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

void shutdownStream(RTSPClient* rtspClient, int exitCode) {
	(void)exitCode;

//  UsageEnvironment& env = rtspClient->envir(); // alias
  StreamClientState& scs = ((ourRTSPClient*)rtspClient)->scs; // alias

	int channel = ((ourRTSPClient*)rtspClient)->_channel;
	while(pthread_mutex_lock(&g_context[channel].mutex) != 0 && errno == EINTR){};

  // First, check whether any subsessions have still to be closed:
  if (scs.session != NULL) { 
    Boolean someSubsessionsWereActive = False;
    MediaSubsessionIterator iter(*scs.session);
    MediaSubsession* subsession;

    while ((subsession = iter.next()) != NULL) {
      if (subsession->sink != NULL) {
	Medium::close(subsession->sink);
	subsession->sink = NULL;

	if (subsession->rtcpInstance() != NULL) {
	  subsession->rtcpInstance()->setByeHandler(NULL, NULL); // in case the server sends a RTCP "BYE" while handling "TEARDOWN"
	}

	someSubsessionsWereActive = True;
      }
    }

    if (someSubsessionsWereActive) {
      // Send a RTSP "TEARDOWN" command, to tell the server to shutdown the stream.
      // Don't bother handling the response to the "TEARDOWN".
      rtspClient->sendTeardownCommand(*scs.session, NULL);
    }
  }

//  env << *rtspClient << "Closing the stream.\n";
  Medium::close(rtspClient);
    // Note that this will also cause this stream's "StreamClientState" structure to get reclaimed.

  //if (--rtspClientCount == 0) 
  {
    // The final stream has ended, so exit the application now.
    // (Of course, if you're embedding this code into your own application, you might want to comment this out,
    // and replace it with "eventLoopWatchVariable = 1;", so that we leave the LIVE555 event loop, and continue running "main()".)
   	// exit(exitCode);
	g_context[channel].shutdown		= true;
	g_context[channel].eventLoopVar	= 1;
  }

  while(pthread_mutex_unlock(&g_context[channel].mutex) != 0 && errno == EINTR){};
}

AVIFileSink::AVIFileSink(UsageEnvironment& env,
			 MediaSession& inputSession,
			 char const* outputFileName,
			 unsigned bufferSize,
			 unsigned short movieWidth, unsigned short movieHeight,
			 unsigned movieFPS, Boolean packetLossCompensate)
  : Medium(env), fInputSession(inputSession),
    fIndexRecordsHead(NULL), fIndexRecordsTail(NULL), fNumIndexRecords(0),
    fBufferSize(bufferSize), fPacketLossCompensate(packetLossCompensate),
    fAreCurrentlyBeingPlayed(False), fNumSubsessions(0), fNumBytesWritten(0),
    fHaveCompletedOutputFile(False),
    fMovieWidth(movieWidth), fMovieHeight(movieHeight), fMovieFPS(movieFPS) {
  fOutFid = OpenOutputFile(env, outputFileName);
  if (fOutFid == NULL) return;

  // Set up I/O state for each input subsession:
  MediaSubsessionIterator iter(fInputSession);
  MediaSubsession* subsession;
  while ((subsession = iter.next()) != NULL) {
    // Ignore subsessions without a data source:
    FramedSource* subsessionSource = subsession->readSource();
    if (subsessionSource == NULL) continue;

    // If "subsession's" SDP description specified screen dimension
    // or frame rate parameters, then use these.
    if (subsession->videoWidth() != 0) {
      fMovieWidth = subsession->videoWidth();
    }
    if (subsession->videoHeight() != 0) {
      fMovieHeight = subsession->videoHeight();
    }
    if (subsession->videoFPS() != 0) {
      fMovieFPS = subsession->videoFPS();
    }

    AVISubsessionIOState* ioState
      = new AVISubsessionIOState(*this, *subsession);
    subsession->miscPtr = (void*)ioState;

    // Also set a 'BYE' handler for this subsession's RTCP instance:
    if (subsession->rtcpInstance() != NULL) {
      subsession->rtcpInstance()->setByeHandler(onRTCPBye, ioState);
    }

    ++fNumSubsessions;
  }

  // Begin by writing an AVI header:
  addFileHeader_AVI();
}

void CAimer39RTSPClient::shutdownStream( RTSPClient* rtspClient ) 
{
	UsageEnvironment& env = rtspClient->envir(); // alias
	StreamClientState& scs = ((ourRTSPClient*)rtspClient)->scs; // alias

	CAimer39RTSPClient * arc = findClient( (ourRTSPClient*)rtspClient );

	if ( NULL == arc ) {
		env << "some how the system in to a dangerous situation!" << "\n";
		return;
	}

	// First, check whether any subsessions have still to be closed:
	if (scs.session != NULL) { 
		Boolean someSubsessionsWereActive = False;
		MediaSubsessionIterator iter(*scs.session);
		MediaSubsession* subsession;

		while ((subsession = iter.next()) != NULL) {
			if (subsession->sink != NULL) {
				Medium::close(subsession->sink);
				subsession->sink = NULL;

				if (subsession->rtcpInstance() != NULL) {
					subsession->rtcpInstance()->setByeHandler(NULL, NULL); // in case the server sends a RTCP "BYE" while handling "TEARDOWN"
				}

				someSubsessionsWereActive = True;
			}
		}

		if (someSubsessionsWereActive) {
			// Send a RTSP "TEARDOWN" command, to tell the server to shutdown the stream.
			// Don't bother handling the response to the "TEARDOWN".
			rtspClient->sendTeardownCommand(*scs.session, NULL);
		}
	}

	env << *rtspClient << "Closing the stream.\n";
	arc->m_bIsShutDown = true;
	if (arc->m_pFinishCallback)	arc->m_pFinishCallback(arc->m_pCallBackParam, arc); //call back inform stream over
	Medium::close(rtspClient);
	// Note that this will also cause this stream's "StreamClientState" structure to get reclaimed.
}

// Used to shut down and close a stream (including its "RTSPClient" object):
void StreamClient::shutdownStream()
{
    // First, check whether any subsessions have still to be closed:
    if (_state.session != nullptr) {
        Boolean someSubsessionsWereActive = False;
        MediaSubsessionIterator iter(*_state.session);
        MediaSubsession * subsession = nullptr;

        while ((subsession = iter.next()) != nullptr) {
            if (subsession->sink != nullptr) {
                Medium::close(subsession->sink);
                subsession->sink = nullptr;

                if (subsession->rtcpInstance() != nullptr) {
                    subsession->rtcpInstance()->setByeHandler(nullptr, nullptr); // in case the server sends a RTCP "BYE" while handling "TEARDOWN"
                }
                someSubsessionsWereActive = True;
            }
        }

        if (someSubsessionsWereActive) {
            sendTeardown(*_state.session);
        }
    }

    crLogN("StreamClient::shutdownStream() Closing the stream.");
    Medium::close(this);
    // Note that this will also cause this stream's "StreamClientState" structure to get reclaimed.

//    if (--rtspClientCount == 0) {
//        // The final stream has ended, so exit the application now.
//        // (Of course, if you're embedding this code into your own application, you might want to comment this out,
//        // and replace it with "eventLoopWatchVariable = 1;", so that we leave the LIVE555 event loop, and continue running "main()".)
//        exit(exitCode);
//    }
    exit(0);
}

//.........这里部分代码省略.........
				}
			}
		}
		if (!madeProgress) 
			break;

		// Perform additional 'setup' on each subsession, before playing them:
		pThis->SetRtspStatus( RTSPStatus_Setup );
		unsigned nResponseCode = NULL;
		BOOL bSetupSuccess = setupStreams( &nResponseCode );
		if ( !bSetupSuccess )
		{
			// setup失败！
			if ( RTSPResp_Error_Server_Full == nResponseCode )
			{
				pThis->SetRtspStatus( RTSPStatus_Error_Server_Full );
			}
			else
			{
				pThis->SetRtspStatus( RTSPStatus_Idle );
			}
			break;
		}
		// Create output files:
		
		if ( true  ) 
		{
				// Create and start "FileSink"s for each subsession: 
				madeProgress = False;
				iter.reset();
				while ((subsession = iter.next()) != NULL) 
				{
					if (subsession->readSource() == NULL) continue; // was not initiated

					MediaSink *pDecodeSink = 0;
					if (strcmp(subsession->mediumName(), "video") == 0 )
					{
						int nBandWidth = subsession->GetBandWidth();

						if ( strcmp(subsession->codecName(), "MP4V-ES") == 0 )
						{
							CMpeg4StreamDecodeSink *pMsds = CMpeg4StreamDecodeSink::CreateNew( *env, 20000, nBandWidth );
							 pDecodeSink = pMsds;
							
						}
						else if ( strcmp( subsession->codecName(), "H264" ) == 0 )
						{
							 CH264StreamDecodeSink *pHsds = CH264StreamDecodeSink::CreateNew( *env, 20000, nBandWidth );
							 pDecodeSink = pHsds;
						}
						else
						{
							continue;
						}
					}				

					subsession->sink = pDecodeSink;
					if (subsession->sink == NULL) 
					{
						*env << "Failed to create CH264StreamDecodeSink \""  << "\n";
					} 


					subsession->sink->startPlaying(*(subsession->readSource()),
						subsessionAfterPlaying,
						subsession);

					// Also set a handler to be called if a RTCP "BYE" arrives
					// for this subsession:
					if (subsession->rtcpInstance() != NULL) 
					{
						subsession->rtcpInstance()->setByeHandler(subsessionByeHandler,
							subsession);
					}

					// 发送NAT探测包。
					unsigned char temp[112] = {0};
					temp[0] = 0x80;
					subsession->rtpSource()->RTPgs()->output( *env, 0,temp, 112 );

					madeProgress = True;
				}
			}


		// Finally, start playing each subsession, to start the data flow:
		pThis->SetRtspStatus( RTSPStatus_Play );
		startPlayingStreams();


		pThis->SetRtspStatus( RTSPStatus_Running );
		// 传入结束标志指针。 
		env->taskScheduler().doEventLoop( &pThis->m_runFlag ); 

		pThis->SetRtspStatus( RTSPStatus_Idle );

	} while(0);	

	return 0;
}

bool MtkRTSPClient::handSetup(char* resultString)
{
	CHECK_NULL_COND(session, false); 
	CHECK_NULL_COND(rtsp::env, false);

	bool bSuccess = false;
	
	// Then, setup the "RTPSource"s for the session:
	MediaSubsessionIterator iter(*(session));
	MediaSubsession *subsession = NULL;
	while ((subsession = iter.next()) != NULL) 
	{					
		if (subsession->readSource() == NULL) 
		{
			LOG_ERR("warning");
			continue; // was not initiated
		}

		if (subsession->sink != NULL)/*already be set*/
		{
			continue;
		}

		unsigned int type = getBufType(subsession);
		if (type == 0)
		{
			LOG_ERR("error type=%d", type);
			continue;
		}
		
		{
			iSetupCount--;
			/*set mediay info*/
			setMediaInfo(subsession, type);
		}

		CmpbSink *sink = NULL;
		if ((type != mediatype_audio) && (strcmp(subsession->codecName(), "H264") == 0))
		{
			sink = CmpbH264Sink::createNew(*env, *subsession, type, fileSinkBufferSize);
		}
        else if ((type == mediatype_audio) && 
                    ((stMediaInfo.audioCodec == MEDIACODEC_AC3) || 
                     (stMediaInfo.audioCodec == MEDIACODEC_EAC3) ||
                     (stMediaInfo.audioCodec == MEDIACODEC_MPEG4_GENERIC)))
		{
			sink = CmpbAACSink::createNew(*env, *subsession, type, fileSinkBufferSize);
		}
        else if ((type == mediatype_audio) && (stMediaInfo.audioCodec == MEDIACODEC_MP4A_LATM))
		{
			sink = CmpbLATMSink::createNew(*env, *subsession, type, fileSinkBufferSize);
		}
		else
		{
			sink = CmpbSink::createNew(*env, *subsession, type, fileSinkBufferSize);
		}
		subsession->sink = sink;
		if (subsession->sink == NULL) 
		{
			LOG_ERR("error!"); 
		} 
		else 
		{		
#if 0 /*this should be remove to cmpb sink*/           
			if ((type != mediatype_audio) && (strcmp(subsession->codecName(), "MP4V-ES") == 0)
				&& (subsession->fmtp_config() != NULL)) 
			{
			    // For MPEG-4 video RTP streams, the 'config' information
			    // from the SDP description contains useful VOL etc. headers.
			    // Insert this data at the front of the output file:
			    unsigned configLen;
			    unsigned char* configData
			      = parseGeneralConfigStr(subsession->fmtp_config(), configLen);
			    struct timeval timeNow;
			    gettimeofday(&timeNow, NULL);
			    sink->sendData(configData, configLen, timeNow);
			    delete[] configData;
		  	}
#endif			
			subsession->sink->startPlaying(*(subsession->readSource()),
												subsessionAfterPlaying,
													subsession);
			// Also set a handler to be called if a RTCP "BYE" arrives
			// for this subsession:
			if (subsession->rtcpInstance() != NULL) 
			{
				subsession->rtcpInstance()->setByeHandler(subsessionAfterPlaying, subsession);
			}

			bSuccess = true;
		}

		break;

	}

	if (iSetupCount == 0)
	{
		mediaInfoReady(); 
	}
//.........这里部分代码省略.........

void shutdownStream(RTSPClient* rtspClient, int exitCode) {
	OUTPUT_DEBUG_STRING("%s \n", __FUNCTION__);
	
	OUTPUT_DEBUG_STRING("open times = %d close times = %d\n", open_times, ++close_times);


	UsageEnvironment& env = rtspClient->envir(); // alias
	StreamClientState& scs = ((ourRTSPClient*)rtspClient)->scs; // alias

	// First, check whether any subsessions have still to be closed:
	if (scs.session != NULL) { 
		Boolean someSubsessionsWereActive = False;
		MediaSubsessionIterator iter(*scs.session);
		MediaSubsession* subsession;

		while ((subsession = iter.next()) != NULL) {
			if (subsession->sink != NULL) {
				//subsession->sink->stopPlaying();
				Medium::close(subsession->sink);
				subsession->sink = NULL;

				OUTPUT_DEBUG_STRING("22 add = %d, del = %d\n", sum_add, ++sum_del);
				
				if (subsession->rtcpInstance() != NULL) {
					subsession->rtcpInstance()->setByeHandler(NULL, NULL); // in case the server sends a RTCP "BYE" while handling "TEARDOWN"
				}
				someSubsessionsWereActive = True;
			}
		}

		if (someSubsessionsWereActive) {
			// Send a RTSP "TEARDOWN" command, to tell the server to shutdown the stream.
			// Don't bother handling the response to the "TEARDOWN".
			//gEnv->taskScheduler().unscheduleDelayedTask(scs.streamTimerTask);
			rtspClient->sendTeardownCommand(*scs.session, NULL);
			
		}
		
		//Medium::close(scs.session);
		//gEnv->taskScheduler().unscheduleDelayedTask(scs.streamTimerTask);
		Medium::close(rtspClient);
		rtspClient = NULL;
		
		//gEnv->reclaim(); //del by huguohu
		//gEnv = NULL;
		if(gScheduler != NULL){
			delete gScheduler; 
			gScheduler = NULL;	
		}

	}

	

	

	


	//env << *rtspClient << "Closing the stream.\n";
	//Medium::close(scs.session);
	

	// Note that this will also cause this stream's "StreamClientState" structure to get reclaimed.


	// 
	// 	if (--rtspClientCount == 0) {
	// 		// The final stream has ended, so exit the application now.
	// 		// (Of course, if you're embedding this code into your own application, you might want to comment this out,
	// 		// and replace it with "eventLoopWatchVariable = 1;", so that we leave the LIVE555 event loop, and continue running "main()".)
	// 		//exit(exitCode);
	// 	}

	
}

//.........这里部分代码省略.........
    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) 
          {
            // Set the RTP source's input buffer size as specified:
            int socketNum= subsession->rtpSource()->RTPgs()->socketNum();
            unsigned curBufferSize= getReceiveBufferSize(*m_env, socketNum);
            unsigned newBufferSize= setReceiveBufferTo(*m_env, socketNum, socketInputBufferSize);
            XBMC->Log(LOG_DEBUG,  "Changed socket receive buffer size for the %s %s %d %d",
            subsession->mediumName(),subsession->codecName(),curBufferSize,newBufferSize);
          }
        }
      }
    } 
    else 
    {
      if (subsession->clientPortNum() == 0) 
      {
        XBMC->Log(LOG_DEBUG, "No client port was specified for the %s %s",subsession->mediumName(),subsession->codecName());
      } 
      else 
      {
        madeProgress = True;
      }
    }
  }
  if (!madeProgress) 
  {
    shutdown();
    return false;
  }
  
  // Perform additional 'setup' on each subsession, before playing them:
  if (!setupStreams())
  {
    return false;
  }

  // Create output files:
  // Create and start "FileSink"s for each subsession:
  madeProgress = False;
  iter.reset();
  while ((subsession = iter.next()) != NULL) 
  {
    if (subsession->readSource() == NULL) continue; // was not initiated
    
    // Mediaportal:
    CMemorySink* fileSink= CMemorySink::createNew(*m_env, *m_buffer, fileSinkBufferSize);
    // XBMC test via file:
    //FileSink* fileSink = FileSink::createNew(*m_env, m_outFileName, fileSinkBufferSize, false); //oneFilePerFrame

    subsession->sink = fileSink;
    if (subsession->sink == NULL) 
    {
      XBMC->Log(LOG_DEBUG, "Failed to create FileSink %s",m_env->getResultMsg());
      shutdown();
      return false;
    } 
    XBMC->Log(LOG_DEBUG, "Created output sink: %s", m_outFileName);
    subsession->sink->startPlaying(*(subsession->readSource()),my_subsessionAfterPlaying,subsession);
    
    // Also set a handler to be called if a RTCP "BYE" arrives
    // for this subsession:
    if (subsession->rtcpInstance() != NULL) 
    {
      subsession->rtcpInstance()->setByeHandler(my_subsessionByeHandler,subsession);
    }
    madeProgress = True;
  }

  return true;
}

//.........这里部分代码省略.........
                      if (result != STATUS_OK)
                      {
                        // error occured
                        TCHAR *message = FormatString(METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("error while setup subsession"));
                        this->LogRtspMessage(LOGGER_ERROR, message);
                        FREE_MEM(message);
                      }
                      else
                      {
                        this->logger.Log(LOGGER_WARNING, METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("subsession audio codec not supported"));
                      }
                    }
                    else if (_tcsncicmp(subSessionName, _T("video"), 5) == 0)
                    {
                      // video
                      this->logger.Log(LOGGER_VERBOSE, _T("%s: %s: video subsession '%s'"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, subSessionName);
                      result |= (!rtspClient->setupMediaSubsession(*subsession));

                      if (result != STATUS_OK)
                      {
                        // error occured
                        TCHAR *message = FormatString(METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("error while setup subsession"));
                        this->LogRtspMessage(LOGGER_ERROR, message);
                        FREE_MEM(message);
                      }
                      else
                      {
                        if (_tcsncicmp(subSessionCodecName, _T("MP2T"), 4) == 0)
                        {
                          // MPEG2 Transport Stream
                          // set new RTSP source
                          this->rtspSource = subsession->rtpSource();

                          if (subsession->rtcpInstance() != NULL)
                          {
                            this->logger.Log(LOGGER_VERBOSE, METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("set subsession 'Bye' handler"));
                            subsession->rtcpInstance()->setByeHandler(SubsessionByeHandler, this);
                          }
                        }
                        else if (_tcsncicmp(subSessionCodecName, _T("H264"), 4) == 0)
                        {
                          // H264 codec, HD TV
                          this->logger.Log(LOGGER_ERROR, METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("H264 not supported"));
                          result = STATUS_ERROR;
                        }
                        else
                        {
                          // SD TV
                          this->logger.Log(LOGGER_ERROR, METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("other subsession video codec than MP2T not supported"));
                          result = STATUS_ERROR;
                        }
                      }
                    }
                    else
                    {
                      this->logger.Log(LOGGER_WARNING, _T("%s: %s: unknown subsession '%s', ignored"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, subSessionName);
                    }
                  }
                  else
                  {
                    this->logger.Log(LOGGER_WARNING, _T("%s: %s: subsession '%s' doesn't have RTP source"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, subSessionName);
                  }
                }

                // free subsession name and codec name
                FREE_MEM(subSessionName);

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.........
				      fileSinkBufferSize,
				      movieWidth, movieHeight,
				      movieFPS,
				      packetLossCompensate);
      if (aviOut == NULL) {
	*env << "Failed to create AVI file sink for stdout: " << env->getResultMsg();
	shutdown();
      }

      aviOut->startPlaying(sessionAfterPlaying, NULL);
    } else {
      // Create and start "FileSink"s for each subsession:
      madeProgress = False;
      iter.reset();
      while ((subsession = iter.next()) != NULL) {
	if (subsession->readSource() == NULL) continue; // was not initiated

	// Create an output file for each desired stream:
	char outFileName[1000];
	if (singleMedium == NULL) {
	  // Output file name is
	  //     "<filename-prefix><medium_name>-<codec_name>-<counter>"
	  static unsigned streamCounter = 0;
	  snprintf(outFileName, sizeof outFileName, "%s%s-%s-%d",
		   fileNamePrefix, subsession->mediumName(),
		   subsession->codecName(), ++streamCounter);
	} else {
	  sprintf(outFileName, "stdout");
	}
	FileSink* fileSink;
	if (strcmp(subsession->mediumName(), "audio") == 0 &&
	    (strcmp(subsession->codecName(), "AMR") == 0 ||
	     strcmp(subsession->codecName(), "AMR-WB") == 0)) {
	  // For AMR audio streams, we use a special sink that inserts AMR frame hdrs:
	  fileSink = AMRAudioFileSink::createNew(*env, outFileName,
						 fileSinkBufferSize, oneFilePerFrame);
	} else if (strcmp(subsession->mediumName(), "video") == 0 &&
	    (strcmp(subsession->codecName(), "H264") == 0)) {
	  // For H.264 video stream, we use a special sink that insert start_codes:
	  fileSink = H264VideoFileSink::createNew(*env, outFileName,
						 fileSinkBufferSize, oneFilePerFrame);
	} else {
	  // Normal case:
	  fileSink = FileSink::createNew(*env, outFileName,
					 fileSinkBufferSize, oneFilePerFrame);
	}
	subsession->sink = fileSink;
	if (subsession->sink == NULL) {
	  *env << "Failed to create FileSink for \"" << outFileName
		  << "\": " << env->getResultMsg() << "\n";
	} else {
	  if (singleMedium == NULL) {
	    *env << "Created output file: \"" << outFileName << "\"\n";
	  } else {
	    *env << "Outputting data from the \"" << subsession->mediumName()
			<< "/" << subsession->codecName()
			<< "\" subsession to 'stdout'\n";
	  }

	  if (strcmp(subsession->mediumName(), "video") == 0 &&
	      strcmp(subsession->codecName(), "MP4V-ES") == 0 &&
	      subsession->fmtp_config() != NULL) {
	    // For MPEG-4 video RTP streams, the 'config' information
	    // from the SDP description contains useful VOL etc. headers.
	    // Insert this data at the front of the output file:
	    unsigned configLen;
	    unsigned char* configData
	      = parseGeneralConfigStr(subsession->fmtp_config(), configLen);
	    struct timeval timeNow;
	    gettimeofday(&timeNow, NULL);
	    fileSink->addData(configData, configLen, timeNow);
	    delete[] configData;
	  }

	  subsession->sink->startPlaying(*(subsession->readSource()),
					 subsessionAfterPlaying,
					 subsession);

	  // Also set a handler to be called if a RTCP "BYE" arrives
	  // for this subsession:
	  if (subsession->rtcpInstance() != NULL) {
	    subsession->rtcpInstance()->setByeHandler(subsessionByeHandler,
						      subsession);
	  }

	  madeProgress = True;
	}
      }
      if (!madeProgress) shutdown();
    }
  }

  // Finally, start playing each subsession, to start the data flow:

  startPlayingStreams();

  env->taskScheduler().doEventLoop(); // does not return

  return 0; // only to prevent compiler warning
}