testProgs/playCommon.cpp File Reference

#include "playCommon.hh"
#include "BasicUsageEnvironment.hh"
#include "GroupsockHelper.hh"
#include <signal.h>

Include dependency graph for playCommon.cpp:

Go to the source code of this file.

Data Structures
class	qosMeasurementRecord
Defines
#define	USE_SIGNALS   1
Functions
void	continueAfterOPTIONS (RTSPClient *client, int resultCode, char *resultString)
void	continueAfterDESCRIBE (RTSPClient *client, int resultCode, char *resultString)
void	continueAfterSETUP (RTSPClient *client, int resultCode, char *resultString)
void	continueAfterPLAY (RTSPClient *client, int resultCode, char *resultString)
void	continueAfterTEARDOWN (RTSPClient *client, int resultCode, char *resultString)
void	setupStreams ()
void	closeMediaSinks ()
void	subsessionAfterPlaying (void *clientData)
void	subsessionByeHandler (void *clientData)
void	sessionAfterPlaying (void *clientData=NULL)
void	sessionTimerHandler (void *clientData)
void	shutdown (int exitCode=1)
void	signalHandlerShutdown (int sig)
void	checkForPacketArrival (void *clientData)
void	checkInterPacketGaps (void *clientData)
void	beginQOSMeasurement ()
void	usage ()
int	main (int argc, char **argv)
static void	periodicQOSMeasurement (void *clientData)
static void	scheduleNextQOSMeasurement ()
void	printQOSData (int exitCode)
Variables
char const *	progName
UsageEnvironment *	env
Medium *	ourClient = NULL
Authenticator *	ourAuthenticator = NULL
char const *	streamURL = NULL
MediaSession *	session = NULL
TaskToken	sessionTimerTask = NULL
TaskToken	arrivalCheckTimerTask = NULL
TaskToken	interPacketGapCheckTimerTask = NULL
TaskToken	qosMeasurementTimerTask = NULL
Boolean	createReceivers = True
Boolean	outputQuickTimeFile = False
Boolean	generateMP4Format = False
QuickTimeFileSink *	qtOut = NULL
Boolean	outputAVIFile = False
AVIFileSink *	aviOut = NULL
Boolean	audioOnly = False
Boolean	videoOnly = False
char const *	singleMedium = NULL
int	verbosityLevel = 1
double	duration = 0
double	durationSlop = -1.0
double	initialSeekTime = 0.0f
float	scale = 1.0f
double	endTime
unsigned	interPacketGapMaxTime = 0
unsigned	totNumPacketsReceived = ~0
Boolean	playContinuously = False
int	simpleRTPoffsetArg = -1
Boolean	sendOptionsRequest = True
Boolean	sendOptionsRequestOnly = False
Boolean	oneFilePerFrame = False
Boolean	notifyOnPacketArrival = False
Boolean	streamUsingTCP = False
unsigned short	desiredPortNum = 0
portNumBits	tunnelOverHTTPPortNum = 0
char *	username = NULL
char *	password = NULL
char *	proxyServerName = NULL
unsigned short	proxyServerPortNum = 0
unsigned char	desiredAudioRTPPayloadFormat = 0
char *	mimeSubtype = NULL
unsigned short	movieWidth = 240
Boolean	movieWidthOptionSet = False
unsigned short	movieHeight = 180
Boolean	movieHeightOptionSet = False
unsigned	movieFPS = 15
Boolean	movieFPSOptionSet = False
char const *	fileNamePrefix = ""
unsigned	fileSinkBufferSize = 100000
unsigned	socketInputBufferSize = 0
Boolean	packetLossCompensate = False
Boolean	syncStreams = False
Boolean	generateHintTracks = False
unsigned	qosMeasurementIntervalMS = 0
timeval	startTime
MediaSubsession *	subsession
Boolean	madeProgress = False
static qosMeasurementRecord *	qosRecordHead = NULL
static unsigned	nextQOSMeasurementUSecs
Boolean	areAlreadyShuttingDown = False
int	shutdownExitCode

---

Define Documentation

#define USE_SIGNALS   1

Definition at line 32 of file playCommon.cpp.

---

Function Documentation

void beginQOSMeasurement

(

)

Definition at line 1029 of file playCommon.cpp.

References qosMeasurementRecord::fNext, iter, MediaSubsessionIterator::next(), nextQOSMeasurementUSecs, NULL, qosRecordHead, MediaSubsession::rtpSource(), scheduleNextQOSMeasurement(), session, and subsession.

Referenced by continueAfterPLAY().

                           {
  // 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();
    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 checkForPacketArrival

(

void *

clientData

)

Definition at line 1175 of file playCommon.cpp.

References arrivalCheckTimerTask, aviOut, env, RTPSource::hasBeenSynchronizedUsingRTCP(), iter, MediaSubsessionIterator::next(), notifyOnPacketArrival, NULL, RTPReceptionStatsDB::numActiveSourcesSinceLastReset(), AVIFileSink::numActiveSubsessions(), QuickTimeFileSink::numActiveSubsessions(), qtOut, RTPSource::receptionStatsDB(), MediaSubsession::rtpSource(), TaskScheduler::scheduleDelayedTask(), session, subsession, syncStreams, and UsageEnvironment::taskScheduler().

Referenced by continueAfterPLAY().

                                                 {
  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

)

Definition at line 1233 of file playCommon.cpp.

References env, interPacketGapCheckTimerTask, interPacketGapMaxTime, iter, MediaSubsessionIterator::next(), NULL, RTPSource::receptionStatsDB(), MediaSubsession::rtpSource(), TaskScheduler::scheduleDelayedTask(), session, sessionAfterPlaying(), subsession, UsageEnvironment::taskScheduler(), totNumPacketsReceived, and RTPReceptionStatsDB::totNumPacketsReceived().

Referenced by continueAfterPLAY().

                                                {
  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 closeMediaSinks

(

)

Definition at line 855 of file playCommon.cpp.

References aviOut, Medium::close(), iter, MediaSubsessionIterator::next(), NULL, qtOut, session, MediaSubsession::sink, and subsession.

Referenced by continueAfterTEARDOWN().

                       {
  Medium::close(qtOut);
  Medium::close(aviOut);

  if (session == NULL) return;
  MediaSubsessionIterator iter(*session);
  MediaSubsession* subsession;
  while ((subsession = iter.next()) != NULL) {
    Medium::close(subsession->sink);
    subsession->sink = NULL;
  }
}

void continueAfterDESCRIBE	(	RTSPClient *	client,
		int	resultCode,
		char *	resultString
	)

Definition at line 534 of file playCommon.cpp.

References MediaSubsession::clientPortNum(), MediaSubsession::codecName(), MediaSession::createNew(), createReceivers, desiredPortNum, env, False, fileSinkBufferSize, getReceiveBufferSize(), UsageEnvironment::getResultMsg(), MediaSession::hasSubsessions(), MediaSubsession::initiate(), iter, madeProgress, MediaSubsession::mediumName(), MediaSubsessionIterator::next(), NULL, RTPSource::RTPgs(), MediaSubsession::rtpSource(), session, MediaSubsession::setClientPortNum(), RTPSource::setPacketReorderingThresholdTime(), setReceiveBufferTo(), setupStreams(), shutdown(), simpleRTPoffsetArg, singleMedium, socketInputBufferSize, Socket::socketNum(), streamURL, subsession, and True.

                                                                            {
  if (resultCode != 0) {
    *env << "Failed to get a SDP description for the URL \"" << streamURL << "\": " << resultString << "\n";
    shutdown();
  }

  char* sdpDescription = resultString;
  *env << "Opened URL \"" << streamURL << "\", 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., no \"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();
}

void continueAfterOPTIONS	(	RTSPClient *	client,
		int	resultCode,
		char *	resultString
	)

Definition at line 519 of file playCommon.cpp.

References clientProtocolName, continueAfterDESCRIBE(), env, getSDPDescription(), sendOptionsRequestOnly, and shutdown().

                                                                           {
  if (sendOptionsRequestOnly) {
    if (resultCode != 0) {
      *env << clientProtocolName << " \"OPTIONS\" request failed: " << resultString << "\n";
    } else {
      *env << clientProtocolName << " \"OPTIONS\" request returned: " << resultString << "\n";
    }
    shutdown();
  }
  delete[] resultString;

  // Next, get a SDP description for the stream:
  getSDPDescription(continueAfterDESCRIBE);
}

void continueAfterPLAY	(	RTSPClient *	client,
		int	resultCode,
		char *	resultString
	)

Definition at line 802 of file playCommon.cpp.

References beginQOSMeasurement(), checkForPacketArrival(), checkInterPacketGaps(), createReceivers, duration, durationSlop, endTime, env, False, initialSeekTime, NULL, MediaSession::playEndTime(), MediaSession::playStartTime(), qosMeasurementIntervalMS, scale, TaskScheduler::scheduleDelayedTask(), session, sessionTimerHandler(), sessionTimerTask, shutdown(), UsageEnvironment::taskScheduler(), and True.

Referenced by sessionAfterPlaying(), setupNextSubsession(), and setupStreams().

                                                                        {
  if (resultCode != 0) {
    *env << "Failed to start playing session: " << resultString << "\n";
    shutdown();
  } else {
    *env << "Started playing session\n";
  }

  if (qosMeasurementIntervalMS > 0) {
    // Begin periodic QOS measurements:
    beginQOSMeasurement();
  }

  // Figure out how long to delay (if at all) before shutting down, or
  // repeating the playing
  Boolean timerIsBeingUsed = False;
  double secondsToDelay = duration;
  if (duration > 0) {
    // First, adjust "duration" based on any change to the play range (that was specified in the "PLAY" response):
    double rangeAdjustment = (session->playEndTime() - session->playStartTime()) - (endTime - initialSeekTime);
    if (duration + rangeAdjustment > 0.0) duration += rangeAdjustment;

    timerIsBeingUsed = True;
    double absScale = scale > 0 ? scale : -scale; // ASSERT: scale != 0
    secondsToDelay = duration/absScale + durationSlop;

    int64_t uSecsToDelay = (int64_t)(secondsToDelay*1000000.0);
    sessionTimerTask = env->taskScheduler().scheduleDelayedTask(uSecsToDelay, (TaskFunc*)sessionTimerHandler, (void*)NULL);
  }

  char const* actionString
    = createReceivers? "Receiving streamed data":"Data is being streamed";
  if (timerIsBeingUsed) {
    *env << actionString
                << " (for up to " << secondsToDelay
                << " seconds)...\n";
  } else {
#ifdef USE_SIGNALS
    pid_t ourPid = getpid();
    *env << actionString
                << " (signal with \"kill -HUP " << (int)ourPid
                << "\" or \"kill -USR1 " << (int)ourPid
                << "\" to terminate)...\n";
#else
    *env << actionString << "...\n";
#endif
  }

  // Watch for incoming packets (if desired):
  checkForPacketArrival(NULL);
  checkInterPacketGaps(NULL);
}

void continueAfterSETUP	(	RTSPClient *	client,
		int	resultCode,
		char *	resultString
	)

Definition at line 638 of file playCommon.cpp.

References MediaSubsession::clientPortNum(), MediaSubsession::codecName(), env, UsageEnvironment::getResultMsg(), madeProgress, MediaSubsession::mediumName(), setupStreams(), subsession, and True.

                                                                         {
  if (resultCode == 0) {
      *env << "Setup \"" << subsession->mediumName()
           << "/" << subsession->codecName()
           << "\" subsession (client ports " << subsession->clientPortNum()
           << "-" << subsession->clientPortNum()+1 << ")\n";
      madeProgress = True;
  } else {
    *env << "Failed to setup \"" << subsession->mediumName()
         << "/" << subsession->codecName()
         << "\" subsession: " << env->getResultMsg() << "\n";
  }

  // Set up the next subsession, if any:
  setupStreams();
}

void continueAfterTEARDOWN	(	RTSPClient *	client,
		int	resultCode,
		char *	resultString
	)

Definition at line 1157 of file playCommon.cpp.

References Medium::close(), closeMediaSinks(), exit, ourAuthenticator, ourClient, session, and shutdownExitCode.

Referenced by shutdown().

                                                                                    {
  // Now that we've stopped any more incoming data from arriving, close our output files:
  closeMediaSinks();
  Medium::close(session);

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

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

int main	(	int	argc,
		char **	argv
	)

Definition at line 124 of file playCommon.cpp.

References allowProxyServers, audioOnly, clientProtocolName, continueAfterOPTIONS(), controlConnectionUsesTCP, createClient(), BasicUsageEnvironment::createNew(), BasicTaskScheduler::createNew(), createReceivers, NetAddress::data(), desiredAudioRTPPayloadFormat, desiredPortNum, TaskScheduler::doEventLoop(), duration, durationSlop, env, False, fileNamePrefix, fileSinkBufferSize, NetAddressList::firstAddress(), generateHintTracks, generateMP4Format, getOptions(), UsageEnvironment::getResultMsg(), initialSeekTime, interPacketGapMaxTime, mimeSubtype, movieFPS, movieFPSOptionSet, movieHeight, movieHeightOptionSet, movieWidth, movieWidthOptionSet, notifyOnPacketArrival, NULL, NetAddressList::numAddresses(), oneFilePerFrame, ourAuthenticator, ourClient, outputAVIFile, outputQuickTimeFile, packetLossCompensate, password, playContinuously, progName, proxyServerName, proxyServerPortNum, qosMeasurementIntervalMS, ReceivingInterfaceAddr, scale, sendOptionsRequest, sendOptionsRequestOnly, shutdown(), signalHandlerShutdown(), simpleRTPoffsetArg, singleMedium, socketInputBufferSize, startTime, streamURL, streamUsingTCP, syncStreams, UsageEnvironment::taskScheduler(), True, tunnelOverHTTPPortNum, usage(), username, verbosityLevel, and videoOnly.

                                {
  // 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

  // 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
      if (argc < 4) usage(); // there's no argv[3] (for the "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;
        }
      }

      ourAuthenticator = new Authenticator(username, password);
      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(); // there must be exactly one "rtsp://" URL at the end
  if (outputQuickTimeFile && outputAVIFile) {
    *env << "The -i and -q (or -4) options cannot both be used!\n";
    usage();
  }
  Boolean outputCompositeFile = outputQuickTimeFile || outputAVIFile;
  if (!createReceivers && outputCompositeFile) {
    *env << "The -r and -q (or -4 or -i) options 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 options cannot both be used!\n";
    usage();
  }
  if (sendOptionsRequestOnly && !sendOptionsRequest) {
    *env << "The -o and -O options cannot both be used!\n";
    usage();
  }
  if (tunnelOverHTTPPortNum > 0) {
    if (streamUsingTCP) {
      *env << "The -t and -T options 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;
  }

  streamURL = argv[1];

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

  if (sendOptionsRequest) {
    // Begin by sending an "OPTIONS" command:
    getOptions(continueAfterOPTIONS);
  } else {
    continueAfterOPTIONS(NULL, 0, NULL);
  }

  // All subsequent activity takes place within the event loop:
  env->taskScheduler().doEventLoop(); // does not return

  return 0; // only to prevent compiler warning
}

static void periodicQOSMeasurement

(

void *

clientData

)

[static]

Definition at line 977 of file playCommon.cpp.

References NULL, qosRecordHead, and scheduleNextQOSMeasurement().

Referenced by scheduleNextQOSMeasurement().

                                                         {
  struct timeval timeNow;
  gettimeofday(&timeNow, NULL);

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

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

void printQOSData

(

int

exitCode

)

Definition at line 1052 of file playCommon.cpp.

References MediaSubsession::codecName(), env, qosMeasurementRecord::fNext, iter, qosMeasurementRecord::kbits_per_second_max, qosMeasurementRecord::kbits_per_second_min, qosMeasurementRecord::kBytesTotal, qosMeasurementRecord::measurementEndTime, MediaSubsession::mediumName(), MediaSubsessionIterator::next(), NULL, qosMeasurementRecord::packet_loss_fraction_max, qosMeasurementRecord::packet_loss_fraction_min, qosMeasurementIntervalMS, qosRecordHead, RTPSource::receptionStatsDB(), MediaSubsession::rtpSource(), session, subsession, qosMeasurementRecord::totNumPacketsExpected, totNumPacketsReceived, qosMeasurementRecord::totNumPacketsReceived, and True.

Referenced by shutdown().

                                {
  *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;
}

static void scheduleNextQOSMeasurement

(

)

[static]

Definition at line 965 of file playCommon.cpp.

References env, nextQOSMeasurementUSecs, NULL, periodicQOSMeasurement(), qosMeasurementIntervalMS, qosMeasurementTimerTask, TaskScheduler::scheduleDelayedTask(), and UsageEnvironment::taskScheduler().

Referenced by beginQOSMeasurement(), and periodicQOSMeasurement().

                                         {
  nextQOSMeasurementUSecs += qosMeasurementIntervalMS*1000;
  struct timeval timeNow;
  gettimeofday(&timeNow, NULL);
  unsigned timeNowUSecs = timeNow.tv_sec*1000000 + timeNow.tv_usec;
  unsigned usecsToDelay = nextQOSMeasurementUSecs - timeNowUSecs;
     // Note: This works even when nextQOSMeasurementUSecs wraps around

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

void sessionAfterPlaying

(

void *

clientData = NULL

)

Definition at line 900 of file playCommon.cpp.

References arrivalCheckTimerTask, continueAfterPLAY(), endTime, env, initialSeekTime, interPacketGapCheckTimerTask, NULL, playContinuously, qosMeasurementTimerTask, scale, session, sessionTimerTask, shutdown(), startPlayingSession(), UsageEnvironment::taskScheduler(), totNumPacketsReceived, and TaskScheduler::unscheduleDelayedTask().

Referenced by checkInterPacketGaps(), sessionTimerHandler(), setupStreams(), and subsessionAfterPlaying().

                                               {
  if (!playContinuously) {
    shutdown(0);
  } else {
    // We've been asked to play the stream(s) over again.
    // First, reset state from the current session:
    if (env != NULL) {
      env->taskScheduler().unscheduleDelayedTask(sessionTimerTask);
      env->taskScheduler().unscheduleDelayedTask(arrivalCheckTimerTask);
      env->taskScheduler().unscheduleDelayedTask(interPacketGapCheckTimerTask);
      env->taskScheduler().unscheduleDelayedTask(qosMeasurementTimerTask);
    }
    totNumPacketsReceived = ~0;

    startPlayingSession(session, initialSeekTime, endTime, scale, continueAfterPLAY);
  }
}

void sessionTimerHandler

(

void *

clientData

)

Definition at line 918 of file playCommon.cpp.

References NULL, sessionAfterPlaying(), and sessionTimerTask.

Referenced by continueAfterPLAY().

                                               {
  sessionTimerTask = NULL;

  sessionAfterPlaying();
}

void setupStreams

(

)

Definition at line 655 of file playCommon.cpp.

References FileSink::addData(), aviOut, MediaSubsession::clientPortNum(), MediaSubsession::codecName(), continueAfterPLAY(), continueAfterSETUP(), FileSink::createNew(), H264VideoFileSink::createNew(), AMRAudioFileSink::createNew(), AVIFileSink::createNew(), QuickTimeFileSink::createNew(), createReceivers, duration, endTime, env, False, fileNamePrefix, fileSinkBufferSize, MediaSubsession::fmtp_config(), MediaSubsession::fmtp_spropparametersets(), generateHintTracks, generateMP4Format, UsageEnvironment::getResultMsg(), initialSeekTime, iter, madeProgress, MediaSubsession::mediumName(), movieFPS, movieHeight, movieWidth, MediaSubsessionIterator::next(), NULL, oneFilePerFrame, outputAVIFile, outputQuickTimeFile, packetLossCompensate, parseGeneralConfigStr(), MediaSession::playEndTime(), qtOut, MediaSubsession::readSource(), MediaSubsession::rtcpInstance(), scale, session, sessionAfterPlaying(), RTCPInstance::setByeHandler(), setupSubsession(), shutdown(), singleMedium, MediaSubsession::sink, MediaSink::startPlaying(), AVIFileSink::startPlaying(), QuickTimeFileSink::startPlaying(), startPlayingSession(), streamUsingTCP, subsession, subsessionAfterPlaying(), subsessionByeHandler(), syncStreams, and True.

Referenced by continueAfterDESCRIBE(), and continueAfterSETUP().

                    {
  static MediaSubsessionIterator* setupIter = NULL;
  if (setupIter == NULL) setupIter = new MediaSubsessionIterator(*session);
  while ((subsession = setupIter->next()) != NULL) {
    // We have another subsession left to set up:
    if (subsession->clientPortNum() == 0) continue; // port # was not set

    setupSubsession(subsession, streamUsingTCP, continueAfterSETUP);
    return;
  }

  // We're done setting up subsessions.
  delete setupIter;
  if (!madeProgress) shutdown();

  // 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;
      MediaSubsessionIterator iter(*session);
      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,
                                                  subsession->fmtp_spropparametersets(),
                                                  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:
  if (duration == 0) {
    if (scale > 0) duration = session->playEndTime() - initialSeekTime; // use SDP end time
    else if (scale < 0) duration = initialSeekTime;
  }
  if (duration < 0) duration = 0.0;

  endTime = initialSeekTime;
  if (scale > 0) {
    if (duration <= 0) endTime = -1.0f;
    else endTime = initialSeekTime + duration;
  } else {
    endTime = initialSeekTime - duration;
    if (endTime < 0) endTime = 0.0f;
  }

  startPlayingSession(session, initialSeekTime, endTime, scale, continueAfterPLAY);
}

void shutdown

(

int

exitCode = 1

)

Definition at line 1133 of file playCommon.cpp.

References areAlreadyShuttingDown, arrivalCheckTimerTask, continueAfterTEARDOWN(), env, interPacketGapCheckTimerTask, NULL, printQOSData(), qosMeasurementIntervalMS, qosMeasurementTimerTask, session, sessionTimerTask, shutdownExitCode, UsageEnvironment::taskScheduler(), tearDownSession(), True, and TaskScheduler::unscheduleDelayedTask().

Referenced by continueAfterDESCRIBE(), continueAfterOPTIONS(), continueAfterPLAY(), main(), sessionAfterPlaying(), setupStreams(), signalHandlerShutdown(), and usage().

                            {
  if (areAlreadyShuttingDown) return; // in case we're called after receiving a RTCP "BYE" while in the middle of a "TEARDOWN".
  areAlreadyShuttingDown = True;

  shutdownExitCode = 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);
  }

  // Teardown, then shutdown, any outstanding RTP/RTCP subsessions
  if (session != NULL) {
    tearDownSession(session, continueAfterTEARDOWN);
  } else {
    continueAfterTEARDOWN(NULL, 0, NULL);
  }
}

void signalHandlerShutdown

(

int

sig

)

Definition at line 1170 of file playCommon.cpp.

References env, and shutdown().

Referenced by main().

                                        {
  *env << "Got shutdown signal\n";
  shutdown(0);
}

void subsessionAfterPlaying

(

void *

clientData

)

Definition at line 868 of file playCommon.cpp.

References Medium::close(), iter, MediaSubsessionIterator::next(), NULL, MediaSubsession::parentSession(), session, sessionAfterPlaying(), MediaSubsession::sink, and subsession.

Referenced by setupStreams(), and subsessionByeHandler().

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

  // Next, check whether *all* subsessions' streams 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' streams have now been closed
  sessionAfterPlaying();
}

void subsessionByeHandler

(

void *

clientData

)

Definition at line 885 of file playCommon.cpp.

References MediaSubsession::codecName(), env, MediaSubsession::mediumName(), NULL, startTime, subsession, and subsessionAfterPlaying().

Referenced by setupStreams().

                                            {
  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 usage

(

)

Definition at line 112 of file playCommon.cpp.

References allowProxyServers, controlConnectionUsesTCP, env, progName, shutdown(), and supportCodecSelection.

             {
  *env << "Usage: " << progName
       << " [-p <startPortNum>] [-r|-q|-4|-i] [-a|-v] [-V] [-d <duration>] [-D <max-inter-packet-gap-time> [-c] [-S <offset>] [-n] [-O]"
           << (controlConnectionUsesTCP ? " [-t|-T <http-port>]" : "")
       << " [-u <username> <password>"
           << (allowProxyServers ? " [<proxy-server> [<proxy-server-port>]]" : "")
       << "]" << (supportCodecSelection ? " [-A <audio-codec-rtp-payload-format-code>|-M <mime-subtype-name>]" : "")
       << " [-s <initial-seek-time>] [-z <scale>]"
       << " [-w <width> -h <height>] [-f <frames-per-second>] [-y] [-H] [-Q [<measurement-interval>]] [-F <filename-prefix>] [-b <file-sink-buffer-size>] [-B <input-socket-buffer-size>] [-I <input-interface-ip-address>] [-m] <url> (or " << progName << " -o [-V] <url>)\n";
  shutdown();
}

---

Variable Documentation

Boolean areAlreadyShuttingDown = False

Definition at line 1131 of file playCommon.cpp.

Referenced by shutdown().

TaskToken arrivalCheckTimerTask = NULL

Definition at line 61 of file playCommon.cpp.

Referenced by checkForPacketArrival(), sessionAfterPlaying(), and shutdown().

Boolean audioOnly = False

Definition at line 70 of file playCommon.cpp.

Referenced by main().

AVIFileSink* aviOut = NULL

Definition at line 69 of file playCommon.cpp.

Referenced by checkForPacketArrival(), closeMediaSinks(), and setupStreams().

Boolean createReceivers = True

Definition at line 64 of file playCommon.cpp.

Referenced by continueAfterDESCRIBE(), continueAfterPLAY(), main(), and setupStreams().

unsigned char desiredAudioRTPPayloadFormat = 0

Definition at line 94 of file playCommon.cpp.

Referenced by createClient(), and main().

unsigned short desiredPortNum = 0

Definition at line 88 of file playCommon.cpp.

Referenced by continueAfterDESCRIBE(), getSDPDescription(), and main().

double duration = 0

Definition at line 74 of file playCommon.cpp.

double durationSlop = -1.0

Definition at line 75 of file playCommon.cpp.

Referenced by continueAfterPLAY(), and main().

double endTime

Definition at line 78 of file playCommon.cpp.

Referenced by continueAfterPLAY(), sessionAfterPlaying(), and setupStreams().

UsageEnvironment* env

Definition at line 55 of file playCommon.cpp.

char const* fileNamePrefix = ""

Definition at line 102 of file playCommon.cpp.

Referenced by main(), and setupStreams().

unsigned fileSinkBufferSize = 100000

Definition at line 103 of file playCommon.cpp.

Referenced by continueAfterDESCRIBE(), main(), and setupStreams().

Boolean generateHintTracks = False

Definition at line 107 of file playCommon.cpp.

Referenced by main(), and setupStreams().

Boolean generateMP4Format = False

Definition at line 66 of file playCommon.cpp.

Referenced by main(), and setupStreams().

double initialSeekTime = 0.0f

Definition at line 76 of file playCommon.cpp.

Referenced by continueAfterPLAY(), main(), sessionAfterPlaying(), and setupStreams().

TaskToken interPacketGapCheckTimerTask = NULL

Definition at line 62 of file playCommon.cpp.

Referenced by checkInterPacketGaps(), sessionAfterPlaying(), and shutdown().

unsigned interPacketGapMaxTime = 0

Definition at line 79 of file playCommon.cpp.

Referenced by checkInterPacketGaps(), and main().

Boolean madeProgress = False

Definition at line 637 of file playCommon.cpp.

Referenced by continueAfterDESCRIBE(), continueAfterSETUP(), and setupStreams().

char* mimeSubtype = NULL

Definition at line 95 of file playCommon.cpp.

Referenced by createClient(), and main().

unsigned movieFPS = 15

Definition at line 100 of file playCommon.cpp.

Referenced by main(), and setupStreams().

Boolean movieFPSOptionSet = False

Definition at line 101 of file playCommon.cpp.

Referenced by main().

unsigned short movieHeight = 180

Definition at line 98 of file playCommon.cpp.

Referenced by main(), and setupStreams().

Boolean movieHeightOptionSet = False

Definition at line 99 of file playCommon.cpp.

Referenced by main().

unsigned short movieWidth = 240

Definition at line 96 of file playCommon.cpp.

Referenced by main(), and setupStreams().

Boolean movieWidthOptionSet = False

Definition at line 97 of file playCommon.cpp.

Referenced by main().

unsigned nextQOSMeasurementUSecs [static]

Definition at line 963 of file playCommon.cpp.

Referenced by beginQOSMeasurement(), and scheduleNextQOSMeasurement().

Boolean notifyOnPacketArrival = False

Definition at line 86 of file playCommon.cpp.

Referenced by checkForPacketArrival(), and main().

Boolean oneFilePerFrame = False

Definition at line 85 of file playCommon.cpp.

Referenced by main(), and setupStreams().

Authenticator* ourAuthenticator = NULL

Definition at line 57 of file playCommon.cpp.

Referenced by continueAfterTEARDOWN(), getOptions(), getSDPDescription(), main(), setupSubsession(), startPlayingSession(), and tearDownSession().

Medium* ourClient = NULL

Definition at line 56 of file playCommon.cpp.

Referenced by continueAfterTEARDOWN(), and main().

Boolean outputAVIFile = False

Definition at line 68 of file playCommon.cpp.

Referenced by main(), and setupStreams().

Boolean outputQuickTimeFile = False

Definition at line 65 of file playCommon.cpp.

Referenced by main(), and setupStreams().

Boolean packetLossCompensate = False

Definition at line 105 of file playCommon.cpp.

Referenced by main(), and setupStreams().

char* password = NULL

Definition at line 91 of file playCommon.cpp.

Referenced by RTSPServer::RTSPClientSession::authenticationOK(), getSDPDescription(), SIPClient::invite(), main(), RTSPClient::openConnection(), UserAuthenticationDatabase::removeUserRecord(), RTSPClient::sendRequest(), and UserAuthenticationDatabase::~UserAuthenticationDatabase().

Boolean playContinuously = False

Definition at line 81 of file playCommon.cpp.

Referenced by main(), and sessionAfterPlaying().

char const* progName

Definition at line 54 of file playCommon.cpp.

Referenced by main(), and usage().

char* proxyServerName = NULL

Definition at line 92 of file playCommon.cpp.

Referenced by getSDPDescription(), and main().

unsigned short proxyServerPortNum = 0

Definition at line 93 of file playCommon.cpp.

Referenced by getSDPDescription(), and main().

unsigned qosMeasurementIntervalMS = 0

Definition at line 108 of file playCommon.cpp.

Referenced by continueAfterPLAY(), main(), printQOSData(), scheduleNextQOSMeasurement(), and shutdown().

TaskToken qosMeasurementTimerTask = NULL

Definition at line 63 of file playCommon.cpp.

Referenced by scheduleNextQOSMeasurement(), sessionAfterPlaying(), and shutdown().

qosMeasurementRecord* qosRecordHead = NULL [static]

Definition at line 959 of file playCommon.cpp.

Referenced by beginQOSMeasurement(), periodicQOSMeasurement(), and printQOSData().

QuickTimeFileSink* qtOut = NULL

Definition at line 67 of file playCommon.cpp.

Referenced by checkForPacketArrival(), closeMediaSinks(), and setupStreams().

float scale = 1.0f

Definition at line 77 of file playCommon.cpp.

Referenced by continueAfterPLAY(), RTSPServer::RTSPClientSession::handleCmd_PLAY(), main(), sessionAfterPlaying(), and setupStreams().

Boolean sendOptionsRequest = True

Definition at line 83 of file playCommon.cpp.

Referenced by main().

Boolean sendOptionsRequestOnly = False

Definition at line 84 of file playCommon.cpp.

Referenced by continueAfterOPTIONS(), and main().

MediaSession* session = NULL

Definition at line 59 of file playCommon.cpp.

Referenced by beginQOSMeasurement(), checkForPacketArrival(), checkInterPacketGaps(), closeMediaSinks(), continueAfterDESCRIBE(), continueAfterPLAY(), continueAfterTEARDOWN(), RTSPServer::RTSPClientSession::handleAlternativeRequestByte(), RTSPServer::RTSPClientSession::handleCmd_DESCRIBE(), RTSPServerSupportingHTTPStreaming::RTSPClientSessionSupportingHTTPStreaming::handleHTTPCmd_StreamingGET(), RTSPClient::handlePLAYResponse(), RTSPServer::RTSPClientSession::incomingRequestHandler(), printQOSData(), RTSPClient::sendGetParameterCommand(), RTSPClient::sendPauseCommand(), RTSPClient::sendPlayCommand(), RTSPClient::sendRecordCommand(), RTSPClient::sendSetParameterCommand(), RTSPClient::sendTeardownCommand(), sessionAfterPlaying(), RTSPClient::sessionURL(), setupStreams(), shutdown(), startPlayingSession(), subsessionAfterPlaying(), and tearDownSession().

TaskToken sessionTimerTask = NULL

Definition at line 60 of file playCommon.cpp.

Referenced by continueAfterPLAY(), sessionAfterPlaying(), sessionTimerHandler(), and shutdown().

int shutdownExitCode

Definition at line 1132 of file playCommon.cpp.

Referenced by continueAfterTEARDOWN(), and shutdown().

int simpleRTPoffsetArg = -1

Definition at line 82 of file playCommon.cpp.

Referenced by continueAfterDESCRIBE(), and main().

char const* singleMedium = NULL

Definition at line 72 of file playCommon.cpp.

Referenced by continueAfterDESCRIBE(), main(), and setupStreams().

unsigned socketInputBufferSize = 0

Definition at line 104 of file playCommon.cpp.

Referenced by continueAfterDESCRIBE(), and main().

struct timeval startTime

Definition at line 110 of file playCommon.cpp.

Referenced by main(), and subsessionByeHandler().

char const* streamURL = NULL

Definition at line 58 of file playCommon.cpp.

Referenced by continueAfterDESCRIBE(), getSDPDescription(), and main().

Boolean streamUsingTCP = False

Definition at line 87 of file playCommon.cpp.

Referenced by main(), RTSPClient::sendRequest(), and setupStreams().

MediaSubsession* subsession

Definition at line 636 of file playCommon.cpp.

Boolean syncStreams = False

Definition at line 106 of file playCommon.cpp.

Referenced by checkForPacketArrival(), main(), and setupStreams().

unsigned totNumPacketsReceived = ~0

Definition at line 80 of file playCommon.cpp.

Referenced by checkInterPacketGaps(), qosMeasurementRecord::periodicQOSMeasurement(), printQOSData(), and sessionAfterPlaying().

portNumBits tunnelOverHTTPPortNum = 0

Definition at line 89 of file playCommon.cpp.

Referenced by createClient(), and main().

char* username = NULL

Definition at line 90 of file playCommon.cpp.

Referenced by RTSPServer::RTSPClientSession::authenticationOK(), getSDPDescription(), SIPClient::invite(), main(), RTSPClient::openConnection(), and RTSPClient::sendRequest().

int verbosityLevel = 1

Definition at line 73 of file playCommon.cpp.

Referenced by main().

Boolean videoOnly = False

Definition at line 71 of file playCommon.cpp.

Referenced by main().

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