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Diffstat (limited to 'fdk-aac/libDRCdec/src/drcDec_gainDecoder.cpp')
| -rw-r--r-- | fdk-aac/libDRCdec/src/drcDec_gainDecoder.cpp | 445 |
1 files changed, 445 insertions, 0 deletions
diff --git a/fdk-aac/libDRCdec/src/drcDec_gainDecoder.cpp b/fdk-aac/libDRCdec/src/drcDec_gainDecoder.cpp new file mode 100644 index 0000000..ca81fad --- /dev/null +++ b/fdk-aac/libDRCdec/src/drcDec_gainDecoder.cpp @@ -0,0 +1,445 @@ +/* ----------------------------------------------------------------------------- +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; +} |