/* ----------------------------------------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2013 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 ----------------------------------------------------------------------------------------------------------- */ /***************************** MPEG-4 AAC Decoder ************************** Author(s): Daniel Homm Description: ******************************************************************************/ #include "tpdec_lib.h" #include "tp_data.h" void CProgramConfig_Reset(CProgramConfig *pPce) { pPce->elCounter = 0; } void CProgramConfig_Init(CProgramConfig *pPce) { FDKmemclear(pPce, sizeof(CProgramConfig)); #ifdef TP_PCE_ENABLE pPce->SamplingFrequencyIndex = 0xf; #endif } int CProgramConfig_IsValid ( const CProgramConfig *pPce ) { return ( (pPce->isValid) ? 1 : 0); } #ifdef TP_PCE_ENABLE void CProgramConfig_Read( CProgramConfig *pPce, HANDLE_FDK_BITSTREAM bs, UINT alignmentAnchor ) { int i; pPce->NumEffectiveChannels = 0; pPce->NumChannels = 0; pPce->ElementInstanceTag = (UCHAR) FDKreadBits(bs,4); pPce->Profile = (UCHAR) FDKreadBits(bs,2); pPce->SamplingFrequencyIndex = (UCHAR) FDKreadBits(bs,4); pPce->NumFrontChannelElements = (UCHAR) FDKreadBits(bs,4); pPce->NumSideChannelElements = (UCHAR) FDKreadBits(bs,4); pPce->NumBackChannelElements = (UCHAR) FDKreadBits(bs,4); pPce->NumLfeChannelElements = (UCHAR) FDKreadBits(bs,2); pPce->NumAssocDataElements = (UCHAR) FDKreadBits(bs,3); pPce->NumValidCcElements = (UCHAR) FDKreadBits(bs,4); if ((pPce->MonoMixdownPresent = (UCHAR) FDKreadBits(bs,1)) != 0) { pPce->MonoMixdownElementNumber = (UCHAR) FDKreadBits(bs,4); } if ((pPce->StereoMixdownPresent = (UCHAR) FDKreadBits(bs,1)) != 0) { pPce->StereoMixdownElementNumber = (UCHAR) FDKreadBits(bs,4); } if ((pPce->MatrixMixdownIndexPresent = (UCHAR) FDKreadBits(bs,1)) != 0) { pPce->MatrixMixdownIndex = (UCHAR) FDKreadBits(bs,2); pPce->PseudoSurroundEnable = (UCHAR) FDKreadBits(bs,1); } for (i=0; i < pPce->NumFrontChannelElements; i++) { pPce->FrontElementIsCpe[i] = (UCHAR) FDKreadBits(bs,1); pPce->FrontElementTagSelect[i] = (UCHAR) FDKreadBits(bs,4); pPce->NumChannels += pPce->FrontElementIsCpe[i] ? 2 : 1; } for (i=0; i < pPce->NumSideChannelElements; i++) { pPce->SideElementIsCpe[i] = (UCHAR) FDKreadBits(bs,1); pPce->SideElementTagSelect[i] = (UCHAR) FDKreadBits(bs,4); pPce->NumChannels += pPce->SideElementIsCpe[i] ? 2 : 1; } for (i=0; i < pPce->NumBackChannelElements; i++) { pPce->BackElementIsCpe[i] = (UCHAR) FDKreadBits(bs,1); pPce->BackElementTagSelect[i] = (UCHAR) FDKreadBits(bs,4); pPce->NumChannels += pPce->BackElementIsCpe[i] ? 2 : 1; } pPce->NumEffectiveChannels = pPce->NumChannels; for (i=0; i < pPce->NumLfeChannelElements; i++) { pPce->LfeElementTagSelect[i] = (UCHAR) FDKreadBits(bs,4); pPce->NumChannels += 1; } for (i=0; i < pPce->NumAssocDataElements; i++) { pPce->AssocDataElementTagSelect[i] = (UCHAR) FDKreadBits(bs,4); } for (i=0; i < pPce->NumValidCcElements; i++) { pPce->CcElementIsIndSw[i] = (UCHAR) FDKreadBits(bs,1); pPce->ValidCcElementTagSelect[i] = (UCHAR) FDKreadBits(bs,4); } FDKbyteAlign(bs, alignmentAnchor); pPce->CommentFieldBytes = (UCHAR) FDKreadBits(bs,8); for (i=0; i < pPce->CommentFieldBytes; i++) { UCHAR text; text = (UCHAR)FDKreadBits(bs,8); if (i < PC_COMMENTLENGTH) { pPce->Comment[i] = text; } } pPce->isValid = 1; } #endif /* TP_PCE_ENABLE */ /** * \brief get implicit audio channel type for given channelConfig and MPEG ordered channel index * \param channelConfig MPEG channelConfiguration from 1 upto 7 * \param index MPEG channel order index * \return audio channel type. */ void getImplicitAudioChannelTypeAndIndex( AUDIO_CHANNEL_TYPE *chType, UCHAR *chIndex, UINT channelConfig, UINT index ) { if (index < 3) { *chType = ACT_FRONT; *chIndex = index; } else { switch (channelConfig) { case MODE_1_2_1: case MODE_1_2_2: case MODE_1_2_2_1: switch (index) { case 3: case 4: *chType = ACT_BACK; *chIndex = index - 3; break; case 5: *chType = ACT_LFE; *chIndex = 0; break; } break; case MODE_1_2_2_2_1: switch (index) { case 3: case 4: *chType = ACT_SIDE; *chIndex = index - 3; break; case 5: case 6: *chType = ACT_BACK; *chIndex = index - 5; break; case 7: *chType = ACT_LFE; *chIndex = 0; break; } break; default: *chType = ACT_NONE; break; } } } int CProgramConfig_LookupElement( CProgramConfig *pPce, const UINT channelConfig, const UINT tag, const UINT channelIdx, UCHAR chMapping[], AUDIO_CHANNEL_TYPE chType[], UCHAR chIndex[], UCHAR *elMapping, MP4_ELEMENT_ID elList[], MP4_ELEMENT_ID elType ) { if (channelConfig > 0) { /* Constant channel mapping must have been set during initialization. */ if ( elType == ID_SCE || elType == ID_CPE || elType == ID_LFE ) { *elMapping = pPce->elCounter; if (elList[pPce->elCounter] != elType) { /* Not in the list */ return 0; } /* Assume all front channels */ getImplicitAudioChannelTypeAndIndex(&chType[channelIdx], &chIndex[channelIdx], channelConfig, channelIdx); if (elType == ID_CPE) { chType[channelIdx+1] = chType[channelIdx]; chIndex[channelIdx+1] = chIndex[channelIdx]+1; } pPce->elCounter++; } /* Accept all non-channel elements, too. */ return 1; } else { #ifdef TP_PCE_ENABLE if (!pPce->isValid) #endif /* TP_PCE_ENABLE */ { /* Implicit channel mapping. */ if ( elType == ID_SCE || elType == ID_CPE || elType == ID_LFE ) { /* Store all channel element IDs */ elList[pPce->elCounter] = elType; *elMapping = pPce->elCounter++; } } #ifdef TP_PCE_ENABLE else { /* Accept the additional channel(s), only if the tag is in the lists */ int isCpe = 0, i; int cc = 0, fc = 0, sc = 0, bc = 0, lc = 0, ec = 0; /* Channel and element counters */ switch (elType) { case ID_CPE: isCpe = 1; case ID_SCE: /* search in front channels */ for (i = 0; i < pPce->NumFrontChannelElements; i++) { if (isCpe == pPce->FrontElementIsCpe[i] && pPce->FrontElementTagSelect[i] == tag) { chMapping[cc] = channelIdx; chType[cc] = ACT_FRONT; chIndex[cc] = fc; if (isCpe) { chMapping[cc+1] = channelIdx+1; chType[cc+1] = ACT_FRONT; chIndex[cc+1] = fc+1; } *elMapping = ec; return 1; } ec++; if (pPce->FrontElementIsCpe[i]) { cc+=2; fc+=2; } else { cc++; fc++; } } /* search in side channels */ for (i = 0; i < pPce->NumSideChannelElements; i++) { if (isCpe == pPce->SideElementIsCpe[i] && pPce->SideElementTagSelect[i] == tag) { chMapping[cc] = channelIdx; chType[cc] = ACT_SIDE; chIndex[cc] = sc; if (isCpe) { chMapping[cc+1] = channelIdx+1; chType[cc+1] = ACT_SIDE; chIndex[cc+1] = sc+1; } *elMapping = ec; return 1; } ec++; if (pPce->SideElementIsCpe[i]) { cc+=2; sc+=2; } else { cc++; sc++; } } /* search in back channels */ for (i = 0; i < pPce->NumBackChannelElements; i++) { if (isCpe == pPce->BackElementIsCpe[i] && pPce->BackElementTagSelect[i] == tag) { chMapping[cc] = channelIdx; chType[cc] = ACT_BACK; chIndex[cc] = bc; if (isCpe) { chMapping[cc+1] = channelIdx+1; chType[cc+1] = ACT_BACK; chIndex[cc+1] = bc+1; } *elMapping = ec; return 1; } ec++; if (pPce->BackElementIsCpe[i]) { cc+=2; bc+=2; } else { cc++; bc++; } } break; case ID_LFE: /* Initialize channel counter and element counter */ cc = pPce->NumEffectiveChannels; ec = pPce->NumFrontChannelElements+ pPce->NumSideChannelElements + pPce->NumBackChannelElements; /* search in lfe channels */ for (i = 0; i < pPce->NumLfeChannelElements; i++) { if ( pPce->LfeElementTagSelect[i] == tag ) { chMapping[cc] = channelIdx; *elMapping = ec; chType[cc] = ACT_LFE; chIndex[cc] = lc; return 1; } ec++; cc++; lc++; } break; /* Non audio elements */ case ID_CCE: /* search in cce channels */ for (i = 0; i < pPce->NumValidCcElements; i++) { if (pPce->ValidCcElementTagSelect[i] == tag) { return 1; } } break; case ID_DSE: /* search associated data elements */ for (i = 0; i < pPce->NumAssocDataElements; i++) { if (pPce->AssocDataElementTagSelect[i] == tag) { return 1; } } break; default: return 0; } return 0; /* not found in any list */ } #endif /* TP_PCE_ENABLE */ } return 1; } #ifdef TP_PCE_ENABLE int CProgramConfig_GetElementTable( const CProgramConfig *pPce, MP4_ELEMENT_ID elList[], const INT elListSize ) { int i, el = 0; if ( elListSize < pPce->NumFrontChannelElements + pPce->NumSideChannelElements + pPce->NumBackChannelElements + pPce->NumLfeChannelElements ) { return 0; } for (i=0; i < pPce->NumFrontChannelElements; i++) { elList[el++] = (pPce->FrontElementIsCpe[i]) ? ID_CPE : ID_SCE; } for (i=0; i < pPce->NumSideChannelElements; i++) { elList[el++] = (pPce->SideElementIsCpe[i]) ? ID_CPE : ID_SCE; } for (i=0; i < pPce->NumBackChannelElements; i++) { elList[el++] = (pPce->BackElementIsCpe[i]) ? ID_CPE : ID_SCE; } for (i=0; i < pPce->NumLfeChannelElements; i++) { elList[el++] = ID_LFE; } return el; } #endif static AUDIO_OBJECT_TYPE getAOT(HANDLE_FDK_BITSTREAM bs) { int tmp = 0; tmp = FDKreadBits(bs,5); if (tmp == AOT_ESCAPE) { int tmp2 = FDKreadBits(bs,6); tmp = 32 + tmp2; } return (AUDIO_OBJECT_TYPE)tmp; } static INT getSampleRate(HANDLE_FDK_BITSTREAM bs, UCHAR *index, int nBits) { INT sampleRate; int idx; idx = FDKreadBits(bs, nBits); if( idx == (1<m_frameLengthFlag = FDKreadBits(bs,1); self->m_dependsOnCoreCoder = FDKreadBits(bs,1); if( self->m_dependsOnCoreCoder ) self->m_coreCoderDelay = FDKreadBits(bs,14); self->m_extensionFlag = FDKreadBits(bs,1); if( asc->m_channelConfiguration == 0 ) { CProgramConfig_Read(&asc->m_progrConfigElement, bs, ascStartAnchor); } if ((asc->m_aot == AOT_AAC_SCAL) || (asc->m_aot == AOT_ER_AAC_SCAL)) { self->m_layer = FDKreadBits(bs,3); } if (self->m_extensionFlag) { if (asc->m_aot == AOT_ER_BSAC) { self->m_numOfSubFrame = FDKreadBits(bs,5); self->m_layerLength = FDKreadBits(bs,11); } if ((asc->m_aot == AOT_ER_AAC_LC) || (asc->m_aot == AOT_ER_AAC_LTP) || (asc->m_aot == AOT_ER_AAC_SCAL) || (asc->m_aot == AOT_ER_AAC_LD)) { asc->m_vcb11Flag = FDKreadBits(bs,1); /* aacSectionDataResilienceFlag */ asc->m_rvlcFlag = FDKreadBits(bs,1); /* aacScalefactorDataResilienceFlag */ asc->m_hcrFlag = FDKreadBits(bs,1); /* aacSpectralDataResilienceFlag */ } self->m_extensionFlag3 = FDKreadBits(bs,1); } return (ErrorStatus); } #endif /* TP_GA_ENABLE */ #ifdef TP_ELD_ENABLE static INT ld_sbr_header( const CSAudioSpecificConfig *asc, HANDLE_FDK_BITSTREAM hBs, CSTpCallBacks *cb ) { const int channelConfiguration = asc->m_channelConfiguration; int i = 0; INT error = 0; if (channelConfiguration == 2) { error = cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, AOT_ER_AAC_ELD, ID_CPE, i++); } else { error = cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, AOT_ER_AAC_ELD, ID_SCE, i++); } switch ( channelConfiguration ) { case 5: error |= cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, AOT_ER_AAC_ELD, ID_CPE, i++); case 3: error |= cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, AOT_ER_AAC_ELD, ID_CPE, i++); break; case 7: error |= cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, AOT_ER_AAC_ELD, ID_SCE, i++); case 6: error |= cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, AOT_ER_AAC_ELD, ID_CPE, i++); case 4: error |= cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, AOT_ER_AAC_ELD, ID_CPE, i++); break; } return error; } static TRANSPORTDEC_ERROR EldSpecificConfig_Parse( CSAudioSpecificConfig *asc, HANDLE_FDK_BITSTREAM hBs, CSTpCallBacks *cb ) { TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK; CSEldSpecificConfig *esc = &asc->m_sc.m_eldSpecificConfig; ASC_ELD_EXT_TYPE eldExtType; int eldExtLen, len, cnt; FDKmemclear(esc, sizeof(CSEldSpecificConfig)); esc->m_frameLengthFlag = FDKreadBits(hBs, 1 ); if (esc->m_frameLengthFlag) { asc->m_samplesPerFrame = 480; } else { asc->m_samplesPerFrame = 512; } asc->m_vcb11Flag = FDKreadBits(hBs, 1 ); asc->m_rvlcFlag = FDKreadBits(hBs, 1 ); asc->m_hcrFlag = FDKreadBits(hBs, 1 ); esc->m_sbrPresentFlag = FDKreadBits(hBs, 1 ); if (esc->m_sbrPresentFlag == 1) { esc->m_sbrSamplingRate = FDKreadBits(hBs, 1 ); /* 0: single rate, 1: dual rate */ esc->m_sbrCrcFlag = FDKreadBits(hBs, 1 ); asc->m_extensionSamplingFrequency = asc->m_samplingFrequency << esc->m_sbrSamplingRate; if (cb->cbSbr != NULL){ if ( 0 != ld_sbr_header(asc, hBs, cb) ) { return TRANSPORTDEC_PARSE_ERROR; } } } esc->m_useLdQmfTimeAlign = 0; /* new ELD syntax */ /* parse ExtTypeConfigData */ while ((eldExtType = (ASC_ELD_EXT_TYPE)FDKreadBits(hBs, 4 )) != ELDEXT_TERM) { eldExtLen = len = FDKreadBits(hBs, 4 ); if ( len == 0xf ) { len = FDKreadBits(hBs, 8 ); eldExtLen += len; if ( len == 0xff ) { len = FDKreadBits(hBs, 16 ); eldExtLen += len; } } switch (eldExtType) { case ELDEXT_LDSAC: esc->m_useLdQmfTimeAlign = 1; if (cb->cbSsc != NULL) { ErrorStatus = (TRANSPORTDEC_ERROR)cb->cbSsc( cb->cbSscData, hBs, asc->m_aot, asc->m_samplingFrequency, 1, /* muxMode */ len ); } else { ErrorStatus = TRANSPORTDEC_UNSUPPORTED_FORMAT; } if (ErrorStatus != TRANSPORTDEC_OK) { goto bail; } break; default: for(cnt=0; cntm_aot = AOT_NONE; asc->m_samplingFrequencyIndex = 0xf; asc->m_epConfig = -1; asc->m_extensionAudioObjectType = AOT_NULL_OBJECT; #ifdef TP_PCE_ENABLE CProgramConfig_Init(&asc->m_progrConfigElement); #endif } TRANSPORTDEC_ERROR AudioSpecificConfig_Parse( CSAudioSpecificConfig *self, HANDLE_FDK_BITSTREAM bs, int fExplicitBackwardCompatible, CSTpCallBacks *cb ) { TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK; UINT ascStartAnchor = FDKgetValidBits(bs); int frameLengthFlag = -1; AudioSpecificConfig_Init(self); self->m_aot = getAOT(bs); self->m_samplingFrequency = getSampleRate(bs, &self->m_samplingFrequencyIndex, 4); if (self->m_samplingFrequency <= 0) { return TRANSPORTDEC_PARSE_ERROR; } self->m_channelConfiguration = FDKreadBits(bs,4); /* SBR extension ( explicit non-backwards compatible mode ) */ self->m_sbrPresentFlag = 0; self->m_psPresentFlag = 0; if ( self->m_aot == AOT_SBR || self->m_aot == AOT_PS ) { self->m_extensionAudioObjectType = AOT_SBR; self->m_sbrPresentFlag = 1; if ( self->m_aot == AOT_PS ) { self->m_psPresentFlag = 1; } self->m_extensionSamplingFrequency = getSampleRate(bs, &self->m_extensionSamplingFrequencyIndex, 4); self->m_aot = getAOT(bs); } else { self->m_extensionAudioObjectType = AOT_NULL_OBJECT; } /* Parse whatever specific configs */ switch (self->m_aot) { #ifdef TP_GA_ENABLE case AOT_AAC_LC: case AOT_ER_AAC_LC: case AOT_ER_AAC_LD: case AOT_ER_AAC_SCAL: case AOT_ER_BSAC: if ((ErrorStatus = GaSpecificConfig_Parse(&self->m_sc.m_gaSpecificConfig, self, bs, ascStartAnchor)) != TRANSPORTDEC_OK ) { return (ErrorStatus); } frameLengthFlag = self->m_sc.m_gaSpecificConfig.m_frameLengthFlag; break; #endif /* TP_GA_ENABLE */ case AOT_MPEGS: if (cb->cbSsc != NULL) { cb->cbSsc( cb->cbSscData, bs, self->m_aot, self->m_samplingFrequency, 1, 0 /* don't know the length */ ); } else { return TRANSPORTDEC_UNSUPPORTED_FORMAT; } break; #ifdef TP_ELD_ENABLE case AOT_ER_AAC_ELD: if ((ErrorStatus = EldSpecificConfig_Parse(self, bs, cb)) != TRANSPORTDEC_OK ) { return (ErrorStatus); } frameLengthFlag = self->m_sc.m_eldSpecificConfig.m_frameLengthFlag; self->m_sbrPresentFlag = self->m_sc.m_eldSpecificConfig.m_sbrPresentFlag; self->m_extensionSamplingFrequency = (self->m_sc.m_eldSpecificConfig.m_sbrSamplingRate+1) * self->m_samplingFrequency; break; #endif /* TP_ELD_ENABLE */ default: return TRANSPORTDEC_UNSUPPORTED_FORMAT; break; } /* Frame length */ switch (self->m_aot) { #if defined(TP_GA_ENABLE) || defined(TP_USAC_ENABLE) case AOT_AAC_LC: case AOT_ER_AAC_LC: case AOT_ER_AAC_SCAL: case AOT_ER_BSAC: /*case AOT_USAC:*/ if (!frameLengthFlag) self->m_samplesPerFrame = 1024; else self->m_samplesPerFrame = 960; break; #endif /* TP_GA_ENABLE */ #if defined(TP_GA_ENABLE) case AOT_ER_AAC_LD: if (!frameLengthFlag) self->m_samplesPerFrame = 512; else self->m_samplesPerFrame = 480; break; #endif /* defined(TP_GA_ENABLE) */ default: break; } switch (self->m_aot) { case AOT_ER_AAC_LC: case AOT_ER_AAC_LD: case AOT_ER_AAC_ELD: case AOT_ER_AAC_SCAL: case AOT_ER_CELP: case AOT_ER_HVXC: case AOT_ER_BSAC: self->m_epConfig = FDKreadBits(bs,2); if (self->m_epConfig > 1) { return TRANSPORTDEC_UNSUPPORTED_FORMAT; // EPCONFIG; } break; default: break; } return (ErrorStatus); }