testProgs/vobStreamer.cpp File Reference

#include "liveMedia.hh"
#include "AC3AudioStreamFramer.hh"
#include "BasicUsageEnvironment.hh"
#include "GroupsockHelper.hh"

Include dependency graph for vobStreamer.cpp:

Go to the source code of this file.

Functions
void	usage ()
void	play ()
int	main (int argc, char const **argv)
void	afterPlaying (void *clientData)
Variables
char const *	programName
Boolean	iFramesOnly = False
unsigned const	VOB_AUDIO = 1<<0
unsigned const	VOB_VIDEO = 1<<1
unsigned	mediaToStream = VOB_AUDIO|VOB_VIDEO
char const **	inputFileNames
char const **	curInputFileName
Boolean	haveReadOneFile = False
UsageEnvironment *	env
MPEG1or2Demux *	mpegDemux
AC3AudioStreamFramer *	audioSource = NULL
FramedSource *	videoSource = NULL
RTPSink *	audioSink = NULL
RTCPInstance *	audioRTCP = NULL
RTPSink *	videoSink = NULL
RTCPInstance *	videoRTCP = NULL
RTSPServer *	rtspServer = NULL
unsigned short const	defaultRTSPServerPortNum = 554
unsigned short	rtspServerPortNum = defaultRTSPServerPortNum
Groupsock *	rtpGroupsockAudio
Groupsock *	rtcpGroupsockAudio
Groupsock *	rtpGroupsockVideo
Groupsock *	rtcpGroupsockVideo

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Function Documentation

void afterPlaying

(

void *

clientData

)

Definition at line 224 of file vobStreamer.cpp.

References audioSink, audioSource, Medium::close(), curInputFileName, env, FramedSource::isCurrentlyAwaitingData(), mpegDemux, NULL, 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 != NULL && audioSource->isCurrentlyAwaitingData()) ||
      (videoSource != NULL && videoSource->isCurrentlyAwaitingData())) {
    return;
  }

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

  if (audioSink != NULL) audioSink->stopPlaying();
  if (videoSink != NULL) 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.

  // Move to the next file name (if any):
  ++curInputFileName;

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

int main	(	int	argc,
		char const **	argv
	)

Definition at line 66 of file vobStreamer.cpp.

References RTSPServer::addServerMediaSession(), ServerMediaSession::addSubsession(), audioRTCP, audioSink, chooseRandomIPv4SSMAddress(), PassiveServerMediaSubsession::createNew(), ServerMediaSession::createNew(), RTSPServer::createNew(), MPEG1or2VideoRTPSink::createNew(), RTCPInstance::createNew(), AC3AudioRTPSink::createNew(), BasicUsageEnvironment::createNew(), BasicTaskScheduler::createNew(), curInputFileName, TaskScheduler::doEventLoop(), env, exit, UsageEnvironment::getResultMsg(), iFramesOnly, inputFileNames, mediaToStream, Groupsock::multicastSendOnly(), NULL, play(), programName, rtcpGroupsockAudio, rtcpGroupsockVideo, rtpGroupsockAudio, rtpGroupsockVideo, rtspServer, rtspServerPortNum, RTSPServer::rtspURL(), UsageEnvironment::taskScheduler(), True, usage(), videoRTCP, videoSink, VOB_AUDIO, and VOB_VIDEO.

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

  // Parse command-line options:
  // (Unfortunately we can't use getopt() here; Windoze doesn't have it)
  programName = argv[0];
  while (argc > 2) {
    char const* const opt = argv[1];
    if (opt[0] != '-') break;
    switch (opt[1]) {

    case 'i': { // transmit video I-frames only
      iFramesOnly = True;
      break;
    }

    case 'a': { // transmit audio, but not video
      mediaToStream &=~ VOB_VIDEO;
      break;
    }

    case 'v': { // transmit video, but not audio
      mediaToStream &=~ VOB_AUDIO;
      break;
    }

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

    default: {
      usage();
      break;
    }
    }

    ++argv; --argc;
  }
  if (argc < 2) usage();
  if (mediaToStream == 0) {
    *env << "The -a and -v flags cannot both be used!\n";
    usage();
  }
  if (iFramesOnly && (mediaToStream&VOB_VIDEO) == 0) {
    *env << "Warning: Because we're not streaming video, the -i flag has no effect.\n";
  }

  inputFileNames = &argv[1];
  curInputFileName = inputFileNames;

  // Create 'groupsocks' for RTP and RTCP:
  struct in_addr destinationAddress;
  destinationAddress.s_addr = chooseRandomIPv4SSMAddress(*env);

  const unsigned short rtpPortNumAudio = 4444;
  const unsigned short rtcpPortNumAudio = rtpPortNumAudio+1;
  const unsigned short rtpPortNumVideo = 8888;
  const unsigned short rtcpPortNumVideo = rtpPortNumVideo+1;
  const unsigned char ttl = 255;

  const Port rtpPortAudio(rtpPortNumAudio);
  const Port rtcpPortAudio(rtcpPortNumAudio);
  const Port rtpPortVideo(rtpPortNumVideo);
  const Port rtcpPortVideo(rtcpPortNumVideo);

  const unsigned maxCNAMElen = 100;
  unsigned char CNAME[maxCNAMElen+1];
  gethostname((char*)CNAME, maxCNAMElen);
  CNAME[maxCNAMElen] = '\0'; // just in case

  if (mediaToStream&VOB_AUDIO) {
    rtpGroupsockAudio
      = new Groupsock(*env, destinationAddress, rtpPortAudio, ttl);
    rtpGroupsockAudio->multicastSendOnly(); // because we're a SSM source

    // Create an 'AC3 Audio RTP' sink from the RTP 'groupsock':
    audioSink
      = AC3AudioRTPSink::createNew(*env, rtpGroupsockAudio, 96, 0);
    // set the RTP timestamp frequency 'for real' later

    // Create (and start) a 'RTCP instance' for this RTP sink:
    rtcpGroupsockAudio
      = new Groupsock(*env, destinationAddress, rtcpPortAudio, ttl);
    rtcpGroupsockAudio->multicastSendOnly(); // because we're a SSM source
    const unsigned estimatedSessionBandwidthAudio
      = 160; // in kbps; for RTCP b/w share
    audioRTCP = RTCPInstance::createNew(*env, rtcpGroupsockAudio,
                                        estimatedSessionBandwidthAudio, CNAME,
                                        audioSink, NULL /* we're a server */,
                                        True /* we're a SSM source */);
    // Note: This starts RTCP running automatically
  }

  if (mediaToStream&VOB_VIDEO) {
    rtpGroupsockVideo
      = new Groupsock(*env, destinationAddress, rtpPortVideo, ttl);
    rtpGroupsockVideo->multicastSendOnly(); // because we're a SSM source

    // 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:
    rtcpGroupsockVideo
      = new Groupsock(*env, destinationAddress, rtcpPortVideo, ttl);
    rtcpGroupsockVideo->multicastSendOnly(); // because we're a SSM source
    const unsigned estimatedSessionBandwidthVideo
      = 4500; // in kbps; for RTCP b/w share
    videoRTCP = RTCPInstance::createNew(*env, rtcpGroupsockVideo,
                                        estimatedSessionBandwidthVideo, CNAME,
                                        videoSink, NULL /* we're a server */,
                                        True /* we're a SSM source */);
    // Note: This starts RTCP running automatically
  }

  if (rtspServer == NULL) {
    rtspServer = RTSPServer::createNew(*env, rtspServerPortNum);
    if (rtspServer == NULL) {
      *env << "Failed to create RTSP server: " << env->getResultMsg() << "\n";
      *env << "To change the RTSP server's port number, use the \"-p <port number>\" option.\n";
      exit(1);
    }
    ServerMediaSession* sms
      = ServerMediaSession::createNew(*env, "vobStream", *curInputFileName,
             "Session streamed by \"vobStreamer\"", True /*SSM*/);
    if (audioSink != NULL) sms->addSubsession(PassiveServerMediaSubsession::createNew(*audioSink, audioRTCP));
    if (videoSink != NULL) sms->addSubsession(PassiveServerMediaSubsession::createNew(*videoSink, videoRTCP));
    rtspServer->addServerMediaSession(sms);

    *env << "Created RTSP server.\n";

    // Display our "rtsp://" URL, for clients to connect to:
    char* url = rtspServer->rtspURL(sms);
    *env << "Access this stream using the URL:\n\t" << url << "\n";
    delete[] url;
  }

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

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

  return 0; // only to prevent compiler warning
}

void play

(

)

void usage

(

)

Definition at line 57 of file vobStreamer.cpp.

References env, exit, and programName.

             {
  *env << "usage: " << programName << " [-i] [-a|-v] "
          "[-p <RTSP-server-port-number>] "
          "<VOB-file>...<VOB-file>\n";
  exit(1);
}

---

Variable Documentation

RTCPInstance* audioRTCP = NULL

Definition at line 45 of file vobStreamer.cpp.

Referenced by main().

RTPSink* audioSink = NULL

Definition at line 44 of file vobStreamer.cpp.

AC3AudioStreamFramer* audioSource = NULL

Definition at line 42 of file vobStreamer.cpp.

char const** curInputFileName

Definition at line 37 of file vobStreamer.cpp.

Referenced by afterPlaying(), and main().

unsigned short const defaultRTSPServerPortNum = 554

Definition at line 49 of file vobStreamer.cpp.

UsageEnvironment* env

Definition at line 40 of file vobStreamer.cpp.

Boolean haveReadOneFile = False

Definition at line 38 of file vobStreamer.cpp.

Boolean iFramesOnly = False

Definition at line 30 of file vobStreamer.cpp.

char const** inputFileNames

Definition at line 36 of file vobStreamer.cpp.

Referenced by main().

unsigned mediaToStream = VOB_AUDIO|VOB_VIDEO

Definition at line 34 of file vobStreamer.cpp.

Referenced by main().

MPEG1or2Demux* mpegDemux

Definition at line 41 of file vobStreamer.cpp.

char const* programName

Definition at line 27 of file vobStreamer.cpp.

Groupsock* rtcpGroupsockAudio

Definition at line 53 of file vobStreamer.cpp.

Referenced by main().

Groupsock* rtcpGroupsockVideo

Definition at line 55 of file vobStreamer.cpp.

Referenced by main().

Groupsock* rtpGroupsockAudio

Definition at line 52 of file vobStreamer.cpp.

Referenced by main().

Groupsock* rtpGroupsockVideo

Definition at line 54 of file vobStreamer.cpp.

Referenced by main().

RTSPServer* rtspServer = NULL

Definition at line 48 of file vobStreamer.cpp.

Referenced by announceStream(), and main().

unsigned short rtspServerPortNum = defaultRTSPServerPortNum

Definition at line 50 of file vobStreamer.cpp.

Referenced by main().

RTCPInstance* videoRTCP = NULL

Definition at line 47 of file vobStreamer.cpp.

Referenced by main().

RTPSink* videoSink = NULL

Definition at line 46 of file vobStreamer.cpp.

FramedSource* videoSource = NULL

Definition at line 43 of file vobStreamer.cpp.

unsigned const VOB_AUDIO = 1<<0

Definition at line 32 of file vobStreamer.cpp.

Referenced by main().

unsigned const VOB_VIDEO = 1<<1

Definition at line 33 of file vobStreamer.cpp.

Referenced by main().

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