C++ MediaSubsession::readSource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;
}

AVIFileSink::~AVIFileSink() {
  completeOutputFile();

  // Then, stop streaming and delete each active "AVISubsessionIOState":
  MediaSubsessionIterator iter(fInputSession);
  MediaSubsession* subsession;
  while ((subsession = iter.next()) != NULL) {
    if (subsession->readSource() != NULL) subsession->readSource()->stopGettingFrames();

    AVISubsessionIOState* ioState
      = (AVISubsessionIOState*)(subsession->miscPtr);
    if (ioState == NULL) continue;

    delete ioState;
  }

  // Then, delete the index records:
  AVIIndexRecord* cur = fIndexRecordsHead;
  while (cur != NULL) {
    AVIIndexRecord* next = cur->next();
    delete cur;
    cur = next;
  }

  // Finally, close our output file:
  CloseOutputFile(fOutFid);
}

void beginQOSMeasurement() {
	// Set up a measurement record for each active subsession:
	struct timeval startTime;
	gettimeofday(&startTime, NULL);
	nextQOSMeasurementUSecs = startTime.tv_sec*1000000 + startTime.tv_usec;
	qosMeasurementRecord* qosRecordTail = NULL;
	MediaSubsessionIterator iter(*session);
	MediaSubsession* subsession;
	while ((subsession = iter.next()) != NULL) {
		RTPSource* src = subsession->rtpSource();
#ifdef SUPPORT_REAL_RTSP
		if (session->isRealNetworksRDT) src = (RTPSource*)(subsession->readSource()); // hack
#endif
		if (src == NULL) continue;

		qosMeasurementRecord* qosRecord
			= new qosMeasurementRecord(startTime, src);
		if (qosRecordHead == NULL) qosRecordHead = qosRecord;
		if (qosRecordTail != NULL) qosRecordTail->fNext = qosRecord;
		qosRecordTail  = qosRecord;
	}

	// Then schedule the first of the periodic measurements:
	scheduleNextQOSMeasurement();
}

Boolean AVIFileSink::continuePlaying() {
  // Run through each of our input session's 'subsessions',
  // asking for a frame from each one:
  Boolean haveActiveSubsessions = False;
  MediaSubsessionIterator iter(fInputSession);
  MediaSubsession* subsession;
  while ((subsession = iter.next()) != NULL) {
    FramedSource* subsessionSource = subsession->readSource();
    if (subsessionSource == NULL) continue;

    if (subsessionSource->isCurrentlyAwaitingData()) continue;

    AVISubsessionIOState* ioState
      = (AVISubsessionIOState*)(subsession->miscPtr);
    if (ioState == NULL) continue;

    haveActiveSubsessions = True;
    unsigned char* toPtr = ioState->fBuffer->dataEnd();
    unsigned toSize = ioState->fBuffer->bytesAvailable();
    subsessionSource->getNextFrame(toPtr, toSize,
				   afterGettingFrame, ioState,
				   onSourceClosure, ioState);
  }
  if (!haveActiveSubsessions) {
    envir().setResultMsg("No subsessions are currently active");
    return False;
  }

  return True;
}

AVISubsessionIOState::AVISubsessionIOState(AVIFileSink& sink,
				     MediaSubsession& subsession)
  : fOurSink(sink), fOurSubsession(subsession),
    fMaxBytesPerSecond(0), fIsVideo(False), fIsAudio(False), fIsByteSwappedAudio(False), fNumFrames(0) {
  fBuffer = new SubsessionBuffer(fOurSink.fBufferSize);
  fPrevBuffer = sink.fPacketLossCompensate
    ? new SubsessionBuffer(fOurSink.fBufferSize) : NULL;

  FramedSource* subsessionSource = subsession.readSource();
  fOurSourceIsActive = subsessionSource != NULL;

  fPrevPresentationTime.tv_sec = 0;
  fPrevPresentationTime.tv_usec = 0;
}

//.........这里部分代码省略.........
					madeProgress = True;
				}
			}
		}
		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,

void printQOSData(int exitCode) {
	if (exitCode != 0 && statusCode == 0) statusCode = 2;
	*env << "begin_QOS_statistics\n";
	*env << "server_availability\t" << (statusCode == 1 ? 0 : 100) << "\n";
	*env << "stream_availability\t" << (statusCode == 0 ? 100 : 0) << "\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();
#ifdef SUPPORT_REAL_RTSP
			if (session->isRealNetworksRDT) src = (RTPSource*)(subsession->readSource()); // hack
#endif
			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" << 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";

#ifdef SUPPORT_REAL_RTSP
				if (session->isRealNetworksRDT) {
					RealRDTSource* rdt = (RealRDTSource*)src;
					*env << "inter_packet_gap_ms_min\t" << rdt->minInterPacketGapUS()/1000.0 << "\n";
					struct timeval totalGaps = rdt->totalInterPacketGaps();
					double totalGapsMS = totalGaps.tv_sec*1000.0 + totalGaps.tv_usec/1000.0;
					unsigned totNumPacketsReceived = rdt->totNumPacketsReceived();
					*env << "inter_packet_gap_ms_ave\t"
						<< (totNumPacketsReceived == 0 ? 0.0 : totalGapsMS/totNumPacketsReceived) << "\n";
					*env << "inter_packet_gap_ms_max\t" << rdt->maxInterPacketGapUS()/1000.0 << "\n";
				} else {
#endif
					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";
					}
#ifdef SUPPORT_REAL_RTSP
				}
#endif

				curQOSRecord = curQOSRecord->fNext;
			}
		}
	}

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

//.........这里部分代码省略.........
    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());

	// Set the OS's socket receive buffer sufficiently large to avoid
	// incoming packets getting dropped between successive reads from this
	// subsession's demuxer.  Depending on the bitrate(s) that you expect,
	// you may wish to tweak the "desiredReceiveBufferSize" values above.
	int rtpSocketNum = subsession->rtpSource()->RTPgs()->socketNum();
	int receiveBufferSize
	  = increaseReceiveBufferTo(*env, rtpSocketNum,
				    desiredReceiveBufferSize);
	if (verbose > 0) {
	  fprintf(stderr, "Increased %s socket receive buffer to %d bytes \n",
		  subsession->mediumName(), receiveBufferSize);
	}

	if (rtspClient != NULL) {
	  // Issue a RTSP "SETUP" command on the chosen subsession:
	  if (!rtspClient->setupMediaSubsession(*subsession, False,
						rtsp_transport_tcp)) break;
	  if (!strcmp(subsession->mediumName(), "audio"))
	    audiofound = 1;
	  if (!strcmp(subsession->mediumName(), "video"))
            videofound = 1;
	}
      }
    }

    if (rtspClient != NULL) {
      // Issue a RTSP aggregate "PLAY" command on the whole session:
      if (!rtspClient->playMediaSession(*mediaSession)) break;
    } else if (sipClient != NULL) {
      sipClient->sendACK(); // to start the stream flowing
    }

    // Now that the session is ready to be read, do additional
    // MPlayer codec-specific initialization on each subsession:
    iter.reset();
    while ((subsession = iter.next()) != NULL) {
      if (subsession->readSource() == NULL) continue; // not reading this

      unsigned flags = 0;
      if (strcmp(subsession->mediumName(), "audio") == 0) {
	rtpState->audioBufferQueue
	  = new ReadBufferQueue(subsession, demuxer, "audio");
	rtpState->audioBufferQueue->otherQueue = &(rtpState->videoBufferQueue);
	rtpCodecInitialize_audio(demuxer, subsession, flags);
      } else if (strcmp(subsession->mediumName(), "video") == 0) {
	rtpState->videoBufferQueue
	  = new ReadBufferQueue(subsession, demuxer, "video");
	rtpState->videoBufferQueue->otherQueue = &(rtpState->audioBufferQueue);
	rtpCodecInitialize_video(demuxer, subsession, flags);
      }
      rtpState->flags |= flags;
    }
    success = True;
  } while (0);
  if (!success) return NULL; // an error occurred

  // Hack: If audio and video are demuxed together on a single RTP stream,
  // then create a new "demuxer_t" structure to allow the higher-level
  // code to recognize this:
  if (demux_is_multiplexed_rtp_stream(demuxer)) {
    stream_t* s = new_ds_stream(demuxer->video);
    demuxer_t* od = demux_open(opts, s, DEMUXER_TYPE_UNKNOWN,
			       opts->audio_id, opts->video_id, opts->sub_id,
                               NULL);
    demuxer = new_demuxers_demuxer(od, od, od);
  }

  return demuxer;
}

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.........

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.........

//.........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;
}

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.........
      // Create a "QuickTimeFileSink", to write to 'stdout':
      qtOut = QuickTimeFileSink::createNew(*env, *session, "stdout",
					   fileSinkBufferSize,
					   movieWidth, movieHeight,
					   movieFPS,
					   packetLossCompensate,
					   syncStreams,
					   generateHintTracks,
					   generateMP4Format);
      if (qtOut == NULL) {
	*env << "Failed to create QuickTime file sink for stdout: " << env->getResultMsg();
	shutdown();
      }

      qtOut->startPlaying(sessionAfterPlaying, NULL);
    } else if (outputAVIFile) {
      // Create an "AVIFileSink", to write to 'stdout':
      aviOut = AVIFileSink::createNew(*env, *session, "stdout",
				      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) {