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authorThe Android Open Source Project <initial-contribution@android.com>2012-07-11 10:15:24 -0700
committerThe Android Open Source Project <initial-contribution@android.com>2012-07-11 10:15:24 -0700
commit2228e360595641dd906bf1773307f43d304f5b2e (patch)
tree57f3d390ebb0782cc0de0fb984c8ea7e45b4f386 /libMpegTPEnc/src/tpenc_latm.cpp
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Change-Id: If584e579464f28b97d50e51fc76ba654a5536c54
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+
+/* -----------------------------------------------------------------------------------------------------------
+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;
+}
+
+
+
+
+
+