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Fix USAC time domain limiter latency at config change. 

We have observed quality problems regarding config switching for USAC streams. Crossfading did not consider the USAC time domain limiter latency correctly. The limiter memory still contained the last part of the frame before the config change.
With this patch we were able to improve the quality by moving the limiter processing to the end of the processing chain (crossfade -> DRC -> limiter). By that we don't have to consider the limiter latency at the crossfader anymore and can resolve the quality issue.
Bug: 176246647
Test: atest android.media.cts.DecoderTestAacFormat android.media.cts.DecoderTestXheAac android.media.cts.DecoderTestAacDrc
Change-Id: I0dfd3b76ff2b0daf495ad406283f56a39982ad8f

Change-Id: I26f5da65ef8344602007e180e837820c6a25f173
This commit is contained in:
Fraunhofer IIS FDK 2020-04-17 15:07:13 +02:00 committed by Ray Essick
parent 82f6f3dac6
commit 09e7c40a3a
5 changed files with 156 additions and 141 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

4
libAACdec/src/aac_ram.cpp
View File

@ -1,7 +1,7 @@
/* ----------------------------------------------------------------------------- /* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android Software License for The Fraunhofer FDK AAC Codec Library for Android
© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten © Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved. Forschung e.V. All rights reserved.
1. INTRODUCTION 1. INTRODUCTION
@ -148,7 +148,7 @@ C_ALLOC_MEM(CplxPredictionData, CCplxPredictionData, 1)
/*! The buffer holds time samples for the crossfade in case of an USAC DASH IPF /*! The buffer holds time samples for the crossfade in case of an USAC DASH IPF
config change Dimension: (8) config change Dimension: (8)
*/ */
C_ALLOC_MEM2(TimeDataFlush, INT_PCM, TIME_DATA_FLUSH_SIZE, (8)) C_ALLOC_MEM2(TimeDataFlush, PCM_DEC, TIME_DATA_FLUSH_SIZE, (8))
/* @} */ /* @} */

4
libAACdec/src/aac_ram.h
View File

@ -1,7 +1,7 @@
/* ----------------------------------------------------------------------------- /* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android Software License for The Fraunhofer FDK AAC Codec Library for Android
© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten © Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved. Forschung e.V. All rights reserved.
1. INTRODUCTION 1. INTRODUCTION
@ -132,7 +132,7 @@ H_ALLOC_MEM(CplxPredictionData, CCplxPredictionData)
H_ALLOC_MEM(SpectralCoeffs, FIXP_DBL) H_ALLOC_MEM(SpectralCoeffs, FIXP_DBL)
H_ALLOC_MEM(SpecScale, SHORT) H_ALLOC_MEM(SpecScale, SHORT)
H_ALLOC_MEM(TimeDataFlush, INT_PCM) H_ALLOC_MEM(TimeDataFlush, PCM_DEC)
H_ALLOC_MEM_OVERLAY(WorkBufferCore1, CWorkBufferCore1) H_ALLOC_MEM_OVERLAY(WorkBufferCore1, CWorkBufferCore1)
H_ALLOC_MEM_OVERLAY(WorkBufferCore2, FIXP_DBL) H_ALLOC_MEM_OVERLAY(WorkBufferCore2, FIXP_DBL)

23
libAACdec/src/aacdecoder.cpp
View File

@ -568,7 +568,7 @@ static int CProgramConfigElement_Read(HANDLE_FDK_BITSTREAM bs,
\return Error code \return Error code
*/ */
LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_PrepareCrossFade( LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_PrepareCrossFade(
const INT_PCM *pTimeData, INT_PCM **pTimeDataFlush, const INT numChannels, const PCM_DEC *pTimeData, PCM_DEC **pTimeDataFlush, const INT numChannels,
const INT frameSize, const INT interleaved) { const INT frameSize, const INT interleaved) {
int i, ch, s1, s2; int i, ch, s1, s2;
AAC_DECODER_ERROR ErrorStatus; AAC_DECODER_ERROR ErrorStatus;
@ -584,7 +584,7 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_PrepareCrossFade(
} }
for (ch = 0; ch < numChannels; ch++) { for (ch = 0; ch < numChannels; ch++) {
const INT_PCM *pIn = &pTimeData[ch * s1]; const PCM_DEC *pIn = &pTimeData[ch * s1];
for (i = 0; i < TIME_DATA_FLUSH_SIZE; i++) { for (i = 0; i < TIME_DATA_FLUSH_SIZE; i++) {
pTimeDataFlush[ch][i] = *pIn; pTimeDataFlush[ch][i] = *pIn;
pIn += s2; pIn += s2;
@ -606,7 +606,7 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_PrepareCrossFade(
\return Error code \return Error code
*/ */
LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_ApplyCrossFade( LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_ApplyCrossFade(
INT_PCM *pTimeData, INT_PCM **pTimeDataFlush, const INT numChannels, PCM_DEC *pTimeData, PCM_DEC **pTimeDataFlush, const INT numChannels,
const INT frameSize, const INT interleaved) { const INT frameSize, const INT interleaved) {
int i, ch, s1, s2; int i, ch, s1, s2;
AAC_DECODER_ERROR ErrorStatus; AAC_DECODER_ERROR ErrorStatus;
@ -622,15 +622,15 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_ApplyCrossFade(
} }
for (ch = 0; ch < numChannels; ch++) { for (ch = 0; ch < numChannels; ch++) {
INT_PCM *pIn = &pTimeData[ch * s1]; PCM_DEC *pIn = &pTimeData[ch * s1];
for (i = 0; i < TIME_DATA_FLUSH_SIZE; i++) { for (i = 0; i < TIME_DATA_FLUSH_SIZE; i++) {
FIXP_SGL alpha = (FIXP_SGL)i FIXP_SGL alpha = (FIXP_SGL)i
<< (FRACT_BITS - 1 - TIME_DATA_FLUSH_SIZE_SF); << (FRACT_BITS - 1 - TIME_DATA_FLUSH_SIZE_SF);
FIXP_DBL time = FX_PCM2FX_DBL(*pIn); FIXP_DBL time = PCM_DEC2FIXP_DBL(*pIn);
FIXP_DBL timeFlush = FX_PCM2FX_DBL(pTimeDataFlush[ch][i]); FIXP_DBL timeFlush = PCM_DEC2FIXP_DBL(pTimeDataFlush[ch][i]);
*pIn = (INT_PCM)(FIXP_PCM)FX_DBL2FX_PCM( *pIn = FIXP_DBL2PCM_DEC(timeFlush - fMult(timeFlush, alpha) +
timeFlush - fMult(timeFlush, alpha) + fMult(time, alpha)); fMult(time, alpha));
pIn += s2; pIn += s2;
} }
} }
@ -753,7 +753,12 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_PreRollExtensionPayloadParse(
/* We are interested in preroll AUs if an explicit or an implicit config /* We are interested in preroll AUs if an explicit or an implicit config
* change is signalized in other words if the build up status is set. */ * change is signalized in other words if the build up status is set. */
if (self->buildUpStatus == AACDEC_USAC_BUILD_UP_ON) { if (self->buildUpStatus == AACDEC_USAC_BUILD_UP_ON) {
self->applyCrossfade |= FDKreadBit(hBs); UCHAR applyCrossfade = FDKreadBit(hBs);
if (applyCrossfade) {
self->applyCrossfade |= AACDEC_CROSSFADE_BITMASK_PREROLL;
} else {
self->applyCrossfade &= ~AACDEC_CROSSFADE_BITMASK_PREROLL;
}
FDKreadBit(hBs); /* reserved */ FDKreadBit(hBs); /* reserved */
/* Read num preroll AU's */ /* Read num preroll AU's */
*numPrerollAU = escapedValue(hBs, 2, 4, 0); *numPrerollAU = escapedValue(hBs, 2, 4, 0);

18
libAACdec/src/aacdecoder.h
View File

@ -1,7 +1,7 @@
/* ----------------------------------------------------------------------------- /* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android Software License for The Fraunhofer FDK AAC Codec Library for Android
© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten © Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved. Forschung e.V. All rights reserved.
1. INTRODUCTION 1. INTRODUCTION
@ -172,6 +172,12 @@ enum {
AACDEC_RSV60_BUILD_UP_IDLE_IN_BAND = 5 AACDEC_RSV60_BUILD_UP_IDLE_IN_BAND = 5
}; };
#define AACDEC_CROSSFADE_BITMASK_OFF \
((UCHAR)0) /*!< No cross-fade between frames shall be applied at next \
config change. */
#define AACDEC_CROSSFADE_BITMASK_PREROLL \
((UCHAR)1 << 1) /*!< applyCrossfade is signaled in AudioPreRoll */
typedef struct { typedef struct {
/* Usac Extension Elements */ /* Usac Extension Elements */
USAC_EXT_ELEMENT_TYPE usacExtElementType[(3)]; USAC_EXT_ELEMENT_TYPE usacExtElementType[(3)];
@ -325,7 +331,7 @@ This structure is allocated once for each CPE. */
UINT loudnessInfoSetPosition[3]; UINT loudnessInfoSetPosition[3];
SCHAR defaultTargetLoudness; SCHAR defaultTargetLoudness;
INT_PCM PCM_DEC
*pTimeDataFlush[((8) * 2)]; /*!< Pointer to the flushed time data which *pTimeDataFlush[((8) * 2)]; /*!< Pointer to the flushed time data which
will be used for the crossfade in case of will be used for the crossfade in case of
an USAC DASH IPF config change */ an USAC DASH IPF config change */
@ -341,8 +347,8 @@ This structure is allocated once for each CPE. */
start position in the start position in the
bitstream */ bitstream */
INT accessUnit; /*!< Number of the actual processed preroll accessUnit */ INT accessUnit; /*!< Number of the actual processed preroll accessUnit */
UCHAR applyCrossfade; /*!< if set crossfade for seamless stream switching is UCHAR applyCrossfade; /*!< If any bit is set, cross-fade for seamless stream
applied */ switching is applied */
FDK_SignalDelay usacResidualDelay; /*!< Delay residual signal to compensate FDK_SignalDelay usacResidualDelay; /*!< Delay residual signal to compensate
for eSBR delay of DMX signal in case of for eSBR delay of DMX signal in case of
@ -439,12 +445,12 @@ LINKSPEC_H AAC_DECODER_ERROR CAacDecoder_FreeMem(HANDLE_AACDECODER self,
/* Prepare crossfade for USAC DASH IPF config change */ /* Prepare crossfade for USAC DASH IPF config change */
LINKSPEC_H AAC_DECODER_ERROR CAacDecoder_PrepareCrossFade( LINKSPEC_H AAC_DECODER_ERROR CAacDecoder_PrepareCrossFade(
const INT_PCM *pTimeData, INT_PCM **pTimeDataFlush, const INT numChannels, const PCM_DEC *pTimeData, PCM_DEC **pTimeDataFlush, const INT numChannels,
const INT frameSize, const INT interleaved); const INT frameSize, const INT interleaved);
/* Apply crossfade for USAC DASH IPF config change */ /* Apply crossfade for USAC DASH IPF config change */
LINKSPEC_H AAC_DECODER_ERROR CAacDecoder_ApplyCrossFade( LINKSPEC_H AAC_DECODER_ERROR CAacDecoder_ApplyCrossFade(
INT_PCM *pTimeData, INT_PCM **pTimeDataFlush, const INT numChannels, PCM_DEC *pTimeData, PCM_DEC **pTimeDataFlush, const INT numChannels,
const INT frameSize, const INT interleaved); const INT frameSize, const INT interleaved);
/* Set flush and build up mode */ /* Set flush and build up mode */

248
libAACdec/src/aacdecoder_lib.cpp
View File

@ -1155,6 +1155,8 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame(HANDLE_AACDECODER self,
int applyCrossfade = 1; /* flag indicates if flushing was possible */ int applyCrossfade = 1; /* flag indicates if flushing was possible */
PCM_DEC *pTimeData2; PCM_DEC *pTimeData2;
PCM_AAC *pTimeData3; PCM_AAC *pTimeData3;
INT pcmLimiterScale = 0;
INT interleaved = 0;
if (self == NULL) { if (self == NULL) {
return AAC_DEC_INVALID_HANDLE; return AAC_DEC_INVALID_HANDLE;
@ -1800,8 +1802,7 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame(HANDLE_AACDECODER self,
} }
if (self->streamInfo.extAot != AOT_AAC_SLS) { if (self->streamInfo.extAot != AOT_AAC_SLS) {
INT pcmLimiterScale = 0; interleaved = 0;
INT interleaved = 0;
interleaved |= (self->sbrEnabled) ? 1 : 0; interleaved |= (self->sbrEnabled) ? 1 : 0;
interleaved |= (self->mpsEnableCurr) ? 1 : 0; interleaved |= (self->mpsEnableCurr) ? 1 : 0;
PCMDMX_ERROR dmxErr = PCMDMX_OK; PCMDMX_ERROR dmxErr = PCMDMX_OK;
@ -1832,145 +1833,38 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame(HANDLE_AACDECODER self,
* predictable behavior and thus maybe produce strange output. */ * predictable behavior and thus maybe produce strange output. */
ErrorStatus = AAC_DEC_DECODE_FRAME_ERROR; ErrorStatus = AAC_DEC_DECODE_FRAME_ERROR;
} }
}
pcmLimiterScale += PCM_OUT_HEADROOM;
if (flags & AACDEC_CLRHIST) {
if (!(self->flags[0] & AC_USAC)) {
/* Reset DRC data */
aacDecoder_drcReset(self->hDrcInfo);
/* Delete the delayed signal. */
pcmLimiter_Reset(self->hLimiter);
}
}
/* Set applyExtGain if DRC processing is enabled and if
progRefLevelPresent is present for the first time. Consequences: The
headroom of the output signal can be set to AACDEC_DRC_GAIN_SCALING
only for audio formats which support legacy DRC Level Normalization.
For all other audio formats the headroom of the output
signal is set to PCM_OUT_HEADROOM. */
if (self->hDrcInfo->enable &&
(self->hDrcInfo->progRefLevelPresent == 1)) {
self->hDrcInfo->applyExtGain |= 1;
}
/* Check whether time data buffer is large enough. */
if (timeDataSize <
(self->streamInfo.numChannels * self->streamInfo.frameSize)) {
ErrorStatus = AAC_DEC_OUTPUT_BUFFER_TOO_SMALL;
goto bail;
}
if (self->limiterEnableCurr) {
/* use workBufferCore2 buffer for interleaving */
PCM_LIM *pInterleaveBuffer;
int blockLength = self->streamInfo.frameSize;
/* Set actual signal parameters */
pcmLimiter_SetNChannels(self->hLimiter, self->streamInfo.numChannels);
pcmLimiter_SetSampleRate(self->hLimiter, self->streamInfo.sampleRate);
if ((self->streamInfo.numChannels == 1) || (self->sbrEnabled) ||
(self->mpsEnableCurr)) {
pInterleaveBuffer = (PCM_LIM *)pTimeData2;
} else {
pInterleaveBuffer = (PCM_LIM *)self->workBufferCore2;
/* applyLimiter requests for interleaved data */
/* Interleave ouput buffer */
FDK_interleave(pTimeData2, pInterleaveBuffer,
self->streamInfo.numChannels, blockLength,
self->streamInfo.frameSize);
}
FIXP_DBL *pGainPerSample = NULL;
if (self->hDrcInfo->enable && self->hDrcInfo->applyExtGain) {
pGainPerSample = self->workBufferCore1;
if ((INT)GetRequiredMemWorkBufferCore1() <
(INT)(self->streamInfo.frameSize * sizeof(FIXP_DBL))) {
ErrorStatus = AAC_DEC_UNKNOWN;
goto bail;
}
pcmLimiterScale = applyDrcLevelNormalization(
self->hDrcInfo, (PCM_DEC *)pInterleaveBuffer, self->extGain,
pGainPerSample, pcmLimiterScale, self->extGainDelay,
self->streamInfo.frameSize, self->streamInfo.numChannels, 1, 1);
}
pcmLimiter_Apply(self->hLimiter, pInterleaveBuffer, pTimeData,
pGainPerSample, pcmLimiterScale,
self->streamInfo.frameSize);
{
/* Announce the additional limiter output delay */
self->streamInfo.outputDelay += pcmLimiter_GetDelay(self->hLimiter);
}
} else {
if (self->hDrcInfo->enable && self->hDrcInfo->applyExtGain) {
pcmLimiterScale = applyDrcLevelNormalization(
self->hDrcInfo, pTimeData2, self->extGain, NULL,
pcmLimiterScale, self->extGainDelay, self->streamInfo.frameSize,
self->streamInfo.numChannels,
(interleaved || (self->streamInfo.numChannels == 1))
? 1
: self->streamInfo.frameSize,
0);
}
/* If numChannels = 1 we do not need interleaving. The same applies if
SBR or MPS are used, since their output is interleaved already
(resampled or not) */
if ((self->streamInfo.numChannels == 1) || (self->sbrEnabled) ||
(self->mpsEnableCurr)) {
scaleValuesSaturate(
pTimeData, pTimeData2,
self->streamInfo.frameSize * self->streamInfo.numChannels,
pcmLimiterScale);
} else {
scaleValuesSaturate(
(INT_PCM *)self->workBufferCore2, pTimeData2,
self->streamInfo.frameSize * self->streamInfo.numChannels,
pcmLimiterScale);
/* Interleave ouput buffer */
FDK_interleave((INT_PCM *)self->workBufferCore2, pTimeData,
self->streamInfo.numChannels,
self->streamInfo.frameSize,
self->streamInfo.frameSize);
}
}
} /* if (self->streamInfo.extAot != AOT_AAC_SLS)*/
if (self->flags[0] & AC_USAC) { if (self->flags[0] & AC_USAC) {
if (self->flushStatus == AACDEC_USAC_DASH_IPF_FLUSH_ON && if (self->flushStatus == AACDEC_USAC_DASH_IPF_FLUSH_ON &&
!(flags & AACDEC_CONCEAL)) { !(flags & AACDEC_CONCEAL)) {
CAacDecoder_PrepareCrossFade(pTimeData, self->pTimeDataFlush, CAacDecoder_PrepareCrossFade(pTimeData2, self->pTimeDataFlush,
self->streamInfo.numChannels, self->streamInfo.numChannels,
self->streamInfo.frameSize, 1); self->streamInfo.frameSize, interleaved);
} }
/* prepare crossfade buffer for fade in */ /* prepare crossfade buffer for fade in */
if (!applyCrossfade && self->applyCrossfade && if (!applyCrossfade &&
(self->applyCrossfade != AACDEC_CROSSFADE_BITMASK_OFF) &&
!(flags & AACDEC_CONCEAL)) { !(flags & AACDEC_CONCEAL)) {
for (int ch = 0; ch < self->streamInfo.numChannels; ch++) { for (int ch = 0; ch < self->streamInfo.numChannels; ch++) {
for (int i = 0; i < TIME_DATA_FLUSH_SIZE; i++) { for (int i = 0; i < TIME_DATA_FLUSH_SIZE; i++) {
self->pTimeDataFlush[ch][i] = 0; self->pTimeDataFlush[ch][i] = (PCM_DEC)0;
} }
} }
applyCrossfade = 1; applyCrossfade = 1;
} }
if (applyCrossfade && self->applyCrossfade && if (applyCrossfade &&
(self->applyCrossfade != AACDEC_CROSSFADE_BITMASK_OFF) &&
!(accessUnit < numPrerollAU) && !(accessUnit < numPrerollAU) &&
(self->buildUpStatus == AACDEC_USAC_BUILD_UP_ON)) { (self->buildUpStatus == AACDEC_USAC_BUILD_UP_ON)) {
CAacDecoder_ApplyCrossFade(pTimeData, self->pTimeDataFlush, CAacDecoder_ApplyCrossFade(pTimeData2, self->pTimeDataFlush,
self->streamInfo.numChannels, self->streamInfo.numChannels,
self->streamInfo.frameSize, 1); self->streamInfo.frameSize, interleaved);
self->applyCrossfade = 0; self->applyCrossfade =
AACDEC_CROSSFADE_BITMASK_OFF; /* disable cross-fade between frames
at nect config change */
} }
} }
@ -2012,6 +1906,116 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame(HANDLE_AACDECODER self,
((self->flushStatus == AACDEC_USAC_DASH_IPF_FLUSH_ON) && ((self->flushStatus == AACDEC_USAC_DASH_IPF_FLUSH_ON) &&
!(flags & AACDEC_CONCEAL))); !(flags & AACDEC_CONCEAL)));
if (self->streamInfo.extAot != AOT_AAC_SLS) {
pcmLimiterScale += PCM_OUT_HEADROOM;
if (flags & AACDEC_CLRHIST) {
if (!(self->flags[0] & AC_USAC)) {
/* Reset DRC data */
aacDecoder_drcReset(self->hDrcInfo);
/* Delete the delayed signal. */
pcmLimiter_Reset(self->hLimiter);
}
}
/* Set applyExtGain if DRC processing is enabled and if progRefLevelPresent
is present for the first time. Consequences: The headroom of the output
signal can be set to AACDEC_DRC_GAIN_SCALING only for audio formats which
support legacy DRC Level Normalization. For all other audio formats the
headroom of the output signal is set to PCM_OUT_HEADROOM. */
if (self->hDrcInfo->enable && (self->hDrcInfo->progRefLevelPresent == 1)) {
self->hDrcInfo->applyExtGain |= 1;
}
/* Check whether time data buffer is large enough. */
if (timeDataSize <
(self->streamInfo.numChannels * self->streamInfo.frameSize)) {
ErrorStatus = AAC_DEC_OUTPUT_BUFFER_TOO_SMALL;
goto bail;
}
if (self->limiterEnableCurr) {
/* use workBufferCore2 buffer for interleaving */
PCM_LIM *pInterleaveBuffer;
int blockLength = self->streamInfo.frameSize;
/* Set actual signal parameters */
pcmLimiter_SetNChannels(self->hLimiter, self->streamInfo.numChannels);
pcmLimiter_SetSampleRate(self->hLimiter, self->streamInfo.sampleRate);
if ((self->streamInfo.numChannels == 1) || (self->sbrEnabled) ||
(self->mpsEnableCurr)) {
pInterleaveBuffer = (PCM_LIM *)pTimeData2;
} else {
pInterleaveBuffer = (PCM_LIM *)self->workBufferCore2;
/* applyLimiter requests for interleaved data */
/* Interleave ouput buffer */
FDK_interleave(pTimeData2, pInterleaveBuffer,
self->streamInfo.numChannels, blockLength,
self->streamInfo.frameSize);
}
FIXP_DBL *pGainPerSample = NULL;
if (self->hDrcInfo->enable && self->hDrcInfo->applyExtGain) {
pGainPerSample = self->workBufferCore1;
if ((INT)GetRequiredMemWorkBufferCore1() <
(INT)(self->streamInfo.frameSize * sizeof(FIXP_DBL))) {
ErrorStatus = AAC_DEC_UNKNOWN;
goto bail;
}
pcmLimiterScale = applyDrcLevelNormalization(
self->hDrcInfo, (PCM_DEC *)pInterleaveBuffer, self->extGain,
pGainPerSample, pcmLimiterScale, self->extGainDelay,
self->streamInfo.frameSize, self->streamInfo.numChannels, 1, 1);
}
pcmLimiter_Apply(self->hLimiter, pInterleaveBuffer, pTimeData,
pGainPerSample, pcmLimiterScale,
self->streamInfo.frameSize);
{
/* Announce the additional limiter output delay */
self->streamInfo.outputDelay += pcmLimiter_GetDelay(self->hLimiter);
}
} else {
if (self->hDrcInfo->enable && self->hDrcInfo->applyExtGain) {
pcmLimiterScale = applyDrcLevelNormalization(
self->hDrcInfo, pTimeData2, self->extGain, NULL, pcmLimiterScale,
self->extGainDelay, self->streamInfo.frameSize,
self->streamInfo.numChannels,
(interleaved || (self->streamInfo.numChannels == 1))
? 1
: self->streamInfo.frameSize,
0);
}
/* If numChannels = 1 we do not need interleaving. The same applies if SBR
or MPS are used, since their output is interleaved already (resampled or
not) */
if ((self->streamInfo.numChannels == 1) || (self->sbrEnabled) ||
(self->mpsEnableCurr)) {
scaleValuesSaturate(
pTimeData, pTimeData2,
self->streamInfo.frameSize * self->streamInfo.numChannels,
pcmLimiterScale);
} else {
scaleValuesSaturate(
(INT_PCM *)self->workBufferCore2, pTimeData2,
self->streamInfo.frameSize * self->streamInfo.numChannels,
pcmLimiterScale);
/* Interleave ouput buffer */
FDK_interleave((INT_PCM *)self->workBufferCore2, pTimeData,
self->streamInfo.numChannels, self->streamInfo.frameSize,
self->streamInfo.frameSize);
}
}
} /* if (self->streamInfo.extAot != AOT_AAC_SLS)*/
bail: bail:
/* error in renderer part occurred, ErrorStatus was set to invalid output */ /* error in renderer part occurred, ErrorStatus was set to invalid output */