffset, const int numChannelsProcessed, |
| const int processSingleTimeslot, FIXP_DBL* audioIOBufferReal[], |
| FIXP_DBL* audioIOBufferImag[]); |
| |
| DRC_ERROR |
| drcDec_GainDecoder_Conceal(HANDLE_DRC_GAIN_DECODER hGainDec, |
| HANDLE_UNI_DRC_CONFIG hUniDrcConfig, |
| HANDLE_UNI_DRC_GAIN hUniDrcGain); |
| |
| DRC_ERROR |
| drcDec_GainDecoder_SetLoudnessNormalizationGainDb( |
| HANDLE_DRC_GAIN_DECODER hGainDec, FIXP_DBL loudnessNormalizationGainDb); |
| |
| int drcDec_GainDecoder_GetFrameSize(HANDLE_DRC_GAIN_DECODER hGainDec); |
| |
| int drcDec_GainDecoder_GetDeltaTminDefault(HANDLE_DRC_GAIN_DECODER hGainDec); |
| |
| void drcDec_GainDecoder_SetChannelGains(HANDLE_DRC_GAIN_DECODER hGainDec, |
| const int numChannels, |
| const int frameSize, |
| const FIXP_DBL* channelGainDb, |
| const int audioBufferChannelOffset, |
| FIXP_DBL* audioBuffer); |
| |
| #endif |