mediasession.cpp

H.264 RTSP 串流(live 555)視窗版本

  // Look up the codec name and timestamp frequency for known (static)
  // RTP payload formats.
  char const* temp = NULL;
  switch (rtpPayloadType) {
  case 0: {temp = "PCMU"; freq = 8000; nCh = 1; break;}
  case 2: {temp = "G726-32"; freq = 8000; nCh = 1; break;}
  case 3: {temp = "GSM"; freq = 8000; nCh = 1; break;}
  case 4: {temp = "G723"; freq = 8000; nCh = 1; break;}
  case 5: {temp = "DVI4"; freq = 8000; nCh = 1; break;}
  case 6: {temp = "DVI4"; freq = 16000; nCh = 1; break;}
  case 7: {temp = "LPC"; freq = 8000; nCh = 1; break;}
  case 8: {temp = "PCMA"; freq = 8000; nCh = 1; break;}
  case 9: {temp = "G722"; freq = 8000; nCh = 1; break;}
  case 10: {temp = "L16"; freq = 44100; nCh = 2; break;}
  case 11: {temp = "L16"; freq = 44100; nCh = 1; break;}
  case 12: {temp = "QCELP"; freq = 8000; nCh = 1; break;}
  case 14: {temp = "MPA"; freq = 90000; nCh = 1; break;}
    // 'number of channels' is actually encoded in the media stream
  case 15: {temp = "G728"; freq = 8000; nCh = 1; break;}
  case 16: {temp = "DVI4"; freq = 11025; nCh = 1; break;}
  case 17: {temp = "DVI4"; freq = 22050; nCh = 1; break;}
  case 18: {temp = "G729"; freq = 8000; nCh = 1; break;}
  case 25: {temp = "CELB"; freq = 90000; nCh = 1; break;}
  case 26: {temp = "JPEG"; freq = 90000; nCh = 1; break;}
  case 28: {temp = "NV"; freq = 90000; nCh = 1; break;}
  case 31: {temp = "H261"; freq = 90000; nCh = 1; break;}
  case 32: {temp = "MPV"; freq = 90000; nCh = 1; break;}
  case 33: {temp = "MP2T"; freq = 90000; nCh = 1; break;}
  case 34: {temp = "H263"; freq = 90000; nCh = 1; break;}
  };

  return strDup(temp);
}

unsigned MediaSession::guessRTPTimestampFrequency(char const* mediumName,
						  char const* codecName) {
  // By default, we assume that audio sessions use a frequency of 8000,
  // video sessions use a frequency of 90000,
  // and text sessions use a frequency of 1000.
  // Begin by checking for known exceptions to this rule
  // (where the frequency is known unambiguously (e.g., not like "DVI4"))
  if (strcmp(codecName, "L16") == 0) return 44100;
  if (strcmp(codecName, "MPA") == 0
      || strcmp(codecName, "MPA-ROBUST") == 0
      || strcmp(codecName, "X-MP3-DRAFT-00")) return 90000;

  // Now, guess default values:
  if (strcmp(mediumName, "video") == 0) return 90000;
  else if (strcmp(mediumName, "text") == 0) return 1000;
  return 8000; // for "audio", and any other medium
}

Boolean MediaSession
::initiateByMediaType(char const* mimeType,
		      MediaSubsession*& resultSubsession,
		      int useSpecialRTPoffset) {
  // Look through this session's subsessions for media that match "mimeType"
  resultSubsession = NULL;
  MediaSubsessionIterator iter(*this);
  MediaSubsession* subsession;
  while ((subsession = iter.next()) != NULL) {
    Boolean wasAlreadyInitiated = subsession->readSource() != NULL;
    if (!wasAlreadyInitiated) {
      // Try to create a source for this subsession:
      if (!subsession->initiate(useSpecialRTPoffset)) return False;
    }

    // Make sure the source's MIME type is one that we handle:
    if (strcmp(subsession->readSource()->MIMEtype(), mimeType) != 0) {
      if (!wasAlreadyInitiated) subsession->deInitiate();
      continue;
    }

    resultSubsession = subsession;
    break; // use this
  }

  if (resultSubsession == NULL) {
    envir().setResultMsg("Session has no usable media subsession");
    return False;
  }

  return True;
}


////////// MediaSubsessionIterator //////////

MediaSubsessionIterator::MediaSubsessionIterator(MediaSession& session)
  : fOurSession(session) {
  reset();
}

MediaSubsessionIterator::~MediaSubsessionIterator() {
}

MediaSubsession* MediaSubsessionIterator::next() {
  MediaSubsession* result = fNextPtr;

  if (fNextPtr != NULL) fNextPtr = fNextPtr->fNext;

  return result;
}

void MediaSubsessionIterator::reset() {
  fNextPtr = fOurSession.fSubsessionsHead;
}

////////// MediaSubsession //////////

MediaSubsession::MediaSubsession(MediaSession& parent)
  : sessionId(NULL), serverPortNum(0), sink(NULL), miscPtr(NULL),
    fParent(parent), fNext(NULL),
    fConnectionEndpointName(NULL),
    fClientPortNum(0), fRTPPayloadFormat(0xFF),
    fSavedSDPLines(NULL), fMediumName(NULL), fCodecName(NULL), fProtocolName(NULL),
    fRTPTimestampFrequency(0), fControlPath(NULL),
    fSourceFilterAddr(parent.sourceFilterAddr()), fBandwidth(0),
    fAuxiliarydatasizelength(0), fConstantduration(0), fConstantsize(0),
    fCRC(0), fCtsdeltalength(0), fDe_interleavebuffersize(0), fDtsdeltalength(0),
    fIndexdeltalength(0), fIndexlength(0), fInterleaving(0), fMaxdisplacement(0),
    fObjecttype(0), fOctetalign(0), fProfile_level_id(0), fRobustsorting(0),
    fSizelength(0), fStreamstateindication(0), fStreamtype(0),
    fCpresent(False), fRandomaccessindication(False),
    fConfig(NULL), fMode(NULL), fSpropParameterSets(NULL),
    fPlayStartTime(0.0), fPlayEndTime(0.0),
    fVideoWidth(0), fVideoHeight(0), fVideoFPS(0), fNumChannels(1), fScale(1.0f), fNPT_PTS_Offset(0.0f),
    fRTPSocket(NULL), fRTCPSocket(NULL),
    fRTPSource(NULL), fRTCPInstance(NULL), fReadSource(NULL) {
  rtpInfo.seqNum = 0; rtpInfo.timestamp = 0; rtpInfo.infoIsNew = False;
}

MediaSubsession::~MediaSubsession() {
  deInitiate();

  delete[] fConnectionEndpointName; delete[] fSavedSDPLines;
  delete[] fMediumName; delete[] fCodecName; delete[] fProtocolName;
  delete[] fControlPath; delete[] fConfig; delete[] fMode; delete[] fSpropParameterSets;

  delete fNext;
}

double MediaSubsession::playStartTime() const {
  if (fPlayStartTime > 0) return fPlayStartTime;

  return fParent.playStartTime();
}

double MediaSubsession::playEndTime() const {
  if (fPlayEndTime > 0) return fPlayEndTime;

  return fParent.playEndTime();
}

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;
      NoReuse dummy; // ensures that our new ephemeral port number won't be one that's already in use

      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;
	  Groupsock* existing = (Groupsock*)socketHashTable->Add((char const*)key, fRTPSocket);
	  delete existing; // in case it wasn't NULL
	  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.
	  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;
	  Groupsock* existing = (Groupsock*)socketHashTable->Add((char const*)key, fRTPSocket);
	  delete existing; // in case it wasn't NULL
	  continue;
	}
      }

      // Clean up the socket hash table (and contents):
      Groupsock* oldGS;
      while ((oldGS = (Groupsock*)socketHashTable->RemoveNext()) != NULL) {
	delete oldGS;
      }
      delete socketHashTable;

      if (!success) break; // a fatal error occurred trying to create the RTP and RTCP sockets; we can't continue
    }

    // Try to use a big receive buffer for RTP - at least 0.1 second of
    // specified bandwidth and at least 50 KB
    unsigned rtpBufSize = fBandwidth * 25 / 2; // 1 kbps * 0.1 s = 12.5 bytes
    if (rtpBufSize < 50 * 1024)
      rtpBufSize = 50 * 1024;
    increaseReceiveBufferTo(env(), fRTPSocket->socketNum(), rtpBufSize);

    // 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);
    }

    // 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
	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, "MP4V-ES") == 0) { // MPEG-4 Elem Str vid
	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,