diff options
| -rw-r--r-- | libAACdec/src/aac_ram.cpp | 4 | ||||
| -rw-r--r-- | libAACdec/src/aac_ram.h | 4 | ||||
| -rw-r--r-- | libAACdec/src/aacdecoder.cpp | 23 | ||||
| -rw-r--r-- | libAACdec/src/aacdecoder.h | 18 | ||||
| -rw-r--r-- | libAACdec/src/aacdecoder_lib.cpp | 248 |
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 */ |