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Diffstat (limited to 'libMpegTPEnc/src/tpenc_latm.cpp')
| -rw-r--r-- | libMpegTPEnc/src/tpenc_latm.cpp | 882 |
1 files changed, 882 insertions, 0 deletions
diff --git a/libMpegTPEnc/src/tpenc_latm.cpp b/libMpegTPEnc/src/tpenc_latm.cpp new file mode 100644 index 0000000..54fd717 --- /dev/null +++ b/libMpegTPEnc/src/tpenc_latm.cpp @@ -0,0 +1,882 @@ + +/* ----------------------------------------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2012 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 Encoder ************************** + + Author(s): + Description: + +******************************************************************************/ + +#include "tpenc_latm.h" + + +#include "genericStds.h" + +static const short celpFrameLengthTable[64] = { + 154, 170, 186, 147, 156, 165, 114, 120, + 186, 126, 132, 138, 142, 146, 154, 166, + 174, 182, 190, 198, 206, 210, 214, 110, + 114, 118, 120, 122, 218, 230, 242, 254, + 266, 278, 286, 294, 318, 342, 358, 374, + 390, 406, 422, 136, 142, 148, 154, 160, + 166, 170, 174, 186, 198, 206, 214, 222, + 230, 238, 216, 160, 280, 338, 0, 0 +}; + +/******* + write value to transport stream + first two bits define the size of the value itself + then the value itself, with a size of 0-3 bytes +*******/ +static +UINT transportEnc_LatmWriteValue(HANDLE_FDK_BITSTREAM hBs, int value) +{ + UCHAR valueBytes = 4; + unsigned int bitsWritten = 0; + int i; + + if ( value < (1<<8) ) { + valueBytes = 1; + } else if ( value < (1<<16) ) { + valueBytes = 2; + } else if ( value < (1<<24) ) { + valueBytes = 3; + } else { + valueBytes = 4; + } + + FDKwriteBits(hBs, valueBytes-1, 2 ); /* size of value in Bytes */ + for (i=0; i<valueBytes; i++) { + /* write most significant Byte first */ + FDKwriteBits(hBs, (UCHAR)(value>>((valueBytes-1-i)<<3)), 8); + } + + bitsWritten = (valueBytes<<3)+2; + + return bitsWritten; +} + +static +UINT transportEnc_LatmCountFixBitDemandHeader ( HANDLE_LATM_STREAM hAss ) +{ + int bitDemand = 0; + int insertSetupData = 0 ; + + /* only if start of new latm frame */ + if (hAss->subFrameCnt==0) + { + /* AudioSyncStream */ + + if (hAss->tt == TT_MP4_LOAS) { + bitDemand += 11 ; /* syncword */ + bitDemand += 13 ; /* audioMuxLengthBytes */ + } + + /* AudioMuxElement*/ + + /* AudioMuxElement::Stream Mux Config */ + if (hAss->muxConfigPeriod > 0) { + insertSetupData = (hAss->latmFrameCounter == 0); + } else { + insertSetupData = 0; + } + + if (hAss->tt != TT_MP4_LATM_MCP0) { + /* AudioMuxElement::useSameStreamMux Flag */ + bitDemand+=1; + + if( insertSetupData ) { + bitDemand += hAss->streamMuxConfigBits; + } + } + + /* AudioMuxElement::otherDataBits */ + bitDemand += 8*hAss->otherDataLenBytes; + + /* AudioMuxElement::ByteAlign */ + if ( bitDemand % 8 ) { + hAss->fillBits = 8 - (bitDemand % 8); + bitDemand += hAss->fillBits ; + } else { + hAss->fillBits = 0; + } + } + + return bitDemand ; +} + +static +UINT transportEnc_LatmCountVarBitDemandHeader ( HANDLE_LATM_STREAM hAss , unsigned int streamDataLength ) +{ + int bitDemand = 0; + int prog, layer; + + /* Payload Length Info*/ + if( hAss->allStreamsSameTimeFraming ) { + for( prog=0; prog<hAss->noProgram; prog++ ) { + for( layer=0; layer<LATM_MAX_LAYERS; layer++ ) { + LATM_LAYER_INFO *p_linfo = &(hAss->m_linfo[prog][layer]); + + if( p_linfo->streamID >= 0 ) { + switch( p_linfo->frameLengthType ) { + case 0: + if ( streamDataLength > 0 ) { + streamDataLength -= bitDemand ; + while( streamDataLength >= (255<<3) ) { + bitDemand+=8; + streamDataLength -= (255<<3); + } + bitDemand += 8; + } + break; + + case 1: + case 4: + case 6: + bitDemand += 2; + break; + + default: + return 0; + } + } + } + } + } else { + /* there are many possibilities to use this mechanism. */ + switch( hAss->varMode ) { + case LATMVAR_SIMPLE_SEQUENCE: { + /* Use the sequence generated by the encoder */ + //int streamCntPosition = transportEnc_SetWritePointer( hAss->hAssemble, 0 ); + //int streamCntPosition = FDKgetValidBits( hAss->hAssemble ); + bitDemand+=4; + + hAss->varStreamCnt = 0; + for( prog=0; prog<hAss->noProgram; prog++ ) { + for( layer=0; layer<LATM_MAX_LAYERS; layer++ ) { + LATM_LAYER_INFO *p_linfo = &(hAss->m_linfo[prog][layer]); + + if( p_linfo->streamID >= 0 ) { + + bitDemand+=4; /* streamID */ + switch( p_linfo->frameLengthType ) { + case 0: + streamDataLength -= bitDemand ; + while( streamDataLength >= (255<<3) ) { + bitDemand+=8; + streamDataLength -= (255<<3); + } + + bitDemand += 8; + break; + /*bitDemand += 1; endFlag + break;*/ + + case 1: + case 4: + case 6: + + break; + + default: + return 0; + } + hAss->varStreamCnt++; + } + } + } + bitDemand+=4; + //transportEnc_UpdateBitstreamField( hAss->hAssemble, streamCntPosition, hAss->varStreamCnt-1, 4 ); + //UINT pos = streamCntPosition-FDKgetValidBits(hAss->hAssemble); + //FDKpushBack( hAss->hAssemble, pos); + //FDKwriteBits( hAss->hAssemble, hAss->varStreamCnt-1, 4); + //FDKpushFor( hAss->hAssemble, pos-4); + } + break; + + default: + return 0; + } + } + + return bitDemand ; +} + +TRANSPORTENC_ERROR +CreateStreamMuxConfig( + HANDLE_LATM_STREAM hAss, + HANDLE_FDK_BITSTREAM hBs, + int bufferFullness, + CSTpCallBacks *cb + ) +{ + INT streamIDcnt, tmp; + int layer, prog; + + USHORT coreFrameOffset=0; + + hAss->audioMuxVersionA = 0; /* for future extensions */ + hAss->streamMuxConfigBits = 0; + + FDKwriteBits( hBs, hAss->audioMuxVersion, 1 ); /* audioMuxVersion */ + hAss->streamMuxConfigBits += 1; + + if ( hAss->audioMuxVersion == 1 ) { + FDKwriteBits( hBs, hAss->audioMuxVersionA, 1 ); /* audioMuxVersionA */ + hAss->streamMuxConfigBits+=1; + } + + if ( hAss->audioMuxVersionA == 0 ) + { + if ( hAss->audioMuxVersion == 1 ) { + hAss->streamMuxConfigBits+= transportEnc_LatmWriteValue( hBs, hAss->taraBufferFullness );/* taraBufferFullness */ + } + FDKwriteBits( hBs, hAss->allStreamsSameTimeFraming ? 1:0, 1 ); /* allStreamsSameTimeFraming */ + FDKwriteBits( hBs, hAss->noSubframes-1, 6 ); /* Number of Subframes */ + FDKwriteBits( hBs, hAss->noProgram-1, 4 ); /* Number of Programs */ + + hAss->streamMuxConfigBits+=11; + + streamIDcnt = 0; + for( prog=0; prog<hAss->noProgram; prog++ ) { + int transLayer = 0; + + FDKwriteBits( hBs, hAss->noLayer[prog]-1, 3 ); + hAss->streamMuxConfigBits+=3; + + for( layer=0; layer<LATM_MAX_LAYERS; layer++ ) { + LATM_LAYER_INFO *p_linfo = &(hAss->m_linfo[prog][layer]); + CODER_CONFIG *p_lci = hAss->config[prog][layer]; + + p_linfo->streamID = -1; + + if( hAss->config[prog][layer] != NULL ) { + int useSameConfig = 0; + + if( transLayer > 0 ) { + FDKwriteBits( hBs, useSameConfig ? 1 : 0, 1 ); + hAss->streamMuxConfigBits+=1; + } + if( (useSameConfig == 0) || (transLayer==0) ) { + UINT bits; + + if ( hAss->audioMuxVersion == 1 ) { + FDKpushFor(hBs, 2); /* align to ASC, even if we do not know the length of the "ascLen" field yet */ + } + + bits = FDKgetValidBits( hBs ); + + transportEnc_writeASC( + hBs, + hAss->config[prog][layer], + cb + ); + + bits = FDKgetValidBits( hBs ) - bits; + + if ( hAss->audioMuxVersion == 1 ) { + FDKpushBack(hBs, bits+2); + hAss->streamMuxConfigBits += transportEnc_LatmWriteValue( hBs, bits ); + transportEnc_writeASC( + hBs, + hAss->config[prog][layer], + cb + ); + } + + hAss->streamMuxConfigBits += bits; /* add asc length to smc summary */ + } + transLayer++; + + if( !hAss->allStreamsSameTimeFraming ) { + if( streamIDcnt >= LATM_MAX_STREAM_ID ) + return TRANSPORTENC_INVALID_CONFIG; + } + p_linfo->streamID = streamIDcnt++; + + switch( p_lci->aot ) { + case AOT_AAC_MAIN : + case AOT_AAC_LC : + case AOT_AAC_SSR : + case AOT_AAC_LTP : + case AOT_AAC_SCAL : + case AOT_ER_AAC_LD : + case AOT_ER_AAC_ELD : + case AOT_USAC: + case AOT_RSVD50: + p_linfo->frameLengthType = 0; + + FDKwriteBits( hBs, p_linfo->frameLengthType, 3 ); /* frameLengthType */ + FDKwriteBits( hBs, bufferFullness, 8 ); /* bufferFullness */ + hAss->streamMuxConfigBits+=11; + + if ( !hAss->allStreamsSameTimeFraming ) { + CODER_CONFIG *p_lci_prev = hAss->config[prog][layer-1]; + if ( ((p_lci->aot == AOT_AAC_SCAL) || (p_lci->aot == AOT_ER_AAC_SCAL)) && + ((p_lci_prev->aot == AOT_CELP) || (p_lci_prev->aot == AOT_ER_CELP)) ) { + FDKwriteBits( hBs, coreFrameOffset, 6 ); /* coreFrameOffset */ + hAss->streamMuxConfigBits+=6; + } + } + break; + + case AOT_TWIN_VQ: + p_linfo->frameLengthType = 1; + tmp = ( (p_lci->bitsFrame+7) >> 3 ) - 20; /* transmission frame length in bytes */ + if( (tmp < 0) ) { + return TRANSPORTENC_INVALID_TRANSMISSION_FRAME_LENGTH; + } + FDKwriteBits( hBs, p_linfo->frameLengthType, 3 ); /* frameLengthType */ + FDKwriteBits( hBs, tmp, 9 ); + hAss->streamMuxConfigBits+=12; + + p_linfo->frameLengthBits = (tmp+20) << 3; + break; + + case AOT_CELP: + p_linfo->frameLengthType = 4; + FDKwriteBits( hBs, p_linfo->frameLengthType, 3 ); /* frameLengthType */ + hAss->streamMuxConfigBits+=3; + { + int i; + for( i=0; i<62; i++ ) { + if( celpFrameLengthTable[i] == p_lci->bitsFrame ) + break; + } + if( i>=62 ) { + return TRANSPORTENC_INVALID_CELP_FRAME_LENGTH; + } + + FDKwriteBits( hBs, i, 6 ); /* CELPframeLengthTabelIndex */ + hAss->streamMuxConfigBits+=6; + } + p_linfo->frameLengthBits = p_lci->bitsFrame; + break; + + case AOT_HVXC: + p_linfo->frameLengthType = 6; + FDKwriteBits( hBs, p_linfo->frameLengthType, 3 ); /* frameLengthType */ + hAss->streamMuxConfigBits+=3; + { + int i; + + if( p_lci->bitsFrame == 40 ) { + i = 0; + } else if( p_lci->bitsFrame == 80 ) { + i = 1; + } else { + return TRANSPORTENC_INVALID_FRAME_BITS; + } + FDKwriteBits( hBs, i, 1 ); /* HVXCframeLengthTableIndex */ + hAss->streamMuxConfigBits+=1; + } + p_linfo->frameLengthBits = p_lci->bitsFrame; + break; + + case AOT_NULL_OBJECT: + default: + return TRANSPORTENC_INVALID_AOT; + } + } + } + } + + FDKwriteBits( hBs, (hAss->otherDataLenBytes>0) ? 1:0, 1 ); /* otherDataPresent */ + hAss->streamMuxConfigBits+=1; + + if( hAss->otherDataLenBytes > 0 ) { + + INT otherDataLenTmp = hAss->otherDataLenBytes; + INT escCnt = 0; + INT otherDataLenEsc = 1; + + while(otherDataLenTmp) { + otherDataLenTmp >>= 8; + escCnt ++; + } + + do { + otherDataLenTmp = (hAss->otherDataLenBytes>>(escCnt*8)) & 0xFF; + escCnt--; + otherDataLenEsc = escCnt>0; + + FDKwriteBits( hBs, otherDataLenEsc, 1 ); + FDKwriteBits( hBs, otherDataLenTmp, 8 ); + hAss->streamMuxConfigBits+=9; + } while(otherDataLenEsc); + } + + { + USHORT crcCheckPresent=0; + USHORT crcCheckSum=0; + + FDKwriteBits( hBs, crcCheckPresent, 1 ); /* crcCheckPresent */ + hAss->streamMuxConfigBits+=1; + if ( crcCheckPresent ){ + FDKwriteBits( hBs, crcCheckSum, 8 ); /* crcCheckSum */ + hAss->streamMuxConfigBits+=8; + } + } + + } else { /* if ( audioMuxVersionA == 0 ) */ + + /* for future extensions */ + + } + + return TRANSPORTENC_OK; +} + + +static TRANSPORTENC_ERROR +WriteAuPayloadLengthInfo( HANDLE_FDK_BITSTREAM hBitStream, int AuLengthBits ) +{ + int restBytes; + + if( AuLengthBits % 8 ) + return TRANSPORTENC_INVALID_AU_LENGTH; + + while( AuLengthBits >= 255*8 ) { + FDKwriteBits( hBitStream, 255, 8 ); /* 255 shows incomplete AU */ + AuLengthBits -= (255*8); + } + + restBytes = (AuLengthBits) >> 3; + FDKwriteBits( hBitStream, restBytes, 8 ); + + return TRANSPORTENC_OK; +} + +static +TRANSPORTENC_ERROR transportEnc_LatmSetNrOfSubframes( HANDLE_LATM_STREAM hAss, + INT noSubframes_next) /* nr of access units / payloads within a latm frame */ +{ + /* sanity chk */ + if (noSubframes_next < 1 || noSubframes_next > MAX_NR_OF_SUBFRAMES) { + return TRANSPORTENC_LATM_INVALID_NR_OF_SUBFRAMES; + } + + hAss->noSubframes_next = noSubframes_next; + + /* if at start then we can take over the value immediately, otherwise we have to wait for the next SMC */ + if ( (hAss->subFrameCnt == 0) && (hAss->latmFrameCounter == 0) ) { + hAss->noSubframes = noSubframes_next; + } + + return TRANSPORTENC_OK; +} + +static +int allStreamsSameTimeFraming( HANDLE_LATM_STREAM hAss, UCHAR noProgram, UCHAR noLayer[] /* return */ ) +{ + int prog, layer; + + signed int lastNoSamples = -1; + signed int minFrameSamples = FDK_INT_MAX; + signed int maxFrameSamples = 0; + + signed int highestSamplingRate = -1; + + for( prog=0; prog<noProgram; prog++ ) { + noLayer[prog] = 0; + + for( layer=0; layer<LATM_MAX_LAYERS; layer++ ) + { + if( hAss->config[prog][layer] != NULL ) + { + INT hsfSamplesFrame; + + noLayer[prog]++; + + if( highestSamplingRate < 0 ) + highestSamplingRate = hAss->config[prog][layer]->samplingRate; + + hsfSamplesFrame = hAss->config[prog][layer]->samplesPerFrame * highestSamplingRate / hAss->config[prog][layer]->samplingRate; + + if( hsfSamplesFrame <= minFrameSamples ) minFrameSamples = hsfSamplesFrame; + if( hsfSamplesFrame >= maxFrameSamples ) maxFrameSamples = hsfSamplesFrame; + + if( lastNoSamples == -1 ) { + lastNoSamples = hsfSamplesFrame; + } else { + if( hsfSamplesFrame != lastNoSamples ) { + return 0; + } + } + } + } + } + + return 1; +} + +/** + * Initialize LATM/LOAS Stream and add layer 0 at program 0. + */ +static +TRANSPORTENC_ERROR transportEnc_InitLatmStream( HANDLE_LATM_STREAM hAss, + int fractDelayPresent, + signed int muxConfigPeriod, /* insert setup data every muxConfigPeriod frames */ + UINT audioMuxVersion, + TRANSPORT_TYPE tt + ) +{ + TRANSPORTENC_ERROR ErrorStatus = TRANSPORTENC_OK; + + if (hAss == NULL) + return TRANSPORTENC_INVALID_PARAMETER; + + hAss->tt = tt; + + hAss->noProgram = 1; + + hAss->audioMuxVersion = audioMuxVersion; + + /* Fill noLayer array using hAss->config */ + hAss->allStreamsSameTimeFraming = allStreamsSameTimeFraming( hAss, hAss->noProgram, hAss->noLayer ); + /* Only allStreamsSameTimeFraming==1 is supported */ + FDK_ASSERT(hAss->allStreamsSameTimeFraming); + + hAss->fractDelayPresent = fractDelayPresent; + hAss->otherDataLenBytes = 0; + + hAss->varMode = LATMVAR_SIMPLE_SEQUENCE; + + /* initialize counters */ + hAss->subFrameCnt = 0; + hAss->noSubframes = DEFAULT_LATM_NR_OF_SUBFRAMES; + hAss->noSubframes_next = DEFAULT_LATM_NR_OF_SUBFRAMES; + + /* sync layer related */ + hAss->audioMuxLengthBytes = 0; + + hAss->latmFrameCounter = 0; + hAss->muxConfigPeriod = muxConfigPeriod; + + return ErrorStatus; +} + + +/** + * + */ +UINT transportEnc_LatmCountTotalBitDemandHeader ( HANDLE_LATM_STREAM hAss , unsigned int streamDataLength ) +{ + UINT bitDemand = 0; + + switch (hAss->tt) { + case TT_MP4_LOAS: + case TT_MP4_LATM_MCP0: + case TT_MP4_LATM_MCP1: + if (hAss->subFrameCnt == 0) { + bitDemand = transportEnc_LatmCountFixBitDemandHeader ( hAss ); + } + bitDemand += transportEnc_LatmCountVarBitDemandHeader ( hAss , streamDataLength /*- bitDemand*/); + break; + default: + break; + } + + return bitDemand; +} + +static TRANSPORTENC_ERROR +AdvanceAudioMuxElement ( + HANDLE_LATM_STREAM hAss, + HANDLE_FDK_BITSTREAM hBs, + int auBits, + int bufferFullness, + CSTpCallBacks *cb + ) +{ + TRANSPORTENC_ERROR ErrorStatus = TRANSPORTENC_OK; + int insertMuxSetup; + + /* Insert setup data to assemble Buffer */ + if (hAss->subFrameCnt == 0) + { + if (hAss->muxConfigPeriod > 0) { + insertMuxSetup = (hAss->latmFrameCounter == 0); + } else { + insertMuxSetup = 0; + } + + if (hAss->tt != TT_MP4_LATM_MCP0) { + if( insertMuxSetup ) { + FDKwriteBits( hBs, 0, 1 ); /* useSameStreamMux useNewStreamMux */ + CreateStreamMuxConfig(hAss, hBs, bufferFullness, cb); + if (ErrorStatus != TRANSPORTENC_OK) + return ErrorStatus; + } else { + FDKwriteBits( hBs, 1, 1 ); /* useSameStreamMux */ + } + } + } + + /* PayloadLengthInfo */ + { + int prog, layer; + + for (prog = 0; prog < hAss->noProgram; prog++) { + for (layer = 0; layer < hAss->noLayer[prog]; layer++) { + ErrorStatus = WriteAuPayloadLengthInfo( hBs, auBits ); + if (ErrorStatus != TRANSPORTENC_OK) + return ErrorStatus; + } + } + } + /* At this point comes the access unit. */ + + return TRANSPORTENC_OK; +} + +TRANSPORTENC_ERROR +transportEnc_LatmWrite ( + HANDLE_LATM_STREAM hAss, + HANDLE_FDK_BITSTREAM hBs, + int auBits, + int bufferFullness, + CSTpCallBacks *cb + ) +{ + TRANSPORTENC_ERROR ErrorStatus; + + if (hAss->subFrameCnt == 0) { + /* Start new frame */ + FDKresetBitbuffer(hBs, BS_WRITER); + } + + hAss->latmSubframeStart = FDKgetValidBits(hBs); + + /* Insert syncword and syncword distance + - only if loas + - we must update the syncword distance (=audiomuxlengthbytes) later + */ + if( hAss->tt == TT_MP4_LOAS && hAss->subFrameCnt == 0) + { + /* Start new LOAS frame */ + FDKwriteBits( hBs, 0x2B7, 11 ); + hAss->audioMuxLengthBytes = 0; + hAss->audioMuxLengthBytesPos = FDKgetValidBits( hBs ); /* store read pointer position */ + FDKwriteBits( hBs, hAss->audioMuxLengthBytes, 13 ); + } + + ErrorStatus = AdvanceAudioMuxElement( + hAss, + hBs, + auBits, + bufferFullness, + cb + ); + + if (ErrorStatus != TRANSPORTENC_OK) + return ErrorStatus; + + return ErrorStatus; +} + +void transportEnc_LatmAdjustSubframeBits(HANDLE_LATM_STREAM hAss, + int *bits) +{ + /* Substract bits from possible previous subframe */ + *bits -= hAss->latmSubframeStart; + /* Add fill bits */ + if (hAss->subFrameCnt == 0) + *bits += hAss->fillBits; +} + + +void transportEnc_LatmGetFrame(HANDLE_LATM_STREAM hAss, + HANDLE_FDK_BITSTREAM hBs, + int *bytes) +{ + + hAss->subFrameCnt++; + if (hAss->subFrameCnt >= hAss->noSubframes) + { + + /* Add LOAS frame length if required. */ + if (hAss->tt == TT_MP4_LOAS) + { + int latmBytes; + + latmBytes = (FDKgetValidBits(hBs)+7) >> 3; + + /* write length info into assembler buffer */ + hAss->audioMuxLengthBytes = latmBytes - 3; /* 3=Syncword + length */ + { + FDK_BITSTREAM tmpBuf; + + FDKinitBitStream( &tmpBuf, hBs->hBitBuf.Buffer, hBs->hBitBuf.bufSize, 0, BS_WRITER ) ; + FDKpushFor( &tmpBuf, hAss->audioMuxLengthBytesPos ); + FDKwriteBits( &tmpBuf, hAss->audioMuxLengthBytes, 13 ); + FDKsyncCache( &tmpBuf ); + } + } + + /* Write AudioMuxElement byte alignment fill bits */ + FDKwriteBits(hBs, 0, hAss->fillBits); + + FDK_ASSERT( (FDKgetValidBits(hBs) % 8) == 0); + + hAss->subFrameCnt = 0; + + FDKsyncCache(hBs); + *bytes = (FDKgetValidBits(hBs) + 7)>>3; + //FDKfetchBuffer(hBs, buffer, (UINT*)bytes); + + if (hAss->muxConfigPeriod > 0) + { + hAss->latmFrameCounter++; + + if (hAss->latmFrameCounter >= hAss->muxConfigPeriod) { + hAss->latmFrameCounter = 0; + hAss->noSubframes = hAss->noSubframes_next; + } + } + } else { + /* No data this time */ + *bytes = 0; + } +} + +/** + * Init LATM/LOAS + */ +TRANSPORTENC_ERROR transportEnc_Latm_Init( + HANDLE_LATM_STREAM hAss, + HANDLE_FDK_BITSTREAM hBs, + CODER_CONFIG *layerConfig, + UINT audioMuxVersion, + TRANSPORT_TYPE tt, + CSTpCallBacks *cb + ) +{ + TRANSPORTENC_ERROR ErrorStatus; + int fractDelayPresent = 0; + int prog, layer; + + int setupDataDistanceFrames = layerConfig->headerPeriod; + + FDK_ASSERT(setupDataDistanceFrames>=0); + + for (prog=0; prog<LATM_MAX_PROGRAMS; prog++) { + for (layer=0; layer<LATM_MAX_LAYERS; layer++) { + hAss->config[prog][layer] = NULL; + hAss->m_linfo[prog][layer].streamID = -1; + } + } + + hAss->config[0][0] = layerConfig; + hAss->m_linfo[0][0].streamID = 0; + + ErrorStatus = transportEnc_InitLatmStream( hAss, + fractDelayPresent, + setupDataDistanceFrames, + (audioMuxVersion)?1:0, + tt + ); + if (ErrorStatus != TRANSPORTENC_OK) + goto bail; + + ErrorStatus = transportEnc_LatmSetNrOfSubframes( + hAss, + layerConfig->nSubFrames + ); + if (ErrorStatus != TRANSPORTENC_OK) + goto bail; + + /* Get the size of the StreamMuxConfig somehow */ + AdvanceAudioMuxElement(hAss, hBs, 0, 0, cb); + //CreateStreamMuxConfig(hAss, hBs, 0); + +bail: + return ErrorStatus; +} + + + + + + |