| /* ----------------------------------------------------------------------------- |
| 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: |
| |
| *******************************************************************************/ |
| |
| #include "drcDec_types.h" |
| #include "drcDec_gainDecoder.h" |
| #include "drcGainDec_preprocess.h" |
| #include "drcGainDec_init.h" |
| #include "drcGainDec_process.h" |
| #include "drcDec_tools.h" |
| |
| /*******************************************/ |
| /* static functions */ |
| /*******************************************/ |
| |
| static int _fitsLocation(DRC_INSTRUCTIONS_UNI_DRC* pInst, |
| const GAIN_DEC_LOCATION drcLocation) { |
| int downmixId = pInst->drcApplyToDownmix ? pInst->downmixId[0] : 0; |
| switch (drcLocation) { |
| case GAIN_DEC_DRC1: |
| return (downmixId == 0); |
| case GAIN_DEC_DRC1_DRC2: |
| return ((downmixId == 0) || (downmixId == DOWNMIX_ID_ANY_DOWNMIX)); |
| case GAIN_DEC_DRC2: |
| return (downmixId == DOWNMIX_ID_ANY_DOWNMIX); |
| case GAIN_DEC_DRC3: |
| return ((downmixId != 0) && (downmixId != DOWNMIX_ID_ANY_DOWNMIX)); |
| case GAIN_DEC_DRC2_DRC3: |
| return (downmixId != 0); |
| } |
| return 0; |
| } |
| |
| static void _setChannelGains(HANDLE_DRC_GAIN_DECODER hGainDec, |
| const int numChannelGains, |
| const FIXP_DBL* channelGainDb) { |
| int i, channelGain_e; |
| FIXP_DBL channelGain; |
| FDK_ASSERT(numChannelGains <= 8); |
| for (i = 0; i < numChannelGains; i++) { |
| if (channelGainDb[i] == (FIXP_DBL)MINVAL_DBL) { |
| hGainDec->channelGain[i] = (FIXP_DBL)0; |
| } else { |
| /* add loudness normalisation gain (dB) to channel gain (dB) */ |
| FIXP_DBL tmp_channelGainDb = (channelGainDb[i] >> 1) + |
| (hGainDec->loudnessNormalisationGainDb >> 2); |
| tmp_channelGainDb = |
| SATURATE_LEFT_SHIFT(tmp_channelGainDb, 1, DFRACT_BITS); |
| channelGain = dB2lin(tmp_channelGainDb, 8, &channelGain_e); |
| hGainDec->channelGain[i] = scaleValue(channelGain, channelGain_e - 8); |
| } |
| } |
| } |
| |
| /*******************************************/ |
| /* public functions */ |
| /*******************************************/ |
| |
| DRC_ERROR |
| drcDec_GainDecoder_Open(HANDLE_DRC_GAIN_DECODER* phGainDec) { |
| DRC_GAIN_DECODER* hGainDec = NULL; |
| |
| hGainDec = (DRC_GAIN_DECODER*)FDKcalloc(1, sizeof(DRC_GAIN_DECODER)); |
| if (hGainDec == NULL) return DE_MEMORY_ERROR; |
| |
| hGainDec->multiBandActiveDrcIndex = -1; |
| hGainDec->channelGainActiveDrcIndex = -1; |
| |
| *phGainDec = hGainDec; |
| |
| return DE_OK; |
| } |
| |
| DRC_ERROR |
| drcDec_GainDecoder_Init(HANDLE_DRC_GAIN_DECODER hGainDec, const int frameSize, |
| const int sampleRate) { |
| DRC_ERROR err = DE_OK; |
| |
| err = initGainDec(hGainDec, frameSize, sampleRate); |
| if (err) return err; |
| |
| initDrcGainBuffers(hGainDec->frameSize, &hGainDec->drcGainBuffers); |
| |
| return err; |
| } |
| |
| DRC_ERROR |
| drcDec_GainDecoder_SetCodecDependentParameters( |
| HANDLE_DRC_GAIN_DECODER hGainDec, const DELAY_MODE delayMode, |
| const int timeDomainSupported, |
| const SUBBAND_DOMAIN_MODE subbandDomainSupported) { |
| if ((delayMode != DM_REGULAR_DELAY) && (delayMode != DM_LOW_DELAY)) { |
| return DE_NOT_OK; |
| } |
| hGainDec->delayMode = delayMode; |
| hGainDec->timeDomainSupported = timeDomainSupported; |
| hGainDec->subbandDomainSupported = subbandDomainSupported; |
| |
| return DE_OK; |
| } |
| |
| DRC_ERROR |
| drcDec_GainDecoder_Config(HANDLE_DRC_GAIN_DECODER hGainDec, |
| HANDLE_UNI_DRC_CONFIG hUniDrcConfig, |
| const UCHAR numSelectedDrcSets, |
| const SCHAR* selectedDrcSetIds, |
| const UCHAR* selectedDownmixIds) { |
| DRC_ERROR err = DE_OK; |
| int a; |
| |
| hGainDec->nActiveDrcs = 0; |
| hGainDec->multiBandActiveDrcIndex = -1; |
| hGainDec->channelGainActiveDrcIndex = -1; |
| for (a = 0; a < numSelectedDrcSets; a++) { |
| err = initActiveDrc(hGainDec, hUniDrcConfig, selectedDrcSetIds[a], |
| selectedDownmixIds[a]); |
| if (err) return err; |
| } |
| |
| err = initActiveDrcOffset(hGainDec); |
| if (err) return err; |
| |
| return err; |
| } |
| |
| DRC_ERROR |
| drcDec_GainDecoder_Close(HANDLE_DRC_GAIN_DECODER* phGainDec) { |
| if (*phGainDec != NULL) { |
| FDKfree(*phGainDec); |
| *phGainDec = NULL; |
| } |
| |
| return DE_OK; |
| } |
| |
| 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) { |
| DRC_ERROR err = DE_OK; |
| int a, c; |
| |
| /* lnbPointer is the index on the most recent node buffer */ |
| hGainDec->drcGainBuffers.lnbPointer++; |
| if (hGainDec->drcGainBuffers.lnbPointer >= NUM_LNB_FRAMES) |
| hGainDec->drcGainBuffers.lnbPointer = 0; |
| |
| for (a = 0; a < hGainDec->nActiveDrcs; a++) { |
| /* prepare gain interpolation of sequences used by copying and modifying |
| * nodes in node buffers */ |
| err = prepareDrcGain(hGainDec, hUniDrcGain, compress, boost, |
| loudnessNormalizationGainDb, a); |
| if (err) return err; |
| } |
| |
| for (a = 0; a < MAX_ACTIVE_DRCS; a++) { |
| for (c = 0; c < 8; c++) { |
| hGainDec->activeDrc[a] |
| .lnbIndexForChannel[c][hGainDec->drcGainBuffers.lnbPointer] = |
| -1; /* "no DRC processing" */ |
| } |
| hGainDec->activeDrc[a].subbandGainsReady = 0; |
| } |
| |
| for (c = 0; c < 8; c++) { |
| hGainDec->drcGainBuffers |
| .channelGain[c][hGainDec->drcGainBuffers.lnbPointer] = |
| FL2FXCONST_DBL(1.0f / (float)(1 << 8)); |
| } |
| |
| return err; |
| } |
| |
| /* create gain sequence out of gain sequences of last frame for concealment and |
| * flushing */ |
| DRC_ERROR |
| drcDec_GainDecoder_Conceal(HANDLE_DRC_GAIN_DECODER hGainDec, |
| HANDLE_UNI_DRC_CONFIG hUniDrcConfig, |
| HANDLE_UNI_DRC_GAIN hUniDrcGain) { |
| int seq, gainSequenceCount; |
| DRC_COEFFICIENTS_UNI_DRC* pCoef = |
| selectDrcCoefficients(hUniDrcConfig, LOCATION_SELECTED); |
| if (pCoef == NULL) return DE_OK; |
| |
| gainSequenceCount = fMin(pCoef->gainSequenceCount, (UCHAR)12); |
| |
| for (seq = 0; seq < gainSequenceCount; seq++) { |
| int lastNodeIndex = 0; |
| FIXP_SGL lastGainDb = (FIXP_SGL)0; |
| |
| lastNodeIndex = hUniDrcGain->nNodes[seq] - 1; |
| if ((lastNodeIndex >= 0) && (lastNodeIndex < 16)) { |
| lastGainDb = hUniDrcGain->gainNode[seq][lastNodeIndex].gainDb; |
| } |
| |
| hUniDrcGain->nNodes[seq] = 1; |
| if (lastGainDb > (FIXP_SGL)0) { |
| hUniDrcGain->gainNode[seq][0].gainDb = |
| FX_DBL2FX_SGL(fMult(FL2FXCONST_SGL(0.9f), lastGainDb)); |
| } else { |
| hUniDrcGain->gainNode[seq][0].gainDb = |
| FX_DBL2FX_SGL(fMult(FL2FXCONST_SGL(0.98f), lastGainDb)); |
| } |
| hUniDrcGain->gainNode[seq][0].time = hGainDec->frameSize - 1; |
| } |
| return DE_OK; |
| } |
| |
| 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) { |
| int c, i; |
| |
| if (hGainDec->channelGainActiveDrcIndex >= 0) { |
| /* channel gains will be applied in drcDec_GainDecoder_ProcessTimeDomain or |
| * drcDec_GainDecoder_ProcessSubbandDomain, respectively. */ |
| _setChannelGains(hGainDec, numChannels, channelGainDb); |
| |
| if (!hGainDec->status) { /* overwrite previous channel gains at startup */ |
| DRC_GAIN_BUFFERS* pDrcGainBuffers = &hGainDec->drcGainBuffers; |
| for (c = 0; c < numChannels; c++) { |
| for (i = 0; i < NUM_LNB_FRAMES; i++) { |
| pDrcGainBuffers->channelGain[c][i] = hGainDec->channelGain[c]; |
| } |
| } |
| hGainDec->status = 1; |
| } |
| } else { |
| /* smooth and apply channel gains */ |
| FIXP_DBL prevChannelGain[8]; |
| for (c = 0; c < numChannels; c++) { |
| prevChannelGain[c] = hGainDec->channelGain[c]; |
| } |
| |
| _setChannelGains(hGainDec, numChannels, channelGainDb); |
| |
| if (!hGainDec->status) { /* overwrite previous channel gains at startup */ |
| for (c = 0; c < numChannels; c++) |
| prevChannelGain[c] = hGainDec->channelGain[c]; |
| hGainDec->status = 1; |
| } |
| |
| for (c = 0; c < numChannels; c++) { |
| INT n_min = fMin(fMin(CntLeadingZeros(prevChannelGain[c]), |
| CntLeadingZeros(hGainDec->channelGain[c])) - |
| 1, |
| 9); |
| FIXP_DBL gain = prevChannelGain[c] << n_min; |
| FIXP_DBL stepsize = ((hGainDec->channelGain[c] << n_min) - gain); |
| if (stepsize != (FIXP_DBL)0) { |
| if (frameSize == 1024) |
| stepsize = stepsize >> 10; |
| else |
| stepsize = (LONG)stepsize / frameSize; |
| } |
| n_min = 9 - n_min; |
| #ifdef FUNCTION_drcDec_GainDecoder_SetChannelGains_func1 |
| drcDec_GainDecoder_SetChannelGains_func1(audioBuffer, gain, stepsize, |
| n_min, frameSize); |
| #else |
| for (i = 0; i < frameSize; i++) { |
| audioBuffer[i] = fMultDiv2(audioBuffer[i], gain) << n_min; |
| gain += stepsize; |
| } |
| #endif |
| audioBuffer += audioBufferChannelOffset; |
| } |
| } |
| } |
| |
| 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 err = DE_OK; |
| int a; |
| |
| if (!hGainDec->timeDomainSupported) { |
| return DE_NOT_OK; |
| } |
| |
| for (a = 0; a < hGainDec->nActiveDrcs; a++) { |
| if (!_fitsLocation(hGainDec->activeDrc[a].pInst, drcLocation)) continue; |
| |
| /* Apply DRC */ |
| err = processDrcTime(hGainDec, a, delaySamples, channelOffset, |
| drcChannelOffset, numChannelsProcessed, |
| timeDataChannelOffset, audioIOBuffer); |
| if (err) return err; |
| } |
| |
| return err; |
| } |
| |
| DRC_ERROR |
| drcDec_GainDecoder_ProcessSubbandDomain( |
| HANDLE_DRC_GAIN_DECODER hGainDec, const int delaySamples, |
| const GAIN_DEC_LOCATION drcLocation, const int channelOffset, |
| const int drcChannelOffset, const int numChannelsProcessed, |
| const int processSingleTimeslot, FIXP_DBL* audioIOBufferReal[], |
| FIXP_DBL* audioIOBufferImag[]) { |
| DRC_ERROR err = DE_OK; |
| int a; |
| |
| if (hGainDec->subbandDomainSupported == SDM_OFF) { |
| return DE_NOT_OK; |
| } |
| |
| for (a = 0; a < hGainDec->nActiveDrcs; a++) { |
| if (!_fitsLocation(hGainDec->activeDrc[a].pInst, drcLocation)) continue; |
| |
| /* Apply DRC */ |
| err = processDrcSubband(hGainDec, a, delaySamples, channelOffset, |
| drcChannelOffset, numChannelsProcessed, |
| processSingleTimeslot, audioIOBufferReal, |
| audioIOBufferImag); |
| if (err) return err; |
| } |
| |
| return err; |
| } |
| |
| DRC_ERROR |
| drcDec_GainDecoder_SetLoudnessNormalizationGainDb( |
| HANDLE_DRC_GAIN_DECODER hGainDec, FIXP_DBL loudnessNormalizationGainDb) { |
| hGainDec->loudnessNormalisationGainDb = loudnessNormalizationGainDb; |
| |
| return DE_OK; |
| } |
| |
| int drcDec_GainDecoder_GetFrameSize(HANDLE_DRC_GAIN_DECODER hGainDec) { |
| if (hGainDec == NULL) return -1; |
| |
| return hGainDec->frameSize; |
| } |
| |
| int drcDec_GainDecoder_GetDeltaTminDefault(HANDLE_DRC_GAIN_DECODER hGainDec) { |
| if (hGainDec == NULL) return -1; |
| |
| return hGainDec->deltaTminDefault; |
| } |