testProgs/testMPEG1or2AudioVideoStreamer.cpp File Reference

#include "liveMedia.hh"
#include "BasicUsageEnvironment.hh"
#include "GroupsockHelper.hh"

Include dependency graph for testMPEG1or2AudioVideoStreamer.cpp:

Go to the source code of this file.

Functions
void	play ()
int	main (int argc, char **argv)
void	afterPlaying (void *clientData)
Variables
UsageEnvironment *	env
char const *	inputFileName = "test.mpg"
MPEG1or2Demux *	mpegDemux
FramedSource *	audioSource
FramedSource *	videoSource
RTPSink *	audioSink
RTPSink *	videoSink
Boolean const	isSSM = False
Boolean	iFramesOnly = False

---

Function Documentation

void afterPlaying

(

void *

clientData

)

Definition at line 153 of file testMPEG1or2AudioVideoStreamer.cpp.

References audioSink, audioSource, Medium::close(), env, FramedSource::isCurrentlyAwaitingData(), mpegDemux, play(), MediaSink::stopPlaying(), videoSink, and videoSource.

                                    {
  // One of the sinks has ended playing.
  // Check whether any of the sources have a pending read.  If so,
  // wait until its sink ends playing also:
  if (audioSource->isCurrentlyAwaitingData()
      || videoSource->isCurrentlyAwaitingData()) return;

  // Now that both sinks have ended, close both input sources,
  // and start playing again:
  *env << "...done reading from file\n";

  audioSink->stopPlaying();
  videoSink->stopPlaying();
      // ensures that both are shut down
  Medium::close(audioSource);
  Medium::close(videoSource);
  Medium::close(mpegDemux);
  // Note: This also closes the input file that this source read from.

  // Start playing once again:
  play();
}

int main	(	int	argc,
		char **	argv
	)

Definition at line 52 of file testMPEG1or2AudioVideoStreamer.cpp.

References RTSPServer::addServerMediaSession(), ServerMediaSession::addSubsession(), audioRTCP, audioSink, PassiveServerMediaSubsession::createNew(), ServerMediaSession::createNew(), RTSPServer::createNew(), MPEG1or2VideoRTPSink::createNew(), RTCPInstance::createNew(), MPEG1or2AudioRTPSink::createNew(), BasicUsageEnvironment::createNew(), BasicTaskScheduler::createNew(), TaskScheduler::doEventLoop(), env, UsageEnvironment::getResultMsg(), inputFileName, isSSM, Groupsock::multicastSendOnly(), NULL, our_inet_addr(), play(), rtcpGroupsockAudio, rtcpGroupsockVideo, rtpGroupsockAudio, rtpGroupsockVideo, rtspServer, RTSPServer::rtspURL(), UsageEnvironment::taskScheduler(), videoRTCP, and videoSink.

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

  // Create 'groupsocks' for RTP and RTCP:
  char* destinationAddressStr
#ifdef USE_SSM
    = "232.255.42.42";
#else
    = "239.255.42.42";
  // Note: This is a multicast address.  If you wish to stream using
  // unicast instead, then replace this string with the unicast address
  // of the (single) destination.  (You may also need to make a similar
  // change to the receiver program.)
#endif
  const unsigned short rtpPortNumAudio = 6666;
  const unsigned short rtcpPortNumAudio = rtpPortNumAudio+1;
  const unsigned short rtpPortNumVideo = 8888;
  const unsigned short rtcpPortNumVideo = rtpPortNumVideo+1;
  const unsigned char ttl = 7; // low, in case routers don't admin scope

  struct in_addr destinationAddress;
  destinationAddress.s_addr = our_inet_addr(destinationAddressStr);
  const Port rtpPortAudio(rtpPortNumAudio);
  const Port rtcpPortAudio(rtcpPortNumAudio);
  const Port rtpPortVideo(rtpPortNumVideo);
  const Port rtcpPortVideo(rtcpPortNumVideo);

  Groupsock rtpGroupsockAudio(*env, destinationAddress, rtpPortAudio, ttl);
  Groupsock rtcpGroupsockAudio(*env, destinationAddress, rtcpPortAudio, ttl);
  Groupsock rtpGroupsockVideo(*env, destinationAddress, rtpPortVideo, ttl);
  Groupsock rtcpGroupsockVideo(*env, destinationAddress, rtcpPortVideo, ttl);
#ifdef USE_SSM
  rtpGroupsockAudio.multicastSendOnly();
  rtcpGroupsockAudio.multicastSendOnly();
  rtpGroupsockVideo.multicastSendOnly();
  rtcpGroupsockVideo.multicastSendOnly();
#endif

  // Create a 'MPEG Audio RTP' sink from the RTP 'groupsock':
  audioSink = MPEG1or2AudioRTPSink::createNew(*env, &rtpGroupsockAudio);

  // Create (and start) a 'RTCP instance' for this RTP sink:
  const unsigned estimatedSessionBandwidthAudio = 160; // in kbps; for RTCP b/w share
  const unsigned maxCNAMElen = 100;
  unsigned char CNAME[maxCNAMElen+1];
  gethostname((char*)CNAME, maxCNAMElen);
  CNAME[maxCNAMElen] = '\0'; // just in case
#ifdef IMPLEMENT_RTSP_SERVER
  RTCPInstance* audioRTCP =
#endif
    RTCPInstance::createNew(*env, &rtcpGroupsockAudio,
                            estimatedSessionBandwidthAudio, CNAME,
                            audioSink, NULL /* we're a server */, isSSM);
  // Note: This starts RTCP running automatically

  // Create a 'MPEG Video RTP' sink from the RTP 'groupsock':
  videoSink = MPEG1or2VideoRTPSink::createNew(*env, &rtpGroupsockVideo);

  // Create (and start) a 'RTCP instance' for this RTP sink:
  const unsigned estimatedSessionBandwidthVideo = 4500; // in kbps; for RTCP b/w share
#ifdef IMPLEMENT_RTSP_SERVER
  RTCPInstance* videoRTCP =
#endif
    RTCPInstance::createNew(*env, &rtcpGroupsockVideo,
                              estimatedSessionBandwidthVideo, CNAME,
                              videoSink, NULL /* we're a server */, isSSM);
  // Note: This starts RTCP running automatically

#ifdef IMPLEMENT_RTSP_SERVER
  RTSPServer* rtspServer = RTSPServer::createNew(*env);
  // Note that this (attempts to) start a server on the default RTSP server
  // port: 554.  To use a different port number, add it as an extra
  // (optional) parameter to the "RTSPServer::createNew()" call above.
  if (rtspServer == NULL) {
    *env << "Failed to create RTSP server: " << env->getResultMsg() << "\n";
    exit(1);
  }
  ServerMediaSession* sms
    = ServerMediaSession::createNew(*env, "testStream", inputFileName,
                   "Session streamed by \"testMPEG1or2AudioVideoStreamer\"",
                                           isSSM);
  sms->addSubsession(PassiveServerMediaSubsession::createNew(*audioSink, audioRTCP));
  sms->addSubsession(PassiveServerMediaSubsession::createNew(*videoSink, videoRTCP));
  rtspServer->addServerMediaSession(sms);

  char* url = rtspServer->rtspURL(sms);
  *env << "Play this stream using the URL \"" << url << "\"\n";
  delete[] url;
#endif

  // Finally, start the streaming:
  *env << "Beginning streaming...\n";
  play();

  env->taskScheduler().doEventLoop(); // does not return

  return 0; // only to prevent compiler warning
}

void play

(

)

---

Variable Documentation

RTPSink* audioSink

Definition at line 31 of file testMPEG1or2AudioVideoStreamer.cpp.

FramedSource* audioSource

Definition at line 29 of file testMPEG1or2AudioVideoStreamer.cpp.

UsageEnvironment* env

Definition at line 26 of file testMPEG1or2AudioVideoStreamer.cpp.

Boolean iFramesOnly = False

Definition at line 50 of file testMPEG1or2AudioVideoStreamer.cpp.

Referenced by main().

char const* inputFileName = "test.mpg"

Definition at line 27 of file testMPEG1or2AudioVideoStreamer.cpp.

Boolean const isSSM = False

Definition at line 41 of file testMPEG1or2AudioVideoStreamer.cpp.

MPEG1or2Demux* mpegDemux

Definition at line 28 of file testMPEG1or2AudioVideoStreamer.cpp.

Referenced by afterPlaying().

RTPSink* videoSink

Definition at line 32 of file testMPEG1or2AudioVideoStreamer.cpp.

Referenced by afterPlaying(), and main().

FramedSource* videoSource

Definition at line 30 of file testMPEG1or2AudioVideoStreamer.cpp.

Referenced by afterPlaying().

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