void rtpCodecInitialize_audio(demuxer_t* demuxer,
MediaSubsession* subsession,
unsigned& flags) {
flags = 0;
// Create a dummy audio stream header
// to make the main MPlayer code happy:
sh_audio_t* sh_audio = new_sh_audio(demuxer,0);
WAVEFORMATEX* wf = (WAVEFORMATEX*)calloc(1,sizeof(WAVEFORMATEX));
sh_audio->wf = wf;
demux_stream_t* d_audio = demuxer->audio;
d_audio->sh = sh_audio; sh_audio->ds = d_audio;
d_audio->id = sh_audio->aid;
wf->nChannels = subsession->numChannels();
// Map known audio MIME types to the WAVEFORMATEX parameters
// that this program uses. (Note that not all types need all
// of the parameters to be set.)
wf->nSamplesPerSec
= subsession->rtpSource()->timestampFrequency(); // by default
if (strcmp(subsession->codecName(), "MPA") == 0 ||
strcmp(subsession->codecName(), "MPA-ROBUST") == 0 ||
strcmp(subsession->codecName(), "X-MP3-DRAFT-00") == 0) {
wf->wFormatTag = sh_audio->format = 0x55;
// Note: 0x55 is for layer III, but should work for I,II also
wf->nSamplesPerSec = 0; // sample rate is deduced from the data
} else if (strcmp(subsession->codecName(), "AC3") == 0) {
wf->wFormatTag = sh_audio->format = 0x2000;
wf->nSamplesPerSec = 0; // sample rate is deduced from the data
} else if (strcmp(subsession->codecName(), "X-ASF-PF") == 0) {
;
} else if (strcmp(subsession->codecName(), "L16") == 0) {
wf->wFormatTag = sh_audio->format = 0x736f7774; // "twos"
wf->nBlockAlign = 1;
wf->wBitsPerSample = 16;
wf->cbSize = 0;
} else if (strcmp(subsession->codecName(), "L8") == 0) {
wf->wFormatTag = sh_audio->format = 0x20776172; // "raw "
wf->nBlockAlign = 1;
wf->wBitsPerSample = 8;
wf->cbSize = 0;
} else if (strcmp(subsession->codecName(), "PCMU") == 0) {
wf->wFormatTag = sh_audio->format = 0x7;
wf->nAvgBytesPerSec = 8000;
wf->nBlockAlign = 1;
wf->wBitsPerSample = 8;
wf->cbSize = 0;
} else if (strcmp(subsession->codecName(), "PCMA") == 0) {
wf->wFormatTag = sh_audio->format = 0x6;
wf->nAvgBytesPerSec = 8000;
wf->nBlockAlign = 1;
wf->wBitsPerSample = 8;
wf->cbSize = 0;
} else if (strcmp(subsession->codecName(), "AMR") == 0) {
wf->wFormatTag = sh_audio->format = mmioFOURCC('s','a','m','r');
} else if (strcmp(subsession->codecName(), "AMR-WB") == 0) {
wf->wFormatTag = sh_audio->format = mmioFOURCC('s','a','w','b');
} else if (strcmp(subsession->codecName(), "GSM") == 0) {
wf->wFormatTag = sh_audio->format = mmioFOURCC('a','g','s','m');
wf->nAvgBytesPerSec = 1650;
wf->nBlockAlign = 33;
wf->wBitsPerSample = 16;
wf->cbSize = 0;
} else if (strcmp(subsession->codecName(), "QCELP") == 0) {
wf->wFormatTag = sh_audio->format = mmioFOURCC('Q','c','l','p');
wf->nAvgBytesPerSec = 1750;
wf->nBlockAlign = 35;
wf->wBitsPerSample = 16;
wf->cbSize = 0;
} else if (strcmp(subsession->codecName(), "MP4A-LATM") == 0) {
wf->wFormatTag = sh_audio->format = mmioFOURCC('m','p','4','a');
// For the codec to work correctly, it needs "AudioSpecificConfig"
// data, which is parsed from the "StreamMuxConfig" string that
// was present (hopefully) in the SDP description:
unsigned codecdata_len;
sh_audio->codecdata
= parseStreamMuxConfigStr(subsession->fmtp_config(),
codecdata_len);
sh_audio->codecdata_len = codecdata_len;
//faad doesn't understand LATM's data length field, so omit it
((MPEG4LATMAudioRTPSource*)subsession->rtpSource())->omitLATMDataLengthField();
} else if (strcmp(subsession->codecName(), "MPEG4-GENERIC") == 0) {
wf->wFormatTag = sh_audio->format = mmioFOURCC('m','p','4','a');
// For the codec to work correctly, it needs "AudioSpecificConfig"
// data, which was present (hopefully) in the SDP description:
unsigned codecdata_len;
sh_audio->codecdata
= parseGeneralConfigStr(subsession->fmtp_config(),
codecdata_len);
sh_audio->codecdata_len = codecdata_len;
} else if (strcmp(subsession->codecName(), "X-QT") == 0 ||
strcmp(subsession->codecName(), "X-QUICKTIME") == 0) {
// QuickTime generic RTP format, as described in
// http://developer.apple.com/quicktime/icefloe/dispatch026.html
// We can't initialize this stream until we've received the first packet
// that has QuickTime "sdAtom" information in the header. So, keep
// reading packets until we get one:
unsigned char* packetData; unsigned packetDataLen; float pts;
QuickTimeGenericRTPSource* qtRTPSource
= (QuickTimeGenericRTPSource*)(subsession->rtpSource());
unsigned fourcc, numChannels;
do {
if (!awaitRTPPacket(demuxer, demuxer->audio,
packetData, packetDataLen, pts)) {
return;
}
} while (!parseQTState_audio(qtRTPSource->qtState, fourcc, numChannels));
wf->wFormatTag = sh_audio->format = fourcc;
wf->nChannels = numChannels;
if (qtRTPSource->qtState.sdAtomSize > 33) {
wf->wBitsPerSample = qtRTPSource->qtState.sdAtom[27];
wf->nSamplesPerSec = qtRTPSource->qtState.sdAtom[32]<<8|qtRTPSource->qtState.sdAtom[33];
}
uint8_t *pos = (uint8_t*)qtRTPSource->qtState.sdAtom + 52;
uint8_t *endpos = (uint8_t*)qtRTPSource->qtState.sdAtom
+ qtRTPSource->qtState.sdAtomSize;
while (pos+8 < endpos) {
unsigned atomLength = pos[0]<<24 | pos[1]<<16 | pos[2]<<8 | pos[3];
if (atomLength == 0 || atomLength > endpos-pos) break;
if (!memcmp(pos+4, "wave", 4) && fourcc==mmioFOURCC('Q','D','M','2') &&
atomLength > 8 &&
atomLength <= INT_MAX) {
sh_audio->codecdata = (unsigned char*) malloc(atomLength-8);
if (sh_audio->codecdata) {
memcpy(sh_audio->codecdata, pos+8, atomLength-8);
sh_audio->codecdata_len = atomLength-8;
}
break;
}
pos += atomLength;
}
} else {
fprintf(stderr,
"Unknown MPlayer format code for MIME type \"audio/%s\"\n",
subsession->codecName());
}
}
int CLvRtspClient::InitAudio( MediaSubsession *pss )
{_STT();
if ( !m_pRtspClient )
{ setLastError( -200, oexT( "Invalid rtsp client object" ) );
return 0;
} // end if
if ( !pss )
{ setLastError( -201, oexT( "Invalid video object" ) );
return 0;
} // end if
// Create receiver for stream
if ( !pss->initiate() )
{ setLastError( -203, sqbind::oex2std( oexMks( oexT( "initiate() video stream failed : " ), oexT( " : " ), m_pEnv->getResultMsg() ) ) );
return 0;
} // end if
if ( !pss->rtpSource() )
{ setLastError( -204, sqbind::oex2std( oexMks( oexT( "RTP source is null : " ), oexT( " : " ), m_pEnv->getResultMsg() ) ) );
return 0;
} // end if
if ( oex::CStr8( "MP4A-LATM" ) == pss->codecName() )
{
((MPEG4LATMAudioRTPSource*)pss->rtpSource())->omitLATMDataLengthField();
const char *pCfg = pss->fmtp_config();
if ( pCfg )
{ unsigned elen = 0;
unsigned char *pExtra = parseStreamMuxConfigStr( pCfg, elen );
if ( pExtra && elen )
m_extraAudio.AppendBuffer( (const char*)pExtra, elen );
} // end if
} // end if
else if ( oex::CStr8( "MPEG4-GENERIC" ) == pss->codecName() )
{
const char *pCfg = pss->fmtp_config();
if ( pCfg )
{ unsigned elen = 0;
unsigned char *pExtra = parseGeneralConfigStr( pCfg, elen );
if ( pExtra && elen )
m_extraAudio.AppendBuffer( (const char*)pExtra, elen );
} // end if
} // end if
// Read extradata
const char *props = pss->fmtp_spropparametersets();
if ( props )
{ oex::TList< oex::CStr8 > lst = oex::CParser::Explode( props, oexT( "," ) );
for ( oex::TList< oex::CStr8 >::iterator it; lst.Next( it ); )
{ m_extraAudio.AppendBuffer( "\x00\x00\x01", 3 );
m_extraAudio.Mem().appendString( oex::CBase64::Decode( *it ) );
} // end for
} // end if
// Set minimum rx buffer size
if ( 2000000 > m_nRxBufferSize )
m_nRxBufferSize = 2000000;
// Set rx buffer size
int sn = pss->rtpSource()->RTPgs()->socketNum();
increaseReceiveBufferTo( *m_pEnv, sn, m_nRxBufferSize );
pss->rtpSource()->setPacketReorderingThresholdTime( 2000000 );
if ( pss->codecName() )
m_sAudioCodec = oexMbToStrPtr( pss->codecName() );
if ( !m_pRtspClient->setupMediaSubsession( *pss, False, False ) )
{ setLastError( -205, sqbind::oex2std( oexMks( oexT( "setupMediaSubsession() failed : " ), oexT( " : " ), m_pEnv->getResultMsg() ) ) );
return 0;
} // end if
// Save away important audio parameters
m_nAudioNumChannels = pss->numChannels();
m_nAudioRate = pss->rtpTimestampFrequency();
m_nAudioBps = 0;
m_pAs = new CAudioSink( *m_pEnv );
if ( !m_pAs )
{ setLastError( -206, sqbind::oex2std( oexMks( oexT( "CAudioSink::createNew() failed : " ), oexT( " : " ), m_pEnv->getResultMsg() ) ) );
return 0;
} // end if
m_pAs->setDataEvent( &m_evtData );
m_pAsPss = pss;
return 1;
}
