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/**********
This library is free software; you can redistribute it and/or modify it under
the terms of the GNU Lesser General Public License as published by the
Free Software Foundation; either version 2.1 of the License, or (at your
option) any later version. (See <http://www.gnu.org/copyleft/lesser.html>.)
This library is distributed in the hope that it will be useful, but WITHOUT
ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS
FOR A PARTICULAR PURPOSE. See the GNU Lesser General Public License for
more details.
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
**********/
// Copyright (c) 1996-2007, Live Networks, Inc. All rights reserved
// A test program that reads a MPEG-1 or 2 Program Stream file,
// splits it into Audio and Video Elementary Streams,
// and streams both using RTP, through a remote Darwin Streaming Server.
// main program
////////// NOTE //////////
// This demo software is provided only as a courtesy to those developers who - for whatever reason - wish
// to send outgoing streams through a separate Darwin Streaming Server. However, it is not necessary to use
// a Darwin Streaming Server in order to serve streams using RTP/RTSP. Instead, the "LIVE555 Streaming Media"
// includes its own RTSP/RTP server implementation, which you should use instead. For tips on using our
// RTSP/RTP server implementation, see the "testOnDemandRTSPServer" demo application, and/or the
// "live555MediaServer" application (in the "mediaServer") directory.
//////////////////////////
#include "liveMedia.hh"
#include "BasicUsageEnvironment.hh"
UsageEnvironment* env;
char const* inputFileName = "test.mpg";
char const* remoteStreamName = "test.sdp"; // the stream name, as served by the DSS
MPEG1or2Demux* mpegDemux;
FramedSource* audioSource;
FramedSource* videoSource;
RTPSink* audioSink;
RTPSink* videoSink;
char const* programName;
// To stream *only* MPEG "I" frames (e.g., to reduce network bandwidth),
// change the following "False" to "True":
Boolean iFramesOnly = False;
void usage() {
*env << "usage: " << programName
<< " <Darwin Streaming Server name or IP address>\n";
exit(1);
}
void play(); // forward
int main(int argc, char** argv) {
// Begin by setting up our usage environment:
TaskScheduler* scheduler = BasicTaskScheduler::createNew();
env = BasicUsageEnvironment::createNew(*scheduler);
// Parse command-line arguments:
programName = argv[0];
if (argc != 2) usage();
char const* dssNameOrAddress = argv[1];
// Create a 'Darwin injector' object:
DarwinInjector* injector = DarwinInjector::createNew(*env, programName);
////////// AUDIO //////////
// Create 'groupsocks' for RTP and RTCP.
// (Note: Because we will actually be streaming through a remote Darwin server,
// via TCP, we just use dummy destination addresses, port numbers, and TTLs here.)
struct in_addr dummyDestAddress;
dummyDestAddress.s_addr = 0;
Groupsock rtpGroupsockAudio(*env, dummyDestAddress, 0, 0);
Groupsock rtcpGroupsockAudio(*env, dummyDestAddress, 0, 0);
// 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
RTCPInstance* audioRTCP =
RTCPInstance::createNew(*env, &rtcpGroupsockAudio,
estimatedSessionBandwidthAudio, CNAME,
audioSink, NULL /* we're a server */);
// Note: This starts RTCP running automatically
// Add these to our 'Darwin injector':
injector->addStream(audioSink, audioRTCP);
////////// END AUDIO //////////
////////// VIDEO //////////
// Create 'groupsocks' for RTP and RTCP.
// (Note: Because we will actually be streaming through a remote Darwin server,
// via TCP, we just use dummy destination addresses, port numbers, and TTLs here.)
Groupsock rtpGroupsockVideo(*env, dummyDestAddress, 0, 0);
Groupsock rtcpGroupsockVideo(*env, dummyDestAddress, 0, 0);
// 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
RTCPInstance* videoRTCP =
RTCPInstance::createNew(*env, &rtcpGroupsockVideo,
estimatedSessionBandwidthVideo, CNAME,
videoSink, NULL /* we're a server */);
// Note: This starts RTCP running automatically
// Add these to our 'Darwin injector':
injector->addStream(videoSink, videoRTCP);
////////// END VIDEO //////////
// Next, specify the destination Darwin Streaming Server:
if (!injector->setDestination(dssNameOrAddress, remoteStreamName,
programName, "LIVE555 Streaming Media")) {
*env << "injector->setDestination() failed: "
<< env->getResultMsg() << "\n";
exit(1);
}
*env << "Play this stream (from the Darwin Streaming Server) using the URL:\n"
<< "\trtsp://" << dssNameOrAddress << "/" << remoteStreamName << "\n";
// Finally, start the streaming:
*env << "Beginning streaming...\n";
play();
env->taskScheduler().doEventLoop(); // does not return
return 0; // only to prevent compiler warning
}
void afterPlaying(void* clientData) {
// 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();
}
void play() {
// Open the input file as a 'byte-stream file source':
ByteStreamFileSource* fileSource
= ByteStreamFileSource::createNew(*env, inputFileName);
if (fileSource == NULL) {
*env << "Unable to open file \"" << inputFileName
<< "\" as a byte-stream file source\n";
exit(1);
}
// We must demultiplex Audio and Video Elementary Streams
// from the input source:
mpegDemux = MPEG1or2Demux::createNew(*env, fileSource);
FramedSource* audioES = mpegDemux->newAudioStream();
FramedSource* videoES = mpegDemux->newVideoStream();
// Create a framer for each Elementary Stream:
audioSource
= MPEG1or2AudioStreamFramer::createNew(*env, audioES);
videoSource
= MPEG1or2VideoStreamFramer::createNew(*env, videoES, iFramesOnly);
// Finally, start playing each sink.
*env << "Beginning to read from file...\n";
videoSink->startPlaying(*videoSource, afterPlaying, videoSink);
audioSink->startPlaying(*audioSource, afterPlaying, audioSink);
}
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