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-rw-r--r--libAACdec/src/aac_ram.cpp4
-rw-r--r--libAACdec/src/aac_ram.h4
-rw-r--r--libAACdec/src/aacdecoder.cpp23
-rw-r--r--libAACdec/src/aacdecoder.h18
-rw-r--r--libAACdec/src/aacdecoder_lib.cpp248
5 files changed, 156 insertions, 141 deletions
diff --git a/libAACdec/src/aac_ram.cpp b/libAACdec/src/aac_ram.cpp
index aa8f6a6..fac1540 100644
--- a/libAACdec/src/aac_ram.cpp
+++ b/libAACdec/src/aac_ram.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
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.
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
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))
/* @} */
diff --git a/libAACdec/src/aac_ram.h b/libAACdec/src/aac_ram.h
index b9b95b7..395b2b2 100644
--- a/libAACdec/src/aac_ram.h
+++ b/libAACdec/src/aac_ram.h
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
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.
1. INTRODUCTION
@@ -132,7 +132,7 @@ H_ALLOC_MEM(CplxPredictionData, CCplxPredictionData)
H_ALLOC_MEM(SpectralCoeffs, FIXP_DBL)
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(WorkBufferCore2, FIXP_DBL)
diff --git a/libAACdec/src/aacdecoder.cpp b/libAACdec/src/aacdecoder.cpp
index c6d1832..c18e5e9 100644
--- a/libAACdec/src/aacdecoder.cpp
+++ b/libAACdec/src/aacdecoder.cpp
@@ -568,7 +568,7 @@ static int CProgramConfigElement_Read(HANDLE_FDK_BITSTREAM bs,
\return Error code
*/
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) {
int i, ch, s1, s2;
AAC_DECODER_ERROR ErrorStatus;
@@ -584,7 +584,7 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_PrepareCrossFade(
}
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++) {
pTimeDataFlush[ch][i] = *pIn;
pIn += s2;
@@ -606,7 +606,7 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_PrepareCrossFade(
\return Error code
*/
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) {
int i, ch, s1, s2;
AAC_DECODER_ERROR ErrorStatus;
@@ -622,15 +622,15 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_ApplyCrossFade(
}
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++) {
FIXP_SGL alpha = (FIXP_SGL)i
<< (FRACT_BITS - 1 - TIME_DATA_FLUSH_SIZE_SF);
- FIXP_DBL time = FX_PCM2FX_DBL(*pIn);
- FIXP_DBL timeFlush = FX_PCM2FX_DBL(pTimeDataFlush[ch][i]);
+ FIXP_DBL time = PCM_DEC2FIXP_DBL(*pIn);
+ FIXP_DBL timeFlush = PCM_DEC2FIXP_DBL(pTimeDataFlush[ch][i]);
- *pIn = (INT_PCM)(FIXP_PCM)FX_DBL2FX_PCM(
- timeFlush - fMult(timeFlush, alpha) + fMult(time, alpha));
+ *pIn = FIXP_DBL2PCM_DEC(timeFlush - fMult(timeFlush, alpha) +
+ fMult(time, alpha));
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
* change is signalized in other words if the build up status is set. */
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 */
/* Read num preroll AU's */
*numPrerollAU = escapedValue(hBs, 2, 4, 0);
diff --git a/libAACdec/src/aacdecoder.h b/libAACdec/src/aacdecoder.h
index bd1f38f..002807f 100644
--- a/libAACdec/src/aacdecoder.h
+++ b/libAACdec/src/aacdecoder.h
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
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.
1. INTRODUCTION
@@ -172,6 +172,12 @@ enum {
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 {
/* Usac Extension Elements */
USAC_EXT_ELEMENT_TYPE usacExtElementType[(3)];
@@ -325,7 +331,7 @@ This structure is allocated once for each CPE. */
UINT loudnessInfoSetPosition[3];
SCHAR defaultTargetLoudness;
- INT_PCM
+ PCM_DEC
*pTimeDataFlush[((8) * 2)]; /*!< Pointer to the flushed time data which
will be used for the crossfade in case of
an USAC DASH IPF config change */
@@ -341,8 +347,8 @@ This structure is allocated once for each CPE. */
start position in the
bitstream */
INT accessUnit; /*!< Number of the actual processed preroll accessUnit */
- UCHAR applyCrossfade; /*!< if set crossfade for seamless stream switching is
- applied */
+ UCHAR applyCrossfade; /*!< If any bit is set, cross-fade for seamless stream
+ switching is applied */
FDK_SignalDelay usacResidualDelay; /*!< Delay residual signal to compensate
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 */
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);
/* Apply crossfade for USAC DASH IPF config change */
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);
/* Set flush and build up mode */
diff --git a/libAACdec/src/aacdecoder_lib.cpp b/libAACdec/src/aacdecoder_lib.cpp
index 6fb7bf1..5efa369 100644
--- a/libAACdec/src/aacdecoder_lib.cpp
+++ b/libAACdec/src/aacdecoder_lib.cpp
@@ -1155,6 +1155,8 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame(HANDLE_AACDECODER self,
int applyCrossfade = 1; /* flag indicates if flushing was possible */
PCM_DEC *pTimeData2;
PCM_AAC *pTimeData3;
+ INT pcmLimiterScale = 0;
+ INT interleaved = 0;
if (self == NULL) {
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) {
- INT pcmLimiterScale = 0;
- INT interleaved = 0;
+ interleaved = 0;
interleaved |= (self->sbrEnabled) ? 1 : 0;
interleaved |= (self->mpsEnableCurr) ? 1 : 0;
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. */
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->flushStatus == AACDEC_USAC_DASH_IPF_FLUSH_ON &&
!(flags & AACDEC_CONCEAL)) {
- CAacDecoder_PrepareCrossFade(pTimeData, self->pTimeDataFlush,
+ CAacDecoder_PrepareCrossFade(pTimeData2, self->pTimeDataFlush,
self->streamInfo.numChannels,
- self->streamInfo.frameSize, 1);
+ self->streamInfo.frameSize, interleaved);
}
/* prepare crossfade buffer for fade in */
- if (!applyCrossfade && self->applyCrossfade &&
+ if (!applyCrossfade &&
+ (self->applyCrossfade != AACDEC_CROSSFADE_BITMASK_OFF) &&
!(flags & AACDEC_CONCEAL)) {
for (int ch = 0; ch < self->streamInfo.numChannels; ch++) {
for (int i = 0; i < TIME_DATA_FLUSH_SIZE; i++) {
- self->pTimeDataFlush[ch][i] = 0;
+ self->pTimeDataFlush[ch][i] = (PCM_DEC)0;
}
}
applyCrossfade = 1;
}
- if (applyCrossfade && self->applyCrossfade &&
+ if (applyCrossfade &&
+ (self->applyCrossfade != AACDEC_CROSSFADE_BITMASK_OFF) &&
!(accessUnit < numPrerollAU) &&
(self->buildUpStatus == AACDEC_USAC_BUILD_UP_ON)) {
- CAacDecoder_ApplyCrossFade(pTimeData, self->pTimeDataFlush,
+ CAacDecoder_ApplyCrossFade(pTimeData2, self->pTimeDataFlush,
self->streamInfo.numChannels,
- self->streamInfo.frameSize, 1);
- self->applyCrossfade = 0;
+ self->streamInfo.frameSize, interleaved);
+ 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) &&
!(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:
/* error in renderer part occurred, ErrorStatus was set to invalid output */
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-27 09:41:07 +0000