mediasession.cpp

流媒体传输协议的实现代码,非常有用.可以支持rtsp mms等流媒体传输协议

  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,
  // and that video sessions use a frequency of 90000.
  // 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;
  return 8000; // for "audio", and any other medium
}

static unsigned computeSeqNumStagger(unsigned staggerSeconds) {
  // To compute the sequence number stagger, assume
  // - one MP3 frame per packet
  // - 1152 samples per MP3 frame (defined by MP3 standard)
  // - a sampling frequency of 44100 Hz
  // (Later, allow this to be parameterized)#####
  unsigned const samplesPerFrame = 1152;
  unsigned samplesPerSecond = 44100;
  double secondsPerFrame = (double)samplesPerFrame/samplesPerSecond;
  return (unsigned)(staggerSeconds/secondsPerFrame);
}

Boolean MediaSession
::initiateByMediaType(char const* mimeType,
		      MediaSubsession*& resultSubsession,
		      PrioritizedRTPStreamSelector*& resultMultiSource,
		      int& resultMultiSourceSessionId,
		      int useSpecialRTPoffset) {
  // Look through this session's subsessions for media that match "mimeType"
  resultSubsession = NULL;
  resultMultiSource = NULL;
  resultMultiSourceSessionId = 0;
  unsigned maxStaggerSeconds = 0;
  MediaSubsessionIterator iter(*this);
  MediaSubsession* subsession;
  while ((subsession = iter.next()) != NULL) {
    if (resultMultiSourceSessionId != 0
	&& subsession->mctSLAPSessionId() != resultMultiSourceSessionId) {
      // We're using a multi-source SLAP session, but this subsession
      // isn't part of it
      continue;
    }

    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;
    }

    if (subsession->mctSLAPSessionId() == 0) {
      // Normal case: a single session
      resultSubsession = subsession;
      break; // use this
    } else {
      // Special case: a multi-source SLAP session
      resultMultiSourceSessionId = subsession->mctSLAPSessionId();
      unsigned subsessionStaggerSeconds = subsession->mctSLAPStagger();
      if (subsessionStaggerSeconds > maxStaggerSeconds) {
	maxStaggerSeconds = subsessionStaggerSeconds;
      }
    }
  }

  if (resultSubsession == NULL && resultMultiSourceSessionId == 0) {
    envir().setResultMsg("Session has no usable media subsession");
    return False;
  }

  if (resultMultiSourceSessionId != 0) {
    // We have a multi-source MCT SLAP session; create a selector for it:
    unsigned seqNumStagger = computeSeqNumStagger(maxStaggerSeconds);
    resultMultiSource
      = PrioritizedRTPStreamSelector::createNew(envir(), seqNumStagger);
    if (resultMultiSource == NULL) return False;
    // Note: each subsession has its own RTCP instance; we don't return them

    // Then run through the subsessions again, adding each of the sources:
    iter.reset();
    while ((subsession = iter.next()) != NULL) {
      if (subsession->mctSLAPSessionId() == resultMultiSourceSessionId) {
	resultMultiSource->addInputRTPStream(subsession->rtpSource(),
					     subsession->rtcpInstance());
      }
    }
  }

  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()),
    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),
    fPlayEndTime(0.0),
    fMCT_SLAP_SessionId(0), fMCT_SLAP_Stagger(0),
    fVideoWidth(0), fVideoHeight(0), fVideoFPS(0), fNumChannels(1), fScale(1.0f),
    fRTPSocket(NULL), fRTCPSocket(NULL),
    fRTPSource(NULL), fRTCPInstance(NULL), fReadSource(NULL) {
#ifdef SUPPORT_REAL_RTSP
    RealInitSDPAttributes(this);
#endif
}

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

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

  delete fNext;
#ifdef SUPPORT_REAL_RTSP
  RealReclaimSDPAttributes(this);
#endif
}

float 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)
    Groupsock* oldGroupsock = NULL;
    Boolean success = False;
    struct in_addr tempAddr;
    tempAddr.s_addr = connectionEndpointAddress();
        // This could get changed later, as a result of a RTSP "SETUP"
    while (1) {
      unsigned short rtpPortNum = fClientPortNum&~1;
      if (isSSM()) {
	fRTPSocket = new Groupsock(env(), tempAddr, fSourceFilterAddr,
				   rtpPortNum);
      } else {
	fRTPSocket = new Groupsock(env(), tempAddr, rtpPortNum, 255);
      }
      if (fRTPSocket == NULL) {
	env().setResultMsg("Failed to create RTP socket");
	break;
      }

      // Get the client port number, to make sure that it's even (for RTP):
      Port clientPort(0);
      if (!getSourcePort(env(), fRTPSocket->socketNum(), clientPort)) {
	break;
      }
      fClientPortNum = ntohs(clientPort.num());

      // If the port number's not even, try again:
      if ((fClientPortNum&1) == 0) {
	success = True;
	break;
      }
      // Try again:
      delete oldGroupsock;
      oldGroupsock = fRTPSocket;
      fClientPortNum = 0;
    }
    delete oldGroupsock;
    if (!success) break;

    // Set our RTCP port to be the RTP port +1
    unsigned short const rtcpPortNum = fClientPortNum|1;
    if (isSSM()) {
      fRTCPSocket = new Groupsock(env(), tempAddr, fSourceFilterAddr,
				  rtcpPortNum);
      // Also, send RTCP packets back to the source via unicast:
      if (fRTCPSocket != NULL) {
	fRTCPSocket->changeDestinationParameters(fSourceFilterAddr,0,~0);
      }
    } 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;
    }

    // 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;
      Boolean doNormalMBitRule = False; // used if "createSimpleRTPSource"
      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, "JPEG") == 0) { // motion JPEG
	fReadSource = fRTPSource
	  = JPEGVideoRTPSource::createNew(env(), fRTPSocket,
					  fRTPPayloadFormat,
					  fRTPTimestampFrequency);
      } 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());