C++ MediaSubsession::setClientPortNum方法代码示例

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

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.........

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 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.........
    {
      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;
          }
        }
      }
      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);

//.........这里部分代码省略.........
  // 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:

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.........