C++ MediaSubsession::initiate方法代码示例

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;
}

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.........

//.........这里部分代码省略.........
		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()
						<< "-" << 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);

						if (socketInputBufferSize > 0) 
						{
							// Set the RTP source's input buffer size as specified:
							int socketNum
								= subsession->rtpSource()->RTPgs()->socketNum();
							unsigned curBufferSize
								= getReceiveBufferSize(*env, socketNum);
							unsigned newBufferSize
								= setReceiveBufferTo(*env, socketNum, socketInputBufferSize);
							*env << "Changed socket receive buffer size for the \""
								<< subsession->mediumName()

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.........

bool MtkRTSPClient::handDescription(char* resultString)
{
	CHECK_NULL_COND(resultString, false);

	
	char* sdpDescription = resultString;
	//LOG_DEBUG("SDP description:%s", sdpDescription);
	
	// Create a media session object from this SDP description:
	session = MediaSession::createNew(*env, sdpDescription);
	if (session == NULL) 
	{
			LOG_ERR("Failed to create a MediaSession object from the SDP description: %s", env->getResultMsg());
			return false;
	} 
	if (!session->hasSubsessions())
	{
			LOG_ERR("This session has no media subsessions (i.e., \"m=\" lines)");
			Medium::close(session);
			session = NULL;
			return false;
	}

	/*
	 *TO DO:GET THE TIME RANGE
	 */
	fStartTime = session->playStartTime();
	if (fStartTime < 0)
	{
		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
		 */
//.........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.........

bool CRTSPClient::OpenStream(char* url)
{
  XBMC->Log(LOG_DEBUG, "CRTSPClient::OpenStream()");
  m_session=NULL;
  
  strcpy(m_url,url);
  // Open the URL, to get a SDP description: 
  char* sdpDescription= getSDPDescriptionFromURL(m_ourClient, url, ""/*username*/, ""/*password*/,""/*proxyServerName*/, 0/*proxyServerPortNum*/,1234/*desiredPortNum*/);
  if (sdpDescription == NULL) 
  {
    XBMC->Log(LOG_DEBUG, "Failed to get a SDP description from URL %s %s",url ,m_env->getResultMsg() );
    shutdown();
    return false;
  }
  XBMC->Log(LOG_DEBUG, "Opened URL %s %s",url,sdpDescription);

  char* range=strstr(sdpDescription,"a=range:npt=");
  if (range!=NULL)
  {
    char *pStart = range+strlen("a=range:npt=");
    char *pEnd = strstr(range,"-") ;
    if (pEnd!=NULL)
    {
      pEnd++ ;
      double Start=atof(pStart) ;
      double End=atof(pEnd) ;

      XBMC->Log(LOG_DEBUG, "rangestart:%f rangeend:%f", Start,End);
      m_duration=(long) ((End-Start)*1000.0);
    }
  }
  // Create a media session object from this SDP description:
  m_session = MediaSession::createNew(*m_env, sdpDescription);
  delete[] sdpDescription;
  if (m_session == NULL) 
  {
    XBMC->Log(LOG_DEBUG, "Failed to create a MediaSession object from the SDP description:%s ",m_env->getResultMsg());
    shutdown();
    return false;
  } 
  else if (!m_session->hasSubsessions()) 
  {
    XBMC->Log(LOG_DEBUG, "This session has no media subsessions");
    shutdown();
    return false;
  }

  // Then, setup the "RTPSource"s for the session:
  MediaSubsessionIterator iter(*m_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) 
      {
        XBMC->Log(LOG_DEBUG, "Ignoring %s %s %s" , subsession->mediumName(),subsession->codecName(),singleMedium);
        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)) 
      {
        XBMC->Log(LOG_DEBUG, "Unable to create receiver for %s %s %s" ,subsession->mediumName(),subsession->codecName(),m_env->getResultMsg());
      } 
      else 
      {
        XBMC->Log(LOG_DEBUG, "Created receiver for type=%s codec=%s ports: %d %d " ,subsession->mediumName(),subsession->codecName(),subsession->clientPortNum(),subsession->clientPortNum()+1 );
        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:
          
          int socketNum= subsession->rtpSource()->RTPgs()->socketNum();
          XBMC->Log(LOG_DEBUG, "rtsp:increaseReceiveBufferTo to 2000000 for s:%d",socketNum);
          increaseReceiveBufferTo( *m_env, socketNum, 2000000 );

          unsigned const thresh = 1000000; // 1 second 
          subsession->rtpSource()->setPacketReorderingThresholdTime(thresh);

          if (socketInputBufferSize > 0) 
//.........这里部分代码省略.........

//.........这里部分代码省略.........
      + strlen(sessionName)
      + strlen(sessionInfo)
      + strlen(remoteRTSPServerAddressStr.val())
      + strlen(sessionName)
      + strlen(sessionInfo)
      + strlen(fApplicationName)
      + strlen(sessionAuthor)
      + strlen(sessionCopyright)
      + fSubstreamSDPSizes;
    unsigned const sdpSessionId = our_random32();
    unsigned const sdpVersion = sdpSessionId;
    sdp = new char[sdpLen];
    sprintf(sdp, sdpFmt,
	    sdpSessionId, sdpVersion, // o= line
	    sessionName, // s= line
	    sessionInfo, // i= line
	    remoteRTSPServerAddressStr.val(), // c= line
	    sessionName, // a=x-qt-text-nam: line
	    sessionInfo, // a=x-qt-text-inf: line
	    fApplicationName, // a=x-qt-text-cmt: line
	    sessionAuthor, // a=x-qt-text-aut: line
	    sessionCopyright // a=x-qt-text-cpy: line
	    );
    char* p = &sdp[strlen(sdp)];
    SubstreamDescriptor* ss;
    for (ss = fHeadSubstream; ss != NULL; ss = ss->next()) {
      sprintf(p, "%s", ss->sdpLines());
      p += strlen(p);
    }

    // Do a RTSP "ANNOUNCE" with this SDP description:
    Authenticator auth;
    Authenticator* authToUse = NULL;
    if (remoteUserName[0] != '\0' || remotePassword[0] != '\0') {
      auth.setUsernameAndPassword(remoteUserName, remotePassword);
      authToUse = &auth;
    }
    fWatchVariable = 0;
    (void)fRTSPClient->sendAnnounceCommand(sdp, genericResponseHandler, authToUse);

    // Now block (but handling events) until we get a response:
    envir().taskScheduler().doEventLoop(&fWatchVariable);

    delete[] fResultString;
    if (fResultCode != 0) break; // an error occurred with the RTSP "ANNOUNCE" command

    // Next, tell the remote server to start receiving the stream from us.
    // (To do this, we first create a "MediaSession" object from the SDP description.)
    fSession = MediaSession::createNew(envir(), sdp);
    if (fSession == NULL) break;

    ss = fHeadSubstream;
    MediaSubsessionIterator iter(*fSession);
    MediaSubsession* subsession;
    ss = fHeadSubstream;
    unsigned streamChannelId = 0;
    while ((subsession = iter.next()) != NULL) {
      if (!subsession->initiate()) break;

      fWatchVariable = 0;
      (void)fRTSPClient->sendSetupCommand(*subsession, genericResponseHandler,
					  True /*streamOutgoing*/,
					  True /*streamUsingTCP*/);
      // Now block (but handling events) until we get a response:
      envir().taskScheduler().doEventLoop(&fWatchVariable);

      delete[] fResultString;
      if (fResultCode != 0) break; // an error occurred with the RTSP "SETUP" command

      // Tell this subsession's RTPSink and RTCPInstance to use
      // the RTSP TCP connection:
      ss->rtpSink()->setStreamSocket(fRTSPClient->socketNum(), streamChannelId++);
      if (ss->rtcpInstance() != NULL) {
	ss->rtcpInstance()->setStreamSocket(fRTSPClient->socketNum(),
					    streamChannelId++);
      }
      ss = ss->next();
    }
    if (subsession != NULL) break; // an error occurred above

    // Tell the RTSP server to start:
    fWatchVariable = 0;
    (void)fRTSPClient->sendPlayCommand(*fSession, genericResponseHandler);

    // Now block (but handling events) until we get a response:
    envir().taskScheduler().doEventLoop(&fWatchVariable);

    delete[] fResultString;
    if (fResultCode != 0) break; // an error occurred with the RTSP "PLAY" command

    // Finally, make sure that the output TCP buffer is a reasonable size:
    increaseSendBufferTo(envir(), fRTSPClient->socketNum(), 100*1024);

    success = True;
  } while (0);

  delete[] sdp;
  delete[] url;
  return success;
}

//.........这里部分代码省略.........
      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;
          }
        }
      }
      while ((groupsock == NULL) && (port <= this->rtspRtpClientPortRangeEnd));
      // Did we find a free port? If not, we fall back to a random port chosen
      // by LIVE555.
      if (groupsock != NULL)
      {
        delete groupsock;
        groupsock = NULL;
        subsession->setClientPortNum(port);
      }
    }

    if (!subsession->initiate() || subsession->rtpSource() == NULL)
    {
      TCHAR *message = FormatString(_T("failed to create receiver for sub-session, 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;
    }

    this->logger.Log(LOGGER_VERBOSE, _T("%s: %s: created receiver for sub-session, medium = %s, codec = %s"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, subSessionName, subSessionCodecName);
    FREE_MEM(subSessionName);
    FREE_MEM(subSessionCodecName);

    // set session ID, doesn't matter what
    subsession->setSessionId(subsession->mediumName());

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

    // set the RTP source's OS socket buffer size as appropriate
    int socketNum = subsession->rtpSource()->RTPgs()->socketNum();
    unsigned int currentBufferSize = getReceiveBufferSize(*this->rtspEnvironment, socketNum);
    if (this->defaultBufferSize > currentBufferSize)
    {
      setReceiveBufferTo(*this->rtspEnvironment, socketNum, this->defaultBufferSize);
      unsigned setBufferSize = getReceiveBufferSize(*this->rtspEnvironment, socketNum);
      if (setBufferSize == this->defaultBufferSize)
      {
        this->logger.Log(LOGGER_VERBOSE, _T("%s: %s: set buffer size for sub-session, previous size = %i, requested size = %i, current size = %i"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, subSessionName, currentBufferSize, this->defaultBufferSize, setBufferSize);
      }

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;
}

//.........这里部分代码省略.........
    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 "

void continueAfterDESCRIBE(RTSPClient*, int resultCode, char* resultString) {
    ALOG(TX_LOG_INFO, TAG,"continueAfterDESCRIBE\n");
  if (resultCode != 0) {
      ALOG(TX_LOG_INFO, TAG,"Failed to get a SDP description for the URL %s, %s\n", streamURL, resultString);
    *env << "Failed to get a SDP description for the URL \"" << streamURL << "\": " << resultString << "\n";
    delete[] resultString;
    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) {
      ALOG(TX_LOG_INFO, TAG, "Failed to create a MediaSession object from the SDP description:");
    shutdown();
  } else if (!session->hasSubsessions()) {
      ALOG(TX_LOG_INFO, TAG, "This session has no media subsessions (i.e., no \"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) {
          ALOG(TX_LOG_INFO, TAG, "Ignoring codecName = %s\n", subsession->codecName());
		  *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)) {
          ALOG(TX_LOG_INFO, TAG, "Unable to create receiver for\n");
	*env << "Unable to create receiver for \"" << subsession->mediumName()
	     << "/" << subsession->codecName()
	     << "\" subsession: " << env->getResultMsg() << "\n";
      } else {
      ALOG(TX_LOG_INFO, TAG, "Created receiver\n");
	*env << "Created receiver for \"" << subsession->mediumName()
	     << "/" << subsession->codecName() << "\" subsession (";
	if (subsession->rtcpIsMuxed()) {
        ALOG(TX_LOG_INFO, TAG, "subsession->rtcpIsMuxed()  client port = %d\n", subsession->clientPortNum());
	  *env << "client port " << subsession->clientPortNum();
	} else {
        ALOG(TX_LOG_INFO, TAG, "subsession->rtcpIsMuxed(),,,,else");
	  *env << "client ports " << subsession->clientPortNum()
	       << "-" << subsession->clientPortNum()+1;
	}
	*env << ")\n";
	madeProgress = True;

	if (subsession->rtpSource() != NULL) {
	  unsigned const thresh = 1000000; // 1 second
	  subsession->rtpSource()->setPacketReorderingThresholdTime(thresh);

	  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) {
            ALOG(TX_LOG_INFO, TAG, "socketInputBufferSize > 0  Changed socket receive buffer size for the\n");
	      *env << "Changed socket receive buffer size for the \""
		   << subsession->mediumName()
		   << "/" << subsession->codecName()
		   << "\" subsession from "
		   << curBufferSize << " to "
		   << newBufferSize << " bytes\n";
	    }
	  }
	}
      }
    } else {
          ALOG(TX_LOG_INFO, TAG, "socketInputBufferSize > 0=====else\n");
      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 {
//.........这里部分代码省略.........