diff options
26 files changed, 725 insertions, 354 deletions
diff --git a/fuzzer/aac_dec_fuzzer.cpp b/fuzzer/aac_dec_fuzzer.cpp index b5545fc..c970197 100644 --- a/fuzzer/aac_dec_fuzzer.cpp +++ b/fuzzer/aac_dec_fuzzer.cpp @@ -118,7 +118,8 @@ void Codec::decodeFrames(UCHAR *data, UINT size) { INT_PCM outputBuf[kMaxOutBufferSize]; do { mErrorCode = - aacDecoder_DecodeFrame(mAacDecoderHandle, outputBuf, sizeof(outputBuf), 0); + aacDecoder_DecodeFrame(mAacDecoderHandle, outputBuf, + kMaxOutBufferSize /*size in number of INT_PCM, not bytes*/, 0); } while (mErrorCode == AAC_DEC_OK); UINT offset = inputSize - valid; data += offset; 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/aacdec_drc.cpp b/libAACdec/src/aacdec_drc.cpp index b6f5b49..760a9ba 100644 --- a/libAACdec/src/aacdec_drc.cpp +++ b/libAACdec/src/aacdec_drc.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 @@ -150,6 +150,19 @@ static INT convert_drcParam(FIXP_DBL param_dbl) { } /*! +\brief Disable DRC + +\self Handle of DRC info + +\return none +*/ +void aacDecoder_drcDisable(HANDLE_AAC_DRC self) { + self->enable = 0; + self->applyExtGain = 0; + self->progRefLevelPresent = 0; +} + +/*! \brief Reset DRC information \self Handle of DRC info diff --git a/libAACdec/src/aacdec_drc.h b/libAACdec/src/aacdec_drc.h index 76a44d6..2bb945d 100644 --- a/libAACdec/src/aacdec_drc.h +++ b/libAACdec/src/aacdec_drc.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 @@ -140,6 +140,8 @@ typedef enum { /** * \brief DRC module interface functions */ +void aacDecoder_drcDisable(HANDLE_AAC_DRC self); + void aacDecoder_drcReset(HANDLE_AAC_DRC self); void aacDecoder_drcInit(HANDLE_AAC_DRC self); diff --git a/libAACdec/src/aacdec_hcrs.cpp b/libAACdec/src/aacdec_hcrs.cpp index 44b32a5..5e3f9ac 100644 --- a/libAACdec/src/aacdec_hcrs.cpp +++ b/libAACdec/src/aacdec_hcrs.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 @@ -173,7 +173,9 @@ void DecodeNonPCWs(HANDLE_FDK_BITSTREAM bs, H_HCR_INFO pHcr) { pHcr->segmentInfo.readDirection = FROM_RIGHT_TO_LEFT; /* Process sets subsequently */ + numSet = fMin(numSet, (UCHAR)MAX_HCR_SETS); for (currentSet = 1; currentSet < numSet; currentSet++) { + /* step 1 */ numCodeword -= *pNumSegment; /* number of remaining non PCWs [for all sets] */ diff --git a/libAACdec/src/aacdecoder.cpp b/libAACdec/src/aacdecoder.cpp index 965631b..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); @@ -1397,6 +1402,31 @@ static void CAacDecoder_DeInit(HANDLE_AACDECODER self, } /*! + * \brief CAacDecoder_AcceptFlags Accept flags and element flags + * + * \param self [o] handle to AACDECODER structure + * \param asc [i] handle to ASC structure + * \param flags [i] flags + * \param elFlags [i] pointer to element flags + * \param streamIndex [i] stream index + * \param elementOffset [i] element offset + * + * \return void + */ +static void CAacDecoder_AcceptFlags(HANDLE_AACDECODER self, + const CSAudioSpecificConfig *asc, + UINT flags, UINT *elFlags, int streamIndex, + int elementOffset) { + { + FDKmemcpy( + self->elFlags, elFlags, + sizeof(*elFlags) * (3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) + 1)); + } + + self->flags[streamIndex] = flags; +} + +/*! * \brief CAacDecoder_CtrlCFGChange Set config change parameters. * * \param self [i] handle to AACDECODER structure @@ -1493,6 +1523,9 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, const int streamIndex = 0; INT flushChannels = 0; + UINT flags; + UINT elFlags[(3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) + 1)]; + if (!self) return AAC_DEC_INVALID_HANDLE; UCHAR downscaleFactor = self->downscaleFactor; @@ -1649,8 +1682,8 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, } /* Set syntax flags */ - self->flags[streamIndex] = 0; - { FDKmemclear(self->elFlags, sizeof(self->elFlags)); } + flags = 0; + { FDKmemclear(elFlags, sizeof(elFlags)); } if ((asc->m_channelConfiguration > 0) || IS_USAC(asc->m_aot)) { if (IS_USAC(asc->m_aot)) { @@ -1700,31 +1733,30 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, } } - self->elFlags[el] |= - (asc->m_sc.m_usacConfig.element[_el].m_noiseFilling) - ? AC_EL_USAC_NOISE - : 0; - self->elFlags[el] |= + elFlags[el] |= (asc->m_sc.m_usacConfig.element[_el].m_noiseFilling) + ? AC_EL_USAC_NOISE + : 0; + elFlags[el] |= (asc->m_sc.m_usacConfig.element[_el].m_stereoConfigIndex > 0) ? AC_EL_USAC_MPS212 : 0; - self->elFlags[el] |= (asc->m_sc.m_usacConfig.element[_el].m_interTes) - ? AC_EL_USAC_ITES - : 0; - self->elFlags[el] |= + elFlags[el] |= (asc->m_sc.m_usacConfig.element[_el].m_interTes) + ? AC_EL_USAC_ITES + : 0; + elFlags[el] |= (asc->m_sc.m_usacConfig.element[_el].m_pvc) ? AC_EL_USAC_PVC : 0; - self->elFlags[el] |= + elFlags[el] |= (asc->m_sc.m_usacConfig.element[_el].usacElementType == ID_USAC_LFE) ? AC_EL_USAC_LFE : 0; - self->elFlags[el] |= + elFlags[el] |= (asc->m_sc.m_usacConfig.element[_el].usacElementType == ID_USAC_LFE) ? AC_EL_LFE : 0; if ((asc->m_sc.m_usacConfig.element[_el].usacElementType == ID_USAC_CPE) && ((self->usacStereoConfigIndex[el] == 0))) { - self->elFlags[el] |= AC_EL_USAC_CP_POSSIBLE; + elFlags[el] |= AC_EL_USAC_CP_POSSIBLE; } } @@ -1791,9 +1823,17 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, downscaleFactorInBS = asc->m_samplingFrequency / asc->m_sc.m_eldSpecificConfig.m_downscaledSamplingFrequency; - if (downscaleFactorInBS == 1 || downscaleFactorInBS == 2 || - downscaleFactorInBS == 3 || downscaleFactorInBS == 4) { + if ((downscaleFactorInBS == 1 || downscaleFactorInBS == 2 || + (downscaleFactorInBS == 3 && + asc->m_sc.m_eldSpecificConfig.m_frameLengthFlag) || + downscaleFactorInBS == 4) && + ((asc->m_samplingFrequency % + asc->m_sc.m_eldSpecificConfig.m_downscaledSamplingFrequency) == + 0)) { downscaleFactor = downscaleFactorInBS; + } else { + downscaleFactorInBS = 1; + downscaleFactor = 1; } } } else { @@ -1838,8 +1878,8 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, if (configMode & AC_CM_ALLOC_MEM) { self->streamInfo.extSamplingRate = asc->m_extensionSamplingFrequency; } - self->flags[streamIndex] |= (asc->m_sbrPresentFlag) ? AC_SBR_PRESENT : 0; - self->flags[streamIndex] |= (asc->m_psPresentFlag) ? AC_PS_PRESENT : 0; + flags |= (asc->m_sbrPresentFlag) ? AC_SBR_PRESENT : 0; + flags |= (asc->m_psPresentFlag) ? AC_PS_PRESENT : 0; if (asc->m_sbrPresentFlag) { self->sbrEnabled = 1; self->sbrEnabledPrev = 1; @@ -1865,51 +1905,47 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, } /* --------- vcb11 ------------ */ - self->flags[streamIndex] |= (asc->m_vcb11Flag) ? AC_ER_VCB11 : 0; + flags |= (asc->m_vcb11Flag) ? AC_ER_VCB11 : 0; /* ---------- rvlc ------------ */ - self->flags[streamIndex] |= (asc->m_rvlcFlag) ? AC_ER_RVLC : 0; + flags |= (asc->m_rvlcFlag) ? AC_ER_RVLC : 0; /* ----------- hcr ------------ */ - self->flags[streamIndex] |= (asc->m_hcrFlag) ? AC_ER_HCR : 0; + flags |= (asc->m_hcrFlag) ? AC_ER_HCR : 0; if (asc->m_aot == AOT_ER_AAC_ELD) { self->mpsEnableCurr = 0; - self->flags[streamIndex] |= AC_ELD; - self->flags[streamIndex] |= - (asc->m_sbrPresentFlag) - ? AC_SBR_PRESENT - : 0; /* Need to set the SBR flag for backward-compatibility - reasons. Even if SBR is not supported. */ - self->flags[streamIndex] |= - (asc->m_sc.m_eldSpecificConfig.m_sbrCrcFlag) ? AC_SBRCRC : 0; - self->flags[streamIndex] |= - (asc->m_sc.m_eldSpecificConfig.m_useLdQmfTimeAlign) ? AC_MPS_PRESENT - : 0; + flags |= AC_ELD; + flags |= (asc->m_sbrPresentFlag) + ? AC_SBR_PRESENT + : 0; /* Need to set the SBR flag for backward-compatibility + reasons. Even if SBR is not supported. */ + flags |= (asc->m_sc.m_eldSpecificConfig.m_sbrCrcFlag) ? AC_SBRCRC : 0; + flags |= (asc->m_sc.m_eldSpecificConfig.m_useLdQmfTimeAlign) + ? AC_MPS_PRESENT + : 0; if (self->mpsApplicable) { self->mpsEnableCurr = asc->m_sc.m_eldSpecificConfig.m_useLdQmfTimeAlign; } } - self->flags[streamIndex] |= (asc->m_aot == AOT_ER_AAC_LD) ? AC_LD : 0; - self->flags[streamIndex] |= (asc->m_epConfig >= 0) ? AC_ER : 0; + flags |= (asc->m_aot == AOT_ER_AAC_LD) ? AC_LD : 0; + flags |= (asc->m_epConfig >= 0) ? AC_ER : 0; if (asc->m_aot == AOT_USAC) { - self->flags[streamIndex] |= AC_USAC; - self->flags[streamIndex] |= - (asc->m_sc.m_usacConfig.element[0].m_stereoConfigIndex > 0) - ? AC_MPS_PRESENT - : 0; + flags |= AC_USAC; + flags |= (asc->m_sc.m_usacConfig.element[0].m_stereoConfigIndex > 0) + ? AC_MPS_PRESENT + : 0; } if (asc->m_aot == AOT_DRM_AAC) { - self->flags[streamIndex] |= AC_DRM | AC_SBRCRC | AC_SCALABLE; + flags |= AC_DRM | AC_SBRCRC | AC_SCALABLE; } if (asc->m_aot == AOT_DRM_SURROUND) { - self->flags[streamIndex] |= - AC_DRM | AC_SBRCRC | AC_SCALABLE | AC_MPS_PRESENT; + flags |= AC_DRM | AC_SBRCRC | AC_SCALABLE | AC_MPS_PRESENT; FDK_ASSERT(!asc->m_psPresentFlag); } if ((asc->m_aot == AOT_AAC_SCAL) || (asc->m_aot == AOT_ER_AAC_SCAL)) { - self->flags[streamIndex] |= AC_SCALABLE; + flags |= AC_SCALABLE; } if ((asc->m_epConfig >= 0) && (asc->m_channelConfiguration <= 0)) { @@ -1960,6 +1996,10 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, if (ascChanged != 0) { *configChanged = 1; } + + CAacDecoder_AcceptFlags(self, asc, flags, elFlags, streamIndex, + elementOffset); + return err; } @@ -1988,7 +2028,7 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, } if (usacStereoConfigIndex == 3) { - self->flags[streamIndex] |= AC_USAC_SCFGI3; + flags |= AC_USAC_SCFGI3; } } break; @@ -2069,14 +2109,14 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, ch = aacChannelsOffset; int _numElements; _numElements = (((8)) + (8)); - if (self->flags[streamIndex] & (AC_RSV603DA | AC_USAC)) { + if (flags & (AC_RSV603DA | AC_USAC)) { _numElements = (int)asc->m_sc.m_usacConfig.m_usacNumElements; } for (int _el = 0; _el < _numElements; _el++) { int el_channels = 0; int el = elementOffset + _el; - if (self->flags[streamIndex] & + if (flags & (AC_ER | AC_LD | AC_ELD | AC_RSV603DA | AC_USAC | AC_RSVD50)) { if (ch >= ascChannels) { break; @@ -2176,15 +2216,14 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, if (self->pAacDecoderStaticChannelInfo[ch]->pOverlapBuffer == NULL) { goto bail; } - if (self->flags[streamIndex] & - (AC_USAC | AC_RSVD50 | AC_RSV603DA /*|AC_BSAC*/)) { + if (flags & (AC_USAC | AC_RSVD50 | AC_RSV603DA /*|AC_BSAC*/)) { self->pAacDecoderStaticChannelInfo[ch]->hArCo = CArco_Create(); if (self->pAacDecoderStaticChannelInfo[ch]->hArCo == NULL) { goto bail; } } - if (!(self->flags[streamIndex] & (AC_USAC | AC_RSV603DA))) { + if (!(flags & (AC_USAC | AC_RSV603DA))) { CPns_UpdateNoiseState( &self->pAacDecoderChannelInfo[ch]->data.aac.PnsData, &self->pAacDecoderStaticChannelInfo[ch]->pnsCurrentSeed, @@ -2195,7 +2234,7 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, chIdx++; } - if (self->flags[streamIndex] & AC_USAC) { + if (flags & AC_USAC) { for (int _ch = 0; _ch < flushChannels; _ch++) { ch = aacChannelsOffset + _ch; if (self->pTimeDataFlush[ch] == NULL) { @@ -2207,7 +2246,7 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, } } - if (self->flags[streamIndex] & (AC_USAC | AC_RSV603DA)) { + if (flags & (AC_USAC | AC_RSV603DA)) { int complexStereoPredPossible = 0; ch = aacChannelsOffset; chIdx = aacChannelsOffsetIdx; @@ -2223,7 +2262,7 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, elCh = 1; } - if (self->elFlags[el2] & AC_EL_USAC_CP_POSSIBLE) { + if (elFlags[el2] & AC_EL_USAC_CP_POSSIBLE) { complexStereoPredPossible = 1; if (self->cpeStaticData[el2] == NULL) { self->cpeStaticData[el2] = GetCpePersistentData(); @@ -2360,9 +2399,6 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, } } - /* Update externally visible copy of flags */ - self->streamInfo.flags = self->flags[0]; - if (*configChanged) { int drcDecSampleRate, drcDecFrameSize; @@ -2383,8 +2419,7 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, if (*configChanged) { if (asc->m_aot == AOT_USAC) { - self->hDrcInfo->enable = 0; - self->hDrcInfo->progRefLevelPresent = 0; + aacDecoder_drcDisable(self->hDrcInfo); } } @@ -2393,6 +2428,12 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, pcmLimiter_SetThreshold(self->hLimiter, FL2FXCONST_DBL(0.89125094f)); } + CAacDecoder_AcceptFlags(self, asc, flags, elFlags, streamIndex, + elementOffset); + + /* Update externally visible copy of flags */ + self->streamInfo.flags = self->flags[0]; + return err; bail: @@ -3194,11 +3235,12 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame( * data in the bitstream. */ self->flags[streamIndex] |= AC_DRC_PRESENT; } else { - self->hDrcInfo->enable = 0; - self->hDrcInfo->progRefLevelPresent = 0; ErrorStatus = AAC_DEC_UNSUPPORTED_FORMAT; } } + if (self->flags[streamIndex] & (AC_USAC | AC_RSV603DA)) { + aacDecoder_drcDisable(self->hDrcInfo); + } /* Create a reverse mapping table */ UCHAR Reverse_chMapping[((8) * 2)]; @@ -3441,11 +3483,12 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame( * data in the bitstream. */ self->flags[streamIndex] |= AC_DRC_PRESENT; } else { - self->hDrcInfo->enable = 0; - self->hDrcInfo->progRefLevelPresent = 0; ErrorStatus = AAC_DEC_UNSUPPORTED_FORMAT; } } + if (self->flags[streamIndex] & (AC_USAC | AC_RSV603DA)) { + aacDecoder_drcDisable(self->hDrcInfo); + } } /* Add additional concealment delay */ 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 0f281eb..9d36d10 100644 --- a/libAACdec/src/aacdecoder_lib.cpp +++ b/libAACdec/src/aacdecoder_lib.cpp @@ -385,21 +385,19 @@ static INT aacDecoder_SbrCallback( return errTp; } -static INT aacDecoder_SscCallback(void *handle, HANDLE_FDK_BITSTREAM hBs, - const AUDIO_OBJECT_TYPE coreCodec, - const INT samplingRate, const INT frameSize, - const INT stereoConfigIndex, - const INT coreSbrFrameLengthIndex, - const INT configBytes, const UCHAR configMode, - UCHAR *configChanged) { +static INT aacDecoder_SscCallback( + void *handle, HANDLE_FDK_BITSTREAM hBs, const AUDIO_OBJECT_TYPE coreCodec, + const INT samplingRate, const INT frameSize, const INT numChannels, + const INT stereoConfigIndex, const INT coreSbrFrameLengthIndex, + const INT configBytes, const UCHAR configMode, UCHAR *configChanged) { SACDEC_ERROR err; TRANSPORTDEC_ERROR errTp; HANDLE_AACDECODER hAacDecoder = (HANDLE_AACDECODER)handle; err = mpegSurroundDecoder_Config( (CMpegSurroundDecoder *)hAacDecoder->pMpegSurroundDecoder, hBs, coreCodec, - samplingRate, frameSize, stereoConfigIndex, coreSbrFrameLengthIndex, - configBytes, configMode, configChanged); + samplingRate, frameSize, numChannels, stereoConfigIndex, + coreSbrFrameLengthIndex, configBytes, configMode, configChanged); switch (err) { case MPS_UNSUPPORTED_CONFIG: @@ -443,12 +441,23 @@ static INT aacDecoder_UniDrcCallback(void *handle, HANDLE_FDK_BITSTREAM hBs, TRANSPORTDEC_ERROR errTp; HANDLE_AACDECODER hAacDecoder = (HANDLE_AACDECODER)handle; DRC_DEC_CODEC_MODE drcDecCodecMode = DRC_DEC_CODEC_MODE_UNDEFINED; + UCHAR dummyBuffer[4] = {0}; + FDK_BITSTREAM dummyBs; + HANDLE_FDK_BITSTREAM hReadBs; if (subStreamIndex != 0) { return TRANSPORTDEC_OK; } - else if (aot == AOT_USAC) { + if (hBs == NULL) { + /* use dummy zero payload to clear memory */ + hReadBs = &dummyBs; + FDKinitBitStream(hReadBs, dummyBuffer, 4, 24); + } else { + hReadBs = hBs; + } + + if (aot == AOT_USAC) { drcDecCodecMode = DRC_DEC_MPEG_D_USAC; } @@ -457,10 +466,10 @@ static INT aacDecoder_UniDrcCallback(void *handle, HANDLE_FDK_BITSTREAM hBs, if (payloadType == 0) /* uniDrcConfig */ { - err = FDK_drcDec_ReadUniDrcConfig(hAacDecoder->hUniDrcDecoder, hBs); + err = FDK_drcDec_ReadUniDrcConfig(hAacDecoder->hUniDrcDecoder, hReadBs); } else /* loudnessInfoSet */ { - err = FDK_drcDec_ReadLoudnessInfoSet(hAacDecoder->hUniDrcDecoder, hBs); + err = FDK_drcDec_ReadLoudnessInfoSet(hAacDecoder->hUniDrcDecoder, hReadBs); hAacDecoder->loudnessInfoSetPosition[1] = payloadStart; hAacDecoder->loudnessInfoSetPosition[2] = fullPayloadLength; } @@ -822,6 +831,9 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_SetParam( case AAC_DRC_ATTENUATION_FACTOR: /* DRC compression factor (where 0 is no and 127 is max compression) */ + if ((value < 0) || (value > 127)) { + return AAC_DEC_SET_PARAM_FAIL; + } errorStatus = aacDecoder_drcSetParam(hDrcInfo, DRC_CUT_SCALE, value); uniDrcErr = FDK_drcDec_SetParam(self->hUniDrcDecoder, DRC_DEC_COMPRESS, value * (FL2FXCONST_DBL(0.5f / 127.0f))); @@ -829,6 +841,9 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_SetParam( case AAC_DRC_BOOST_FACTOR: /* DRC boost factor (where 0 is no and 127 is max boost) */ + if ((value < 0) || (value > 127)) { + return AAC_DEC_SET_PARAM_FAIL; + } errorStatus = aacDecoder_drcSetParam(hDrcInfo, DRC_BOOST_SCALE, value); uniDrcErr = FDK_drcDec_SetParam(self->hUniDrcDecoder, DRC_DEC_BOOST, value * (FL2FXCONST_DBL(0.5f / 127.0f))); @@ -1151,6 +1166,8 @@ 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; @@ -1173,8 +1190,10 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame(HANDLE_AACDECODER self, aacDecoder_FreeMemCallback(self, &asc); self->streamInfo.numChannels = 0; /* 3) restore AudioSpecificConfig */ - transportDec_OutOfBandConfig(self->hInput, asc.config, - (asc.configBits + 7) >> 3, 0); + if (asc.configBits <= (TP_USAC_MAX_CONFIG_LEN << 3)) { + transportDec_OutOfBandConfig(self->hInput, asc.config, + (asc.configBits + 7) >> 3, 0); + } } } @@ -1794,8 +1813,7 @@ 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; @@ -1826,145 +1844,38 @@ 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 */ } } @@ -2006,6 +1917,116 @@ 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 */ diff --git a/libAACdec/src/channel.cpp b/libAACdec/src/channel.cpp index a020034..7e62bfb 100644 --- a/libAACdec/src/channel.cpp +++ b/libAACdec/src/channel.cpp @@ -1,7 +1,7 @@ /* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -265,7 +265,9 @@ void CChannelElement_Decode( stereo prediction since scaling has already been carried out. */ int max_sfb_ste = (INT)(pAacDecoderChannelInfo[L]->icsInfo.max_sfb_ste); - if ((!CP_active) || (CP_active && (max_sfb_ste < noSfbs)) || + if (!(CP_active && (max_sfb_ste == noSfbs)) || + !(CP_active && + !(pAacDecoderChannelInfo[ch]->pDynData->TnsData.Active)) || ((flags & (AC_USAC | AC_RSVD50 | AC_RSV603DA)) && (pAacDecoderChannelInfo[L]->pDynData->specificTo.usac.tns_on_lr == 0))) { diff --git a/libAACenc/src/aacenc_lib.cpp b/libAACenc/src/aacenc_lib.cpp index 0ae329b..caa62c5 100644 --- a/libAACenc/src/aacenc_lib.cpp +++ b/libAACenc/src/aacenc_lib.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 @@ -1242,7 +1242,7 @@ static INT aacenc_SbrCallback(void *self, HANDLE_FDK_BITSTREAM hBs, INT aacenc_SscCallback(void *self, HANDLE_FDK_BITSTREAM hBs, const AUDIO_OBJECT_TYPE coreCodec, const INT samplingRate, const INT frameSize, - const INT stereoConfigIndex, + const INT numChannels, const INT stereoConfigIndex, const INT coreSbrFrameLengthIndex, const INT configBytes, const UCHAR configMode, UCHAR *configChanged) { HANDLE_AACENCODER hAacEncoder = (HANDLE_AACENCODER)self; @@ -1784,8 +1784,8 @@ AACENC_ERROR aacEncEncode(const HANDLE_AACENCODER hAacEncoder, hAacEncoder->nSamplesRead)); INT_PCM *pIn = hAacEncoder->inputBuffer + - (hAacEncoder->inputBufferOffset + hAacEncoder->nSamplesRead) / - hAacEncoder->aacConfig.nChannels; + hAacEncoder->inputBufferOffset / hAacEncoder->aacConfig.nChannels + + hAacEncoder->nSamplesRead / hAacEncoder->extParam.nChannels; newSamples -= (newSamples % hAacEncoder->extParam @@ -1827,12 +1827,13 @@ AACENC_ERROR aacEncEncode(const HANDLE_AACENCODER hAacEncoder, /* clear out until end-of-buffer */ if (nZeros) { + INT_PCM *pIn = + hAacEncoder->inputBuffer + + hAacEncoder->inputBufferOffset / + hAacEncoder->aacConfig.nChannels + + hAacEncoder->nSamplesRead / hAacEncoder->extParam.nChannels; for (i = 0; i < (int)hAacEncoder->extParam.nChannels; i++) { - FDKmemclear(hAacEncoder->inputBuffer + - i * hAacEncoder->inputBufferSizePerChannel + - (hAacEncoder->inputBufferOffset + - hAacEncoder->nSamplesRead) / - hAacEncoder->extParam.nChannels, + FDKmemclear(pIn + i * hAacEncoder->inputBufferSizePerChannel, sizeof(INT_PCM) * nZeros); } hAacEncoder->nZerosAppended += nZeros; diff --git a/libDRCdec/src/drcDec_reader.cpp b/libDRCdec/src/drcDec_reader.cpp index 367a352..b3ec187 100644 --- a/libDRCdec/src/drcDec_reader.cpp +++ b/libDRCdec/src/drcDec_reader.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 @@ -512,10 +512,13 @@ drcDec_readUniDrcGain(HANDLE_FDK_BITSTREAM hBs, fMin(tmpNNodes, (UCHAR)16) * sizeof(GAIN_NODE)); } - hUniDrcGain->uniDrcGainExtPresent = FDKreadBits(hBs, 1); - if (hUniDrcGain->uniDrcGainExtPresent == 1) { - err = _readUniDrcGainExtension(hBs, &(hUniDrcGain->uniDrcGainExtension)); - if (err) return err; + if (pCoef && (gainSequenceCount == + pCoef->gainSequenceCount)) { /* all sequences have been read */ + hUniDrcGain->uniDrcGainExtPresent = FDKreadBits(hBs, 1); + if (hUniDrcGain->uniDrcGainExtPresent == 1) { + err = _readUniDrcGainExtension(hBs, &(hUniDrcGain->uniDrcGainExtension)); + if (err) return err; + } } if (err == DE_OK && gainSequenceCount > 0) { diff --git a/libFDK/src/FDK_hybrid.cpp b/libFDK/src/FDK_hybrid.cpp index 08d32a8..d208abd 100644 --- a/libFDK/src/FDK_hybrid.cpp +++ b/libFDK/src/FDK_hybrid.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 @@ -539,11 +539,11 @@ static void dualChannelFiltering(const FIXP_DBL *const pQmfReal, i6 = pQmfImag[pReadIdx[6]] >> 2; FDK_ASSERT((invert == 0) || (invert == 1)); - mHybridReal[0 + invert] = (r6 + r1) << 1; - mHybridImag[0 + invert] = (i6 + i1) << 1; + mHybridReal[0 + invert] = SATURATE_LEFT_SHIFT((r6 + r1), 1, DFRACT_BITS); + mHybridImag[0 + invert] = SATURATE_LEFT_SHIFT((i6 + i1), 1, DFRACT_BITS); - mHybridReal[1 - invert] = (r6 - r1) << 1; - mHybridImag[1 - invert] = (i6 - i1) << 1; + mHybridReal[1 - invert] = SATURATE_LEFT_SHIFT((r6 - r1), 1, DFRACT_BITS); + mHybridImag[1 - invert] = SATURATE_LEFT_SHIFT((i6 - i1), 1, DFRACT_BITS); } static void fourChannelFiltering(const FIXP_DBL *const pQmfReal, @@ -766,15 +766,15 @@ static void eightChannelFiltering(const FIXP_DBL *const pQmfReal, mHybridReal[3] = pfft[FFT_IDX_R(1)] << sc; mHybridImag[3] = pfft[FFT_IDX_I(1)] << sc; - mHybridReal[4] = pfft[FFT_IDX_R(2)] << sc; - mHybridReal[4] += pfft[FFT_IDX_R(5)] << sc; - mHybridImag[4] = pfft[FFT_IDX_I(2)] << sc; - mHybridImag[4] += pfft[FFT_IDX_I(5)] << sc; + mHybridReal[4] = SATURATE_LEFT_SHIFT( + (pfft[FFT_IDX_R(2)] + pfft[FFT_IDX_R(5)]), sc, DFRACT_BITS); + mHybridImag[4] = SATURATE_LEFT_SHIFT( + (pfft[FFT_IDX_I(2)] + pfft[FFT_IDX_I(5)]), sc, DFRACT_BITS); - mHybridReal[5] = pfft[FFT_IDX_R(3)] << sc; - mHybridReal[5] += pfft[FFT_IDX_R(4)] << sc; - mHybridImag[5] = pfft[FFT_IDX_I(3)] << sc; - mHybridImag[5] += pfft[FFT_IDX_I(4)] << sc; + mHybridReal[5] = SATURATE_LEFT_SHIFT( + (pfft[FFT_IDX_R(3)] + pfft[FFT_IDX_R(4)]), sc, DFRACT_BITS); + mHybridImag[5] = SATURATE_LEFT_SHIFT( + (pfft[FFT_IDX_I(3)] + pfft[FFT_IDX_I(4)]), sc, DFRACT_BITS); } else { for (k = 0; k < 8; k++) { mHybridReal[k] = pfft[FFT_IDX_R(k)] << sc; diff --git a/libFDK/src/dct.cpp b/libFDK/src/dct.cpp index bd26736..35507b5 100644 --- a/libFDK/src/dct.cpp +++ b/libFDK/src/dct.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 @@ -305,9 +305,8 @@ void dct_II( { for (i = 0; i < M; i++) { - tmp[i] = pDat[2 * i] >> 1; /* dit_fft expects 1 bit scaled input values */ - tmp[L - 1 - i] = - pDat[2 * i + 1] >> 1; /* dit_fft expects 1 bit scaled input values */ + tmp[i] = pDat[2 * i] >> 2; + tmp[L - 1 - i] = pDat[2 * i + 1] >> 2; } } @@ -337,15 +336,14 @@ void dct_II( a1 = ((pTmp_0[0] >> 1) + (pTmp_1[0] >> 1)); a2 = ((pTmp_0[1] >> 1) - (pTmp_1[1] >> 1)); - cplxMultDiv2(&accu3, &accu4, (a1 + accu2), -(accu1 + a2), - sin_twiddle[i * inc]); - pDat[L - i] = accu4; - pDat[i] = accu3; + cplxMult(&accu3, &accu4, (accu1 + a2), (a1 + accu2), sin_twiddle[i * inc]); + pDat[L - i] = -accu3; + pDat[i] = accu4; - cplxMultDiv2(&accu3, &accu4, (a1 - accu2), -(accu1 - a2), - sin_twiddle[(M - i) * inc]); - pDat[M + i] = accu4; - pDat[M - i] = accu3; + cplxMult(&accu3, &accu4, (accu1 - a2), (a1 - accu2), + sin_twiddle[(M - i) * inc]); + pDat[M + i] = -accu3; + pDat[M - i] = accu4; /* Create index helper variables for (4*i)*inc indexed equivalent values of * short tables. */ @@ -356,12 +354,12 @@ void dct_II( } } - cplxMultDiv2(&accu1, &accu2, tmp[M], tmp[M + 1], sin_twiddle[(M / 2) * inc]); + cplxMult(&accu1, &accu2, tmp[M], tmp[M + 1], sin_twiddle[(M / 2) * inc]); pDat[L - (M / 2)] = accu2; pDat[M / 2] = accu1; - pDat[0] = (tmp[0] >> 1) + (tmp[1] >> 1); - pDat[M] = fMult(((tmp[0] >> 1) - (tmp[1] >> 1)), + pDat[0] = tmp[0] + tmp[1]; + pDat[M] = fMult(tmp[0] - tmp[1], sin_twiddle[M * inc].v.re); /* cos((PI/(2*L))*M); */ *pDat_e += 2; diff --git a/libMpegTPDec/include/tp_data.h b/libMpegTPDec/include/tp_data.h index b015332..b63087a 100644 --- a/libMpegTPDec/include/tp_data.h +++ b/libMpegTPDec/include/tp_data.h @@ -1,7 +1,7 @@ /* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -368,7 +368,7 @@ typedef INT (*cbCtrlCFGChange_t)(void *, const CCtrlCFGChange *); typedef INT (*cbSsc_t)(void *, HANDLE_FDK_BITSTREAM, const AUDIO_OBJECT_TYPE coreCodec, const INT samplingRate, const INT frameSize, - const INT stereoConfigIndex, + const INT numChannels, const INT stereoConfigIndex, const INT coreSbrFrameLengthIndex, const INT configBytes, const UCHAR configMode, UCHAR *configChanged); diff --git a/libMpegTPDec/src/tpdec_asc.cpp b/libMpegTPDec/src/tpdec_asc.cpp index 82f840e..e46cb32 100644 --- a/libMpegTPDec/src/tpdec_asc.cpp +++ b/libMpegTPDec/src/tpdec_asc.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 @@ -266,11 +266,118 @@ static int CProgramConfig_ReadHeightExt(CProgramConfig *pPce, return (err); } +/** + * \brief Sanity checks for program config element. + * Check order of elements according to ISO/IEC 13818-7:2003(E), + * chapter 8.5.1 + * + * \param pPce pointer to program config element. + * + * \return 0 if successful, otherwise 1. + */ +static int CProgramConfig_Check(CProgramConfig *pPce) { + INT i; + INT err = 0; + INT numBackChannels[3] = {0}; + INT numSideChannels[3] = {0}; + INT numFrontChannels[3] = {0}; + UCHAR *pCpeFront = pPce->FrontElementIsCpe; + UCHAR *pCpeSide = pPce->SideElementIsCpe; + UCHAR *pCpeBack = pPce->BackElementIsCpe; + UCHAR *pHeight; + + pHeight = pPce->BackElementHeightInfo; + for (i = 0; i < pPce->NumBackChannelElements; i++) { + numBackChannels[*pHeight] += pPce->BackElementIsCpe[i] ? 2 : 1; + pHeight++; + } + pHeight = pPce->SideElementHeightInfo; + for (i = 0; i < pPce->NumSideChannelElements; i++) { + numSideChannels[*pHeight] += pPce->SideElementIsCpe[i] ? 2 : 1; + pHeight++; + } + pHeight = pPce->FrontElementHeightInfo; + for (i = 0; i < pPce->NumFrontChannelElements; i++) { + numFrontChannels[*pHeight] += pPce->FrontElementIsCpe[i] ? 2 : 1; + pHeight++; + } + + /* 0 = normal height channels, 1 = top height channels, 2 = bottom height + * channels */ + for (i = 0; i < 3; i++) { + /* if number of channels is odd => first element must be a SCE (front center + * channel) */ + if (numFrontChannels[i] & 1) { + if (*pCpeFront++ == ID_CPE) { + err = 1; + goto bail; + } + numFrontChannels[i]--; + } + while (numFrontChannels[i] > 0) { + /* must be CPE or paired SCE */ + if (*pCpeFront++ == ID_SCE) { + if (*pCpeFront++ == ID_CPE) { + err = 1; + goto bail; + } + } + numFrontChannels[i] -= 2; + }; + + /* in case that a top center surround channel (Ts) is transmitted the number + * of channels can be odd */ + if (i != 1) { + /* number of channels must be even */ + if (numSideChannels[i] & 1) { + err = 1; + goto bail; + } + while (numSideChannels[i] > 0) { + /* must be CPE or paired SCE */ + if (*pCpeSide++ == ID_SCE) { + if (*pCpeSide++ == ID_CPE) { + err = 1; + goto bail; + } + } + numSideChannels[i] -= 2; + }; + } + + while (numBackChannels[i] > 1) { + /* must be CPE or paired SCE */ + if (*pCpeBack++ == ID_SCE) { + if (*pCpeBack++ == ID_CPE) { + err = 1; + goto bail; + } + } + numBackChannels[i] -= 2; + }; + /* if number of channels is odd => last element must be a SCE (back center + * channel) */ + if (numBackChannels[i]) { + if (*pCpeBack++ == ID_CPE) { + err = 1; + goto bail; + } + } + } + +bail: + + return err; +} + void CProgramConfig_Read(CProgramConfig *pPce, HANDLE_FDK_BITSTREAM bs, UINT alignmentAnchor) { - int i, err = 0; + int i; int commentBytes; + UCHAR tag, isCpe; + UCHAR checkElementTagSelect[3][PC_FSB_CHANNELS_MAX] = {{0}}; + pPce->isValid = 1; pPce->NumEffectiveChannels = 0; pPce->NumChannels = 0; pPce->ElementInstanceTag = (UCHAR)FDKreadBits(bs, 4); @@ -297,28 +404,60 @@ void CProgramConfig_Read(CProgramConfig *pPce, HANDLE_FDK_BITSTREAM bs, } for (i = 0; i < pPce->NumFrontChannelElements; i++) { - pPce->FrontElementIsCpe[i] = (UCHAR)FDKreadBits(bs, 1); - pPce->FrontElementTagSelect[i] = (UCHAR)FDKreadBits(bs, 4); + pPce->FrontElementIsCpe[i] = isCpe = (UCHAR)FDKreadBits(bs, 1); + pPce->FrontElementTagSelect[i] = tag = (UCHAR)FDKreadBits(bs, 4); pPce->NumChannels += pPce->FrontElementIsCpe[i] ? 2 : 1; + + /* Check element instance tag according to ISO/IEC 13818-7:2003(E), + * chapter 8.2.1.1 */ + if (checkElementTagSelect[isCpe][tag] == 0) { + checkElementTagSelect[isCpe][tag] = 1; + } else { + pPce->isValid = 0; + } } for (i = 0; i < pPce->NumSideChannelElements; i++) { - pPce->SideElementIsCpe[i] = (UCHAR)FDKreadBits(bs, 1); - pPce->SideElementTagSelect[i] = (UCHAR)FDKreadBits(bs, 4); + pPce->SideElementIsCpe[i] = isCpe = (UCHAR)FDKreadBits(bs, 1); + pPce->SideElementTagSelect[i] = tag = (UCHAR)FDKreadBits(bs, 4); pPce->NumChannels += pPce->SideElementIsCpe[i] ? 2 : 1; + + /* Check element instance tag according to ISO/IEC 13818-7:2003(E), + * chapter 8.2.1.1 */ + if (checkElementTagSelect[isCpe][tag] == 0) { + checkElementTagSelect[isCpe][tag] = 1; + } else { + pPce->isValid = 0; + } } for (i = 0; i < pPce->NumBackChannelElements; i++) { - pPce->BackElementIsCpe[i] = (UCHAR)FDKreadBits(bs, 1); - pPce->BackElementTagSelect[i] = (UCHAR)FDKreadBits(bs, 4); + pPce->BackElementIsCpe[i] = isCpe = (UCHAR)FDKreadBits(bs, 1); + pPce->BackElementTagSelect[i] = tag = (UCHAR)FDKreadBits(bs, 4); pPce->NumChannels += pPce->BackElementIsCpe[i] ? 2 : 1; + + /* Check element instance tag according to ISO/IEC 13818-7:2003(E), + * chapter 8.2.1.1 */ + if (checkElementTagSelect[isCpe][tag] == 0) { + checkElementTagSelect[isCpe][tag] = 1; + } else { + pPce->isValid = 0; + } } pPce->NumEffectiveChannels = pPce->NumChannels; for (i = 0; i < pPce->NumLfeChannelElements; i++) { - pPce->LfeElementTagSelect[i] = (UCHAR)FDKreadBits(bs, 4); + pPce->LfeElementTagSelect[i] = tag = (UCHAR)FDKreadBits(bs, 4); pPce->NumChannels += 1; + + /* Check element instance tag according to ISO/IEC 13818-7:2003(E), + * chapter 8.2.1.1 */ + if (checkElementTagSelect[2][tag] == 0) { + checkElementTagSelect[2][tag] = 1; + } else { + pPce->isValid = 0; + } } for (i = 0; i < pPce->NumAssocDataElements; i++) { @@ -336,7 +475,15 @@ void CProgramConfig_Read(CProgramConfig *pPce, HANDLE_FDK_BITSTREAM bs, commentBytes = pPce->CommentFieldBytes; /* Search for height info extension and read it if available */ - err = CProgramConfig_ReadHeightExt(pPce, bs, &commentBytes, alignmentAnchor); + if (CProgramConfig_ReadHeightExt(pPce, bs, &commentBytes, alignmentAnchor)) { + pPce->isValid = 0; + } + + /* Check order of elements according to ISO / IEC 13818 - 7:2003(E), + * chapter 8.5.1 */ + if (CProgramConfig_Check(pPce)) { + pPce->isValid = 0; + } for (i = 0; i < commentBytes; i++) { UCHAR text; @@ -347,8 +494,6 @@ void CProgramConfig_Read(CProgramConfig *pPce, HANDLE_FDK_BITSTREAM bs, pPce->Comment[i] = text; } } - - pPce->isValid = (err) ? 0 : 1; } /* @@ -1415,7 +1560,7 @@ static TRANSPORTDEC_ERROR EldSpecificConfig_Parse(CSAudioSpecificConfig *asc, cb->cbSscData, hBs, asc->m_aot, asc->m_samplingFrequency << esc->m_sbrSamplingRate, asc->m_samplesPerFrame << esc->m_sbrSamplingRate, - 1, /* stereoConfigIndex */ + asc->m_channelConfiguration, 1, /* stereoConfigIndex */ -1, /* nTimeSlots: read from bitstream */ eldExtLen, asc->configMode, &asc->SacConfigChanged); if (ErrorStatus != TRANSPORTDEC_OK) { @@ -1633,6 +1778,10 @@ static TRANSPORTDEC_ERROR configExtension(CSUsacConfig *usc, int numConfigExtensions; CONFIG_EXT_ID usacConfigExtType; int usacConfigExtLength; + int loudnessInfoSetIndex = + -1; /* index of loudnessInfoSet config extension. -1 if not contained. */ + int tmp_subStreamIndex = 0; + AUDIO_OBJECT_TYPE tmp_aot = AOT_USAC; numConfigExtensions = (int)escapedValue(hBs, 2, 4, 8) + 1; for (int confExtIdx = 0; confExtIdx < numConfigExtensions; confExtIdx++) { @@ -1662,10 +1811,12 @@ static TRANSPORTDEC_ERROR configExtension(CSUsacConfig *usc, ErrorStatus = (TRANSPORTDEC_ERROR)cb->cbUniDrc( cb->cbUniDrcData, hBs, usacConfigExtLength, 1, /* loudnessInfoSet */ - 0, loudnessInfoSetConfigExtensionPosition, AOT_USAC); + tmp_subStreamIndex, loudnessInfoSetConfigExtensionPosition, + tmp_aot); if (ErrorStatus != TRANSPORTDEC_OK) { return ErrorStatus; } + loudnessInfoSetIndex = confExtIdx; } } break; default: @@ -1681,6 +1832,17 @@ static TRANSPORTDEC_ERROR configExtension(CSUsacConfig *usc, FDKpushFor(hBs, usacConfigExtLength); } + if (loudnessInfoSetIndex == -1 && cb->cbUniDrc != NULL) { + /* no loudnessInfoSet contained. Clear the loudnessInfoSet struct by feeding + * an empty config extension */ + ErrorStatus = (TRANSPORTDEC_ERROR)cb->cbUniDrc( + cb->cbUniDrcData, NULL, 0, 1 /* loudnessInfoSet */, tmp_subStreamIndex, + 0, tmp_aot); + if (ErrorStatus != TRANSPORTDEC_OK) { + return ErrorStatus; + } + } + return ErrorStatus; } @@ -1697,6 +1859,8 @@ static TRANSPORTDEC_ERROR UsacRsv60DecoderConfig_Parse( int channelElementIdx = 0; /* index for elements which contain audio channels (sce, cpe, lfe) */ SC_CHANNEL_CONFIG sc_chan_config = {0, 0, 0, 0}; + int uniDrcElement = + -1; /* index of uniDrc extension element. -1 if not contained. */ numberOfElements = (int)escapedValue(hBs, 4, 8, 16) + 1; usc->m_usacNumElements = numberOfElements; @@ -1827,6 +1991,8 @@ static TRANSPORTDEC_ERROR UsacRsv60DecoderConfig_Parse( /* Mps212Config() ISO/IEC FDIS 23003-3 */ if (cb->cbSsc(cb->cbSscData, hBs, asc->m_aot, asc->m_extensionSamplingFrequency, samplesPerFrame, + 1, /* only downmix channels (residual channels are + not counted) */ usc->element[i].m_stereoConfigIndex, usc->m_coreSbrFrameLengthIndex, 0, /* don't know the length */ @@ -1870,6 +2036,10 @@ static TRANSPORTDEC_ERROR UsacRsv60DecoderConfig_Parse( case ID_USAC_EXT: ErrorStatus = extElementConfig(&usc->element[i].extElement, hBs, cb, 0, asc->m_samplesPerFrame, 0, asc->m_aot); + if (usc->element[i].extElement.usacExtElementType == + ID_EXT_ELE_UNI_DRC) { + uniDrcElement = i; + } if (ErrorStatus) { return ErrorStatus; @@ -1898,6 +2068,18 @@ static TRANSPORTDEC_ERROR UsacRsv60DecoderConfig_Parse( } } + if (uniDrcElement == -1 && cb->cbUniDrc != NULL) { + /* no uniDrcConfig contained. Clear the uniDrcConfig struct by feeding an + * empty extension element */ + int subStreamIndex = 0; + ErrorStatus = (TRANSPORTDEC_ERROR)cb->cbUniDrc( + cb->cbUniDrcData, NULL, 0, 0 /* uniDrcConfig */, subStreamIndex, 0, + asc->m_aot); + if (ErrorStatus != TRANSPORTDEC_OK) { + return ErrorStatus; + } + } + return ErrorStatus; } @@ -1984,6 +2166,14 @@ static TRANSPORTDEC_ERROR UsacConfig_Parse(CSAudioSpecificConfig *asc, if (err != TRANSPORTDEC_OK) { return err; } + } else if (cb->cbUniDrc != NULL) { + /* no loudnessInfoSet contained. Clear the loudnessInfoSet struct by feeding + * an empty config extension */ + err = (TRANSPORTDEC_ERROR)cb->cbUniDrc( + cb->cbUniDrcData, NULL, 0, 1 /* loudnessInfoSet */, 0, 0, asc->m_aot); + if (err != TRANSPORTDEC_OK) { + return err; + } } /* sanity check whether number of channels signaled in UsacDecoderConfig() @@ -1996,9 +2186,11 @@ static TRANSPORTDEC_ERROR UsacConfig_Parse(CSAudioSpecificConfig *asc, /* Copy UsacConfig() to asc->m_sc.m_usacConfig.UsacConfig[] buffer. */ INT configSize_bits = (INT)FDKgetValidBits(hBs) - nbits; - StoreConfigAsBitstream(hBs, configSize_bits, - asc->m_sc.m_usacConfig.UsacConfig, - TP_USAC_MAX_CONFIG_LEN); + if (StoreConfigAsBitstream(hBs, configSize_bits, + asc->m_sc.m_usacConfig.UsacConfig, + TP_USAC_MAX_CONFIG_LEN)) { + return TRANSPORTDEC_PARSE_ERROR; + } asc->m_sc.m_usacConfig.UsacConfigBits = fAbs(configSize_bits); return err; @@ -2219,7 +2411,7 @@ TRANSPORTDEC_ERROR AudioSpecificConfig_Parse( case AOT_MPEGS: if (cb->cbSsc != NULL) { if (cb->cbSsc(cb->cbSscData, bs, self->m_aot, self->m_samplingFrequency, - self->m_samplesPerFrame, 1, + self->m_samplesPerFrame, self->m_channelConfiguration, 1, -1, /* nTimeSlots: read from bitstream */ 0, /* don't know the length */ self->configMode, &self->SacConfigChanged)) { @@ -2300,8 +2492,10 @@ TRANSPORTDEC_ERROR AudioSpecificConfig_Parse( /* Copy config() to asc->config[] buffer. */ if ((ErrorStatus == TRANSPORTDEC_OK) && (self->m_aot == AOT_USAC)) { INT configSize_bits = (INT)FDKgetValidBits(bs) - (INT)ascStartAnchor; - StoreConfigAsBitstream(bs, configSize_bits, self->config, - TP_USAC_MAX_CONFIG_LEN); + if (StoreConfigAsBitstream(bs, configSize_bits, self->config, + TP_USAC_MAX_CONFIG_LEN)) { + return TRANSPORTDEC_PARSE_ERROR; + } self->configBits = fAbs(configSize_bits); } @@ -2415,6 +2609,8 @@ static TRANSPORTDEC_ERROR Drm_xHEAACDecoderConfig( cb->cbSscData, hBs, AOT_DRM_USAC, /* syntax differs from MPEG Mps212Config() */ asc->m_extensionSamplingFrequency, samplesPerFrame, + 1, /* only downmix channels (residual channels are not + counted) */ usc->element[elemIdx].m_stereoConfigIndex, usc->m_coreSbrFrameLengthIndex, 0, /* don't know the length */ asc->configMode, &asc->SacConfigChanged); diff --git a/libMpegTPEnc/include/tp_data.h b/libMpegTPEnc/include/tp_data.h index 00de356..464c485 100644 --- a/libMpegTPEnc/include/tp_data.h +++ b/libMpegTPEnc/include/tp_data.h @@ -1,7 +1,7 @@ /* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -368,7 +368,7 @@ typedef INT (*cbCtrlCFGChange_t)(void *, const CCtrlCFGChange *); typedef INT (*cbSsc_t)(void *, HANDLE_FDK_BITSTREAM, const AUDIO_OBJECT_TYPE coreCodec, const INT samplingRate, const INT frameSize, - const INT stereoConfigIndex, + const INT numChannels, const INT stereoConfigIndex, const INT coreSbrFrameLengthIndex, const INT configBytes, const UCHAR configMode, UCHAR *configChanged); diff --git a/libMpegTPEnc/src/tpenc_asc.cpp b/libMpegTPEnc/src/tpenc_asc.cpp index 0b484a0..9591ba8 100644 --- a/libMpegTPEnc/src/tpenc_asc.cpp +++ b/libMpegTPEnc/src/tpenc_asc.cpp @@ -1,7 +1,7 @@ /* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -795,7 +795,7 @@ static int transportEnc_writeELDSpecificConfig(HANDLE_FDK_BITSTREAM hBs, const INT eldExtLen = (cb->cbSsc(cb->cbSscData, NULL, config->aot, config->extSamplingRate, 0, - 0, 0, 0, 0, NULL) + + 0, 0, 0, 0, 0, NULL) + 7) >> 3; INT cnt = eldExtLen; @@ -818,7 +818,7 @@ static int transportEnc_writeELDSpecificConfig(HANDLE_FDK_BITSTREAM hBs, } cb->cbSsc(cb->cbSscData, hBs, config->aot, config->extSamplingRate, 0, 0, 0, - 0, 0, NULL); + 0, 0, 0, NULL); } if (config->downscaleSamplingRate != 0 && diff --git a/libSACdec/include/sac_dec_lib.h b/libSACdec/include/sac_dec_lib.h index 1827504..5aad4e0 100644 --- a/libSACdec/include/sac_dec_lib.h +++ b/libSACdec/include/sac_dec_lib.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 @@ -316,8 +316,8 @@ SACDEC_ERROR mpegSurroundDecoder_Init( SACDEC_ERROR mpegSurroundDecoder_Config( CMpegSurroundDecoder *pMpegSurroundDecoder, HANDLE_FDK_BITSTREAM hBs, AUDIO_OBJECT_TYPE coreCodec, INT samplingRate, INT frameSize, - INT stereoConfigIndex, INT coreSbrFrameLengthIndex, INT configBytes, - const UCHAR configMode, UCHAR *configChanged); + INT numChannels, INT stereoConfigIndex, INT coreSbrFrameLengthIndex, + INT configBytes, const UCHAR configMode, UCHAR *configChanged); SACDEC_ERROR mpegSurroundDecoder_ConfigureQmfDomain( diff --git a/libSACdec/src/sac_dec.cpp b/libSACdec/src/sac_dec.cpp index a7b50df..a26e251 100644 --- a/libSACdec/src/sac_dec.cpp +++ b/libSACdec/src/sac_dec.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 @@ -1098,6 +1098,28 @@ static void SpatialDecApplyBypass(spatialDec *self, FIXP_DBL **hybInputReal, } } +/** + * \brief Set internal error and reset error status + * + * \param self spatialDec handle. + * \param bypassMode pointer to bypassMode. + * \param err error status. + * + * \return error status. + */ +static SACDEC_ERROR SpatialDecSetInternalError(spatialDec *self, + int *bypassMode, + SACDEC_ERROR err) { + *bypassMode = 1; + + if (self->errInt == MPS_OK) { + /* store internal error before it gets overwritten */ + self->errInt = err; + } + + return MPS_OK; +} + /******************************************************************************* Functionname: SpatialDecApplyParameterSets ******************************************************************************* @@ -1118,7 +1140,7 @@ static SACDEC_ERROR SpatialDecApplyParameterSets( const FDK_channelMapDescr *const mapDescr) { SACDEC_ERROR err = MPS_OK; - FIXP_SGL alpha; + FIXP_SGL alpha = FL2FXCONST_SGL(0.0); int ts; int ch; @@ -1141,20 +1163,22 @@ static SACDEC_ERROR SpatialDecApplyParameterSets( ts++, ts_io++) { int currSlot = frame->paramSlot[ps]; + err = (currSlot < ts) ? MPS_WRONG_PARAMETERSETS : MPS_OK; + if (err != MPS_OK) { + err = SpatialDecSetInternalError(self, &bypassMode, err); + } + /* * Get new parameter set */ if (ts == prevSlot + 1) { - err = SpatialDecCalculateM1andM2(self, ps, - frame); /* input: ottCLD, ottICC, ... */ - /* output: M1param(Real/Imag), M2(Real/Imag) */ - if (err != MPS_OK) { - bypassMode = 1; - if (self->errInt == MPS_OK) { - /* store internal error befor it gets overwritten */ - self->errInt = err; + if (bypassMode == 0) { + err = SpatialDecCalculateM1andM2( + self, ps, frame); /* input: ottCLD, ottICC, ... */ + /* output: M1param(Real/Imag), M2(Real/Imag) */ + if (err != MPS_OK) { + err = SpatialDecSetInternalError(self, &bypassMode, err); } - err = MPS_OK; } if ((ps == 0) && (self->bOverwriteM1M2prev != 0)) { @@ -1168,13 +1192,16 @@ static SACDEC_ERROR SpatialDecApplyParameterSets( self->bOverwriteM1M2prev = 0; } - SpatialDecSmoothM1andM2( - self, frame, - ps); /* input: M1param(Real/Imag)(Prev), M2(Real/Imag)(Prev) */ - /* output: M1param(Real/Imag), M2(Real/Imag) */ + if (bypassMode == 0) { + SpatialDecSmoothM1andM2( + self, frame, + ps); /* input: M1param(Real/Imag)(Prev), M2(Real/Imag)(Prev) */ + } /* output: M1param(Real/Imag), M2(Real/Imag) */ } - alpha = FX_DBL2FX_SGL(fDivNorm(ts - prevSlot, currSlot - prevSlot)); + if (bypassMode == 0) { + alpha = FX_DBL2FX_SGL(fDivNorm(ts - prevSlot, currSlot - prevSlot)); + } switch (mode) { case INPUTMODE_QMF_SBR: @@ -1182,15 +1209,17 @@ static SACDEC_ERROR SpatialDecApplyParameterSets( self->bShareDelayWithSBR = 0; /* We got no hybrid delay */ else self->bShareDelayWithSBR = 1; - SpatialDecFeedQMF(self, qmfInDataReal, qmfInDataImag, ts_io, bypassMode, - self->qmfInputReal__FDK, self->qmfInputImag__FDK, - self->numInputChannels); + SpatialDecFeedQMF( + self, qmfInDataReal, qmfInDataImag, ts_io, bypassMode, + self->qmfInputReal__FDK, self->qmfInputImag__FDK, + (bypassMode) ? numInputChannels : self->numInputChannels); break; case INPUTMODE_TIME: self->bShareDelayWithSBR = 0; - SpatialDecQMFAnalysis(self, inData, ts_io, bypassMode, - self->qmfInputReal__FDK, self->qmfInputImag__FDK, - self->numInputChannels); + SpatialDecQMFAnalysis( + self, inData, ts_io, bypassMode, self->qmfInputReal__FDK, + self->qmfInputImag__FDK, + (bypassMode) ? numInputChannels : self->numInputChannels); break; default: break; @@ -1360,7 +1389,7 @@ static SACDEC_ERROR SpatialDecApplyParameterSets( } /* !self->tempShapeConfig == 1 */ } /* !bypassMode */ - if (self->phaseCoding == 1) { + if ((self->phaseCoding == 1) && (bypassMode == 0)) { /* only if bsPhaseCoding == 1 and bsResidualCoding == 0 */ SpatialDecApplyPhase( diff --git a/libSACdec/src/sac_dec_lib.cpp b/libSACdec/src/sac_dec_lib.cpp index 57446f8..d30131f 100644 --- a/libSACdec/src/sac_dec_lib.cpp +++ b/libSACdec/src/sac_dec_lib.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 @@ -700,9 +700,10 @@ bail: SACDEC_ERROR mpegSurroundDecoder_Config( CMpegSurroundDecoder *pMpegSurroundDecoder, HANDLE_FDK_BITSTREAM hBs, AUDIO_OBJECT_TYPE coreCodec, INT samplingRate, INT frameSize, - INT stereoConfigIndex, INT coreSbrFrameLengthIndex, INT configBytes, - const UCHAR configMode, UCHAR *configChanged) { + INT numChannels, INT stereoConfigIndex, INT coreSbrFrameLengthIndex, + INT configBytes, const UCHAR configMode, UCHAR *configChanged) { SACDEC_ERROR err = MPS_OK; + INT nInputChannels; SPATIAL_SPECIFIC_CONFIG spatialSpecificConfig; SPATIAL_SPECIFIC_CONFIG *pSsc = &pMpegSurroundDecoder->spatialSpecificConfigBackup; @@ -716,12 +717,18 @@ SACDEC_ERROR mpegSurroundDecoder_Config( err = SpatialDecParseMps212Config( hBs, &spatialSpecificConfig, samplingRate, coreCodec, stereoConfigIndex, coreSbrFrameLengthIndex); + nInputChannels = spatialSpecificConfig.nInputChannels; pSsc = &spatialSpecificConfig; } else { err = SpatialDecParseMps212Config( hBs, &pMpegSurroundDecoder->spatialSpecificConfigBackup, samplingRate, coreCodec, stereoConfigIndex, coreSbrFrameLengthIndex); + nInputChannels = + pMpegSurroundDecoder->spatialSpecificConfigBackup.nInputChannels; + } + if ((err == MPS_OK) && (numChannels != nInputChannels)) { + err = MPS_PARSE_ERROR; } break; case AOT_ER_AAC_ELD: @@ -731,11 +738,19 @@ SACDEC_ERROR mpegSurroundDecoder_Config( * into temporarily allocated structure */ err = SpatialDecParseSpecificConfig(hBs, &spatialSpecificConfig, configBytes, coreCodec); + nInputChannels = spatialSpecificConfig.nInputChannels; pSsc = &spatialSpecificConfig; } else { err = SpatialDecParseSpecificConfig( hBs, &pMpegSurroundDecoder->spatialSpecificConfigBackup, configBytes, coreCodec); + nInputChannels = + pMpegSurroundDecoder->spatialSpecificConfigBackup.nInputChannels; + } + /* check number of channels for channel_configuration > 0 */ + if ((err == MPS_OK) && (numChannels > 0) && + (numChannels != nInputChannels)) { + err = MPS_PARSE_ERROR; } break; default: diff --git a/libSACdec/src/sac_stp.cpp b/libSACdec/src/sac_stp.cpp index bb66277..b328c82 100644 --- a/libSACdec/src/sac_stp.cpp +++ b/libSACdec/src/sac_stp.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 @@ -369,15 +369,15 @@ SACDEC_ERROR subbandTPApply(spatialDec *self, const SPATIAL_BS_FRAME *frame) { hStpDec->update_old_ener = 1; for (ch = 0; ch < self->numInputChannels; ch++) { hStpDec->oldDryEnerLD64[ch] = - CalcLdData(hStpDec->runDryEner[ch] + ABS_THR__FDK); + CalcLdData(fAddSaturate(hStpDec->runDryEner[ch], ABS_THR__FDK)); } for (ch = 0; ch < self->numOutputChannels; ch++) { if (self->treeConfig == TREE_212) hStpDec->oldWetEnerLD64[ch] = - CalcLdData(hStpDec->runWetEner[ch] + ABS_THR__FDK); + CalcLdData(fAddSaturate(hStpDec->runWetEner[ch], ABS_THR__FDK)); else hStpDec->oldWetEnerLD64[ch] = - CalcLdData(hStpDec->runWetEner[ch] + ABS_THR2__FDK); + CalcLdData(fAddSaturate(hStpDec->runWetEner[ch], ABS_THR2__FDK)); } } else { hStpDec->update_old_ener++; diff --git a/libSBRdec/src/env_calc.cpp b/libSBRdec/src/env_calc.cpp index 0b2f651..ad5edfe 100644 --- a/libSBRdec/src/env_calc.cpp +++ b/libSBRdec/src/env_calc.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 @@ -1398,6 +1398,17 @@ void calculateSbrEnvelope( */ noise_e = (start_pos < no_cols) ? adj_e : final_e; + if (start_pos >= no_cols) { + int diff = h_sbr_cal_env->filtBufferNoise_e - noise_e; + if (diff > 0) { + int s = getScalefactor(h_sbr_cal_env->filtBufferNoise, noSubbands); + if (diff > s) { + final_e += diff - s; + noise_e = final_e; + } + } + } + /* Convert energies to amplitude levels */ @@ -2741,6 +2752,9 @@ static void adjustTimeSlotHQ_GainAndNoise( fMult(direct_ratio, noiseLevel[k]); } + smoothedNoise = fMax(fMin(smoothedNoise, (FIXP_DBL)(MAXVAL_DBL / 2)), + (FIXP_DBL)(MINVAL_DBL / 2)); + /* The next 2 multiplications constitute the actual envelope adjustment of the signal and should be carried out with full accuracy @@ -2930,6 +2944,9 @@ static void adjustTimeSlotHQ( fMult(direct_ratio, noiseLevel[k]); } + smoothedNoise = fMax(fMin(smoothedNoise, (FIXP_DBL)(MAXVAL_DBL / 2)), + (FIXP_DBL)(MINVAL_DBL / 2)); + /* The next 2 multiplications constitute the actual envelope adjustment of the signal and should be carried out with full accuracy diff --git a/libSBRdec/src/sbrdec_drc.cpp b/libSBRdec/src/sbrdec_drc.cpp index 2d73f32..089d046 100644 --- a/libSBRdec/src/sbrdec_drc.cpp +++ b/libSBRdec/src/sbrdec_drc.cpp @@ -1,7 +1,7 @@ /* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -233,14 +233,19 @@ void sbrDecoder_drcApplySlot(HANDLE_SBR_DRC_CHANNEL hDrcData, if (hDrcData->winSequenceCurr != 2) { /* long window */ int j = col + (numQmfSubSamples >> 1); - if (hDrcData->drcInterpolationSchemeCurr == 0) { - INT k = (frameLenFlag) ? 0x4444445 : 0x4000000; + if (j < winBorderToColMap[15]) { + if (hDrcData->drcInterpolationSchemeCurr == 0) { + INT k = (frameLenFlag) ? 0x4444445 : 0x4000000; - alphaValue = (FIXP_DBL)(j * k); - } else { - if (j >= (int)winBorderToColMap[hDrcData->drcInterpolationSchemeCurr]) { - alphaValue = (FIXP_DBL)MAXVAL_DBL; + alphaValue = (FIXP_DBL)(j * k); + } else { + if (j >= + (int)winBorderToColMap[hDrcData->drcInterpolationSchemeCurr]) { + alphaValue = (FIXP_DBL)MAXVAL_DBL; + } } + } else { + alphaValue = (FIXP_DBL)MAXVAL_DBL; } } else { /* short windows */ shortDrc = 1; @@ -254,14 +259,19 @@ void sbrDecoder_drcApplySlot(HANDLE_SBR_DRC_CHANNEL hDrcData, if (hDrcData->winSequenceNext != 2) { /* next: long window */ int j = col - (numQmfSubSamples >> 1); - if (hDrcData->drcInterpolationSchemeNext == 0) { - INT k = (frameLenFlag) ? 0x4444445 : 0x4000000; + if (j < winBorderToColMap[15]) { + if (hDrcData->drcInterpolationSchemeNext == 0) { + INT k = (frameLenFlag) ? 0x4444445 : 0x4000000; - alphaValue = (FIXP_DBL)(j * k); - } else { - if (j >= (int)winBorderToColMap[hDrcData->drcInterpolationSchemeNext]) { - alphaValue = (FIXP_DBL)MAXVAL_DBL; + alphaValue = (FIXP_DBL)(j * k); + } else { + if (j >= + (int)winBorderToColMap[hDrcData->drcInterpolationSchemeNext]) { + alphaValue = (FIXP_DBL)MAXVAL_DBL; + } } + } else { + alphaValue = (FIXP_DBL)MAXVAL_DBL; } fact_mag = hDrcData->nextFact_mag; @@ -289,14 +299,19 @@ void sbrDecoder_drcApplySlot(HANDLE_SBR_DRC_CHANNEL hDrcData, if (hDrcData->winSequenceNext != 2) { /* long window */ int j = col - (numQmfSubSamples >> 1); - if (hDrcData->drcInterpolationSchemeNext == 0) { - INT k = (frameLenFlag) ? 0x4444445 : 0x4000000; + if (j < winBorderToColMap[15]) { + if (hDrcData->drcInterpolationSchemeNext == 0) { + INT k = (frameLenFlag) ? 0x4444445 : 0x4000000; - alphaValue = (FIXP_DBL)(j * k); - } else { - if (j >= (int)winBorderToColMap[hDrcData->drcInterpolationSchemeNext]) { - alphaValue = (FIXP_DBL)MAXVAL_DBL; + alphaValue = (FIXP_DBL)(j * k); + } else { + if (j >= + (int)winBorderToColMap[hDrcData->drcInterpolationSchemeNext]) { + alphaValue = (FIXP_DBL)MAXVAL_DBL; + } } + } else { + alphaValue = (FIXP_DBL)MAXVAL_DBL; } } else { /* short windows */ shortDrc = 1; diff --git a/libSBRenc/src/env_est.cpp b/libSBRenc/src/env_est.cpp index 0eb8425..cc8780a 100644 --- a/libSBRenc/src/env_est.cpp +++ b/libSBRenc/src/env_est.cpp @@ -1,7 +1,7 @@ /* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -1267,6 +1267,7 @@ void FDKsbrEnc_extractSbrEnvelope2( sbrExtrEnv->pre_transient_info[1] = ed->transient_info[1]; /* tran_flag */ hEnvChan->encEnvData.noOfEnvelopes = ed->nEnvelopes = ed->frame_info->nEnvelopes; /* number of envelopes of current frame */ + hEnvChan->encEnvData.currentAmpResFF = (AMP_RES)h_con->initAmpResFF; /* Check if the current frame is divided into one envelope only. If so, set @@ -1274,8 +1275,9 @@ void FDKsbrEnc_extractSbrEnvelope2( */ if ((hEnvChan->encEnvData.hSbrBSGrid->frameClass == FIXFIX) && (ed->nEnvelopes == 1)) { + AMP_RES currentAmpResFF = SBR_AMP_RES_1_5; if (h_con->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) { - /* Note: global_tonaliy_float_value == + /* Note: global_tonality_float_value == ((float)hEnvChan->encEnvData.global_tonality/((INT64)(1)<<(31-(19+2)))/0.524288*(2.0/3.0))); threshold_float_value == ((float)h_con->thresholdAmpResFF_m/((INT64)(1)<<(31-(h_con->thresholdAmpResFF_e)))/0.524288*(2.0/3.0))); @@ -1289,14 +1291,13 @@ void FDKsbrEnc_extractSbrEnvelope2( } else { hEnvChan->encEnvData.currentAmpResFF = SBR_AMP_RES_3_0; } - } else - hEnvChan->encEnvData.currentAmpResFF = SBR_AMP_RES_1_5; + currentAmpResFF = hEnvChan->encEnvData.currentAmpResFF; + } - if (hEnvChan->encEnvData.currentAmpResFF != - hEnvChan->encEnvData.init_sbr_amp_res) { + if (currentAmpResFF != hEnvChan->encEnvData.init_sbr_amp_res) { FDKsbrEnc_InitSbrHuffmanTables( &hEnvChan->encEnvData, &hEnvChan->sbrCodeEnvelope, - &hEnvChan->sbrCodeNoiseFloor, hEnvChan->encEnvData.currentAmpResFF); + &hEnvChan->sbrCodeNoiseFloor, currentAmpResFF); } } else { if (sbrHeaderData->sbr_amp_res != hEnvChan->encEnvData.init_sbr_amp_res) { @@ -1355,6 +1356,13 @@ void FDKsbrEnc_extractSbrEnvelope2( } } + if (h_con->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY && + stereoMode == SBR_SWITCH_LRC && + h_envChan[0]->encEnvData.currentAmpResFF != + h_envChan[1]->encEnvData.currentAmpResFF) { + stereoMode = SBR_LEFT_RIGHT; + } + /* Extract envelope of current frame. */ diff --git a/libSBRenc/src/sbr_encoder.cpp b/libSBRenc/src/sbr_encoder.cpp index c1e083f..c3da072 100644 --- a/libSBRenc/src/sbr_encoder.cpp +++ b/libSBRenc/src/sbr_encoder.cpp @@ -1,7 +1,7 @@ /* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -1450,8 +1450,6 @@ static INT initEnvChannel(HANDLE_SBR_CONFIG_DATA sbrConfigData, params->deltaTAcrossFrames, 0, 0)) return (1); - sbrConfigData->initAmpResFF = params->init_amp_res_FF; - if (FDKsbrEnc_InitSbrHuffmanTables(&hEnv->encEnvData, &hEnv->sbrCodeEnvelope, &hEnv->sbrCodeNoiseFloor, sbrHeaderData->sbr_amp_res)) @@ -1749,6 +1747,7 @@ static INT FDKsbrEnc_EnvInit(HANDLE_SBR_ELEMENT hSbrElement, hSbrElement->sbrHeaderData.sbr_data_extra = 1; hSbrElement->sbrHeaderData.sbr_amp_res = (AMP_RES)params->amp_res; + hSbrElement->sbrConfigData.initAmpResFF = params->init_amp_res_FF; /* header_extra_1 */ hSbrElement->sbrHeaderData.freqScale = params->freqScale; |