C++ MediaSubsession::codecName方法代码示例

bool CRTSPClient::setupStreams()
{
  //setup streams
  XBMC->Log(LOG_DEBUG, "CRTSPClient::setupStreams()");
  Boolean madeProgress=False;
  MediaSubsessionIterator iter(*m_session);
  MediaSubsession *subsession;

  while ((subsession = iter.next()) != NULL) 
  {
    if (subsession->clientPortNum() == 0) continue; // port # was not set

    if (!clientSetupSubsession(m_ourClient, subsession, streamUsingTCP)) 
    {
      XBMC->Log(LOG_DEBUG,  "Failed to setup %s %s %s" ,subsession->mediumName(),subsession->codecName(),m_env->getResultMsg() );;
    } 
    else 
    {
      XBMC->Log(LOG_DEBUG,  "Setup %s %s %d %d" ,subsession->mediumName(),subsession->codecName(),subsession->clientPortNum(),subsession->clientPortNum()+1);;
      madeProgress = True;
    }
  }
  if (!madeProgress) 
  {
    shutdown();
    return false;
  }
  return true;
}

BOOL setupStreams( unsigned *pResponseCode /*= NULL*/ ) 
{
	MediaSubsessionIterator iter(*session);
	MediaSubsession *subsession;
	Boolean madeProgress = False;

	BOOL bResult = TRUE;
	while ((subsession = iter.next()) != NULL) 
	{
		if (subsession->clientPortNum() == 0) continue; // port # was not set

		if ( !clientSetupSubsession(ourClient, subsession, streamUsingTCP, pResponseCode ) ) 
		{
			*env << "Failed to setup \"" << subsession->mediumName()
				<< "/" << subsession->codecName()
				<< "\" subsession: " << env->getResultMsg() << "\n";
			
			bResult = FALSE;
		}
		else 
		{
			*env << "Setup \"" << subsession->mediumName()
				<< "/" << subsession->codecName()
				<< "\" subsession (client ports " << subsession->clientPortNum()
				<< "-" << subsession->clientPortNum()+1 << ")\n";
			madeProgress = True;

			bResult = TRUE;
		}
	}
	//if (!madeProgress) 
	//	return bResult;
	return bResult;
}

DefaultSink::DefaultSink(UsageEnvironment & env,
                         MediaSubsession & subsession,
                         char const * sink_url,
                         char const * stream_id)
        : MediaSink(env),
          _subsession(subsession),
          _receive_buffer(RECEIVE_BUFFER_SIZE),
          _stream_id(stream_id),
          _writer(libc2rtsp::sink::SinkFactory().gen(sink_url)),
          _verbose(true),
          _sprop_parameter_sets(),
          _have_written_first_frame(false)
{
    if (::strcmp(subsession.codecName(), "H264") == 0) {
        // For H.264 video stream, we use a special sink that adds 'start codes',
        // and (at the start) the SPS and PPS NAL units:
        _sprop_parameter_sets.emplace_back(std::string(subsession.fmtp_spropparametersets()));

    } else if (::strcmp(subsession.codecName(), "H265") == 0) {
        // For H.265 video stream, we use a special sink that adds 'start codes',
        // and (at the start) the VPS, SPS, and PPS NAL units:
        _sprop_parameter_sets.emplace_back(std::string(subsession.fmtp_spropvps())); // VPS
        _sprop_parameter_sets.emplace_back(std::string(subsession.fmtp_spropsps())); // SPS
        _sprop_parameter_sets.emplace_back(std::string(subsession.fmtp_sproppps())); // PPS

    } else {
        crLogE("DefaultSink::DefaultSink() Unsupported subsession: {}/{}",
               subsession.mediumName(), subsession.codecName());
        throw std::bad_alloc();
    }
}

ProxyServerMediaSubsession
::ProxyServerMediaSubsession(MediaSubsession& mediaSubsession,
			     portNumBits initialPortNum, Boolean multiplexRTCPWithRTP)
  : OnDemandServerMediaSubsession(mediaSubsession.parentSession().envir(), True/*reuseFirstSource*/,
				  initialPortNum, multiplexRTCPWithRTP),
    fClientMediaSubsession(mediaSubsession), fCodecName(strDup(mediaSubsession.codecName())),
    fNext(NULL), fHaveSetupStream(False) {
}

void setupStreams() {
  MediaSubsessionIterator iter(*session);
  MediaSubsession *subsession;
  Boolean madeProgress = False;

  while ((subsession = iter.next()) != NULL) {
    if (subsession->clientPortNum() == 0) continue; // port # was not set

    if (!clientSetupSubsession(ourClient, subsession, streamUsingTCP)) {
      *env << "Failed to setup \"" << subsession->mediumName()
		    << "/" << subsession->codecName()
		    << "\" subsession: " << env->getResultMsg() << "\n";
    } else {
      *env << "Setup \"" << subsession->mediumName()
		    << "/" << subsession->codecName()
		    << "\" subsession (client ports " << subsession->clientPortNum()
		    << "-" << subsession->clientPortNum()+1 << ")\n";
      madeProgress = True;
    }
  }
  if (!madeProgress) shutdown();
}

void subsessionByeHandler(void* clientData) {
  struct timeval timeNow;
  gettimeofday(&timeNow, NULL);
  unsigned secsDiff = timeNow.tv_sec - startTime.tv_sec;

  MediaSubsession* subsession = (MediaSubsession*)clientData;
  *env << "Received RTCP \"BYE\" on \"" << subsession->mediumName()
	<< "/" << subsession->codecName()
	<< "\" subsession (after " << secsDiff
	<< " seconds)\n";

  // Act now as if the subsession had closed:
  subsessionAfterPlaying(subsession);
}

void ProxyServerMediaSession::continueAfterDESCRIBE(char const* sdpDescription) {
    describeCompletedFlag = 1;

    // Create a (client) "MediaSession" object from the stream's SDP description ("resultString"), then iterate through its
    // "MediaSubsession" objects, to set up corresponding "ServerMediaSubsession" objects that we'll use to serve the stream's tracks.
    do {
        fClientMediaSession = MediaSession::createNew(envir(), sdpDescription);
        if (fClientMediaSession == NULL) break;

        MediaSubsessionIterator iter(*fClientMediaSession);
        for (MediaSubsession* mss = iter.next(); mss != NULL; mss = iter.next()) {
            ServerMediaSubsession* smss = new ProxyServerMediaSubsession(*mss);
            addSubsession(smss);
            if (fVerbosityLevel > 0) {
                envir() << *this << " added new \"ProxyServerMediaSubsession\" for "
                        << mss->protocolName() << "/" << mss->mediumName() << "/" << mss->codecName() << " track\n";
            }
        }
    } while (0);
}

//.........这里部分代码省略.........
		fStartTime = 0.0f;
	}

	fEndTime= session->playEndTime();
	if (fEndTime <= 0)
	{
		fEndTime = -1.0f;
	}

	{
		/*send setup requesst count*/
		iSetupCount = 0;
	}
	
	// Then, setup the "RTPSource"s for the session:
	MediaSubsessionIterator iter(*(session));
	MediaSubsession *subsession = NULL;
	RtspReqSender *senderSave = pRtspReqSender->getNext();
	if (senderSave == NULL)
	{
		LOG_ERR("error");
		return false;
	}
	CmdSenderDecorator *senderMove = pRtspReqSender;
	
	while ((subsession = iter.next()) != NULL)
	{
		if (!subsession->initiate(-1))
		{
			LOG_ERR("warning");
			continue;
		}

		if (subsession->rtpSource() != NULL)
		{
#if 0			
			// Because we're saving the incoming data, rather than playing
		  	// it in real time, allow an especially large time threshold
		  	// (1 second) for reordering misordered incoming packets:
			unsigned const thresh = 1000000; // 1 second
		  	subsession->rtpSource()->setPacketReorderingThresholdTime(thresh);
#endif
#if 0
			// Set the RTP source's OS socket buffer size as appropriate - either if we were explicitly asked (using -B),
		  	// or if the desired FileSink buffer size happens to be larger than the current OS socket buffer size.
		  	// (The latter case is a heuristic, on the assumption that if the user asked for a large FileSink buffer size,
		  	// then the input data rate may be large enough to justify increasing the OS socket buffer size also.)
			int socketNum = subsession->rtpSource()->RTPgs()->socketNum();
			unsigned curBufferSize = getReceiveBufferSize(*env, socketNum);
			LOG_DEBUG("old receive buffer size:%d", curBufferSize);
			if (fileSinkBufferSize > curBufferSize) 
			{
			    unsigned newBufferSize = setReceiveBufferTo(*env, socketNum, fileSinkBufferSize);
				LOG_DEBUG("new receive buffer size:%d", newBufferSize);
			}
#else		
			int socketNum = subsession->rtpSource()->RTPgs()->socketNum();
		 	unsigned newBufferSize = setReceiveBufferTo(*env, socketNum, maxBufSize);
			LOG_DEBUG("new receive buffer size:%d", newBufferSize);
#endif
		}
		
		if (subsession->readSource() == NULL) 
		{
			LOG_ERR("warning");
			continue; // was not initiated
		}

		/*
		 *TO DO:SET UP SUBSESSION
		 */
		SetupSender *setupSender = new SetupSender(*senderSave);
		if (setupSender == NULL)
		{
			LOG_ERR("warning");
			continue;
		}
        
		sender->RecordSender(setupSender);
		senderMove->setNext(setupSender);
		senderMove = setupSender;
		setupSender->setRspHandler(respHandler);
		setupSender->setSubsession(subsession);

		if (bUseTcp == true)
		{
			if (subsession->clientPortNum() != 0)
			{
				LOG_DEBUG("sub session %p using tcp port :%d!", subsession, subsession->clientPortNum());
				setupSender->setParam(false, true, false);
			}
		}

		iSetupCount++;
					
		LOG_DEBUG("subsession, name:%s, codec:%s", subsession->mediumName(), subsession->codecName());		
	}

	return true;
}

void printQOSData(int exitCode) {
  *env << "begin_QOS_statistics\n";
  
  // Print out stats for each active subsession:
  qosMeasurementRecord* curQOSRecord = qosRecordHead;
  if (session != NULL) {
    MediaSubsessionIterator iter(*session);
    MediaSubsession* subsession;
    while ((subsession = iter.next()) != NULL) {
      RTPSource* src = subsession->rtpSource();
      if (src == NULL) continue;
      
      *env << "subsession\t" << subsession->mediumName()
	   << "/" << subsession->codecName() << "\n";
      
      unsigned numPacketsReceived = 0, numPacketsExpected = 0;
      
      if (curQOSRecord != NULL) {
	numPacketsReceived = curQOSRecord->totNumPacketsReceived;
	numPacketsExpected = curQOSRecord->totNumPacketsExpected;
      }
      *env << "num_packets_received\t" << numPacketsReceived << "\n";
      *env << "num_packets_lost\t" << int(numPacketsExpected - numPacketsReceived) << "\n";
      
      if (curQOSRecord != NULL) {
	unsigned secsDiff = curQOSRecord->measurementEndTime.tv_sec
	  - curQOSRecord->measurementStartTime.tv_sec;
	int usecsDiff = curQOSRecord->measurementEndTime.tv_usec
	  - curQOSRecord->measurementStartTime.tv_usec;
	double measurementTime = secsDiff + usecsDiff/1000000.0;
	*env << "elapsed_measurement_time\t" << measurementTime << "\n";
	
	*env << "kBytes_received_total\t" << curQOSRecord->kBytesTotal << "\n";
	
	*env << "measurement_sampling_interval_ms\t" << qosMeasurementIntervalMS << "\n";
	
	if (curQOSRecord->kbits_per_second_max == 0) {
	  // special case: we didn't receive any data:
	  *env <<
	    "kbits_per_second_min\tunavailable\n"
	    "kbits_per_second_ave\tunavailable\n"
	    "kbits_per_second_max\tunavailable\n";
	} else {
	  *env << "kbits_per_second_min\t" << curQOSRecord->kbits_per_second_min << "\n";
	  *env << "kbits_per_second_ave\t"
	       << (measurementTime == 0.0 ? 0.0 : 8*curQOSRecord->kBytesTotal/measurementTime) << "\n";
	  *env << "kbits_per_second_max\t" << curQOSRecord->kbits_per_second_max << "\n";
	}
	
	*env << "packet_loss_percentage_min\t" << 100*curQOSRecord->packet_loss_fraction_min << "\n";
	double packetLossFraction = numPacketsExpected == 0 ? 1.0
	  : 1.0 - numPacketsReceived/(double)numPacketsExpected;
	if (packetLossFraction < 0.0) packetLossFraction = 0.0;
	*env << "packet_loss_percentage_ave\t" << 100*packetLossFraction << "\n";
	*env << "packet_loss_percentage_max\t"
	     << (packetLossFraction == 1.0 ? 100.0 : 100*curQOSRecord->packet_loss_fraction_max) << "\n";
	
	RTPReceptionStatsDB::Iterator statsIter(src->receptionStatsDB());
	// Assume that there's only one SSRC source (usually the case):
	RTPReceptionStats* stats = statsIter.next(True);
	if (stats != NULL) {
	  *env << "inter_packet_gap_ms_min\t" << stats->minInterPacketGapUS()/1000.0 << "\n";
	  struct timeval totalGaps = stats->totalInterPacketGaps();
	  double totalGapsMS = totalGaps.tv_sec*1000.0 + totalGaps.tv_usec/1000.0;
	  unsigned totNumPacketsReceived = stats->totNumPacketsReceived();
	  *env << "inter_packet_gap_ms_ave\t"
	       << (totNumPacketsReceived == 0 ? 0.0 : totalGapsMS/totNumPacketsReceived) << "\n";
	  *env << "inter_packet_gap_ms_max\t" << stats->maxInterPacketGapUS()/1000.0 << "\n";
	}
	
	curQOSRecord = curQOSRecord->fNext;
      }
    }
  }

  *env << "end_QOS_statistics\n";
  delete qosRecordHead;
}

void continueAfterDESCRIBE(RTSPClient*, int resultCode, char* resultString) {
  if (resultCode != 0) {
    *env << "Failed to get a SDP description from URL \"" << streamURL << "\": " << resultString << "\n";
    shutdown();
  }

  char* sdpDescription = resultString;
  *env << "Opened URL \"" << streamURL << "\", returning a SDP description:\n" << sdpDescription << "\n";

  // Create a media session object from this SDP description:
  session = MediaSession::createNew(*env, sdpDescription);
  delete[] sdpDescription;
  if (session == NULL) {
    *env << "Failed to create a MediaSession object from the SDP description: " << env->getResultMsg() << "\n";
    shutdown();
  } else if (!session->hasSubsessions()) {
    *env << "This session has no media subsessions (i.e., \"m=\" lines)\n";
    shutdown();
  }

  // Then, setup the "RTPSource"s for the session:
  MediaSubsessionIterator iter(*session);
  MediaSubsession *subsession;
  Boolean madeProgress = False;
  char const* singleMediumToTest = singleMedium;
  while ((subsession = iter.next()) != NULL) {
    // If we've asked to receive only a single medium, then check this now:
    if (singleMediumToTest != NULL) {
      if (strcmp(subsession->mediumName(), singleMediumToTest) != 0) {
		  *env << "Ignoring \"" << subsession->mediumName()
			  << "/" << subsession->codecName()
			  << "\" subsession, because we've asked to receive a single " << singleMedium
			  << " session only\n";
	continue;
      } else {
	// Receive this subsession only
	singleMediumToTest = "xxxxx";
	    // this hack ensures that we get only 1 subsession of this type
      }
    }

    if (desiredPortNum != 0) {
      subsession->setClientPortNum(desiredPortNum);
      desiredPortNum += 2;
    }

    if (createReceivers) {
      if (!subsession->initiate(simpleRTPoffsetArg)) {
	*env << "Unable to create receiver for \"" << subsession->mediumName()
	     << "/" << subsession->codecName()
	     << "\" subsession: " << env->getResultMsg() << "\n";
      } else {
	*env << "Created receiver for \"" << subsession->mediumName()
	     << "/" << subsession->codecName()
	     << "\" subsession (client ports " << subsession->clientPortNum()
	     << "-" << subsession->clientPortNum()+1 << ")\n";
	madeProgress = True;
	
	if (subsession->rtpSource() != NULL) {
	  // Because we're saving the incoming data, rather than playing
	  // it in real time, allow an especially large time threshold
	  // (1 second) for reordering misordered incoming packets:
	  unsigned const thresh = 1000000; // 1 second
	  subsession->rtpSource()->setPacketReorderingThresholdTime(thresh);
	  
	  // Set the RTP source's OS socket buffer size as appropriate - either if we were explicitly asked (using -B),
	  // or if the desired FileSink buffer size happens to be larger than the current OS socket buffer size.
	  // (The latter case is a heuristic, on the assumption that if the user asked for a large FileSink buffer size,
	  // then the input data rate may be large enough to justify increasing the OS socket buffer size also.)
	  int socketNum = subsession->rtpSource()->RTPgs()->socketNum();
	  unsigned curBufferSize = getReceiveBufferSize(*env, socketNum);
	  if (socketInputBufferSize > 0 || fileSinkBufferSize > curBufferSize) {
	    unsigned newBufferSize = socketInputBufferSize > 0 ? socketInputBufferSize : fileSinkBufferSize;
	    newBufferSize = setReceiveBufferTo(*env, socketNum, newBufferSize);
	    if (socketInputBufferSize > 0) { // The user explicitly asked for the new socket buffer size; announce it:
	      *env << "Changed socket receive buffer size for the \""
		   << subsession->mediumName()
		   << "/" << subsession->codecName()
		   << "\" subsession from "
		   << curBufferSize << " to "
		   << newBufferSize << " bytes\n";
	    }
	  }
	}
      }
    } else {
      if (subsession->clientPortNum() == 0) {
	*env << "No client port was specified for the \""
	     << subsession->mediumName()
	     << "/" << subsession->codecName()
	     << "\" subsession.  (Try adding the \"-p <portNum>\" option.)\n";
      } else {
		madeProgress = True;
      }
    }
  }
  if (!madeProgress) shutdown();

  // Perform additional 'setup' on each subsession, before playing them:
  setupStreams();
//.........这里部分代码省略.........

int CMPIPTV_RTSP::OpenConnection(void)
{
  this->logger.Log(LOGGER_INFO, METHOD_START_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME);
  this->isRtspSessionSetup = false;

  // LIVE555 works with char, not with TCHAR
  char *tempRtspUrl = ConvertToMultiByte(this->rtspUrl);
  if (tempRtspUrl == NULL)
  {
    return STATUS_ERROR;
  }

  // start LIVE555 worker thread
  this->rtspSchedulerThreadHandle = CreateThread( 
    NULL,                                   // default security attributes
    0,                                      // use default stack size  
    &CMPIPTV_RTSP::RtspSchedulerWorker,     // thread function name
    this,                                   // argument to thread function 
    0,                                      // use default creation flags 
    &this->rtspSchedulerThreadId);          // returns the thread identifier
  if (this->rtspSchedulerThreadHandle == NULL)
  {
    this->logger.Log(LOGGER_ERROR, _T("%s: %s: failed to create RTSP scheduler thread, error = %i"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, GetLastError());
    return STATUS_ERROR;
  }

  this->rtspClient = MPRTSPClient::createNew(this, *this->rtspEnvironment, tempRtspUrl);
  FREE_MEM(tempRtspUrl);
  if (
    this->rtspClient == NULL ||
    SendRtspCommand(METHOD_OPEN_CONNECTION_NAME, _T("OPTIONS")) != STATUS_OK ||
    SendRtspCommand(METHOD_OPEN_CONNECTION_NAME, _T("DESCRIBE")) != STATUS_OK
  )
  {
    CloseConnection();
    return STATUS_ERROR;
  }

  this->rtspSession = MediaSession::createNew(*this->rtspEnvironment, this->rtspResponseResultString);
  if (this->rtspSession == NULL || !this->rtspSession->hasSubsessions())
  {
    this->LogRtspMessage(LOGGER_ERROR, METHOD_OPEN_CONNECTION_NAME, this->rtspSession == NULL ? _T("failed to create session") : _T("session doesn't have sub-sessions"));
    CloseConnection();
    return STATUS_ERROR;
  }

  // Setup the RTP source for the session. Only one sub-session expected/supported.
  MediaSubsessionIterator iter(*this->rtspSession);
  MediaSubsession *subsession = NULL;
  FramedSource *rtspSource = NULL;
  while ((subsession = iter.next()) != NULL)
  {
#ifdef _MBCS
    TCHAR *subSessionName = ConvertToMultiByteA(subsession->mediumName());
    TCHAR *subSessionCodecName = ConvertToMultiByteA(subsession->codecName());
#else
    TCHAR *subSessionName = ConvertToUnicodeA(subsession->mediumName());
    TCHAR *subSessionCodecName = ConvertToUnicodeA(subsession->codecName());
#endif

    if (_tcsncicmp(subSessionName, _T("video"), 5) != 0 || _tcsncicmp(subSessionCodecName, _T("MP2T"), 4) != 0)
    {
      TCHAR *message = FormatString(_T("sub-session medium or codec not supported, medium = %s, codec = %s"), subSessionName, subSessionCodecName);
      this->LogRtspMessage(LOGGER_ERROR, METHOD_OPEN_CONNECTION_NAME, message);
      FREE_MEM(message);
      FREE_MEM(subSessionName);
      FREE_MEM(subSessionCodecName);
      continue;
    }

    // If a client port is configured, find a free pair of ports in the range.
    // The first port is used for RTP; the second port is used for RTCP. Once
    // we find one free port, we assume the next one is also free.
    if (this->rtspRtpClientPortRangeStart > 0)
    {
      struct in_addr destinationAddress;
      destinationAddress.s_addr = our_inet_addr("127.0.0.1");
      unsigned int port = this->rtspRtpClientPortRangeStart;
      Groupsock *groupsock = NULL;
      do
      {
        this->logger.Log(LOGGER_VERBOSE, _T("%s: %s: RTP client port %u"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, port);

        // special construction force not reuse same UDP port
        {
          NoReuse noReuse(*this->rtspEnvironment);
          groupsock = new Groupsock(*this->rtspEnvironment, destinationAddress, port, 1);
        }

        if (groupsock == NULL || groupsock->socketNum() == -1)
        {
          this->logger.Log(LOGGER_WARNING, _T("%s: %s: RTP client port %u occupied, trying next even port"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, port);
          port += 2;
          if (groupsock != NULL)
          {
            delete groupsock;
            groupsock = NULL;
          }
        }
      }
//.........这里部分代码省略.........

//.........这里部分代码省略.........

		*env << "Opened URL \"" << pThis->m_strRTSPUrlA.c_str()
			<< "\", returning a SDP description:\n" << sdpDescription << "\n";

		// Create a media session object from this SDP description:
		session = MediaSession::createNew(*env, sdpDescription);
		delete[] sdpDescription;
		if (session == NULL) 
		{
			*env << "Failed to create a MediaSession object from the SDP description: " << env->getResultMsg() << "\n";
			pThis->SetRtspStatus( RTSPStatus_Error_Connect_Srv );
			break;
		} 
		else if (!session->hasSubsessions()) 
		{
			*env << "This session has no media subsessions (i.e., \"m=\" lines)\n";
			pThis->SetRtspStatus( RTSPStatus_Error_Connect_Srv );
			break;
		}

		// Then, setup the "RTPSource"s for the session:
		MediaSubsessionIterator iter(*session);
		MediaSubsession *subsession;
		Boolean madeProgress = False;
		char const* singleMediumToTest = singleMedium;
		while ((subsession = iter.next()) != NULL) 
		{
			// If we've asked to receive only a single medium, then check this now:
			if (singleMediumToTest != NULL) 
			{
				if (strcmp(subsession->mediumName(), singleMediumToTest) != 0) 
				{
					*env << "Ignoring \"" << subsession->mediumName()
						<< "/" << subsession->codecName()
						<< "\" subsession, because we've asked to receive a single " << singleMedium
						<< " session only\n";
					continue;
				} 
				else 
				{
					// Receive this subsession only
					singleMediumToTest = "xxxxx";
					// this hack ensures that we get only 1 subsession of this type
				}
			}

			desiredPortNum = 0;
			if (desiredPortNum != 0) 
			{
				subsession->setClientPortNum(desiredPortNum);
				desiredPortNum += 2;
			}

			if (true) 
			{
				if (!subsession->initiate(simpleRTPoffsetArg)) 
				{
					*env << "Unable to create receiver for \"" << subsession->mediumName()
						<< "/" << subsession->codecName()
						<< "\" subsession: " << env->getResultMsg() << "\n";
				} 
				else 
				{
					*env << "Created receiver for \"" << subsession->mediumName()
						<< "/" << subsession->codecName()
						<< "\" subsession (client ports " << subsession->clientPortNum()

extern "C" demuxer_t* demux_open_rtp(demuxer_t* demuxer) {
  struct MPOpts *opts = demuxer->opts;
  Boolean success = False;
  do {
    TaskScheduler* scheduler = BasicTaskScheduler::createNew();
    if (scheduler == NULL) break;
    UsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler);
    if (env == NULL) break;

    RTSPClient* rtspClient = NULL;
    SIPClient* sipClient = NULL;

    if (demuxer == NULL || demuxer->stream == NULL) break;  // shouldn't happen
    demuxer->stream->eof = 0; // just in case

    // Look at the stream's 'priv' field to see if we were initiated
    // via a SDP description:
    char* sdpDescription = (char*)(demuxer->stream->priv);
    if (sdpDescription == NULL) {
      // We weren't given a SDP description directly, so assume that
      // we were given a RTSP or SIP URL:
      char const* protocol = demuxer->stream->streaming_ctrl->url->protocol;
      char const* url = demuxer->stream->streaming_ctrl->url->url;
      extern int verbose;
      if (strcmp(protocol, "rtsp") == 0) {
	rtspClient = RTSPClient::createNew(*env, verbose, "MPlayer");
	if (rtspClient == NULL) {
	  fprintf(stderr, "Failed to create RTSP client: %s\n",
		  env->getResultMsg());
	  break;
	}
	sdpDescription = openURL_rtsp(rtspClient, url);
      } else { // SIP
	unsigned char desiredAudioType = 0; // PCMU (use 3 for GSM)
	sipClient = SIPClient::createNew(*env, desiredAudioType, NULL,
					 verbose, "MPlayer");
	if (sipClient == NULL) {
	  fprintf(stderr, "Failed to create SIP client: %s\n",
		  env->getResultMsg());
	  break;
	}
	sipClient->setClientStartPortNum(8000);
	sdpDescription = openURL_sip(sipClient, url);
      }

      if (sdpDescription == NULL) {
	fprintf(stderr, "Failed to get a SDP description from URL \"%s\": %s\n",
		url, env->getResultMsg());
	break;
      }
    }

    // Now that we have a SDP description, create a MediaSession from it:
    MediaSession* mediaSession = MediaSession::createNew(*env, sdpDescription);
    if (mediaSession == NULL) break;


    // Create a 'RTPState' structure containing the state that we just created,
    // and store it in the demuxer's 'priv' field, for future reference:
    RTPState* rtpState = new RTPState;
    rtpState->sdpDescription = sdpDescription;
    rtpState->rtspClient = rtspClient;
    rtpState->sipClient = sipClient;
    rtpState->mediaSession = mediaSession;
    rtpState->audioBufferQueue = rtpState->videoBufferQueue = NULL;
    rtpState->flags = 0;
    rtpState->firstSyncTime.tv_sec = rtpState->firstSyncTime.tv_usec = 0;
    demuxer->priv = rtpState;

    int audiofound = 0, videofound = 0;
    // Create RTP receivers (sources) for each subsession:
    MediaSubsessionIterator iter(*mediaSession);
    MediaSubsession* subsession;
    unsigned desiredReceiveBufferSize;
    while ((subsession = iter.next()) != NULL) {
      // Ignore any subsession that's not audio or video:
      if (strcmp(subsession->mediumName(), "audio") == 0) {
	if (audiofound) {
	  fprintf(stderr, "Additional subsession \"audio/%s\" skipped\n", subsession->codecName());
	  continue;
	}
	desiredReceiveBufferSize = 100000;
      } else if (strcmp(subsession->mediumName(), "video") == 0) {
	if (videofound) {
	  fprintf(stderr, "Additional subsession \"video/%s\" skipped\n", subsession->codecName());
	  continue;
	}
	desiredReceiveBufferSize = 2000000;
      } else {
	continue;
      }

      if (rtsp_port)
          subsession->setClientPortNum (rtsp_port);

      if (!subsession->initiate()) {
	fprintf(stderr, "Failed to initiate \"%s/%s\" RTP subsession: %s\n", subsession->mediumName(), subsession->codecName(), env->getResultMsg());
      } else {
	fprintf(stderr, "Initiated \"%s/%s\" RTP subsession on port %d\n", subsession->mediumName(), subsession->codecName(), subsession->clientPortNum());

//.........这里部分代码省略.........

int CMPIPTV_RTSP::OpenConnection(void)
{
  this->logger.Log(LOGGER_INFO, METHOD_START_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME);
  int result = STATUS_OK;

  this->rtspClient = RTSPClient::createNew(*this->rtspEnvironment);
  result |= (this->rtspClient == NULL);

  if (result == STATUS_OK)
  {
    // RTSPClient works with char, not with TCHAR
    char *tempRtspUrl = ConvertToMultiByte(this->rtspUrl);
    result |= (tempRtspUrl == NULL);
    if (result == STATUS_OK)
    {
      char* optionsResult = this->rtspClient->sendOptionsCmd(tempRtspUrl, NULL, NULL, NULL, this->receiveDataTimeout / 2000);
      result |= (optionsResult == NULL);

      if (result != STATUS_OK)
      {
        TCHAR *message = FormatString(METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("error occured while sending OPTIONS command"));
        this->LogRtspMessage(LOGGER_ERROR, message);
        FREE_MEM(message);
      }
      else
      {
        TCHAR *message = FormatString(METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("OPTIONS result"));
        this->LogFullRtspMessage(LOGGER_VERBOSE, message, optionsResult);
        FREE_MEM(message);

        char *describeResult = this->rtspClient->describeURL(tempRtspUrl, NULL, FALSE, this->receiveDataTimeout / 2000);
        result |= (describeResult == NULL);

        if (result != STATUS_OK)
        {
          TCHAR *message = FormatString(METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("error occured while sending DESCRIBE command"));
          this->LogRtspMessage(LOGGER_ERROR, message);
          FREE_MEM(message);
        }
        else
        {
          TCHAR *message = FormatString(METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("DESCRIBE result"));
          this->LogFullRtspMessage(LOGGER_VERBOSE, message, describeResult);
          FREE_MEM(message);

          this->rtspSession = MediaSession::createNew(*this->rtspEnvironment, describeResult);
          result |= (this->rtspSession == NULL);

          if (result != STATUS_OK)
          {
            TCHAR *message = FormatString(METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("error occured while creating new session"));
            this->LogRtspMessage(LOGGER_ERROR, message);
            FREE_MEM(message);
          }
          else
          {
            result |= (!this->rtspSession->hasSubsessions());
            if (result != STATUS_OK)
            {
              this->logger.Log(LOGGER_ERROR, METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("session doesn't have subsessions"));
            }
            else
            {
              // Then, setup the "RTPSource"s for the session:
              MediaSubsessionIterator iter(*this->rtspSession);
              MediaSubsession *subsession = NULL;

              while ((result == STATUS_OK) && ((subsession = iter.next()) != NULL))
              {
                char *tempSubSessionName = (char *)subsession->mediumName();
                char *tempSubSessionCodecName = (char *)subsession->codecName();
#ifdef _MBCS
                TCHAR *subSessionName = ConvertToMultiByteA(tempSubSessionName);
                TCHAR *subSessionCodecName = ConvertToMultiByteA(tempSubSessionCodecName);
#else
                TCHAR *subSessionName = ConvertToUnicodeA(tempSubSessionName);
                TCHAR *subSessionCodecName = ConvertToUnicodeA(tempSubSessionCodecName);
#endif
                if (!subsession->initiate())
                {
                  result = STATUS_ERROR;
                  TCHAR *message = FormatString(_T("%s: %s: unable to create receiver for subsession '%s', codec '%s'"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, subSessionName, subSessionCodecName);
                  this->LogRtspMessage(LOGGER_ERROR, message);
                  FREE_MEM(message);
                }
                else
                {
                  this->logger.Log(LOGGER_VERBOSE, _T("%s: %s: created receiver for subsession '%s', codec '%s'"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, subSessionName, subSessionCodecName);

                  // set session ID, doesn't matter what
                  subsession->sessionId = tempSubSessionName;

                  if (subsession->rtpSource() != NULL)
                  {
                    // because we're saving the incoming data, rather than playing
                    // it in real time, allow an especially large time threshold
                    // (1 second) for reordering misordered incoming packets:
                    unsigned const thresh = 1000000; // 1 second
                    subsession->rtpSource()->setPacketReorderingThresholdTime(thresh);

//.........这里部分代码省略.........

//.........这里部分代码省略.........
    = getSDPDescriptionFromURL(ourClient, url, username, password,
			       proxyServerName, proxyServerPortNum,
			       desiredPortNum);
  if (sdpDescription == NULL) {
    *env << "Failed to get a SDP description from URL \"" << url
		<< "\": " << env->getResultMsg() << "\n";
    shutdown();
  }

  *env << "Opened URL \"" << url
	  << "\", returning a SDP description:\n" << sdpDescription << "\n";

  // Create a media session object from this SDP description:
  session = MediaSession::createNew(*env, sdpDescription);
  delete[] sdpDescription;
  if (session == NULL) {
    *env << "Failed to create a MediaSession object from the SDP description: " << env->getResultMsg() << "\n";
    shutdown();
  } else if (!session->hasSubsessions()) {
    *env << "This session has no media subsessions (i.e., \"m=\" lines)\n";
    shutdown();
  }

  // Then, setup the "RTPSource"s for the session:
  MediaSubsessionIterator iter(*session);
  MediaSubsession *subsession;
  Boolean madeProgress = False;
  char const* singleMediumToTest = singleMedium;
  while ((subsession = iter.next()) != NULL) {
    // If we've asked to receive only a single medium, then check this now:
    if (singleMediumToTest != NULL) {
      if (strcmp(subsession->mediumName(), singleMediumToTest) != 0) {
		  *env << "Ignoring \"" << subsession->mediumName()
			  << "/" << subsession->codecName()
			  << "\" subsession, because we've asked to receive a single " << singleMedium
			  << " session only\n";
	continue;
      } else {
	// Receive this subsession only
	singleMediumToTest = "xxxxx";
	    // this hack ensures that we get only 1 subsession of this type
      }
    }

    if (desiredPortNum != 0) {
      subsession->setClientPortNum(desiredPortNum);
      desiredPortNum += 2;
    }

    if (createReceivers) {
      if (!subsession->initiate(simpleRTPoffsetArg)) {
	*env << "Unable to create receiver for \"" << subsession->mediumName()
	     << "/" << subsession->codecName()
	     << "\" subsession: " << env->getResultMsg() << "\n";
      } else {
	*env << "Created receiver for \"" << subsession->mediumName()
	     << "/" << subsession->codecName()
	     << "\" subsession (client ports " << subsession->clientPortNum()
	     << "-" << subsession->clientPortNum()+1 << ")\n";
	madeProgress = True;
	
	if (subsession->rtpSource() != NULL) {
	  // Because we're saving the incoming data, rather than playing
	  // it in real time, allow an especially large time threshold
	  // (1 second) for reordering misordered incoming packets:
	  unsigned const thresh = 1000000; // 1 second