diff options
27 files changed, 439 insertions, 472 deletions
diff --git a/documentation/aacDecoder.pdf b/documentation/aacDecoder.pdf Binary files differindex cc7cf41..3d4699e 100644 --- a/documentation/aacDecoder.pdf +++ b/documentation/aacDecoder.pdf diff --git a/documentation/aacEncoder.pdf b/documentation/aacEncoder.pdf Binary files differindex 77b8f4c..a47708a 100644 --- a/documentation/aacEncoder.pdf +++ b/documentation/aacEncoder.pdf 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/aacdecoder.cpp b/libAACdec/src/aacdecoder.cpp index c18e5e9..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 @@ -1417,11 +1486,7 @@ static void CAacDecoder_AcceptFlags(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, UINT flags, UINT *elFlags, int streamIndex, int elementOffset) { - { - FDKmemcpy( - self->elFlags, elFlags, - sizeof(*elFlags) * (3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) + 1)); - } + FDKmemcpy(self->elFlags, elFlags, sizeof(self->elFlags)); self->flags[streamIndex] = flags; } @@ -1524,8 +1589,14 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, 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; @@ -1709,7 +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++) { @@ -1729,7 +1800,7 @@ 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; } } @@ -1865,7 +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; } } @@ -1881,13 +1952,13 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, 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 */ @@ -1914,7 +1985,7 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, flags |= (asc->m_hcrFlag) ? AC_ER_HCR : 0; if (asc->m_aot == AOT_ER_AAC_ELD) { - self->mpsEnableCurr = 0; + mpsEnableCurr = 0; flags |= AC_ELD; flags |= (asc->m_sbrPresentFlag) ? AC_SBR_PRESENT @@ -1925,7 +1996,7 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, ? AC_MPS_PRESENT : 0; if (self->mpsApplicable) { - self->mpsEnableCurr = asc->m_sc.m_eldSpecificConfig.m_useLdQmfTimeAlign; + mpsEnableCurr = asc->m_sc.m_eldSpecificConfig.m_useLdQmfTimeAlign; } } flags |= (asc->m_aot == AOT_ER_AAC_LD) ? AC_LD : 0; @@ -2006,7 +2077,7 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, /* 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++) { @@ -2043,7 +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); @@ -2071,11 +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); @@ -2430,6 +2501,9 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, 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]; @@ -2968,6 +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 */ diff --git a/libAACdec/src/aacdecoder_lib.cpp b/libAACdec/src/aacdecoder_lib.cpp index 9d36d10..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 @@ -1626,6 +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); 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/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 caa62c5..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 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten +© Copyright 1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -2521,6 +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 b3ec187..b080f50 100644 --- a/libDRCdec/src/drcDec_reader.cpp +++ b/libDRCdec/src/drcDec_reader.cpp @@ -917,7 +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) { @@ -958,7 +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/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/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/src/tpdec_asc.cpp b/libMpegTPDec/src/tpdec_asc.cpp index e46cb32..8f77017 100644 --- a/libMpegTPDec/src/tpdec_asc.cpp +++ b/libMpegTPDec/src/tpdec_asc.cpp @@ -1694,8 +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) { @@ -1712,7 +1711,6 @@ static TRANSPORTDEC_ERROR extElementConfig(CSUsacExtElementConfig *extElement, } } - extElement->usacExtElementType = usacExtElementType; int usacExtElementConfigLength = escapedValue(hBs, 4, 8, 16); extElement->usacExtElementConfigLength = (USHORT)usacExtElementConfigLength; INT bsAnchor; @@ -1746,8 +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. */ @@ -1776,7 +1776,7 @@ 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. */ @@ -1787,7 +1787,7 @@ static TRANSPORTDEC_ERROR configExtension(CSUsacConfig *usc, 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 */ 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/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/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_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 b328c82..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 - 2020 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++; } 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 ad5edfe..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 - 2020 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 */ 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); - } - } + scaleValuesSaturate(&hQmfTransposer->qmfHBEBufReal_F[i][0], + QMF_SYNTH_CHANNELS, shift_ov); + scaleValuesSaturate(&hQmfTransposer->qmfHBEBufImag_F[i][0], + QMF_SYNTH_CHANNELS, 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); - } + 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++) { - 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); - } + /* 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 (int i = 0; i < stopSample - startSample; i++) { + FIXP_DBL accu; - for (i = 0; i < stopSample - startSample; i++) { - accu1 = fMultDiv2(a1r, lowBandReal[i]); - accu1 = (fMultDiv2(a0r, lowBandReal[i + 1]) + accu1); - accu1 = accu1 >> dynamicScale; + accu = fMultDiv2(a1r, lowBandReal[i]) + fMultDiv2(a0r, lowBandReal[i + 1]); + accu = (lowBandReal[i + 2] >> descale) + (accu >> dynscale); - accu1 <<= 1; - accu2 = (lowBandReal[i + 2] >> descale); - qmfBufferReal[i + startSample][hiBand] = accu1 + accu2; + 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 = (lowBandReal[LPC_ORDER + i] >> descale) + (accu1 << 1); - accu2 = (lowBandImag[LPC_ORDER + i] >> descale) + (accu2 << 1); + 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 >> 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_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 |