diff --git a/documentation/aacDecoder.pdf b/documentation/aacDecoder.pdf index cc7cf41..3d4699e 100644 Binary files a/documentation/aacDecoder.pdf and b/documentation/aacDecoder.pdf differ diff --git a/documentation/aacEncoder.pdf b/documentation/aacEncoder.pdf index 77b8f4c..a47708a 100644 Binary files a/documentation/aacEncoder.pdf and b/documentation/aacEncoder.pdf differ 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/include/aacdecoder_lib.h b/libAACdec/include/aacdecoder_lib.h index 56f4ec1..d7928c0 100644 --- a/libAACdec/include/aacdecoder_lib.h +++ b/libAACdec/include/aacdecoder_lib.h @@ -1032,7 +1032,7 @@ LINKSPEC_H AAC_DECODER_ERROR aacDecoder_Fill(HANDLE_AACDECODER self, * \param self AAC decoder handle. * \param pTimeData Pointer to external output buffer where the decoded PCM * samples will be stored into. - * \param timeDataSize Size of external output buffer. + * \param timeDataSize Size of external output buffer in PCM samples. * \param flags Bit field with flags for the decoder: \n * (flags & AACDEC_CONCEAL) == 1: Do concealment. \n * (flags & AACDEC_FLUSH) == 2: Discard input data. Flush 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 @@ -149,6 +149,19 @@ static INT convert_drcParam(FIXP_DBL param_dbl) { return (INT)param_long; } +/*! +\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 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..d5f0cea 100644 --- a/libAACdec/src/aacdecoder.cpp +++ b/libAACdec/src/aacdecoder.cpp @@ -1,7 +1,7 @@ /* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten +© Copyright 1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -494,6 +494,75 @@ static AAC_DECODER_ERROR CDataStreamElement_Read(HANDLE_AACDECODER self, return error; } +static INT findElementInstanceTag( + INT elementTag, MP4_ELEMENT_ID elementId, + CAacDecoderChannelInfo **pAacDecoderChannelInfo, INT nChannels, + MP4_ELEMENT_ID *pElementIdTab, INT nElements) { + int el, chCnt = 0; + + for (el = 0; el < nElements; el++) { + switch (pElementIdTab[el]) { + case ID_CPE: + case ID_SCE: + case ID_LFE: + if ((elementTag == pAacDecoderChannelInfo[chCnt]->ElementInstanceTag) && + (elementId == pElementIdTab[el])) { + return 1; /* element instance tag found */ + } + chCnt += (pElementIdTab[el] == ID_CPE) ? 2 : 1; + break; + default: + break; + } + if (chCnt >= nChannels) break; + if (pElementIdTab[el] == ID_END) break; + } + + return 0; /* element instance tag not found */ +} + +static INT validateElementInstanceTags( + CProgramConfig *pce, CAacDecoderChannelInfo **pAacDecoderChannelInfo, + INT nChannels, MP4_ELEMENT_ID *pElementIdTab, INT nElements) { + if (nChannels >= pce->NumChannels) { + for (int el = 0; el < pce->NumFrontChannelElements; el++) { + if (!findElementInstanceTag(pce->FrontElementTagSelect[el], + pce->FrontElementIsCpe[el] ? ID_CPE : ID_SCE, + pAacDecoderChannelInfo, nChannels, + pElementIdTab, nElements)) { + return 0; /* element instance tag not in raw_data_block() */ + } + } + for (int el = 0; el < pce->NumSideChannelElements; el++) { + if (!findElementInstanceTag(pce->SideElementTagSelect[el], + pce->SideElementIsCpe[el] ? ID_CPE : ID_SCE, + pAacDecoderChannelInfo, nChannels, + pElementIdTab, nElements)) { + return 0; /* element instance tag not in raw_data_block() */ + } + } + for (int el = 0; el < pce->NumBackChannelElements; el++) { + if (!findElementInstanceTag(pce->BackElementTagSelect[el], + pce->BackElementIsCpe[el] ? ID_CPE : ID_SCE, + pAacDecoderChannelInfo, nChannels, + pElementIdTab, nElements)) { + return 0; /* element instance tag not in raw_data_block() */ + } + } + for (int el = 0; el < pce->NumLfeChannelElements; el++) { + if (!findElementInstanceTag(pce->LfeElementTagSelect[el], ID_LFE, + pAacDecoderChannelInfo, nChannels, + pElementIdTab, nElements)) { + return 0; /* element instance tag not in raw_data_block() */ + } + } + } else { + return 0; /* too less decoded audio channels */ + } + + return 1; /* all element instance tags found in raw_data_block() */ +} + /*! \brief Read Program Config Element @@ -568,7 +637,7 @@ 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 +653,7 @@ 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 +675,7 @@ 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 +691,15 @@ 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 +822,12 @@ 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); @@ -1396,6 +1470,27 @@ static void CAacDecoder_DeInit(HANDLE_AACDECODER self, self->samplingRateInfo[subStreamIndex].samplingRate = 0; } +/*! + * \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(self->elFlags)); + + self->flags[streamIndex] = flags; +} + /*! * \brief CAacDecoder_CtrlCFGChange Set config change parameters. * @@ -1493,6 +1588,15 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, const int streamIndex = 0; INT flushChannels = 0; + UINT flags; + /* elFlags[(3*MAX_CHANNELS + (MAX_CHANNELS)/2 + 4 * (MAX_TRACKS) + 1] + where MAX_CHANNELS is (8*2) and MAX_TRACKS is 1 */ + UINT elFlags[(3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) + 1)]; + + UCHAR sbrEnabled = self->sbrEnabled; + UCHAR sbrEnabledPrev = self->sbrEnabledPrev; + UCHAR mpsEnableCurr = self->mpsEnableCurr; + if (!self) return AAC_DEC_INVALID_HANDLE; UCHAR downscaleFactor = self->downscaleFactor; @@ -1649,8 +1753,8 @@ 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)) { @@ -1676,7 +1780,7 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, asc->m_sc.m_usacConfig.m_usacNumElements; } - self->mpsEnableCurr = 0; + mpsEnableCurr = 0; for (int _el = 0; _el < (int)self->pUsacConfig[streamIndex]->m_usacNumElements; _el++) { @@ -1696,35 +1800,34 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, self->usacStereoConfigIndex[el] = asc->m_sc.m_usacConfig.element[_el].m_stereoConfigIndex; if (self->elements[el] == ID_USAC_CPE) { - self->mpsEnableCurr |= self->usacStereoConfigIndex[el] ? 1 : 0; + mpsEnableCurr |= self->usacStereoConfigIndex[el] ? 1 : 0; } } - 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 +1894,17 @@ 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 { @@ -1825,7 +1936,7 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, self->useLdQmfTimeAlign = asc->m_sc.m_eldSpecificConfig.m_useLdQmfTimeAlign; } - if (self->sbrEnabled != asc->m_sbrPresentFlag) { + if (sbrEnabled != asc->m_sbrPresentFlag) { ascChanged = 1; } } @@ -1838,16 +1949,16 @@ 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; + sbrEnabled = 1; + sbrEnabledPrev = 1; } else { - self->sbrEnabled = 0; - self->sbrEnabledPrev = 0; + sbrEnabled = 0; + sbrEnabledPrev = 0; } - if (self->sbrEnabled && asc->m_extensionSamplingFrequency) { + if (sbrEnabled && asc->m_extensionSamplingFrequency) { if (downscaleFactor != 1 && (downscaleFactor)&1) { return AAC_DEC_UNSUPPORTED_SAMPLINGRATE; /* SBR needs an even downscale factor */ @@ -1865,51 +1976,47 @@ 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; + mpsEnableCurr = 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; + 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)) { @@ -1960,13 +2067,17 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, if (ascChanged != 0) { *configChanged = 1; } + + CAacDecoder_AcceptFlags(self, asc, flags, elFlags, streamIndex, + elementOffset); + return err; } /* set AC_USAC_SCFGI3 globally if any usac element uses */ switch (asc->m_aot) { case AOT_USAC: - if (self->sbrEnabled) { + if (sbrEnabled) { for (int _el = 0; _el < (int)self->pUsacConfig[streamIndex]->m_usacNumElements; _el++) { @@ -1988,7 +2099,7 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, } if (usacStereoConfigIndex == 3) { - self->flags[streamIndex] |= AC_USAC_SCFGI3; + flags |= AC_USAC_SCFGI3; } } break; @@ -2003,7 +2114,7 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, */ switch (asc->m_aot) { case AOT_USAC: - if (self->sbrEnabled) { + if (sbrEnabled) { const UCHAR map_sbrRatio_2_nAnaBands[] = {16, 24, 32}; FDK_ASSERT(asc->m_sc.m_usacConfig.m_sbrRatioIndex > 0); @@ -2031,11 +2142,11 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, } break; case AOT_ER_AAC_ELD: - if (self->mpsEnableCurr && + if (mpsEnableCurr && asc->m_sc.m_eldSpecificConfig.m_useLdQmfTimeAlign) { - SAC_INPUT_CONFIG sac_interface = - (self->sbrEnabled && self->hSbrDecoder) ? SAC_INTERFACE_QMF - : SAC_INTERFACE_TIME; + SAC_INPUT_CONFIG sac_interface = (sbrEnabled && self->hSbrDecoder) + ? SAC_INTERFACE_QMF + : SAC_INTERFACE_TIME; mpegSurroundDecoder_ConfigureQmfDomain( (CMpegSurroundDecoder *)self->pMpegSurroundDecoder, sac_interface, (UINT)self->streamInfo.aacSampleRate, asc->m_aot); @@ -2069,14 +2180,14 @@ 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 +2287,14 @@ 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 +2305,7 @@ 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 +2317,7 @@ 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 +2333,7 @@ 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 +2470,6 @@ 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 +2490,7 @@ 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 +2499,15 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, pcmLimiter_SetThreshold(self->hLimiter, FL2FXCONST_DBL(0.89125094f)); } + CAacDecoder_AcceptFlags(self, asc, flags, elFlags, streamIndex, + elementOffset); + self->sbrEnabled = sbrEnabled; + self->sbrEnabledPrev = sbrEnabledPrev; + self->mpsEnableCurr = mpsEnableCurr; + + /* Update externally visible copy of flags */ + self->streamInfo.flags = self->flags[0]; + return err; bail: @@ -2927,6 +3042,24 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame( } /* while ( (type != ID_END) ... ) */ + if (!(self->flags[streamIndex] & + (AC_USAC | AC_RSVD50 | AC_RSV603DA | AC_BSAC | AC_LD | AC_ELD | AC_ER | + AC_SCALABLE)) && + (self->streamInfo.channelConfig == 0) && pce->isValid && + (ErrorStatus == AAC_DEC_OK) && self->frameOK && + !(flags & (AACDEC_CONCEAL | AACDEC_FLUSH))) { + /* Check whether all PCE listed element instance tags are present in + * raw_data_block() */ + if (!validateElementInstanceTags( + &self->pce, self->pAacDecoderChannelInfo, aacChannels, + channel_elements, + fMin(channel_element_count, (int)(sizeof(channel_elements) / + sizeof(*channel_elements))))) { + ErrorStatus = AAC_DEC_DECODE_FRAME_ERROR; + self->frameOK = 0; + } + } + if (!(flags & (AACDEC_CONCEAL | AACDEC_FLUSH))) { /* float decoder checks if bitsLeft is in range 0-7; only prerollAUs are * byteAligned with respect to the first bit */ @@ -3194,11 +3327,12 @@ 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 +3575,12 @@ 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..0c83191 100644 --- a/libAACdec/src/aacdecoder_lib.cpp +++ b/libAACdec/src/aacdecoder_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 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -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); + } } } @@ -1607,6 +1626,11 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame(HANDLE_AACDECODER self, /* set params */ sbrDecoder_SetParam(self->hSbrDecoder, SBR_SYSTEM_BITSTREAM_DELAY, self->sbrParams.bsDelay); + sbrDecoder_SetParam( + self->hSbrDecoder, SBR_FLUSH_DATA, + (flags & AACDEC_FLUSH) | + ((self->flushStatus && !(flags & AACDEC_CONCEAL)) ? AACDEC_FLUSH + : 0)); sbrDecoder_SetParam(self->hSbrDecoder, SBR_SKIP_QMF, 1); @@ -1794,8 +1818,7 @@ 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 +1849,38 @@ 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 +1922,116 @@ 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/libAACdec/src/rvlc.cpp b/libAACdec/src/rvlc.cpp index b7a9be1..0b80364 100644 --- a/libAACdec/src/rvlc.cpp +++ b/libAACdec/src/rvlc.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 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -628,7 +628,7 @@ static void rvlcDecodeBackward(CErRvlcInfo *pRvlc, SHORT *pScfBwd = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd; SHORT *pScfEsc = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfEsc; - UCHAR *pEscEscCnt = &(pRvlc->numDecodedEscapeWordsEsc); + UCHAR escEscCnt = pRvlc->numDecodedEscapeWordsEsc; UCHAR *pEscBwdCnt = &(pRvlc->numDecodedEscapeWordsBwd); pRvlc->pRvlBitCnt_RVL = &(pRvlc->length_of_rvlc_sf_bwd); @@ -636,7 +636,7 @@ static void rvlcDecodeBackward(CErRvlcInfo *pRvlc, *pEscBwdCnt = 0; pRvlc->direction = BWD; - pScfEsc += *pEscEscCnt - 1; /* set pScfEsc to last entry */ + pScfEsc += escEscCnt - 1; /* set pScfEsc to last entry */ pRvlc->firstScf = 0; pRvlc->firstNrg = 0; pRvlc->firstIs = 0; @@ -651,7 +651,7 @@ static void rvlcDecodeBackward(CErRvlcInfo *pRvlc, } dpcm -= TABLE_OFFSET; if ((dpcm == MIN_RVL) || (dpcm == MAX_RVL)) { - if (pRvlc->length_of_rvlc_escapes) { + if ((pRvlc->length_of_rvlc_escapes) || (*pEscBwdCnt >= escEscCnt)) { pRvlc->conceal_min = bnds; return; } else { @@ -694,7 +694,7 @@ static void rvlcDecodeBackward(CErRvlcInfo *pRvlc, } dpcm -= TABLE_OFFSET; if ((dpcm == MIN_RVL) || (dpcm == MAX_RVL)) { - if (pRvlc->length_of_rvlc_escapes) { + if ((pRvlc->length_of_rvlc_escapes) || (*pEscBwdCnt >= escEscCnt)) { pScfBwd[bnds] = position; pRvlc->conceal_min = fMax(0, bnds - offset); return; @@ -731,7 +731,8 @@ static void rvlcDecodeBackward(CErRvlcInfo *pRvlc, } dpcm -= TABLE_OFFSET; if ((dpcm == MIN_RVL) || (dpcm == MAX_RVL)) { - if (pRvlc->length_of_rvlc_escapes) { + if ((pRvlc->length_of_rvlc_escapes) || + (*pEscBwdCnt >= escEscCnt)) { pScfBwd[bnds] = noisenrg; pRvlc->conceal_min = fMax(0, bnds - offset); return; @@ -762,7 +763,7 @@ static void rvlcDecodeBackward(CErRvlcInfo *pRvlc, } dpcm -= TABLE_OFFSET; if ((dpcm == MIN_RVL) || (dpcm == MAX_RVL)) { - if (pRvlc->length_of_rvlc_escapes) { + if ((pRvlc->length_of_rvlc_escapes) || (*pEscBwdCnt >= escEscCnt)) { pScfBwd[bnds] = factor; pRvlc->conceal_min = fMax(0, bnds - offset); return; diff --git a/libAACdec/src/usacdec_acelp.cpp b/libAACdec/src/usacdec_acelp.cpp index a8dadc0..ca1a6a2 100644 --- a/libAACdec/src/usacdec_acelp.cpp +++ b/libAACdec/src/usacdec_acelp.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 @@ -719,7 +719,7 @@ static void ConcealPitchLag(CAcelpStaticMem *acelp_mem, const int PIT_MAX, UCHAR *pold_T0_frac = &acelp_mem->old_T0_frac; if ((int)*pold_T0 >= PIT_MAX) { - *pold_T0 = (UCHAR)(PIT_MAX - 5); + *pold_T0 = (USHORT)(PIT_MAX - 5); } *pT0 = (int)*pold_T0; *pT0_frac = (int)*pold_T0_frac; diff --git a/libAACenc/include/aacenc_lib.h b/libAACenc/include/aacenc_lib.h index 71f7556..f0f23b4 100644 --- a/libAACenc/include/aacenc_lib.h +++ b/libAACenc/include/aacenc_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 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -1643,7 +1643,7 @@ AACENC_ERROR aacEncEncode(const HANDLE_AACENCODER hAacEncoder, * * \return * - AACENC_OK, on succes. - * - AACENC_INIT_ERROR, on failure. + * - AACENC_INVALID_HANDLE, AACENC_INIT_ERROR, on failure. */ AACENC_ERROR aacEncInfo(const HANDLE_AACENCODER hAacEncoder, AACENC_InfoStruct *pInfo); diff --git a/libAACenc/src/aacenc_lib.cpp b/libAACenc/src/aacenc_lib.cpp index 0ae329b..c11db27 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 - 2021 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; @@ -2520,6 +2521,11 @@ AACENC_ERROR aacEncInfo(const HANDLE_AACENCODER hAacEncoder, AACENC_InfoStruct *pInfo) { AACENC_ERROR err = AACENC_OK; + if ((hAacEncoder == NULL) || (pInfo == NULL)) { + err = AACENC_INVALID_HANDLE; + goto bail; + } + FDKmemclear(pInfo, sizeof(AACENC_InfoStruct)); pInfo->confSize = 64; /* pre-initialize */ diff --git a/libDRCdec/src/drcDec_reader.cpp b/libDRCdec/src/drcDec_reader.cpp index 367a352..b080f50 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) { @@ -914,7 +917,7 @@ static void _skipEqCoefficients(HANDLE_FDK_BITSTREAM hBs) { firFilterOrder; int uniqueEqSubbandGainsCount, eqSubbandGainRepresentation, eqSubbandGainCount; - EQ_SUBBAND_GAIN_FORMAT eqSubbandGainFormat; + int eqSubbandGainFormat; eqDelayMaxPresent = FDKreadBits(hBs, 1); if (eqDelayMaxPresent) { @@ -955,7 +958,7 @@ static void _skipEqCoefficients(HANDLE_FDK_BITSTREAM hBs) { uniqueEqSubbandGainsCount = FDKreadBits(hBs, 6); if (uniqueEqSubbandGainsCount > 0) { eqSubbandGainRepresentation = FDKreadBits(hBs, 1); - eqSubbandGainFormat = (EQ_SUBBAND_GAIN_FORMAT)FDKreadBits(hBs, 4); + eqSubbandGainFormat = FDKreadBits(hBs, 4); switch (eqSubbandGainFormat) { case GF_QMF32: eqSubbandGainCount = 32; diff --git a/libFDK/include/nlc_dec.h b/libFDK/include/nlc_dec.h index cca97f1..aded569 100644 --- a/libFDK/include/nlc_dec.h +++ b/libFDK/include/nlc_dec.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 @@ -159,9 +159,6 @@ typedef enum { #ifndef HUFFDEC_PARAMS #define HUFFDEC_PARMS -#define PAIR_SHIFT 4 -#define PAIR_MASK 0xf - #define MAX_ENTRIES 168 #define HANDLE_HUFF_NODE const SHORT(*)[MAX_ENTRIES][2] 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/autocorr2nd.cpp b/libFDK/src/autocorr2nd.cpp index 718a555..8c5673c 100644 --- a/libFDK/src/autocorr2nd.cpp +++ b/libFDK/src/autocorr2nd.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 @@ -102,11 +102,6 @@ amm-info@iis.fraunhofer.de #include "autocorr2nd.h" -/* If the accumulator does not provide enough overflow bits, - products have to be shifted down in the autocorrelation below. */ -#define SHIFT_FACTOR (5) -#define SHIFT >> (SHIFT_FACTOR) - /*! * * \brief Calculate second order autocorrelation using 2 accumulators @@ -126,45 +121,49 @@ INT autoCorr2nd_real( const FIXP_DBL *realBuf = reBuffer; + const int len_scale = fMax(DFRACT_BITS - fNormz((FIXP_DBL)(len / 2)), 1); /* r11r,r22r r01r,r12r r02r */ pReBuf = realBuf - 2; - accu5 = ((fMultDiv2(pReBuf[0], pReBuf[2]) + fMultDiv2(pReBuf[1], pReBuf[3])) - SHIFT); + accu5 = + ((fMultDiv2(pReBuf[0], pReBuf[2]) + fMultDiv2(pReBuf[1], pReBuf[3])) >> + len_scale); pReBuf++; /* len must be even */ - accu1 = fPow2Div2(pReBuf[0]) SHIFT; - accu3 = fMultDiv2(pReBuf[0], pReBuf[1]) SHIFT; + accu1 = fPow2Div2(pReBuf[0]) >> len_scale; + accu3 = fMultDiv2(pReBuf[0], pReBuf[1]) >> len_scale; pReBuf++; for (j = (len - 2) >> 1; j != 0; j--, pReBuf += 2) { - accu1 += ((fPow2Div2(pReBuf[0]) + fPow2Div2(pReBuf[1])) SHIFT); + accu1 += ((fPow2Div2(pReBuf[0]) + fPow2Div2(pReBuf[1])) >> len_scale); - accu3 += ((fMultDiv2(pReBuf[0], pReBuf[1]) + - fMultDiv2(pReBuf[1], pReBuf[2])) SHIFT); + accu3 += + ((fMultDiv2(pReBuf[0], pReBuf[1]) + fMultDiv2(pReBuf[1], pReBuf[2])) >> + len_scale); - accu5 += ((fMultDiv2(pReBuf[0], pReBuf[2]) + - fMultDiv2(pReBuf[1], pReBuf[3])) SHIFT); + accu5 += + ((fMultDiv2(pReBuf[0], pReBuf[2]) + fMultDiv2(pReBuf[1], pReBuf[3])) >> + len_scale); } - accu2 = (fPow2Div2(realBuf[-2]) SHIFT); + accu2 = (fPow2Div2(realBuf[-2]) >> len_scale); accu2 += accu1; - accu1 += (fPow2Div2(realBuf[len - 2]) SHIFT); + accu1 += (fPow2Div2(realBuf[len - 2]) >> len_scale); - accu4 = (fMultDiv2(realBuf[-1], realBuf[-2]) SHIFT); + accu4 = (fMultDiv2(realBuf[-1], realBuf[-2]) >> len_scale); accu4 += accu3; - accu3 += (fMultDiv2(realBuf[len - 1], realBuf[len - 2]) SHIFT); + accu3 += (fMultDiv2(realBuf[len - 1], realBuf[len - 2]) >> len_scale); mScale = CntLeadingZeros( (accu1 | accu2 | fAbs(accu3) | fAbs(accu4) | fAbs(accu5))) - 1; - autoCorrScaling = mScale - 1 - SHIFT_FACTOR; /* -1 because of fMultDiv2*/ + autoCorrScaling = mScale - 1 - len_scale; /* -1 because of fMultDiv2*/ /* Scale to common scale factor */ ac->r11r = accu1 << mScale; @@ -190,7 +189,7 @@ INT autoCorr2nd_cplx( const FIXP_DBL *imBuffer, /*!< Pointer to imag part of input samples */ const int len /*!< Number of input samples (should be smaller than 128) */ ) { - int j, autoCorrScaling, mScale, len_scale; + int j, autoCorrScaling, mScale; FIXP_DBL accu0, accu1, accu2, accu3, accu4, accu5, accu6, accu7, accu8; @@ -199,7 +198,7 @@ INT autoCorr2nd_cplx( const FIXP_DBL *realBuf = reBuffer; const FIXP_DBL *imagBuf = imBuffer; - (len > 64) ? (len_scale = 6) : (len_scale = 5); + const int len_scale = fMax(DFRACT_BITS - fNormz((FIXP_DBL)len), 1); /* r00r, r11r,r22r 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/libFDK/src/nlc_dec.cpp b/libFDK/src/nlc_dec.cpp index 6e98ce0..3733d98 100644 --- a/libFDK/src/nlc_dec.cpp +++ b/libFDK/src/nlc_dec.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 @@ -568,12 +568,12 @@ bail: static ERROR_t huff_decode(HANDLE_FDK_BITSTREAM strm, SCHAR* out_data_1, SCHAR* out_data_2, DATA_TYPE data_type, DIFF_TYPE diff_type_1, DIFF_TYPE diff_type_2, - int num_val, CODING_SCHEME* cdg_scheme, int ldMode) { + int num_val, PAIRING* pairing_scheme, int ldMode) { ERROR_t err = HUFFDEC_OK; + CODING_SCHEME coding_scheme = HUFF_1D; DIFF_TYPE diff_type; int i = 0; - ULONG data = 0; SCHAR pair_vec[28][2]; @@ -596,15 +596,13 @@ static ERROR_t huff_decode(HANDLE_FDK_BITSTREAM strm, SCHAR* out_data_1, int hufYY; /* Coding scheme */ - data = FDKreadBits(strm, 1); - *cdg_scheme = (CODING_SCHEME)(data << PAIR_SHIFT); + coding_scheme = (CODING_SCHEME)FDKreadBits(strm, 1); - if (*cdg_scheme >> PAIR_SHIFT == HUFF_2D) { + if (coding_scheme == HUFF_2D) { if ((out_data_1 != NULL) && (out_data_2 != NULL) && (ldMode == 0)) { - data = FDKreadBits(strm, 1); - *cdg_scheme = (CODING_SCHEME)(*cdg_scheme | data); + *pairing_scheme = (PAIRING)FDKreadBits(strm, 1); } else { - *cdg_scheme = (CODING_SCHEME)(*cdg_scheme | FREQ_PAIR); + *pairing_scheme = FREQ_PAIR; } } @@ -613,7 +611,7 @@ static ERROR_t huff_decode(HANDLE_FDK_BITSTREAM strm, SCHAR* out_data_1, hufYY2 = diff_type_2; } - switch (*cdg_scheme >> PAIR_SHIFT) { + switch (coding_scheme) { case HUFF_1D: p0_flag[0] = (diff_type_1 == DIFF_FREQ); p0_flag[1] = (diff_type_2 == DIFF_FREQ); @@ -634,7 +632,7 @@ static ERROR_t huff_decode(HANDLE_FDK_BITSTREAM strm, SCHAR* out_data_1, case HUFF_2D: - switch (*cdg_scheme & PAIR_MASK) { + switch (*pairing_scheme) { case FREQ_PAIR: if (out_data_1 != NULL) { @@ -843,7 +841,7 @@ ERROR_t EcDataPairDec(DECODER_TYPE DECODER, HANDLE_FDK_BITSTREAM strm, SCHAR* pDataVec[2] = {NULL, NULL}; DIFF_TYPE diff_type[2] = {DIFF_FREQ, DIFF_FREQ}; - CODING_SCHEME cdg_scheme = HUFF_1D; + PAIRING pairing = FREQ_PAIR; DIRECTION direction = BACKWARDS; switch (data_type) { @@ -959,7 +957,7 @@ ERROR_t EcDataPairDec(DECODER_TYPE DECODER, HANDLE_FDK_BITSTREAM strm, } /* Huffman decoding */ err = huff_decode(strm, pDataVec[0], pDataVec[1], data_type, diff_type[0], - diff_type[1], dataBands, &cdg_scheme, + diff_type[1], dataBands, &pairing, (DECODER == SAOC_DECODER)); if (err != HUFFDEC_OK) { return HUFFDEC_NOTOK; @@ -986,8 +984,8 @@ ERROR_t EcDataPairDec(DECODER_TYPE DECODER, HANDLE_FDK_BITSTREAM strm, } } - mixed_time_pair = (diff_type[0] != diff_type[1]) && - ((cdg_scheme & PAIR_MASK) == TIME_PAIR); + mixed_time_pair = + (diff_type[0] != diff_type[1]) && (pairing == TIME_PAIR); if (direction == BACKWARDS) { if (diff_type[0] == DIFF_FREQ) { 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..8f77017 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) { @@ -1549,8 +1694,7 @@ static TRANSPORTDEC_ERROR extElementConfig(CSUsacExtElementConfig *extElement, const AUDIO_OBJECT_TYPE aot) { TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK; - USAC_EXT_ELEMENT_TYPE usacExtElementType = - (USAC_EXT_ELEMENT_TYPE)escapedValue(hBs, 4, 8, 16); + UINT usacExtElementType = escapedValue(hBs, 4, 8, 16); /* recurve extension elements which are invalid for USAC */ if (aot == AOT_USAC) { @@ -1567,7 +1711,6 @@ static TRANSPORTDEC_ERROR extElementConfig(CSUsacExtElementConfig *extElement, } } - extElement->usacExtElementType = usacExtElementType; int usacExtElementConfigLength = escapedValue(hBs, 4, 8, 16); extElement->usacExtElementConfigLength = (USHORT)usacExtElementConfigLength; INT bsAnchor; @@ -1601,8 +1744,10 @@ static TRANSPORTDEC_ERROR extElementConfig(CSUsacExtElementConfig *extElement, } } break; default: + usacExtElementType = ID_EXT_ELE_UNKNOWN; break; } + extElement->usacExtElementType = (USAC_EXT_ELEMENT_TYPE)usacExtElementType; /* Adjust bit stream position. This is required because of byte alignment and * unhandled extensions. */ @@ -1631,14 +1776,18 @@ static TRANSPORTDEC_ERROR configExtension(CSUsacConfig *usc, TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK; int numConfigExtensions; - CONFIG_EXT_ID usacConfigExtType; + UINT 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++) { INT nbits; int loudnessInfoSetConfigExtensionPosition = FDKgetValidBits(hBs); - usacConfigExtType = (CONFIG_EXT_ID)escapedValue(hBs, 4, 8, 16); + usacConfigExtType = escapedValue(hBs, 4, 8, 16); usacConfigExtLength = (int)escapedValue(hBs, 4, 8, 16); /* Start bit position of config extension */ @@ -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/libMpegTPDec/src/tpdec_latm.cpp b/libMpegTPDec/src/tpdec_latm.cpp index 3b71db8..c32be54 100644 --- a/libMpegTPDec/src/tpdec_latm.cpp +++ b/libMpegTPDec/src/tpdec_latm.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 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -591,6 +591,18 @@ bail: return (ErrorStatus); } +static int CLatmDemux_ReadAuChunkLengthInfo(HANDLE_FDK_BITSTREAM bs) { + int len = 0, tmp = 255; + int validBytes = (int)FDKgetValidBits(bs) >> 3; + + while (tmp == 255 && validBytes-- > 0) { + tmp = (int)FDKreadBits(bs, 8); + len += tmp; + } + + return ((tmp == 255) ? -1 : (len << 3)); +} + TRANSPORTDEC_ERROR CLatmDemux_ReadPayloadLengthInfo(HANDLE_FDK_BITSTREAM bs, CLatmDemux *pLatmDemux) { TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK; @@ -602,11 +614,17 @@ TRANSPORTDEC_ERROR CLatmDemux_ReadPayloadLengthInfo(HANDLE_FDK_BITSTREAM bs, FDK_ASSERT(pLatmDemux->m_numLayer[prog] <= LATM_MAX_LAYER); for (UINT lay = 0; lay < pLatmDemux->m_numLayer[prog]; lay++) { LATM_LAYER_INFO *p_linfo = &pLatmDemux->m_linfo[prog][lay]; + int auChunkLengthInfo = 0; switch (p_linfo->m_frameLengthType) { case 0: - p_linfo->m_frameLengthInBits = CLatmDemux_ReadAuChunkLengthInfo(bs); - totalPayloadBits += p_linfo->m_frameLengthInBits; + auChunkLengthInfo = CLatmDemux_ReadAuChunkLengthInfo(bs); + if (auChunkLengthInfo >= 0) { + p_linfo->m_frameLengthInBits = (UINT)auChunkLengthInfo; + totalPayloadBits += p_linfo->m_frameLengthInBits; + } else { + return TRANSPORTDEC_PARSE_ERROR; + } break; case 3: case 5: @@ -627,23 +645,6 @@ TRANSPORTDEC_ERROR CLatmDemux_ReadPayloadLengthInfo(HANDLE_FDK_BITSTREAM bs, return (ErrorStatus); } -int CLatmDemux_ReadAuChunkLengthInfo(HANDLE_FDK_BITSTREAM bs) { - UCHAR endFlag; - int len = 0; - - do { - UCHAR tmp = (UCHAR)FDKreadBits(bs, 8); - endFlag = (tmp < 255); - - len += tmp; - - } while (endFlag == 0); - - len <<= 3; /* convert from bytes to bits */ - - return len; -} - UINT CLatmDemux_GetFrameLengthInBits(CLatmDemux *pLatmDemux, const UINT prog, const UINT layer) { UINT nFrameLenBits = 0; diff --git a/libMpegTPDec/src/tpdec_latm.h b/libMpegTPDec/src/tpdec_latm.h index 6af553d..8b8c971 100644 --- a/libMpegTPDec/src/tpdec_latm.h +++ b/libMpegTPDec/src/tpdec_latm.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 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -151,8 +151,6 @@ typedef struct { AudioPreRoll */ } CLatmDemux; -int CLatmDemux_ReadAuChunkLengthInfo(HANDLE_FDK_BITSTREAM bs); - TRANSPORTDEC_ERROR CLatmDemux_Read(HANDLE_FDK_BITSTREAM bs, CLatmDemux *pLatmDemux, TRANSPORT_TYPE tt, CSTpCallBacks *pTpDecCallbacks, 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/libPCMutils/src/limiter.cpp b/libPCMutils/src/limiter.cpp index 598dc0c..c6b8687 100644 --- a/libPCMutils/src/limiter.cpp +++ b/libPCMutils/src/limiter.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 @@ -322,7 +322,8 @@ TDLIMITER_ERROR pcmLimiter_Apply(TDLimiterPtr limiter, PCM_LIM* samplesIn, (FIXP_DBL)SATURATE_LEFT_SHIFT(tmp, scaling, DFRACT_BITS)); #else samplesOut[j] = (INT_PCM)FX_DBL2FX_PCM((FIXP_DBL)SATURATE_LEFT_SHIFT( - tmp + ((FIXP_DBL)0x8000 >> scaling), scaling, DFRACT_BITS)); + (tmp >> 1) + ((FIXP_DBL)0x8000 >> (scaling + 1)), scaling + 1, + DFRACT_BITS)); #endif } } diff --git a/libPCMutils/src/pcmdmx_lib.cpp b/libPCMutils/src/pcmdmx_lib.cpp index 2070dbc..fca12ce 100644 --- a/libPCMutils/src/pcmdmx_lib.cpp +++ b/libPCMutils/src/pcmdmx_lib.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 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -494,13 +494,40 @@ static PCM_DMX_CHANNEL_MODE getChMode4Plain( return plainChMode; } -static inline UINT getIdxSum(UCHAR numCh) { - UINT result = 0; - int i; - for (i = 1; i < numCh; i += 1) { - result += i; +/** Validates the channel indices of all channels present in the bitstream. + * The channel indices have to be consecutive and unique for each audio channel + *type. + * @param [in] The total number of channels of the given configuration. + * @param [in] The total number of channels of the current audio channel type of + *the given configuration. + * @param [in] Audio channel type to be examined. + * @param [in] Array holding the corresponding channel types for each channel. + * @param [in] Array holding the corresponding channel type indices for each + *channel. + * @returns Returns 1 on success, returns 0 on error. + **/ +static UINT validateIndices(UINT numChannels, UINT numChannelsPlaneAndGrp, + AUDIO_CHANNEL_TYPE aChType, + const AUDIO_CHANNEL_TYPE channelType[], + const UCHAR channelIndices[]) { + for (UINT reqValue = 0; reqValue < numChannelsPlaneAndGrp; reqValue++) { + int found = FALSE; + for (UINT i = 0; i < numChannels; i++) { + if (channelType[i] == aChType) { + if (channelIndices[i] == reqValue) { + if (found == TRUE) { + return 0; /* Found channel index a second time */ + } else { + found = TRUE; /* Found channel index */ + } + } + } + } + if (found == FALSE) { + return 0; /* Did not find channel index */ + } } - return result; + return 1; /* Successfully validated channel indices */ } /** Evaluate a given channel configuration and extract a packed channel mode. In @@ -523,7 +550,6 @@ static PCMDMX_ERROR getChannelMode( UCHAR offsetTable[(8)], /* out */ PCM_DMX_CHANNEL_MODE *chMode /* out */ ) { - UINT idxSum[(3)][(4)]; UCHAR numCh[(3)][(4)]; UCHAR mapped[(8)]; PCM_DMX_SPEAKER_POSITION spkrPos[(8)]; @@ -538,7 +564,6 @@ static PCMDMX_ERROR getChannelMode( FDK_ASSERT(chMode != NULL); /* For details see ISO/IEC 13818-7:2005(E), 8.5.3 Channel configuration */ - FDKmemclear(idxSum, (3) * (4) * sizeof(UINT)); FDKmemclear(numCh, (3) * (4) * sizeof(UCHAR)); FDKmemclear(mapped, (8) * sizeof(UCHAR)); FDKmemclear(spkrPos, (8) * sizeof(PCM_DMX_SPEAKER_POSITION)); @@ -552,19 +577,22 @@ static PCMDMX_ERROR getChannelMode( (channelType[ch] & 0x0F) - 1, 0); /* Assign all undefined channels (ACT_NONE) to front channels. */ numCh[channelType[ch] >> 4][chGrp] += 1; - idxSum[channelType[ch] >> 4][chGrp] += channelIndices[ch]; } - if (numChannels > TWO_CHANNEL) { + + { int chGrp; /* Sanity check on the indices */ for (chGrp = 0; chGrp < (4); chGrp += 1) { int plane; for (plane = 0; plane < (3); plane += 1) { - if (idxSum[plane][chGrp] != getIdxSum(numCh[plane][chGrp])) { + if (numCh[plane][chGrp] == 0) continue; + AUDIO_CHANNEL_TYPE aChType = + (AUDIO_CHANNEL_TYPE)((plane << 4) | ((chGrp + 1) & 0xF)); + if (!validateIndices(numChannels, numCh[plane][chGrp], aChType, + channelType, channelIndices)) { unsigned idxCnt = 0; for (ch = 0; ch < numChannels; ch += 1) { - if (channelType[ch] == - (AUDIO_CHANNEL_TYPE)((plane << 4) | ((chGrp + 1) & 0xF))) { + if (channelType[ch] == aChType) { channelIndices[ch] = idxCnt++; } } 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_bitdec.cpp b/libSACdec/src/sac_bitdec.cpp index 4485ccf..25b3d9e 100644 --- a/libSACdec/src/sac_bitdec.cpp +++ b/libSACdec/src/sac_bitdec.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 @@ -488,12 +488,17 @@ SACDEC_ERROR SpatialDecParseSpecificConfig( pSpatialSpecificConfig->freqRes = (SPATIALDEC_FREQ_RES)freqResTable_LD[bsFreqRes]; - pSpatialSpecificConfig->treeConfig = - (SPATIALDEC_TREE_CONFIG)FDKreadBits(bitstream, 4); + { + UINT treeConfig = FDKreadBits(bitstream, 4); - if (pSpatialSpecificConfig->treeConfig != SPATIALDEC_MODE_RSVD7) { - err = MPS_UNSUPPORTED_CONFIG; - goto bail; + switch (treeConfig) { + case SPATIALDEC_MODE_RSVD7: + pSpatialSpecificConfig->treeConfig = (SPATIALDEC_TREE_CONFIG)treeConfig; + break; + default: + err = MPS_UNSUPPORTED_CONFIG; + goto bail; + } } { 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_process.cpp b/libSACdec/src/sac_process.cpp index 22091a9..33a1647 100644 --- a/libSACdec/src/sac_process.cpp +++ b/libSACdec/src/sac_process.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 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -517,12 +517,11 @@ SACDEC_ERROR SpatialDecApplyM2_Mode212_ResidualsPlusPhaseCoding( maxVal = fAbs(iReal0) | fAbs(iImag0); maxVal |= fAbs(iReal1); - s = fMax(CntLeadingZeros(maxVal) - 1, 0); - s = fMin(s, scale_param_m2); + s = fMin(CntLeadingZeros(maxVal) - 2, scale_param_m2); - mReal0 = iReal0 << s; - mImag0 = iImag0 << s; - mReal1 = iReal1 << s; + mReal0 = scaleValue(iReal0, s); + mImag0 = scaleValue(iImag0, s); + mReal1 = scaleValue(iReal1, s); s = scale_param_m2 - s; @@ -562,12 +561,11 @@ SACDEC_ERROR SpatialDecApplyM2_Mode212_ResidualsPlusPhaseCoding( maxVal = fAbs(iReal0) | fAbs(iImag0); maxVal |= fAbs(iReal1); - s = fMax(CntLeadingZeros(maxVal) - 1, 0); - s = fMin(s, scale_param_m2); + s = fMin(CntLeadingZeros(maxVal) - 2, scale_param_m2); - mReal0 = FX_DBL2FX_SGL(iReal0 << s); - mImag0 = FX_DBL2FX_SGL(iImag0 << s); - mReal1 = FX_DBL2FX_SGL(iReal1 << s); + mReal0 = FX_DBL2FX_SGL(scaleValue(iReal0, s)); + mImag0 = FX_DBL2FX_SGL(scaleValue(iImag0, s)); + mReal1 = FX_DBL2FX_SGL(scaleValue(iReal1, s)); s = scale_param_m2 - s; diff --git a/libSACdec/src/sac_reshapeBBEnv.cpp b/libSACdec/src/sac_reshapeBBEnv.cpp index 272d009..72f4e58 100644 --- a/libSACdec/src/sac_reshapeBBEnv.cpp +++ b/libSACdec/src/sac_reshapeBBEnv.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 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -241,29 +241,56 @@ static inline void combineDryWet(FIXP_DBL *RESTRICT pReal, } } -static inline void slotAmp(FIXP_DBL *RESTRICT slotAmp_dry, - FIXP_DBL *RESTRICT slotAmp_wet, - FIXP_DBL *RESTRICT pHybOutputRealDry, - FIXP_DBL *RESTRICT pHybOutputImagDry, - FIXP_DBL *RESTRICT pHybOutputRealWet, - FIXP_DBL *RESTRICT pHybOutputImagWet, INT cplxBands, - INT hybBands) { - INT qs; +static inline void slotAmp( + FIXP_DBL *RESTRICT slotAmp_dry, INT *RESTRICT slotAmp_dry_e, + FIXP_DBL *RESTRICT slotAmp_wet, INT *RESTRICT slotAmp_wet_e, + FIXP_DBL *RESTRICT pHybOutputRealDry, FIXP_DBL *RESTRICT pHybOutputImagDry, + FIXP_DBL *RESTRICT pHybOutputRealWet, FIXP_DBL *RESTRICT pHybOutputImagWet, + INT cplxBands, INT hybBands) { + INT qs, s1, s2, headroom_dry, headroom_wet; FIXP_DBL dry, wet; + /* headroom can be reduced by 1 bit due to use of fPow2Div2 */ + s1 = DFRACT_BITS - 1 - CntLeadingZeros(hybBands + cplxBands); + headroom_dry = fMin(getScalefactor(pHybOutputRealDry, hybBands), + getScalefactor(pHybOutputImagDry, cplxBands)); + headroom_wet = fMin(getScalefactor(pHybOutputRealWet, hybBands), + getScalefactor(pHybOutputImagWet, cplxBands)); + dry = wet = FL2FXCONST_DBL(0.0f); for (qs = 0; qs < cplxBands; qs++) { - dry = fAddSaturate(dry, fPow2Div2(pHybOutputRealDry[qs] << (1)) + - fPow2Div2(pHybOutputImagDry[qs] << (1))); - wet = fAddSaturate(wet, fPow2Div2(pHybOutputRealWet[qs] << (1)) + - fPow2Div2(pHybOutputImagWet[qs] << (1))); + /* sum up dry part */ + dry += (fPow2Div2(pHybOutputRealDry[qs] << headroom_dry) >> s1); + dry += (fPow2Div2(pHybOutputImagDry[qs] << headroom_dry) >> s1); + /* sum up wet part */ + wet += (fPow2Div2(pHybOutputRealWet[qs] << headroom_wet) >> s1); + wet += (fPow2Div2(pHybOutputImagWet[qs] << headroom_wet) >> s1); } for (; qs < hybBands; qs++) { - dry = fAddSaturate(dry, fPow2Div2(pHybOutputRealDry[qs] << (1))); - wet = fAddSaturate(wet, fPow2Div2(pHybOutputRealWet[qs] << (1))); + dry += (fPow2Div2(pHybOutputRealDry[qs] << headroom_dry) >> s1); + wet += (fPow2Div2(pHybOutputRealWet[qs] << headroom_wet) >> s1); + } + + /* consider fPow2Div2() */ + s1 += 1; + + /* normalize dry part, ensure that exponent is even */ + s2 = fixMax(0, CntLeadingZeros(dry) - 1); + *slotAmp_dry = dry << s2; + *slotAmp_dry_e = s1 - s2 - 2 * headroom_dry; + if (*slotAmp_dry_e & 1) { + *slotAmp_dry = *slotAmp_dry >> 1; + *slotAmp_dry_e += 1; + } + + /* normalize wet part, ensure that exponent is even */ + s2 = fixMax(0, CntLeadingZeros(wet) - 1); + *slotAmp_wet = wet << s2; + *slotAmp_wet_e = s1 - s2 - 2 * headroom_wet; + if (*slotAmp_wet_e & 1) { + *slotAmp_wet = *slotAmp_wet >> 1; + *slotAmp_wet_e += 1; } - *slotAmp_dry = dry >> (2 * (1)); - *slotAmp_wet = wet >> (2 * (1)); } #if defined(__aarch64__) @@ -533,6 +560,7 @@ void SpatialDecReshapeBBEnv(spatialDec *self, const SPATIAL_BS_FRAME *frame, INT ts) { INT ch, scale; INT dryFacSF, slotAmpSF; + INT slotAmp_dry_e, slotAmp_wet_e; FIXP_DBL tmp, dryFac, envShape; FIXP_DBL slotAmp_dry, slotAmp_wet, slotAmp_ratio; FIXP_DBL envDry[MAX_OUTPUT_CHANNELS], envDmx[2]; @@ -594,22 +622,25 @@ void SpatialDecReshapeBBEnv(spatialDec *self, const SPATIAL_BS_FRAME *frame, dryFacSF = SF_SHAPE + 2 * dryFacSF; } + slotAmp_dry_e = slotAmp_wet_e = 0; + /* calculate slotAmp_dry and slotAmp_wet */ - slotAmp(&slotAmp_dry, &slotAmp_wet, &self->hybOutputRealDry__FDK[ch][6], + slotAmp(&slotAmp_dry, &slotAmp_dry_e, &slotAmp_wet, &slotAmp_wet_e, + &self->hybOutputRealDry__FDK[ch][6], &self->hybOutputImagDry__FDK[ch][6], &self->hybOutputRealWet__FDK[ch][6], &self->hybOutputImagWet__FDK[ch][6], cplxBands, hybBands); + /* exponents must be even due to subsequent square root calculation */ + FDK_ASSERT(((slotAmp_dry_e & 1) == 0) && ((slotAmp_wet_e & 1) == 0)); + /* slotAmp_ratio will be scaled by slotAmpSF bits */ if (slotAmp_dry != FL2FXCONST_DBL(0.0f)) { - sc = fixMax(0, CntLeadingZeros(slotAmp_wet) - 1); - sc = sc - (sc & 1); - - slotAmp_wet = sqrtFixp(slotAmp_wet << sc); + slotAmp_wet = sqrtFixp(slotAmp_wet); slotAmp_dry = invSqrtNorm2(slotAmp_dry, &slotAmpSF); slotAmp_ratio = fMult(slotAmp_wet, slotAmp_dry); - slotAmpSF = slotAmpSF - (sc >> 1); + slotAmpSF = slotAmpSF + (slotAmp_wet_e >> 1) - (slotAmp_dry_e >> 1); } /* calculate common scale factor */ diff --git a/libSACdec/src/sac_stp.cpp b/libSACdec/src/sac_stp.cpp index bb66277..0e6affa 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 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -229,15 +229,13 @@ inline void combineSignalCplxScale1(FIXP_DBL *hybOutputRealDry, int n; FIXP_DBL scaleY; for (n = bands - 1; n >= 0; n--) { - scaleY = fMultDiv2(scaleX, *pBP); + scaleY = fMult(scaleX, *pBP); *hybOutputRealDry = SATURATE_LEFT_SHIFT( - (*hybOutputRealDry >> 1) + - (fMultDiv2(*hybOutputRealWet, scaleY) << (SF_SCALE + 1)), - 1, DFRACT_BITS); + (*hybOutputRealDry >> SF_SCALE) + fMult(*hybOutputRealWet, scaleY), + SF_SCALE, DFRACT_BITS); *hybOutputImagDry = SATURATE_LEFT_SHIFT( - (*hybOutputImagDry >> 1) + - (fMultDiv2(*hybOutputImagWet, scaleY) << (SF_SCALE + 1)), - 1, DFRACT_BITS); + (*hybOutputImagDry >> SF_SCALE) + fMult(*hybOutputImagWet, scaleY), + SF_SCALE, DFRACT_BITS); hybOutputRealDry++, hybOutputRealWet++; hybOutputImagDry++, hybOutputImagWet++; pBP++; @@ -252,12 +250,12 @@ inline void combineSignalCplxScale2(FIXP_DBL *hybOutputRealDry, int n; for (n = bands - 1; n >= 0; n--) { - *hybOutputRealDry = - *hybOutputRealDry + - (fMultDiv2(*hybOutputRealWet, scaleX) << (SF_SCALE + 1)); - *hybOutputImagDry = - *hybOutputImagDry + - (fMultDiv2(*hybOutputImagWet, scaleX) << (SF_SCALE + 1)); + *hybOutputRealDry = SATURATE_LEFT_SHIFT( + (*hybOutputRealDry >> SF_SCALE) + fMult(*hybOutputRealWet, scaleX), + SF_SCALE, DFRACT_BITS); + *hybOutputImagDry = SATURATE_LEFT_SHIFT( + (*hybOutputImagDry >> SF_SCALE) + fMult(*hybOutputImagWet, scaleX), + SF_SCALE, DFRACT_BITS); hybOutputRealDry++, hybOutputRealWet++; hybOutputImagDry++, hybOutputImagWet++; } @@ -369,15 +367,15 @@ 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/arm/lpp_tran_arm.cpp b/libSBRdec/src/arm/lpp_tran_arm.cpp deleted file mode 100644 index db1948f..0000000 --- a/libSBRdec/src/arm/lpp_tran_arm.cpp +++ /dev/null @@ -1,159 +0,0 @@ -/* ----------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten -Forschung e.V. All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software -that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding -scheme for digital audio. This FDK AAC Codec software is intended to be used on -a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient -general perceptual audio codecs. AAC-ELD is considered the best-performing -full-bandwidth communications codec by independent studies and is widely -deployed. AAC has been standardized by ISO and IEC as part of the MPEG -specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including -those of Fraunhofer) may be obtained through Via Licensing -(www.vialicensing.com) or through the respective patent owners individually for -the purpose of encoding or decoding bit streams in products that are compliant -with the ISO/IEC MPEG audio standards. Please note that most manufacturers of -Android devices already license these patent claims through Via Licensing or -directly from the patent owners, and therefore FDK AAC Codec software may -already be covered under those patent licenses when it is used for those -licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions -with enhanced sound quality, are also available from Fraunhofer. Users are -encouraged to check the Fraunhofer website for additional applications -information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, -are permitted without payment of copyright license fees provided that you -satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of -the FDK AAC Codec or your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation -and/or other materials provided with redistributions of the FDK AAC Codec or -your modifications thereto in binary form. You must make available free of -charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived -from this library without prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute -the FDK AAC Codec software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating -that you changed the software and the date of any change. For modified versions -of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" -must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK -AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without -limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. -Fraunhofer provides no warranty of patent non-infringement with respect to this -software. - -You may use this FDK AAC Codec software or modifications thereto only for -purposes that are authorized by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright -holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, -including but not limited to the implied warranties of merchantability and -fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, -or consequential damages, including but not limited to procurement of substitute -goods or services; loss of use, data, or profits, or business interruption, -however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of -this software, even if advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------ */ - -/**************************** SBR decoder library ****************************** - - Author(s): Arthur Tritthart - - Description: (ARM optimised) LPP transposer subroutines - -*******************************************************************************/ - -#if defined(__arm__) - -#define FUNCTION_LPPTRANSPOSER_func1 - -#ifdef FUNCTION_LPPTRANSPOSER_func1 - -/* Note: This code requires only 43 cycles per iteration instead of 61 on - * ARM926EJ-S */ -static void lppTransposer_func1(FIXP_DBL *lowBandReal, FIXP_DBL *lowBandImag, - FIXP_DBL **qmfBufferReal, - FIXP_DBL **qmfBufferImag, int loops, int hiBand, - int dynamicScale, int descale, FIXP_SGL a0r, - FIXP_SGL a0i, FIXP_SGL a1r, FIXP_SGL a1i, - const int fPreWhitening, - FIXP_DBL preWhiteningGain, - int preWhiteningGains_sf) { - FIXP_DBL real1, real2, imag1, imag2, accu1, accu2; - - real2 = lowBandReal[-2]; - real1 = lowBandReal[-1]; - imag2 = lowBandImag[-2]; - imag1 = lowBandImag[-1]; - for (int i = 0; i < loops; i++) { - accu1 = fMultDiv2(a0r, real1); - accu2 = fMultDiv2(a0i, imag1); - accu1 = fMultAddDiv2(accu1, a1r, real2); - accu2 = fMultAddDiv2(accu2, a1i, imag2); - real2 = fMultDiv2(a1i, real2); - accu1 = accu1 - accu2; - accu1 = accu1 >> dynamicScale; - - accu2 = fMultAddDiv2(real2, a1r, imag2); - real2 = real1; - imag2 = imag1; - accu2 = fMultAddDiv2(accu2, a0i, real1); - real1 = lowBandReal[i]; - accu2 = fMultAddDiv2(accu2, a0r, imag1); - imag1 = lowBandImag[i]; - accu2 = accu2 >> dynamicScale; - - accu1 <<= 1; - accu2 <<= 1; - accu1 += (real1 >> descale); - accu2 += (imag1 >> descale); - if (fPreWhitening) { - accu1 = scaleValueSaturate(fMultDiv2(accu1, preWhiteningGain), - preWhiteningGains_sf); - accu2 = scaleValueSaturate(fMultDiv2(accu2, preWhiteningGain), - preWhiteningGains_sf); - } - qmfBufferReal[i][hiBand] = accu1; - qmfBufferImag[i][hiBand] = accu2; - } -} -#endif /* #ifdef FUNCTION_LPPTRANSPOSER_func1 */ - -#endif /* __arm__ */ diff --git a/libSBRdec/src/env_calc.cpp b/libSBRdec/src/env_calc.cpp index 0b2f651..cefa612 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 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -664,7 +664,7 @@ static void apply_inter_tes(FIXP_DBL **qmfReal, FIXP_DBL **qmfImag, gain_sf[i] = mult_sf - total_power_low_sf + sf2; gain[i] = sqrtFixp_lookup(gain[i], &gain_sf[i]); if (gain_sf[i] < 0) { - gain[i] >>= -gain_sf[i]; + gain[i] >>= fMin(DFRACT_BITS - 1, -gain_sf[i]); gain_sf[i] = 0; } } else { @@ -683,11 +683,6 @@ static void apply_inter_tes(FIXP_DBL **qmfReal, FIXP_DBL **qmfImag, /* gain[i] = g_inter[i] */ for (i = 0; i < nbSubsample; ++i) { - if (gain_sf[i] < 0) { - gain[i] >>= -gain_sf[i]; - gain_sf[i] = 0; - } - /* calculate: gain[i] = 1.0f + gamma * (gain[i] - 1.0f); */ FIXP_DBL one = (FIXP_DBL)MAXVAL_DBL >> gain_sf[i]; /* to substract this from gain[i] */ @@ -755,23 +750,15 @@ static void apply_inter_tes(FIXP_DBL **qmfReal, FIXP_DBL **qmfImag, int gain_adj_sf = gain_adj_2_sf; for (i = 0; i < nbSubsample; ++i) { - gain[i] = fMult(gain[i], gain_adj); - gain_sf[i] += gain_adj_sf; - - /* limit gain */ - if (gain_sf[i] > INTER_TES_SF_CHANGE) { - gain[i] = (FIXP_DBL)MAXVAL_DBL; - gain_sf[i] = INTER_TES_SF_CHANGE; - } - } - - for (i = 0; i < nbSubsample; ++i) { - /* equalize gain[]'s scale factors */ - gain[i] >>= INTER_TES_SF_CHANGE - gain_sf[i]; + int gain_e = fMax( + fMin(gain_sf[i] + gain_adj_sf - INTER_TES_SF_CHANGE, DFRACT_BITS - 1), + -(DFRACT_BITS - 1)); + FIXP_DBL gain_final = fMult(gain[i], gain_adj); + gain_final = scaleValueSaturate(gain_final, gain_e); for (j = lowSubband; j < highSubband; j++) { - qmfReal[startPos + i][j] = fMult(qmfReal[startPos + i][j], gain[i]); - qmfImag[startPos + i][j] = fMult(qmfImag[startPos + i][j], gain[i]); + qmfReal[startPos + i][j] = fMult(qmfReal[startPos + i][j], gain_final); + qmfImag[startPos + i][j] = fMult(qmfImag[startPos + i][j], gain_final); } } } else { /* gamma_idx == 0 */ @@ -1398,6 +1385,17 @@ 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 +2739,9 @@ 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 +2931,9 @@ 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/hbe.cpp b/libSBRdec/src/hbe.cpp index d210bb6..f2452ea 100644 --- a/libSBRdec/src/hbe.cpp +++ b/libSBRdec/src/hbe.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 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -1400,42 +1400,27 @@ void QmfTransposerApply(HANDLE_HBE_TRANSPOSER hQmfTransposer, if (shift_ov != 0) { for (i = 0; i < HBE_MAX_OUT_SLOTS; i++) { - for (band = 0; band < QMF_SYNTH_CHANNELS; band++) { - if (shift_ov >= 0) { - hQmfTransposer->qmfHBEBufReal_F[i][band] <<= shift_ov; - hQmfTransposer->qmfHBEBufImag_F[i][band] <<= shift_ov; - } else { - hQmfTransposer->qmfHBEBufReal_F[i][band] >>= (-shift_ov); - hQmfTransposer->qmfHBEBufImag_F[i][band] >>= (-shift_ov); - } - } - } - } - - if ((keepStatesSyncedMode == KEEP_STATES_SYNCED_OFF) && shift_ov != 0) { - for (i = timeStep * firstSlotOffsset; i < ov_len; i++) { - for (band = hQmfTransposer->startBand; band < hQmfTransposer->stopBand; - band++) { - if (shift_ov >= 0) { - ppQmfBufferOutReal_F[i][band] <<= shift_ov; - ppQmfBufferOutImag_F[i][band] <<= shift_ov; - } else { - ppQmfBufferOutReal_F[i][band] >>= (-shift_ov); - ppQmfBufferOutImag_F[i][band] >>= (-shift_ov); - } - } + scaleValuesSaturate(&hQmfTransposer->qmfHBEBufReal_F[i][0], + QMF_SYNTH_CHANNELS, shift_ov); + scaleValuesSaturate(&hQmfTransposer->qmfHBEBufImag_F[i][0], + QMF_SYNTH_CHANNELS, shift_ov); } - /* shift lpc filterstates */ - for (i = 0; i < timeStep * firstSlotOffsset + LPC_ORDER; i++) { - for (band = 0; band < (64); band++) { - if (shift_ov >= 0) { - lpcFilterStatesReal[i][band] <<= shift_ov; - lpcFilterStatesImag[i][band] <<= shift_ov; - } else { - lpcFilterStatesReal[i][band] >>= (-shift_ov); - lpcFilterStatesImag[i][band] >>= (-shift_ov); - } + if (keepStatesSyncedMode == KEEP_STATES_SYNCED_OFF) { + int nBands = + fMax(0, hQmfTransposer->stopBand - hQmfTransposer->startBand); + + for (i = timeStep * firstSlotOffsset; i < ov_len; i++) { + scaleValuesSaturate(&ppQmfBufferOutReal_F[i][hQmfTransposer->startBand], + nBands, shift_ov); + scaleValuesSaturate(&ppQmfBufferOutImag_F[i][hQmfTransposer->startBand], + nBands, shift_ov); + } + + /* shift lpc filterstates */ + for (i = 0; i < timeStep * firstSlotOffsset + LPC_ORDER; i++) { + scaleValuesSaturate(&lpcFilterStatesReal[i][0], (64), shift_ov); + scaleValuesSaturate(&lpcFilterStatesImag[i][0], (64), shift_ov); } } } diff --git a/libSBRdec/src/lpp_tran.cpp b/libSBRdec/src/lpp_tran.cpp index 93e1158..68a25bf 100644 --- a/libSBRdec/src/lpp_tran.cpp +++ b/libSBRdec/src/lpp_tran.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 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -132,10 +132,6 @@ amm-info@iis.fraunhofer.de #include "HFgen_preFlat.h" -#if defined(__arm__) -#include "arm/lpp_tran_arm.cpp" -#endif - #define LPC_SCALE_FACTOR 2 /*! @@ -220,19 +216,21 @@ static inline void calc_qmfBufferReal(FIXP_DBL **qmfBufferReal, const FIXP_DBL *const lowBandReal, const int startSample, const int stopSample, const UCHAR hiBand, - const int dynamicScale, const int descale, + const int dynamicScale, const FIXP_SGL a0r, const FIXP_SGL a1r) { - FIXP_DBL accu1, accu2; - int i; + const int dynscale = fixMax(0, dynamicScale - 1) + 1; + const int rescale = -fixMin(0, dynamicScale - 1) + 1; + const int descale = + fixMin(DFRACT_BITS - 1, LPC_SCALE_FACTOR + dynamicScale + rescale); - for (i = 0; i < stopSample - startSample; i++) { - accu1 = fMultDiv2(a1r, lowBandReal[i]); - accu1 = (fMultDiv2(a0r, lowBandReal[i + 1]) + accu1); - accu1 = accu1 >> dynamicScale; + for (int i = 0; i < stopSample - startSample; i++) { + FIXP_DBL accu; - accu1 <<= 1; - accu2 = (lowBandReal[i + 2] >> descale); - qmfBufferReal[i + startSample][hiBand] = accu1 + accu2; + accu = fMultDiv2(a1r, lowBandReal[i]) + fMultDiv2(a0r, lowBandReal[i + 1]); + accu = (lowBandReal[i + 2] >> descale) + (accu >> dynscale); + + qmfBufferReal[i + startSample][hiBand] = + SATURATE_LEFT_SHIFT(accu, rescale, DFRACT_BITS); } } @@ -529,7 +527,7 @@ void lppTransposer( if ((scale > 0) && (result >= (FIXP_DBL)MAXVAL_DBL >> scale)) { resetLPCCoeffs = 1; } else { - alphar[1] = FX_DBL2FX_SGL(scaleValue(result, scale)); + alphar[1] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale)); if ((tmp < FL2FX_DBL(0.0f)) ^ (ac.det < FL2FX_DBL(0.0f))) { alphar[1] = -alphar[1]; } @@ -557,7 +555,7 @@ void lppTransposer( scale)) { resetLPCCoeffs = 1; } else { - alphai[1] = FX_DBL2FX_SGL(scaleValue(result, scale)); + alphai[1] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale)); if ((tmp < FL2FX_DBL(0.0f)) ^ (ac.det < FL2FX_DBL(0.0f))) { alphai[1] = -alphai[1]; } @@ -596,7 +594,7 @@ void lppTransposer( } else { INT scale; FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale); - alphar[0] = FX_DBL2FX_SGL(scaleValue(result, scale + 1)); + alphar[0] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale + 1)); if ((tmp > FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f))) alphar[0] = -alphar[0]; @@ -616,7 +614,7 @@ void lppTransposer( } else { INT scale; FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale); - alphai[0] = FX_DBL2FX_SGL(scaleValue(result, scale + 1)); + alphai[0] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale + 1)); if ((tmp > FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f))) alphai[0] = -alphai[0]; } @@ -659,7 +657,7 @@ void lppTransposer( INT scale; FIXP_DBL result = fDivNorm(fixp_abs(ac.r01r), fixp_abs(ac.r11r), &scale); - k1 = scaleValue(result, scale); + k1 = scaleValueSaturate(result, scale); if (!((ac.r01r < FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f)))) { k1 = -k1; @@ -771,52 +769,50 @@ void lppTransposer( } else { /* bw <= 0 */ if (!useLP) { - int descale = - fixMin(DFRACT_BITS - 1, (LPC_SCALE_FACTOR + dynamicScale)); -#ifdef FUNCTION_LPPTRANSPOSER_func1 - lppTransposer_func1( - lowBandReal + LPC_ORDER + startSample, - lowBandImag + LPC_ORDER + startSample, - qmfBufferReal + startSample, qmfBufferImag + startSample, - stopSample - startSample, (int)hiBand, dynamicScale, descale, a0r, - a0i, a1r, a1i, fPreWhitening, preWhiteningGains[loBand], - preWhiteningGains_exp[loBand] + 1); -#else + const int dynscale = fixMax(0, dynamicScale - 2) + 1; + const int rescale = -fixMin(0, dynamicScale - 2) + 1; + const int descale = fixMin(DFRACT_BITS - 1, + LPC_SCALE_FACTOR + dynamicScale + rescale); + for (i = startSample; i < stopSample; i++) { FIXP_DBL accu1, accu2; - accu1 = (fMultDiv2(a0r, lowBandReal[LPC_ORDER + i - 1]) - - fMultDiv2(a0i, lowBandImag[LPC_ORDER + i - 1]) + - fMultDiv2(a1r, lowBandReal[LPC_ORDER + i - 2]) - - fMultDiv2(a1i, lowBandImag[LPC_ORDER + i - 2])) >> - dynamicScale; - accu2 = (fMultDiv2(a0i, lowBandReal[LPC_ORDER + i - 1]) + - fMultDiv2(a0r, lowBandImag[LPC_ORDER + i - 1]) + - fMultDiv2(a1i, lowBandReal[LPC_ORDER + i - 2]) + - fMultDiv2(a1r, lowBandImag[LPC_ORDER + i - 2])) >> - dynamicScale; + accu1 = ((fMultDiv2(a0r, lowBandReal[LPC_ORDER + i - 1]) - + fMultDiv2(a0i, lowBandImag[LPC_ORDER + i - 1])) >> + 1) + + ((fMultDiv2(a1r, lowBandReal[LPC_ORDER + i - 2]) - + fMultDiv2(a1i, lowBandImag[LPC_ORDER + i - 2])) >> + 1); + accu2 = ((fMultDiv2(a0i, lowBandReal[LPC_ORDER + i - 1]) + + fMultDiv2(a0r, lowBandImag[LPC_ORDER + i - 1])) >> + 1) + + ((fMultDiv2(a1i, lowBandReal[LPC_ORDER + i - 2]) + + fMultDiv2(a1r, lowBandImag[LPC_ORDER + i - 2])) >> + 1); - accu1 = (lowBandReal[LPC_ORDER + i] >> descale) + (accu1 << 1); - accu2 = (lowBandImag[LPC_ORDER + i] >> descale) + (accu2 << 1); + accu1 = + (lowBandReal[LPC_ORDER + i] >> descale) + (accu1 >> dynscale); + accu2 = + (lowBandImag[LPC_ORDER + i] >> descale) + (accu2 >> dynscale); if (fPreWhitening) { - accu1 = scaleValueSaturate( + qmfBufferReal[i][hiBand] = scaleValueSaturate( fMultDiv2(accu1, preWhiteningGains[loBand]), - preWhiteningGains_exp[loBand] + 1); - accu2 = scaleValueSaturate( + preWhiteningGains_exp[loBand] + 1 + rescale); + qmfBufferImag[i][hiBand] = scaleValueSaturate( fMultDiv2(accu2, preWhiteningGains[loBand]), - preWhiteningGains_exp[loBand] + 1); + preWhiteningGains_exp[loBand] + 1 + rescale); + } else { + qmfBufferReal[i][hiBand] = + SATURATE_LEFT_SHIFT(accu1, rescale, DFRACT_BITS); + qmfBufferImag[i][hiBand] = + SATURATE_LEFT_SHIFT(accu2, rescale, DFRACT_BITS); } - qmfBufferReal[i][hiBand] = accu1; - qmfBufferImag[i][hiBand] = accu2; } -#endif } else { FDK_ASSERT(dynamicScale >= 0); calc_qmfBufferReal( qmfBufferReal, &(lowBandReal[LPC_ORDER + startSample - 2]), - startSample, stopSample, hiBand, dynamicScale, - fMin(DFRACT_BITS - 1, (LPC_SCALE_FACTOR + dynamicScale)), a0r, - a1r); + startSample, stopSample, hiBand, dynamicScale, a0r, a1r); } } /* bw <= 0 */ @@ -1066,7 +1062,7 @@ void lppTransposerHBE( if ((scale > 0) && (result >= (FIXP_DBL)MAXVAL_DBL >> scale)) { resetLPCCoeffs = 1; } else { - alphar[1] = FX_DBL2FX_SGL(scaleValue(result, scale)); + alphar[1] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale)); if ((tmp < FL2FX_DBL(0.0f)) ^ (ac.det < FL2FX_DBL(0.0f))) { alphar[1] = -alphar[1]; } @@ -1092,7 +1088,7 @@ void lppTransposerHBE( (result >= /*FL2FXCONST_DBL(1.f)*/ (FIXP_DBL)MAXVAL_DBL >> scale)) { resetLPCCoeffs = 1; } else { - alphai[1] = FX_DBL2FX_SGL(scaleValue(result, scale)); + alphai[1] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale)); if ((tmp < FL2FX_DBL(0.0f)) ^ (ac.det < FL2FX_DBL(0.0f))) { alphai[1] = -alphai[1]; } @@ -1121,7 +1117,7 @@ void lppTransposerHBE( } else { INT scale; FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale); - alphar[0] = FX_DBL2FX_SGL(scaleValue(result, scale + 1)); + alphar[0] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale + 1)); if ((tmp > FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f))) alphar[0] = -alphar[0]; @@ -1140,7 +1136,7 @@ void lppTransposerHBE( } else { INT scale; FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale); - alphai[0] = FX_DBL2FX_SGL(scaleValue(result, scale + 1)); + alphai[0] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale + 1)); if ((tmp > FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f))) { alphai[0] = -alphai[0]; } diff --git a/libSBRdec/src/sbr_dec.cpp b/libSBRdec/src/sbr_dec.cpp index b1fb0da..919e9bb 100644 --- a/libSBRdec/src/sbr_dec.cpp +++ b/libSBRdec/src/sbr_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 @@ -713,7 +713,8 @@ void sbr_dec( } else { /* (flags & SBRDEC_PS_DECODED) */ INT sdiff; - INT scaleFactorHighBand, scaleFactorLowBand_ov, scaleFactorLowBand_no_ov; + INT scaleFactorHighBand, scaleFactorLowBand_ov, scaleFactorLowBand_no_ov, + outScalefactor, outScalefactorR, outScalefactorL; HANDLE_QMF_FILTER_BANK synQmf = &hSbrDec->qmfDomainOutCh->fb; HANDLE_QMF_FILTER_BANK synQmfRight = &hSbrDecRight->qmfDomainOutCh->fb; @@ -744,7 +745,7 @@ void sbr_dec( */ FDK_ASSERT(hSbrDec->qmfDomainInCh->pGlobalConf->nBandsSynthesis <= QMF_MAX_SYNTHESIS_BANDS); - qmfChangeOutScalefactor(synQmfRight, -(8)); + synQmfRight->outScalefactor = synQmf->outScalefactor; FDKmemcpy(synQmfRight->FilterStates, synQmf->FilterStates, 9 * hSbrDec->qmfDomainInCh->pGlobalConf->nBandsSynthesis * sizeof(FIXP_QSS)); @@ -788,9 +789,11 @@ void sbr_dec( FDKmemcpy(&hSbrDecRight->sbrDrcChannel, &hSbrDec->sbrDrcChannel, sizeof(SBRDEC_DRC_CHANNEL)); - for (i = 0; i < synQmf->no_col; i++) { /* ----- no_col loop ----- */ + outScalefactor = maxShift - (8); + outScalefactorL = outScalefactorR = + sbrInDataHeadroom + 1; /* +1: psDiffScale! (MPEG-PS) */ - INT outScalefactorR, outScalefactorL; + for (i = 0; i < synQmf->no_col; i++) { /* ----- no_col loop ----- */ /* qmf timeslot of right channel */ FIXP_DBL *rQmfReal = pWorkBuffer; @@ -815,27 +818,20 @@ void sbr_dec( ? scaleFactorLowBand_ov : scaleFactorLowBand_no_ov, scaleFactorHighBand, synQmf->lsb, synQmf->usb); - - outScalefactorL = outScalefactorR = - 1 + sbrInDataHeadroom; /* psDiffScale! (MPEG-PS) */ } sbrDecoder_drcApplySlot(/* right channel */ &hSbrDecRight->sbrDrcChannel, rQmfReal, rQmfImag, i, synQmfRight->no_col, maxShift); - outScalefactorR += maxShift; - sbrDecoder_drcApplySlot(/* left channel */ &hSbrDec->sbrDrcChannel, *(pLowBandReal + i), *(pLowBandImag + i), i, synQmf->no_col, maxShift); - outScalefactorL += maxShift; - if (!(flags & SBRDEC_SKIP_QMF_SYN)) { - qmfChangeOutScalefactor(synQmf, -(8)); - qmfChangeOutScalefactor(synQmfRight, -(8)); + qmfChangeOutScalefactor(synQmf, outScalefactor); + qmfChangeOutScalefactor(synQmfRight, outScalefactor); qmfSynthesisFilteringSlot( synQmfRight, rQmfReal, /* QMF real buffer */ 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/libSBRdec/src/sbrdec_freq_sca.cpp b/libSBRdec/src/sbrdec_freq_sca.cpp index e187656..daa3554 100644 --- a/libSBRdec/src/sbrdec_freq_sca.cpp +++ b/libSBRdec/src/sbrdec_freq_sca.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 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -765,9 +765,6 @@ resetFreqBandTables(HANDLE_SBR_HEADER_DATA hHeaderData, const UINT flags) { sbrdecUpdateLoRes(hFreq->freqBandTable[0], &nBandsLo, hFreq->freqBandTable[1], nBandsHi); - hFreq->nSfb[0] = nBandsLo; - hFreq->nSfb[1] = nBandsHi; - /* Check index to freqBandTable[0] */ if (!(nBandsLo > 0) || (nBandsLo > (((hHeaderData->numberOfAnalysisBands == 16) @@ -777,6 +774,9 @@ resetFreqBandTables(HANDLE_SBR_HEADER_DATA hHeaderData, const UINT flags) { return SBRDEC_UNSUPPORTED_CONFIG; } + hFreq->nSfb[0] = nBandsLo; + hFreq->nSfb[1] = nBandsHi; + lsb = hFreq->freqBandTable[0][0]; usb = hFreq->freqBandTable[0][nBandsLo]; @@ -814,15 +814,15 @@ resetFreqBandTables(HANDLE_SBR_HEADER_DATA hHeaderData, const UINT flags) { if (intTemp == 0) intTemp = 1; + if (intTemp > MAX_NOISE_COEFFS) { + return SBRDEC_UNSUPPORTED_CONFIG; + } + hFreq->nNfb = intTemp; } hFreq->nInvfBands = hFreq->nNfb; - if (hFreq->nNfb > MAX_NOISE_COEFFS) { - return SBRDEC_UNSUPPORTED_CONFIG; - } - /* Get noise bands */ sbrdecDownSampleLoRes(hFreq->freqBandTableNoise, hFreq->nNfb, hFreq->freqBandTable[0], nBandsLo); diff --git a/libSBRdec/src/sbrdecoder.cpp b/libSBRdec/src/sbrdecoder.cpp index b101a4a..7718695 100644 --- a/libSBRdec/src/sbrdecoder.cpp +++ b/libSBRdec/src/sbrdecoder.cpp @@ -1,7 +1,7 @@ /* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten +© Copyright 1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -961,8 +961,10 @@ SBR_ERROR sbrDecoder_SetParam(HANDLE_SBRDECODER self, const SBRDEC_PARAM param, /* Set sync state UPSAMPLING for the corresponding slot. This switches off bitstream parsing until a new header arrives. */ - hSbrHeader->syncState = UPSAMPLING; - hSbrHeader->status |= SBRDEC_HDR_STAT_UPDATE; + if (hSbrHeader->syncState != SBR_NOT_INITIALIZED) { + hSbrHeader->syncState = UPSAMPLING; + hSbrHeader->status |= SBRDEC_HDR_STAT_UPDATE; + } } } } break; @@ -1371,7 +1373,9 @@ SBR_ERROR sbrDecoder_Parse(HANDLE_SBRDECODER self, HANDLE_FDK_BITSTREAM hBs, } if (headerStatus == HEADER_ERROR) { /* Corrupt SBR info data, do not decode and switch to UPSAMPLING */ - hSbrHeader->syncState = UPSAMPLING; + hSbrHeader->syncState = hSbrHeader->syncState > UPSAMPLING + ? UPSAMPLING + : hSbrHeader->syncState; fDoDecodeSbrData = 0; sbrHeaderPresent = 0; } @@ -1610,7 +1614,9 @@ static SBR_ERROR sbrDecoder_DecodeElement( /* No valid SBR payload available, hence switch to upsampling (in all * headers) */ for (hdrIdx = 0; hdrIdx < ((1) + 1); hdrIdx += 1) { - self->sbrHeader[elementIndex][hdrIdx].syncState = UPSAMPLING; + if (self->sbrHeader[elementIndex][hdrIdx].syncState > UPSAMPLING) { + self->sbrHeader[elementIndex][hdrIdx].syncState = UPSAMPLING; + } } } else { /* Move frame pointer to the next slot which is up to be decoded/applied 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;