playcommon.cpp

linux下的rtsp/rtp/rtcp协议栈源代码； c++写的

}

void subsessionAfterPlaying(void* clientData) {
  // Begin by closing this media subsession:
  MediaSubsession* subsession = (MediaSubsession*)clientData;
  Medium::close(subsession->sink);
  subsession->sink = NULL;

  // Next, check whether *all* subsessions have now been closed:
  MediaSession& session = subsession->parentSession();
  MediaSubsessionIterator iter(session);
  while ((subsession = iter.next()) != NULL) {
    if (subsession->sink != NULL) return; // this subsession is still active
  }

  // All subsessions have now been closed
  sessionAfterPlaying();
}

void subsessionByeHandler(void* clientData) {
  struct timeval timeNow;
  gettimeofday(&timeNow, NULL);
  unsigned secsDiff = timeNow.tv_sec - startTime.tv_sec;

  MediaSubsession* subsession = (MediaSubsession*)clientData;
  *env << "Received RTCP \"BYE\" on \"" << subsession->mediumName()
	<< "/" << subsession->codecName()
	<< "\" subsession (after " << secsDiff
	<< " seconds)\n";

  // Act now as if the subsession had closed:
  subsessionAfterPlaying(subsession);
}

void sessionAfterPlaying(void* /*clientData*/) {
  if (!playContinuously) {
    shutdown(0);
  } else {
    // We've been asked to play the stream(s) over again:
    startPlayingStreams();
  }
}

void sessionTimerHandler(void* /*clientData*/) {
  sessionTimerTask = NULL;

  sessionAfterPlaying();
}

class qosMeasurementRecord {
public:
  qosMeasurementRecord(struct timeval const& startTime, RTPSource* src)
    : fSource(src), fNext(NULL),
      kbits_per_second_min(1e20), kbits_per_second_max(0),
      kBytesTotal(0.0),
      packet_loss_fraction_min(1.0), packet_loss_fraction_max(0.0),
      totNumPacketsReceived(0), totNumPacketsExpected(0) {
    measurementEndTime = measurementStartTime = startTime;

#ifdef SUPPORT_REAL_RTSP
    if (session->isRealNetworksRDT) { // hack for RealMedia sessions (RDT, not RTP)
      RealRDTSource* rdt = (RealRDTSource*)src;
      kBytesTotal = rdt->totNumKBytesReceived();
      totNumPacketsReceived = rdt->totNumPacketsReceived();
      totNumPacketsExpected = totNumPacketsReceived; // because we use TCP
      return;
    }
#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) {
      kBytesTotal = stats->totNumKBytesReceived();
      totNumPacketsReceived = stats->totNumPacketsReceived();
      totNumPacketsExpected = stats->totNumPacketsExpected();
    }
  }
  virtual ~qosMeasurementRecord() { delete fNext; }
    
  void periodicQOSMeasurement(struct timeval const& timeNow);

public:
  RTPSource* fSource;
  qosMeasurementRecord* fNext;

public:
  struct timeval measurementStartTime, measurementEndTime;
  double kbits_per_second_min, kbits_per_second_max;
  double kBytesTotal;
  double packet_loss_fraction_min, packet_loss_fraction_max;
  unsigned totNumPacketsReceived, totNumPacketsExpected;
};

static qosMeasurementRecord* qosRecordHead = NULL;

static void periodicQOSMeasurement(void* clientData); // forward

static unsigned nextQOSMeasurementUSecs;

static void scheduleNextQOSMeasurement() {
  nextQOSMeasurementUSecs += qosMeasurementIntervalMS*1000;
  struct timeval timeNow;
  gettimeofday(&timeNow, NULL);
  unsigned timeNowUSecs = timeNow.tv_sec*1000000 + timeNow.tv_usec;
  unsigned usecsToDelay = nextQOSMeasurementUSecs < timeNowUSecs ? 0
    : nextQOSMeasurementUSecs - timeNowUSecs;

  qosMeasurementTimerTask = env->taskScheduler().scheduleDelayedTask(
     usecsToDelay, (TaskFunc*)periodicQOSMeasurement, (void*)NULL);
}

static void periodicQOSMeasurement(void* /*clientData*/) {
  struct timeval timeNow;
  gettimeofday(&timeNow, NULL);

  for (qosMeasurementRecord* qosRecord = qosRecordHead;
       qosRecord != NULL; qosRecord = qosRecord->fNext) {
    qosRecord->periodicQOSMeasurement(timeNow);
  }

  // Do this again later:
  scheduleNextQOSMeasurement();
}

void qosMeasurementRecord
::periodicQOSMeasurement(struct timeval const& timeNow) {
  unsigned secsDiff = timeNow.tv_sec - measurementEndTime.tv_sec;
  int usecsDiff = timeNow.tv_usec - measurementEndTime.tv_usec;
  double timeDiff = secsDiff + usecsDiff/1000000.0;
  measurementEndTime = timeNow;

#ifdef SUPPORT_REAL_RTSP
  if (session->isRealNetworksRDT) { // hack for RealMedia sessions (RDT, not RTP)
    RealRDTSource* rdt = (RealRDTSource*)fSource;
    double kBytesTotalNow = rdt->totNumKBytesReceived();
    double kBytesDeltaNow = kBytesTotalNow - kBytesTotal;
    kBytesTotal = kBytesTotalNow;

    double kbpsNow = timeDiff == 0.0 ? 0.0 : 8*kBytesDeltaNow/timeDiff;
    if (kbpsNow < 0.0) kbpsNow = 0.0; // in case of roundoff error
    if (kbpsNow < kbits_per_second_min) kbits_per_second_min = kbpsNow;
    if (kbpsNow > kbits_per_second_max) kbits_per_second_max = kbpsNow;

    totNumPacketsReceived = rdt->totNumPacketsReceived();
    totNumPacketsExpected = totNumPacketsReceived; // because we use TCP
    packet_loss_fraction_min = packet_loss_fraction_max = 0.0; // ditto
    return;
  }
#endif
  RTPReceptionStatsDB::Iterator statsIter(fSource->receptionStatsDB());
  // Assume that there's only one SSRC source (usually the case):
  RTPReceptionStats* stats = statsIter.next(True);
  if (stats != NULL) {
    double kBytesTotalNow = stats->totNumKBytesReceived();
    double kBytesDeltaNow = kBytesTotalNow - kBytesTotal;
    kBytesTotal = kBytesTotalNow;

    double kbpsNow = timeDiff == 0.0 ? 0.0 : 8*kBytesDeltaNow/timeDiff;
    if (kbpsNow < 0.0) kbpsNow = 0.0; // in case of roundoff error
    if (kbpsNow < kbits_per_second_min) kbits_per_second_min = kbpsNow;
    if (kbpsNow > kbits_per_second_max) kbits_per_second_max = kbpsNow;

    unsigned totReceivedNow = stats->totNumPacketsReceived();
    unsigned totExpectedNow = stats->totNumPacketsExpected();
    unsigned deltaReceivedNow = totReceivedNow - totNumPacketsReceived;
    unsigned deltaExpectedNow = totExpectedNow - totNumPacketsExpected;
    totNumPacketsReceived = totReceivedNow;
    totNumPacketsExpected = totExpectedNow;

    double lossFractionNow = deltaExpectedNow == 0 ? 0.0
      : 1.0 - deltaReceivedNow/(double)deltaExpectedNow;
    //if (lossFractionNow < 0.0) lossFractionNow = 0.0; //reordering can cause
    if (lossFractionNow < packet_loss_fraction_min) {
      packet_loss_fraction_min = lossFractionNow;
    }
    if (lossFractionNow > packet_loss_fraction_max) {
      packet_loss_fraction_max = lossFractionNow;
    }
  }
}

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();
}

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;
}

void shutdown(int exitCode) {
  if (env != NULL) {
    env->taskScheduler().unscheduleDelayedTask(sessionTimerTask);
    env->taskScheduler().unscheduleDelayedTask(arrivalCheckTimerTask);
    env->taskScheduler().unscheduleDelayedTask(interPacketGapCheckTimerTask);
    env->taskScheduler().unscheduleDelayedTask(qosMeasurementTimerTask);
  }

  if (qosMeasurementIntervalMS > 0) {
    printQOSData(exitCode);
  }

  // Close our output files:
  closeMediaSinks();

  // Teardown, then shutdown, any outstanding RTP/RTCP subsessions
  tearDownStreams();
  Medium::close(session);

  // Finally, shut down our client:
  Medium::close(ourClient);

  // Adios...
  exit(exitCode);
}

void signalHandlerShutdown(int /*sig*/) {
  *env << "Got shutdown signal\n";
  shutdown(0);
}

void checkForPacketArrival(void* /*clientData*/) {
  if (!notifyOnPacketArrival) return; // we're not checking 

  // Check each subsession, to see whether it has received data packets:
  unsigned numSubsessionsChecked = 0;
  unsigned numSubsessionsWithReceivedData = 0;
  unsigned numSubsessionsThatHaveBeenSynced = 0;

  MediaSubsessionIterator iter(*session);
  MediaSubsession* subsession;
  while ((subsession = iter.next()) != NULL) {
    RTPSource* src = subsession->rtpSource();
    if (src == NULL) continue;
    ++numSubsessionsChecked;

    if (src->receptionStatsDB().numActiveSourcesSinceLastReset() > 0) {
      // At least one data packet has arrived
      ++numSubsessionsWithReceivedData;
    }
    if (src->hasBeenSynchronizedUsingRTCP()) {
      ++numSubsessionsThatHaveBeenSynced;
    }
  }

  unsigned numSubsessionsToCheck = numSubsessionsChecked;
  // Special case for "QuickTimeFileSink"s and "AVIFileSink"s:
  // They might not use all of the input sources:
  if (qtOut != NULL) {
    numSubsessionsToCheck = qtOut->numActiveSubsessions();
  } else if (aviOut != NULL) {
    numSubsessionsToCheck = aviOut->numActiveSubsessions();
  }

  Boolean notifyTheUser;
  if (!syncStreams) {
    notifyTheUser = numSubsessionsWithReceivedData > 0; // easy case
  } else {
    notifyTheUser = numSubsessionsWithReceivedData >= numSubsessionsToCheck
      && numSubsessionsThatHaveBeenSynced == numSubsessionsChecked;
    // Note: A subsession with no active sources is considered to be synced
  }
  if (notifyTheUser) {
    struct timeval timeNow;
    gettimeofday(&timeNow, NULL);
	char timestampStr[100];
	sprintf(timestampStr, "%ld%03ld", timeNow.tv_sec, timeNow.tv_usec/1000);
    *env << (syncStreams ? "Synchronized d" : "D")
		<< "ata packets have begun arriving [" << timestampStr << "]\007\n";
    return;
  }

  // No luck, so reschedule this check again, after a delay:
  int uSecsToDelay = 100000; // 100 ms
  arrivalCheckTimerTask
    = env->taskScheduler().scheduleDelayedTask(uSecsToDelay,
			       (TaskFunc*)checkForPacketArrival, NULL);
}

void checkInterPacketGaps(void* /*clientData*/) {
  if (interPacketGapMaxTime == 0) return; // we're not checking 

  // Check each subsession, counting up how many packets have been received:
  unsigned newTotNumPacketsReceived = 0;

  MediaSubsessionIterator iter(*session);
  MediaSubsession* subsession;
  while ((subsession = iter.next()) != NULL) {
    RTPSource* src = subsession->rtpSource();
    if (src == NULL) continue;
    newTotNumPacketsReceived += src->receptionStatsDB().totNumPacketsReceived();
  }

  if (newTotNumPacketsReceived == totNumPacketsReceived) {
    // No additional packets have been received since the last time we
    // checked, so end this stream:
    *env << "Closing session, because we stopped receiving packets.\n";
    interPacketGapCheckTimerTask = NULL;
    sessionAfterPlaying();
  } else {
    totNumPacketsReceived = newTotNumPacketsReceived;
    // Check again, after the specified delay: 
    interPacketGapCheckTimerTask
      = env->taskScheduler().scheduleDelayedTask(interPacketGapMaxTime*1000000,
				 (TaskFunc*)checkInterPacketGaps, NULL);
  }
}