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, (long)(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);
}
}
void ProxyRTSPClient::checkInterPacketGaps_() {
if (fInterPacketGapMaxTime == 0) return; // we're not checking
// Check each subsession, counting up how many packets have been received:
unsigned newTotNumPacketsReceived = 0;
MediaSubsessionIterator iter(*fOurServerMediaSession.fClientMediaSession);
MediaSubsession* subsession;
while ((subsession = iter.next()) != NULL) {
RTPSource* src = subsession->rtpSource();
if (src == NULL) continue;
newTotNumPacketsReceived += src->receptionStatsDB().totNumPacketsReceived();
}
//envir() << *this << "::doLivenessCheck fTotNumPacketsReceived: " << fTotNumPacketsReceived
// << ", newTotNumPacketsReceived: " << newTotNumPacketsReceived << "\n";
if (newTotNumPacketsReceived == fTotNumPacketsReceived) {
// 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";
if (fVerbosityLevel > 0) {
envir() << *this << "::doLivenessCheck last packet received: >" << fInterPacketGapMaxTime
<< " seconds ago. Resetting session\n";
}
continueAfterLivenessCommand(1/*hack*/, fServerSupportsGetParameter);
} else {
fTotNumPacketsReceived = newTotNumPacketsReceived;
// Check again, after the specified delay:
fInterPacketGapsTask = envir().taskScheduler().scheduleDelayedTask(fInterPacketGapMaxTime*MILLION, checkInterPacketGaps, this);
}
}
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;
}
}
}
bool MtkRTSPClient::IsPacketArrived()
{
MediaSubsessionIterator iter(*session);
MediaSubsession* subsession;
while ((subsession = iter.next()) != NULL)
{
RTPSource* src = subsession->rtpSource();
if (src == NULL)
continue;
if (src->receptionStatsDB().numActiveSourcesSinceLastReset() > 0)
{
return true;
}
}
return false;
}
void
DummySink::afterGettingFrame(unsigned frameSize, unsigned numTruncatedBytes,
struct timeval presentationTime, unsigned /*durationInMicroseconds*/) {
#ifndef ANDROID
extern pthread_mutex_t watchdogMutex;
extern struct timeval watchdogTimer;
#endif
if(fSubsession.rtpPayloadFormat() == video_sess_fmt) {
bool marker = false;
int channel = port2channel[fSubsession.clientPortNum()];
RTPSource *rtpsrc = fSubsession.rtpSource();
#ifdef ANDROID // support only single channel
if(channel > 0)
goto dropped;
#endif
if(rtpsrc != NULL) {
marker = rtpsrc->curPacketMarkerBit();
}
play_video(channel,
fReceiveBuffer+MAX_FRAMING_SIZE-video_framing,
frameSize+video_framing, presentationTime,
marker);
#ifdef ANDROID
if(rtspconf->builtin_video_decoder==0
&& rtspconf->builtin_audio_decoder==0)
kickWatchdog(rtspParam->jnienv);
#endif
} else if(fSubsession.rtpPayloadFormat() == audio_sess_fmt) {
play_audio(fReceiveBuffer+MAX_FRAMING_SIZE-audio_framing,
frameSize+audio_framing, presentationTime);
}
#ifndef ANDROID // watchdog is implemented at the Java side
pthread_mutex_lock(&watchdogMutex);
gettimeofday(&watchdogTimer, NULL);
pthread_mutex_unlock(&watchdogMutex);
#endif
dropped:
// Then continue, to request the next frame of data:
continuePlaying();
}
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;
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 printQOSData(int exitCode) {
*env << "begin_QOS_statistics\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();
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" << int(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";
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";
}
curQOSRecord = curQOSRecord->fNext;
}
}
}
*env << "end_QOS_statistics\n";
delete qosRecordHead;
}