Пример #1
0
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;
}
Пример #2
0
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);
}
Пример #3
0
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();
}
Пример #4
0
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;
}
Пример #5
0
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;
}
Пример #6
0
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();
}
Пример #7
0
int CMediaNet::MediaNet_Thread( void * pThisVoid )
{
	CMediaNet *pThis = ( CMediaNet* )pThisVoid;

	do 
	{
		// 开始初始化.
		pThis->SetRtspStatus( RTSPStatus_Init );

		// Begin by setting up our usage environment:
		TaskScheduler* scheduler = BasicTaskScheduler::createNew();
		env = BasicUsageEnvironment::createNew(*scheduler);

		progName = "M_CU";

		string strUrl = pThis->m_strRTSPUrlA;

		gettimeofday(&startTime, NULL);

		unsigned short desiredPortNum = 0;

		// unfortunately we can't use getopt() here, as Windoze doesn't have it

		// Create our client object:
		ourClient = createClient(*env, verbosityLevel, progName);
		if (ourClient == NULL) 
		{
			*env << "Failed to create " << clientProtocolName
				<< " client: " << env->getResultMsg() << "\n";

			pThis->SetRtspStatus( RTSPStatus_Error_Connect_Srv );
			break;
		}

		// 开始获取Opition.
		pThis->SetRtspStatus( RTSPStatus_Opitiion );
		// Begin by sending an "OPTIONS" command:
		char* optionsResponse
			= getOptionsResponse(ourClient, pThis->m_strRTSPUrlA.c_str(), username, password);

		if (optionsResponse == NULL) 
		{
			*env << clientProtocolName << " \"OPTIONS\" request failed: "
				<< env->getResultMsg() << "\n";

			pThis->SetRtspStatus( RTSPStatus_Error_Connect_Srv );
			break;
		} 
		else 
		{
			*env << clientProtocolName << " \"OPTIONS\" request returned: "
				<< optionsResponse << "\n";
		}
		if( optionsResponse )
		{
			delete[] optionsResponse;
		}
			

		// 开始获取Description.
		// Open the URL, to get a SDP description:
		pThis->SetRtspStatus( RTSPStatus_Description );
		char* sdpDescription
			= getSDPDescriptionFromURL(ourClient, pThis->m_strRTSPUrlA.c_str(), username, password,
			proxyServerName, proxyServerPortNum,
			desiredPortNum);
		if (sdpDescription == NULL) 
		{
			*env << "Failed to get a SDP description from URL \"" << pThis->m_strRTSPUrlA.c_str()
				<< "\": " << env->getResultMsg() << "\n";
			pThis->SetRtspStatus( RTSPStatus_Error_Connect_Srv );
			break;
		}

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

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

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

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

			if (true) 
			{
				if (!subsession->initiate(simpleRTPoffsetArg)) 
				{
					*env << "Unable to create receiver for \"" << subsession->mediumName()
						<< "/" << subsession->codecName()
						<< "\" subsession: " << env->getResultMsg() << "\n";
				} 
				else 
				{
					*env << "Created receiver for \"" << subsession->mediumName()
						<< "/" << subsession->codecName()
						<< "\" subsession (client ports " << subsession->clientPortNum()
						<< "-" << 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()
								<< "/" << subsession->codecName()
								<< "\" subsession from "
								<< curBufferSize << " to "
								<< newBufferSize << " bytes\n";
						}
					}
				}
			} 
			else 
			{
				mcu::tlog << _T( "Use port: " ) << (int)subsession->clientPortNum() << endl;
				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) 
			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;
}
Пример #8
0
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;
}
Пример #9
0
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());

	// 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;
}
Пример #10
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(); 
	}

	return bSuccess ;
}
Пример #11
0
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
		 */
		SetupSender *setupSender = new SetupSender(*senderSave);
		if (setupSender == NULL)
		{
			LOG_ERR("warning");
			continue;
		}
        
		sender->RecordSender(setupSender);
		senderMove->setNext(setupSender);
		senderMove = setupSender;
		setupSender->setRspHandler(respHandler);
		setupSender->setSubsession(subsession);

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

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

	return true;
}
Пример #12
0
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) 
          {
            // 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;
}
Пример #13
0
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;
}
Пример #14
0
int main(int argc, char** argv) {
  // Begin by setting up our usage environment:
  TaskScheduler* scheduler = BasicTaskScheduler::createNew();
  env = BasicUsageEnvironment::createNew(*scheduler);

  progName = argv[0];

  gettimeofday(&startTime, NULL);

#ifdef USE_SIGNALS
  // Allow ourselves to be shut down gracefully by a SIGHUP or a SIGUSR1:
  signal(SIGHUP, signalHandlerShutdown);
  signal(SIGUSR1, signalHandlerShutdown);
#endif

  unsigned short desiredPortNum = 0;

  // unfortunately we can't use getopt() here, as Windoze doesn't have it
  while (argc > 2) {
    char* const opt = argv[1];
    if (opt[0] != '-') usage();
    switch (opt[1]) {

    case 'p': { // specify start port number
      int portArg;
      if (sscanf(argv[2], "%d", &portArg) != 1) {
	usage();
      }
      if (portArg <= 0 || portArg >= 65536 || portArg&1) {
	*env << "bad port number: " << portArg
		<< " (must be even, and in the range (0,65536))\n";
	usage();
      }
      desiredPortNum = (unsigned short)portArg;
      ++argv; --argc;
      break;
    }

    case 'r': { // do not receive data (instead, just 'play' the stream(s))
      createReceivers = False;
      break;
    }

    case 'q': { // output a QuickTime file (to stdout)
      outputQuickTimeFile = True;
      break;
    }

    case '4': { // output a 'mp4'-format file (to stdout)
      outputQuickTimeFile = True;
      generateMP4Format = True;
      break;
    }

    case 'i': { // output an AVI file (to stdout)
      outputAVIFile = True;
      break;
    }

    case 'I': { // specify input interface...
      NetAddressList addresses(argv[2]);
      if (addresses.numAddresses() == 0) {
	*env << "Failed to find network address for \"" << argv[2] << "\"";
	break;
      }
      ReceivingInterfaceAddr = *(unsigned*)(addresses.firstAddress()->data());
      ++argv; --argc;
      break;
    }

    case 'a': { // receive/record an audio stream only
      audioOnly = True;
      singleMedium = "audio";
      break;
    }

    case 'v': { // receive/record a video stream only
      videoOnly = True;
      singleMedium = "video";
      break;
    }

    case 'V': { // disable verbose output
      verbosityLevel = 0;
      break;
    }

    case 'd': { // specify duration, or how much to delay after end time
      float arg;
      if (sscanf(argv[2], "%g", &arg) != 1) {
	usage();
      }
      if (argv[2][0] == '-') { // not "arg<0", in case argv[2] was "-0"
	// a 'negative' argument was specified; use this for "durationSlop":
	duration = 0; // use whatever's in the SDP
	durationSlop = -arg;
      } else {
	duration = arg;
	durationSlop = 0;
      }
      ++argv; --argc;
      break;
    }

    case 'D': { // specify maximum number of seconds to wait for packets:
      if (sscanf(argv[2], "%u", &interPacketGapMaxTime) != 1) {
	usage();
      }
      ++argv; --argc;
      break;
    }

    case 'c': { // play continuously
      playContinuously = True;
      break;
    }

    case 'S': { // specify an offset to use with "SimpleRTPSource"s
      if (sscanf(argv[2], "%d", &simpleRTPoffsetArg) != 1) {
	usage();
      }
      if (simpleRTPoffsetArg < 0) {
	*env << "offset argument to \"-S\" must be >= 0\n";
	usage();
      }
      ++argv; --argc;
      break;
    }

    case 'O': { // Don't send an "OPTIONS" request before "DESCRIBE"
      sendOptionsRequest = False;
      break;
    }

    case 'o': { // Send only the "OPTIONS" request to the server
      sendOptionsRequestOnly = True;
      break;
    }

    case 'm': { // output multiple files - one for each frame
      oneFilePerFrame = True;
      break;
    }

    case 'n': { // notify the user when the first data packet arrives
      notifyOnPacketArrival = True;
      break;
    }

    case 't': {
      // stream RTP and RTCP over the TCP 'control' connection
      if (controlConnectionUsesTCP) {
	streamUsingTCP = True;
      } else {
	usage();
      }
      break;
    }

    case 'T': {
      // stream RTP and RTCP over a HTTP connection
      if (controlConnectionUsesTCP) {
	if (argc > 3 && argv[2][0] != '-') {
	  // The next argument is the HTTP server port number:
	  if (sscanf(argv[2], "%hu", &tunnelOverHTTPPortNum) == 1
	      && tunnelOverHTTPPortNum > 0) {
	    ++argv; --argc;
	    break;
	  }
	}
      }

      // If we get here, the option was specified incorrectly:
      usage();
      break;
    }

    case 'u': { // specify a username and password
      username = argv[2];
      password = argv[3];
      argv+=2; argc-=2;
      if (allowProxyServers && argc > 3 && argv[2][0] != '-') {
	// The next argument is the name of a proxy server:
	proxyServerName = argv[2];
	++argv; --argc;

	if (argc > 3 && argv[2][0] != '-') {
	  // The next argument is the proxy server port number:
	  if (sscanf(argv[2], "%hu", &proxyServerPortNum) != 1) {
	    usage();
	  }
	  ++argv; --argc;
	}
      }
      break;
    }

    case 'A': { // specify a desired audio RTP payload format
      unsigned formatArg;
      if (sscanf(argv[2], "%u", &formatArg) != 1
	  || formatArg >= 96) {
	usage();
      }
      desiredAudioRTPPayloadFormat = (unsigned char)formatArg;
      ++argv; --argc;
      break;
    }

    case 'M': { // specify a MIME subtype for a dynamic RTP payload type
      mimeSubtype = argv[2];
      if (desiredAudioRTPPayloadFormat==0) desiredAudioRTPPayloadFormat =96;
      ++argv; --argc;
      break;
    }

    case 'w': { // specify a width (pixels) for an output QuickTime or AVI movie
      if (sscanf(argv[2], "%hu", &movieWidth) != 1) {
	usage();
      }
      movieWidthOptionSet = True;
      ++argv; --argc;
      break;
    }

    case 'h': { // specify a height (pixels) for an output QuickTime or AVI movie
      if (sscanf(argv[2], "%hu", &movieHeight) != 1) {
	usage();
      }
      movieHeightOptionSet = True;
      ++argv; --argc;
      break;
    }

    case 'f': { // specify a frame rate (per second) for an output QT or AVI movie
      if (sscanf(argv[2], "%u", &movieFPS) != 1) {
	usage();
      }
      movieFPSOptionSet = True;
      ++argv; --argc;
      break;
    }

    case 'F': { // specify a prefix for the audio and video output files
      fileNamePrefix = argv[2];
      ++argv; --argc;
      break;
    }

    case 'b': { // specify the size of buffers for "FileSink"s
      if (sscanf(argv[2], "%u", &fileSinkBufferSize) != 1) {
	usage();
      }
      ++argv; --argc;
      break;
    }

    case 'B': { // specify the size of input socket buffers
      if (sscanf(argv[2], "%u", &socketInputBufferSize) != 1) {
	usage();
      }
      ++argv; --argc;
      break;
    }

    // Note: The following option is deprecated, and may someday be removed:
    case 'l': { // try to compensate for packet loss by repeating frames
      packetLossCompensate = True;
      break;
    }

    case 'y': { // synchronize audio and video streams
      syncStreams = True;
      break;
    }

    case 'H': { // generate hint tracks (as well as the regular data tracks)
      generateHintTracks = True;
      break;
    }

    case 'Q': { // output QOS measurements
      qosMeasurementIntervalMS = 1000; // default: 1 second

      if (argc > 3 && argv[2][0] != '-') {
	// The next argument is the measurement interval,
	// in multiples of 100 ms
	if (sscanf(argv[2], "%u", &qosMeasurementIntervalMS) != 1) {
	  usage();
	}
	qosMeasurementIntervalMS *= 100;
	++argv; --argc;
      }
      break;
    }

    case 's': { // specify initial seek time (trick play)
      double arg;
      if (sscanf(argv[2], "%lg", &arg) != 1 || arg < 0) {
	usage();
      }
      initialSeekTime = arg;
      ++argv; --argc;
      break;
    }

    case 'z': { // scale (trick play)
      float arg;
      if (sscanf(argv[2], "%g", &arg) != 1 || arg == 0.0f) {
	usage();
      }
      scale = arg;
      ++argv; --argc;
      break;
    }

    default: {
      usage();
      break;
    }
    }

    ++argv; --argc;
  }
  if (argc != 2) usage();
  if (outputQuickTimeFile && outputAVIFile) {
    *env << "The -i and -q (or -4) flags cannot both be used!\n";
    usage();
  }
  Boolean outputCompositeFile = outputQuickTimeFile || outputAVIFile;
  if (!createReceivers && outputCompositeFile) {
    *env << "The -r and -q (or -4 or -i) flags cannot both be used!\n";
    usage();
  }
  if (outputCompositeFile && !movieWidthOptionSet) {
    *env << "Warning: The -q, -4 or -i option was used, but not -w.  Assuming a video width of "
	 << movieWidth << " pixels\n";
  }
  if (outputCompositeFile && !movieHeightOptionSet) {
    *env << "Warning: The -q, -4 or -i option was used, but not -h.  Assuming a video height of "
	 << movieHeight << " pixels\n";
  }
  if (outputCompositeFile && !movieFPSOptionSet) {
    *env << "Warning: The -q, -4 or -i option was used, but not -f.  Assuming a video frame rate of "
	 << movieFPS << " frames-per-second\n";
  }
  if (audioOnly && videoOnly) {
    *env << "The -a and -v flags cannot both be used!\n";
    usage();
  }
  if (sendOptionsRequestOnly && !sendOptionsRequest) {
    *env << "The -o and -O flags cannot both be used!\n";
    usage();
  }
  if (tunnelOverHTTPPortNum > 0) {
    if (streamUsingTCP) {
      *env << "The -t and -T flags cannot both be used!\n";
      usage();
    } else {
      streamUsingTCP = True;
    }
  }
  if (!createReceivers && notifyOnPacketArrival) {
    *env << "Warning: Because we're not receiving stream data, the -n flag has no effect\n";
  }
  if (durationSlop < 0) {
    // This parameter wasn't set, so use a default value.
    // If we're measuring QOS stats, then don't add any slop, to avoid
    // having 'empty' measurement intervals at the end.
    durationSlop = qosMeasurementIntervalMS > 0 ? 0.0 : 5.0;
  }

  char* url = argv[1];

  // Create our client object:
  ourClient = createClient(*env, verbosityLevel, progName);
  if (ourClient == NULL) {
    *env << "Failed to create " << clientProtocolName
		<< " client: " << env->getResultMsg() << "\n";
    shutdown();
  }

  if (sendOptionsRequest) {
    // Begin by sending an "OPTIONS" command:
    char* optionsResponse
      = getOptionsResponse(ourClient, url, username, password);
    if (sendOptionsRequestOnly) {
      if (optionsResponse == NULL) {
	*env << clientProtocolName << " \"OPTIONS\" request failed: "
	     << env->getResultMsg() << "\n";
      } else {
	*env << clientProtocolName << " \"OPTIONS\" request returned: "
	     << optionsResponse << "\n";
      }
      shutdown();
    }
    delete[] optionsResponse;
  }

  // Open the URL, to get a SDP description:
  char* sdpDescription
    = getSDPDescriptionFromURL(ourClient, url, username, password,
			       proxyServerName, proxyServerPortNum,
			       desiredPortNum);
  if (sdpDescription == NULL) {
    *env << "Failed to get a SDP description from URL \"" << url
		<< "\": " << env->getResultMsg() << "\n";
    shutdown();
  }

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

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

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

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

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

  // Create output files:
  if (createReceivers) {
    if (outputQuickTimeFile) {
      // 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) {
	  *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
}