Boolean MediaSubsession::initiate(int useSpecialRTPoffset) {
if (fReadSource != NULL) return True; // has already been initiated
do {
if (fCodecName == NULL) {
env().setResultMsg("Codec is unspecified");
break;
}
// Create RTP and RTCP 'Groupsocks' on which to receive incoming data.
// (Groupsocks will work even for unicast addresses)
struct in_addr tempAddr;
tempAddr.s_addr = connectionEndpointAddress();
// This could get changed later, as a result of a RTSP "SETUP"
if (fClientPortNum != 0) {
// The sockets' port numbers were specified for us. Use these:
fClientPortNum = fClientPortNum&~1; // even
if (isSSM()) {
fRTPSocket = new Groupsock(env(), tempAddr, fSourceFilterAddr, fClientPortNum);
} else {
fRTPSocket = new Groupsock(env(), tempAddr, fClientPortNum, 255);
}
if (fRTPSocket == NULL) {
env().setResultMsg("Failed to create RTP socket");
break;
}
// Set our RTCP port to be the RTP port +1
portNumBits const rtcpPortNum = fClientPortNum|1;
if (isSSM()) {
fRTCPSocket = new Groupsock(env(), tempAddr, fSourceFilterAddr, rtcpPortNum);
} else {
fRTCPSocket = new Groupsock(env(), tempAddr, rtcpPortNum, 255);
}
if (fRTCPSocket == NULL) {
char tmpBuf[100];
sprintf(tmpBuf, "Failed to create RTCP socket (port %d)", rtcpPortNum);
env().setResultMsg(tmpBuf);
break;
}
} else {
// Port numbers were not specified in advance, so we use ephemeral port numbers.
// Create sockets until we get a port-number pair (even: RTP; even+1: RTCP).
// We need to make sure that we don't keep trying to use the same bad port numbers over and over again.
// so we store bad sockets in a table, and delete them all when we're done.
HashTable* socketHashTable = HashTable::create(ONE_WORD_HASH_KEYS);
if (socketHashTable == NULL) break;
Boolean success = False;
while (1) {
// Create a new socket:
if (isSSM()) {
fRTPSocket = new Groupsock(env(), tempAddr, fSourceFilterAddr, 0);
} else {
fRTPSocket = new Groupsock(env(), tempAddr, 0, 255);
}
if (fRTPSocket == NULL) {
env().setResultMsg("MediaSession::initiate(): unable to create RTP and RTCP sockets");
break;
}
// Get the client port number, and check whether it's even (for RTP):
Port clientPort(0);
if (!getSourcePort(env(), fRTPSocket->socketNum(), clientPort)) {
break;
}
fClientPortNum = ntohs(clientPort.num());
if ((fClientPortNum&1) != 0) { // it's odd
// Record this socket in our table, and keep trying:
unsigned key = (unsigned)fClientPortNum;
socketHashTable->Add((char const*)key, fRTPSocket);
continue;
}
// Make sure we can use the next (i.e., odd) port number, for RTCP:
portNumBits rtcpPortNum = fClientPortNum|1;
if (isSSM()) {
fRTCPSocket = new Groupsock(env(), tempAddr, fSourceFilterAddr, rtcpPortNum);
} else {
fRTCPSocket = new Groupsock(env(), tempAddr, rtcpPortNum, 255);
}
if (fRTCPSocket != NULL) {
// Success! Use these two sockets (and delete any others that we've created):
Groupsock* oldGS;
while ((oldGS = (Groupsock*)socketHashTable->RemoveNext()) != NULL) {
delete oldGS;
}
delete socketHashTable;
success = True;
break;
} else {
// We couldn't create the RTCP socket (perhaps that port number's already in use elsewhere?).
// Record the first socket in our table, and keep trying:
unsigned key = (unsigned)fClientPortNum;
socketHashTable->Add((char const*)key, fRTPSocket);
continue;
}
}
if (!success) break; // a fatal error occurred trying to create the RTP and RTCP sockets; we can't continue
}
// ASSERT: fRTPSocket != NULL && fRTCPSocket != NULL
if (isSSM()) {
// Special case for RTCP SSM: Send RTCP packets back to the source via unicast:
fRTCPSocket->changeDestinationParameters(fSourceFilterAddr,0,~0);
}
//////////////////////////////////////////////////////////////////////////
// 裁剪掉不需要的Source.
// Check "fProtocolName"
if (strcmp(fProtocolName, "UDP") == 0) {
#ifndef CUT_MIN_SIZE
// A UDP-packetized stream (*not* a RTP stream)
fReadSource = BasicUDPSource::createNew(env(), fRTPSocket);
fRTPSource = NULL; // Note!
if (strcmp(fCodecName, "MP2T") == 0) { // MPEG-2 Transport Stream
fReadSource = MPEG2TransportStreamFramer::createNew(env(), fReadSource);
// this sets "durationInMicroseconds" correctly, based on the PCR values
}
#endif
} else {
// Check "fCodecName" against the set of codecs that we support,
// and create our RTP source accordingly
// (Later make this code more efficient, as this set grows #####)
// (Also, add more fmts that can be implemented by SimpleRTPSource#####)
Boolean createSimpleRTPSource = False;
Boolean doNormalMBitRule = False; // used if "createSimpleRTPSource"
if (strcmp(fCodecName, "H264") == 0) {
fReadSource = fRTPSource
= H264VideoRTPSource::createNew(env(), fRTPSocket,
fRTPPayloadFormat,
fRTPTimestampFrequency);
}
else if (strcmp(fCodecName, "MP4V-ES") == 0)
{ // MPEG-4 Elem Str vid
fReadSource = fRTPSource
= MPEG4ESVideoRTPSource::createNew(env(), fRTPSocket,
fRTPPayloadFormat,
fRTPTimestampFrequency);
}
#ifndef CUT_MIN_SIZE
else if (strcmp(fCodecName, "QCELP") == 0) { // QCELP audio
fReadSource =
QCELPAudioRTPSource::createNew(env(), fRTPSocket, fRTPSource,
fRTPPayloadFormat,
fRTPTimestampFrequency);
// Note that fReadSource will differ from fRTPSource in this case
} else if (strcmp(fCodecName, "AMR") == 0) { // AMR audio (narrowband)
fReadSource =
AMRAudioRTPSource::createNew(env(), fRTPSocket, fRTPSource,
fRTPPayloadFormat, 0 /*isWideband*/,
fNumChannels, fOctetalign, fInterleaving,
fRobustsorting, fCRC);
// Note that fReadSource will differ from fRTPSource in this case
} else if (strcmp(fCodecName, "AMR-WB") == 0) { // AMR audio (wideband)
fReadSource =
AMRAudioRTPSource::createNew(env(), fRTPSocket, fRTPSource,
fRTPPayloadFormat, 1 /*isWideband*/,
fNumChannels, fOctetalign, fInterleaving,
fRobustsorting, fCRC);
// Note that fReadSource will differ from fRTPSource in this case
} else if (strcmp(fCodecName, "MPA") == 0) { // MPEG-1 or 2 audio
fReadSource = fRTPSource
= MPEG1or2AudioRTPSource::createNew(env(), fRTPSocket,
fRTPPayloadFormat,
fRTPTimestampFrequency);
} else if (strcmp(fCodecName, "MPA-ROBUST") == 0) { // robust MP3 audio
fRTPSource
= MP3ADURTPSource::createNew(env(), fRTPSocket, fRTPPayloadFormat,
fRTPTimestampFrequency);
if (fRTPSource == NULL) break;
// Add a filter that deinterleaves the ADUs after depacketizing them:
MP3ADUdeinterleaver* deinterleaver
= MP3ADUdeinterleaver::createNew(env(), fRTPSource);
if (deinterleaver == NULL) break;
// Add another filter that converts these ADUs to MP3 frames:
fReadSource = MP3FromADUSource::createNew(env(), deinterleaver);
} else if (strcmp(fCodecName, "X-MP3-DRAFT-00") == 0) {
// a non-standard variant of "MPA-ROBUST" used by RealNetworks
// (one 'ADU'ized MP3 frame per packet; no headers)
fRTPSource
= SimpleRTPSource::createNew(env(), fRTPSocket, fRTPPayloadFormat,
fRTPTimestampFrequency,
"audio/MPA-ROBUST" /*hack*/);
if (fRTPSource == NULL) break;
// Add a filter that converts these ADUs to MP3 frames:
fReadSource = MP3FromADUSource::createNew(env(), fRTPSource,
False /*no ADU header*/);
} else if (strcmp(fCodecName, "MP4A-LATM") == 0) { // MPEG-4 LATM audio
fReadSource = fRTPSource
= MPEG4LATMAudioRTPSource::createNew(env(), fRTPSocket,
fRTPPayloadFormat,
fRTPTimestampFrequency);
} else if (strcmp(fCodecName, "AC3") == 0) { // AC3 audio
fReadSource = fRTPSource
= AC3AudioRTPSource::createNew(env(), fRTPSocket,
fRTPPayloadFormat,
fRTPTimestampFrequency);
} else if (strcmp(fCodecName, "MPEG4-GENERIC") == 0) {
fReadSource = fRTPSource
= MPEG4GenericRTPSource::createNew(env(), fRTPSocket,
fRTPPayloadFormat,
fRTPTimestampFrequency,
fMediumName, fMode,
fSizelength, fIndexlength,
fIndexdeltalength);
} else if (strcmp(fCodecName, "MPV") == 0) { // MPEG-1 or 2 video
fReadSource = fRTPSource
= MPEG1or2VideoRTPSource::createNew(env(), fRTPSocket,
fRTPPayloadFormat,
fRTPTimestampFrequency);
} else if (strcmp(fCodecName, "MP2T") == 0) { // MPEG-2 Transport Stream
fRTPSource = SimpleRTPSource::createNew(env(), fRTPSocket, fRTPPayloadFormat,
fRTPTimestampFrequency, "video/MP2T",
0, False);
fReadSource = MPEG2TransportStreamFramer::createNew(env(), fRTPSource);
// this sets "durationInMicroseconds" correctly, based on the PCR values
} else if (strcmp(fCodecName, "H261") == 0) { // H.261
fReadSource = fRTPSource
= H261VideoRTPSource::createNew(env(), fRTPSocket,
fRTPPayloadFormat,
fRTPTimestampFrequency);
} else if (strcmp(fCodecName, "H263-1998") == 0 ||
strcmp(fCodecName, "H263-2000") == 0) { // H.263+
fReadSource = fRTPSource
= H263plusVideoRTPSource::createNew(env(), fRTPSocket,
fRTPPayloadFormat,
fRTPTimestampFrequency);
} else if (strcmp(fCodecName, "JPEG") == 0) { // motion JPEG
fReadSource = fRTPSource
= JPEGVideoRTPSource::createNew(env(), fRTPSocket,
fRTPPayloadFormat,
fRTPTimestampFrequency,
videoWidth(),
videoHeight());
} else if (strcmp(fCodecName, "X-QT") == 0
|| strcmp(fCodecName, "X-QUICKTIME") == 0) {
// Generic QuickTime streams, as defined in
// <http://developer.apple.com/quicktime/icefloe/dispatch026.html>
char* mimeType
= new char[strlen(mediumName()) + strlen(codecName()) + 2] ;
sprintf(mimeType, "%s/%s", mediumName(), codecName());
fReadSource = fRTPSource
= QuickTimeGenericRTPSource::createNew(env(), fRTPSocket,
fRTPPayloadFormat,
fRTPTimestampFrequency,
mimeType);
delete[] mimeType;
#ifdef SUPPORT_REAL_RTSP
} else if (strcmp(fCodecName, "X-PN-REALAUDIO") == 0 ||
strcmp(fCodecName, "X-PN-MULTIRATE-REALAUDIO-LIVE") == 0 ||
strcmp(fCodecName, "X-PN-REALVIDEO") == 0 ||
strcmp(fCodecName, "X-PN-MULTIRATE-REALVIDEO-LIVE") == 0) {
// A RealNetworks 'RDT' stream (*not* a RTP stream)
fReadSource = RealRDTSource::createNew(env());
fRTPSource = NULL; // Note!
parentSession().isRealNetworksRDT = True;
#endif
}
#endif // CUT_MIN_SIZE
else if ( strcmp(fCodecName, "PCMU") == 0 // PCM u-law audio
|| strcmp(fCodecName, "GSM") == 0 // GSM audio
|| strcmp(fCodecName, "PCMA") == 0 // PCM a-law audio
|| strcmp(fCodecName, "L16") == 0 // 16-bit linear audio
|| strcmp(fCodecName, "MP1S") == 0 // MPEG-1 System Stream
|| strcmp(fCodecName, "MP2P") == 0 // MPEG-2 Program Stream
|| strcmp(fCodecName, "L8") == 0 // 8-bit linear audio
|| strcmp(fCodecName, "G726-16") == 0 // G.726, 16 kbps
|| strcmp(fCodecName, "G726-24") == 0 // G.726, 24 kbps
|| strcmp(fCodecName, "G726-32") == 0 // G.726, 32 kbps
|| strcmp(fCodecName, "G726-40") == 0 // G.726, 40 kbps
|| strcmp(fCodecName, "SPEEX") == 0 // SPEEX audio
) {
createSimpleRTPSource = True;
useSpecialRTPoffset = 0;
} else if (useSpecialRTPoffset >= 0) {
// We don't know this RTP payload format, but try to receive
// it using a 'SimpleRTPSource' with the specified header offset:
createSimpleRTPSource = True;
} else {
env().setResultMsg("RTP payload format unknown or not supported");
break;
}
if (createSimpleRTPSource) {
char* mimeType
= new char[strlen(mediumName()) + strlen(codecName()) + 2] ;
sprintf(mimeType, "%s/%s", mediumName(), codecName());
fReadSource = fRTPSource
= SimpleRTPSource::createNew(env(), fRTPSocket, fRTPPayloadFormat,
fRTPTimestampFrequency, mimeType,
(unsigned)useSpecialRTPoffset,
doNormalMBitRule);
delete[] mimeType;
}
}
if (fReadSource == NULL) {
env().setResultMsg("Failed to create read source");
break;
}
// Finally, create our RTCP instance. (It starts running automatically)
if (fRTPSource != NULL) {
unsigned totSessionBandwidth = 500; // HACK - later get from SDP#####
fRTCPInstance = RTCPInstance::createNew(env(), fRTCPSocket,
totSessionBandwidth,
(unsigned char const*)
fParent.CNAME(),
NULL /* we're a client */,
fRTPSource);
if (fRTCPInstance == NULL) {
env().setResultMsg("Failed to create RTCP instance");
break;
}
}
return True;
} while (0);
delete fRTPSocket; fRTPSocket = NULL;
delete fRTCPSocket; fRTCPSocket = NULL;
Medium::close(fRTCPInstance); fRTCPInstance = NULL;
Medium::close(fReadSource); fReadSource = fRTPSource = NULL;
fClientPortNum = 0;
return False;
}
// IDemux
void CInputStream::UpdateStreams()
{
DisposeStreams();
INPUTSTREAM_IDS streamIDs;
try
{
streamIDs = m_pStruct->GetStreamIds();
}
catch (std::exception &e)
{
DisposeStreams();
CLog::Log(LOGERROR, "CInputStream::UpdateStreams - error GetStreamIds. Reason: %s", e.what());
return;
}
if (streamIDs.m_streamCount > INPUTSTREAM_IDS::MAX_STREAM_COUNT)
{
DisposeStreams();
return;
}
for (unsigned int i=0; i<streamIDs.m_streamCount; i++)
{
INPUTSTREAM_INFO stream;
try
{
stream = m_pStruct->GetStream(streamIDs.m_streamIds[i]);
}
catch (std::exception &e)
{
DisposeStreams();
CLog::Log(LOGERROR, "CInputStream::GetTotalTime - error GetStream. Reason: %s", e.what());
return;
}
if (stream.m_streamType == INPUTSTREAM_INFO::TYPE_NONE)
continue;
std::string codecName(stream.m_codecName);
StringUtils::ToLower(codecName);
AVCodec *codec = avcodec_find_decoder_by_name(codecName.c_str());
if (!codec)
continue;
CDemuxStream *demuxStream;
if (stream.m_streamType == INPUTSTREAM_INFO::TYPE_AUDIO)
{
CDemuxStreamAudio *audioStream = new CDemuxStreamAudio();
audioStream->iChannels = stream.m_Channels;
audioStream->iSampleRate = stream.m_SampleRate;
audioStream->iBlockAlign = stream.m_BlockAlign;
audioStream->iBitRate = stream.m_BitRate;
audioStream->iBitsPerSample = stream.m_BitsPerSample;
demuxStream = audioStream;
}
else if (stream.m_streamType == INPUTSTREAM_INFO::TYPE_VIDEO)
{
CDemuxStreamVideo *videoStream = new CDemuxStreamVideo();
videoStream->iFpsScale = stream.m_FpsScale;
videoStream->iFpsRate = stream.m_FpsRate;
videoStream->iWidth = stream.m_Width;
videoStream->iHeight = stream.m_Height;
videoStream->fAspect = stream.m_Aspect;
videoStream->stereo_mode = "mono";
demuxStream = videoStream;
}
else if (stream.m_streamType == INPUTSTREAM_INFO::TYPE_SUBTITLE)
{
// TODO needs identifier in INPUTSTREAM_INFO
continue;
}
else
continue;
demuxStream->codec = codec->id;
demuxStream->bandwidth = stream.m_Bandwidth;
demuxStream->codecName = stream.m_codecInternalName;
demuxStream->uniqueId = streamIDs.m_streamIds[i];
demuxStream->language[0] = stream.m_language[0];
demuxStream->language[1] = stream.m_language[1];
demuxStream->language[2] = stream.m_language[2];
demuxStream->language[3] = stream.m_language[3];
if (stream.m_ExtraData && stream.m_ExtraSize)
{
demuxStream->ExtraData = new uint8_t[stream.m_ExtraSize];
demuxStream->ExtraSize = stream.m_ExtraSize;
for (unsigned int j=0; j<stream.m_ExtraSize; j++)
demuxStream->ExtraData[j] = stream.m_ExtraData[j];
}
m_streams[demuxStream->uniqueId] = demuxStream;
}
}
void PhVideoDecoder::open(QString fileName)
{
close();
PHDEBUG << fileName;
_currentFrame = PHFRAMEMIN;
if(avformat_open_input(&_formatContext, fileName.toStdString().c_str(), NULL, NULL) < 0) {
emit openFailed();
close();
return;
}
PHDEBUG << "Retrieve stream information";
if (avformat_find_stream_info(_formatContext, NULL) < 0) {
emit openFailed();
close();
return; // Couldn't find stream information
}
// Disable dump for specs
if(PhDebug::logMask() & 1)
av_dump_format(_formatContext, 0, fileName.toStdString().c_str(), 0);
// Find video stream :
for(int i = 0; i < (int)_formatContext->nb_streams; i++) {
AVMediaType streamType = _formatContext->streams[i]->codec->codec_type;
PHDEBUG << i << ":" << streamType;
switch(streamType) {
case AVMEDIA_TYPE_VIDEO:
// Some containers are advertised with several video streams.
// For example, one is the main stream and the other one is just a cover picture (single frame).
// Here we choose the one that has the largest number of frames.
if (!_videoStream || _videoStream->nb_frames < _formatContext->streams[i]->nb_frames) {
_videoStream = _formatContext->streams[i];
}
PHDEBUG << "\t=> video";
break;
case AVMEDIA_TYPE_AUDIO:
if(_useAudio && (_audioStream == NULL))
_audioStream = _formatContext->streams[i];
PHDEBUG << "\t=> audio";
break;
default:
PHDEBUG << "\t=> unknown";
break;
}
}
if(_videoStream == NULL) {
emit openFailed();
close();
return;
}
// Looking for timecode type
_tcType = PhTimeCode::computeTimeCodeType(this->framePerSecond());
PHDEBUG << "size : " << _videoStream->codec->width << "x" << _videoStream->codec->height;
AVCodec * videoCodec = avcodec_find_decoder(_videoStream->codec->codec_id);
if(videoCodec == NULL) {
PHDEBUG << "Unable to find the codec:" << _videoStream->codec->codec_id;
emit openFailed();
close();
return;
}
if (avcodec_open2(_videoStream->codec, videoCodec, NULL) < 0) {
PHDEBUG << "Unable to open the codec:" << _videoStream->codec;
emit openFailed();
close();
return;
}
_videoFrame = av_frame_alloc();
if(_audioStream) {
AVCodec* audioCodec = avcodec_find_decoder(_audioStream->codec->codec_id);
if(audioCodec) {
if(avcodec_open2(_audioStream->codec, audioCodec, NULL) < 0) {
PHDEBUG << "Unable to open audio codec.";
_audioStream = NULL;
}
else {
_audioFrame = av_frame_alloc();
PHDEBUG << "Audio OK.";
}
}
else {
PHDEBUG << "Unable to find codec for audio.";
_audioStream = NULL;
}
}
_fileName = fileName;
emit opened(_tcType, frameIn(), frameLength(), width(), height(), codecName());
}
Boolean MediaSubsession::createSourceObjects(int useSpecialRTPoffset) {
do {
// First, check "fProtocolName"
if (strcmp(fProtocolName, "UDP") == 0) {
// A UDP-packetized stream (*not* a RTP stream)
fReadSource = BasicUDPSource::createNew(env(), fRTPSocket);
fRTPSource = NULL; // Note!
if (strcmp(fCodecName, "MP2T") == 0) { // MPEG-2 Transport Stream
fReadSource = MPEG2TransportStreamFramer::createNew(env(), fReadSource);
// this sets "durationInMicroseconds" correctly, based on the PCR values
}
} else {
// Check "fCodecName" against the set of codecs that we support,
// and create our RTP source accordingly
// (Later make this code more efficient, as this set grows #####)
// (Also, add more fmts that can be implemented by SimpleRTPSource#####)
Boolean createSimpleRTPSource = False; // by default; can be changed below
Boolean doNormalMBitRule = False; // default behavior if "createSimpleRTPSource" is True
if (strcmp(fCodecName, "QCELP") == 0) { // QCELP audio
fReadSource =
QCELPAudioRTPSource::createNew(env(), fRTPSocket, fRTPSource,
fRTPPayloadFormat,
fRTPTimestampFrequency);
// Note that fReadSource will differ from fRTPSource in this case
} else if (strcmp(fCodecName, "AMR") == 0) { // AMR audio (narrowband)
fReadSource =
AMRAudioRTPSource::createNew(env(), fRTPSocket, fRTPSource,
fRTPPayloadFormat, 0 /*isWideband*/,
fNumChannels, fOctetalign, fInterleaving,
fRobustsorting, fCRC);
// Note that fReadSource will differ from fRTPSource in this case
} else if (strcmp(fCodecName, "AMR-WB") == 0) { // AMR audio (wideband)
fReadSource =
AMRAudioRTPSource::createNew(env(), fRTPSocket, fRTPSource,
fRTPPayloadFormat, 1 /*isWideband*/,
fNumChannels, fOctetalign, fInterleaving,
fRobustsorting, fCRC);
// Note that fReadSource will differ from fRTPSource in this case
} else if (strcmp(fCodecName, "MPA") == 0) { // MPEG-1 or 2 audio
fReadSource = fRTPSource
= MPEG1or2AudioRTPSource::createNew(env(), fRTPSocket,
fRTPPayloadFormat,
fRTPTimestampFrequency);
} else if (strcmp(fCodecName, "MPA-ROBUST") == 0) { // robust MP3 audio
fReadSource = fRTPSource
= MP3ADURTPSource::createNew(env(), fRTPSocket, fRTPPayloadFormat,
fRTPTimestampFrequency);
if (fRTPSource == NULL) break;
if (!fReceiveRawMP3ADUs) {
// Add a filter that deinterleaves the ADUs after depacketizing them:
MP3ADUdeinterleaver* deinterleaver
= MP3ADUdeinterleaver::createNew(env(), fRTPSource);
if (deinterleaver == NULL) break;
// Add another filter that converts these ADUs to MP3 frames:
fReadSource = MP3FromADUSource::createNew(env(), deinterleaver);
}
} else if (strcmp(fCodecName, "X-MP3-DRAFT-00") == 0) {
// a non-standard variant of "MPA-ROBUST" used by RealNetworks
// (one 'ADU'ized MP3 frame per packet; no headers)
fRTPSource
= SimpleRTPSource::createNew(env(), fRTPSocket, fRTPPayloadFormat,
fRTPTimestampFrequency,
"audio/MPA-ROBUST" /*hack*/);
if (fRTPSource == NULL) break;
// Add a filter that converts these ADUs to MP3 frames:
fReadSource = MP3FromADUSource::createNew(env(), fRTPSource,
False /*no ADU header*/);
} else if (strcmp(fCodecName, "MP4A-LATM") == 0) { // MPEG-4 LATM audio
fReadSource = fRTPSource
= MPEG4LATMAudioRTPSource::createNew(env(), fRTPSocket,
fRTPPayloadFormat,
fRTPTimestampFrequency);
} else if (strcmp(fCodecName, "VORBIS") == 0) { // Vorbis audio
fReadSource = fRTPSource
= VorbisAudioRTPSource::createNew(env(), fRTPSocket,
fRTPPayloadFormat,
fRTPTimestampFrequency);
} else if (strcmp(fCodecName, "VP8") == 0) { // VP8 video
fReadSource = fRTPSource
= VP8VideoRTPSource::createNew(env(), fRTPSocket,
fRTPPayloadFormat,
fRTPTimestampFrequency);
} else if (strcmp(fCodecName, "AC3") == 0 || strcmp(fCodecName, "EAC3") == 0) { // AC3 audio
fReadSource = fRTPSource
= AC3AudioRTPSource::createNew(env(), fRTPSocket,
fRTPPayloadFormat,
fRTPTimestampFrequency);
} else if (strcmp(fCodecName, "MP4V-ES") == 0) { // MPEG-4 Elementary Stream video
fReadSource = fRTPSource
= MPEG4ESVideoRTPSource::createNew(env(), fRTPSocket,
fRTPPayloadFormat,
fRTPTimestampFrequency);
} else if (strcmp(fCodecName, "MPEG4-GENERIC") == 0) {
fReadSource = fRTPSource
= MPEG4GenericRTPSource::createNew(env(), fRTPSocket,
fRTPPayloadFormat,
fRTPTimestampFrequency,
fMediumName, fMode,
fSizelength, fIndexlength,
fIndexdeltalength);
} else if (strcmp(fCodecName, "MPV") == 0) { // MPEG-1 or 2 video
fReadSource = fRTPSource
= MPEG1or2VideoRTPSource::createNew(env(), fRTPSocket,
fRTPPayloadFormat,
fRTPTimestampFrequency);
} else if (strcmp(fCodecName, "MP2T") == 0) { // MPEG-2 Transport Stream
fRTPSource = SimpleRTPSource::createNew(env(), fRTPSocket, fRTPPayloadFormat,
fRTPTimestampFrequency, "video/MP2T",
0, False);
fReadSource = MPEG2TransportStreamFramer::createNew(env(), fRTPSource);
// this sets "durationInMicroseconds" correctly, based on the PCR values
} else if (strcmp(fCodecName, "H261") == 0) { // H.261
fReadSource = fRTPSource
= H261VideoRTPSource::createNew(env(), fRTPSocket,
fRTPPayloadFormat,
fRTPTimestampFrequency);
} else if (strcmp(fCodecName, "H263-1998") == 0 ||
strcmp(fCodecName, "H263-2000") == 0) { // H.263+
fReadSource = fRTPSource
= H263plusVideoRTPSource::createNew(env(), fRTPSocket,
fRTPPayloadFormat,
fRTPTimestampFrequency);
} else if (strcmp(fCodecName, "H264") == 0) {
fReadSource = fRTPSource
= H264VideoRTPSource::createNew(env(), fRTPSocket,
fRTPPayloadFormat,
fRTPTimestampFrequency);
} else if (strcmp(fCodecName, "DV") == 0) {
fReadSource = fRTPSource
= DVVideoRTPSource::createNew(env(), fRTPSocket,
fRTPPayloadFormat,
fRTPTimestampFrequency);
} else if (strcmp(fCodecName, "JPEG") == 0) { // motion JPEG
fReadSource = fRTPSource
= JPEGVideoRTPSource::createNew(env(), fRTPSocket,
fRTPPayloadFormat,
fRTPTimestampFrequency,
videoWidth(),
videoHeight());
} else if (strcmp(fCodecName, "X-QT") == 0
|| strcmp(fCodecName, "X-QUICKTIME") == 0) {
// Generic QuickTime streams, as defined in
// <http://developer.apple.com/quicktime/icefloe/dispatch026.html>
char* mimeType
= new char[strlen(mediumName()) + strlen(codecName()) + 2] ;
sprintf(mimeType, "%s/%s", mediumName(), codecName());
fReadSource = fRTPSource
= QuickTimeGenericRTPSource::createNew(env(), fRTPSocket,
fRTPPayloadFormat,
fRTPTimestampFrequency,
mimeType);
delete[] mimeType;
} else if ( strcmp(fCodecName, "PCMU") == 0 // PCM u-law audio
|| strcmp(fCodecName, "GSM") == 0 // GSM audio
|| strcmp(fCodecName, "DVI4") == 0 // DVI4 (IMA ADPCM) audio
|| strcmp(fCodecName, "PCMA") == 0 // PCM a-law audio
|| strcmp(fCodecName, "MP1S") == 0 // MPEG-1 System Stream
|| strcmp(fCodecName, "MP2P") == 0 // MPEG-2 Program Stream
|| strcmp(fCodecName, "L8") == 0 // 8-bit linear audio
|| strcmp(fCodecName, "L16") == 0 // 16-bit linear audio
|| strcmp(fCodecName, "L20") == 0 // 20-bit linear audio (RFC 3190)
|| strcmp(fCodecName, "L24") == 0 // 24-bit linear audio (RFC 3190)
|| strcmp(fCodecName, "G726-16") == 0 // G.726, 16 kbps
|| strcmp(fCodecName, "G726-24") == 0 // G.726, 24 kbps
|| strcmp(fCodecName, "G726-32") == 0 // G.726, 32 kbps
|| strcmp(fCodecName, "G726-40") == 0 // G.726, 40 kbps
|| strcmp(fCodecName, "SPEEX") == 0 // SPEEX audio
|| strcmp(fCodecName, "T140") == 0 // T.140 text (RFC 4103)
|| strcmp(fCodecName, "DAT12") == 0 // 12-bit nonlinear audio (RFC 3190)
) {
createSimpleRTPSource = True;
useSpecialRTPoffset = 0;
} else if (useSpecialRTPoffset >= 0) {
// We don't know this RTP payload format, but try to receive
// it using a 'SimpleRTPSource' with the specified header offset:
createSimpleRTPSource = True;
} else {
env().setResultMsg("RTP payload format unknown or not supported");
break;
}
if (createSimpleRTPSource) {
char* mimeType
= new char[strlen(mediumName()) + strlen(codecName()) + 2] ;
sprintf(mimeType, "%s/%s", mediumName(), codecName());
fReadSource = fRTPSource
= SimpleRTPSource::createNew(env(), fRTPSocket, fRTPPayloadFormat,
fRTPTimestampFrequency, mimeType,
(unsigned)useSpecialRTPoffset,
doNormalMBitRule);
delete[] mimeType;
}
}
return True;
} while (0);
return False; // an error occurred
}
const QString& JISX0201CharCodec::name() const
{
return codecName();
}