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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) 1996-2005, Live Networks, Inc. All rights reserved
// A test program that reads a MPEG-2 Transport Stream file,
// and streams it using RTP
// main program
#include "liveMedia.hh"
#include "BasicUsageEnvironment.hh"
#include "GroupsockHelper.hh"
// To stream using "source-specific multicast" (SSM), uncomment the following:
//#define USE_SSM 1
#ifdef USE_SSM
Boolean const isSSM = True;
#else
Boolean const isSSM = False;
#endif
// To set up an internal RTSP server, uncomment the following:
//#define IMPLEMENT_RTSP_SERVER 1
// (Note that this RTSP server works for multicast only)
#define TRANSPORT_PACKET_SIZE 188
#define TRANSPORT_PACKETS_PER_NETWORK_PACKET 7
// The product of these two numbers must be enough to fit within a network packet
UsageEnvironment* env;
char const* inputFileName = "test.ts";
FramedSource* videoSource;
RTPSink* videoSink;
void play(); // forward
int main(int argc, char** argv) {
// 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 rtpPortNum = 1234;
const unsigned short rtcpPortNum = rtpPortNum+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 rtpPort(rtpPortNum);
const Port rtcpPort(rtcpPortNum);
Groupsock rtpGroupsock(*env, destinationAddress, rtpPort, ttl);
Groupsock rtcpGroupsock(*env, destinationAddress, rtcpPort, ttl);
#ifdef USE_SSM
rtpGroupsock.multicastSendOnly();
rtcpGroupsock.multicastSendOnly();
#endif
// Create an appropriate 'RTP sink' from the RTP 'groupsock':
videoSink =
SimpleRTPSink::createNew(*env, &rtpGroupsock, 33, 90000, "video", "mp2t",
1, True, False /*no 'M' bit*/);
// Create (and start) a 'RTCP instance' for this RTP sink:
const unsigned estimatedSessionBandwidth = 5000; // 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* rtcp =
#endif
RTCPInstance::createNew(*env, &rtcpGroupsock,
estimatedSessionBandwidth, 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 \"testMPEG2TransportStreamer\"",
isSSM);
sms->addSubsession(PassiveServerMediaSubsession::createNew(*videoSink, rtcp));
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 afterPlaying(void* /*clientData*/) {
*env << "...done reading from file\n";
Medium::close(videoSource);
// Note that this also closes the input file that this source read from.
play();
}
void play() {
unsigned const inputDataChunkSize
= TRANSPORT_PACKETS_PER_NETWORK_PACKET*TRANSPORT_PACKET_SIZE;
// Open the input file as a 'byte-stream file source':
ByteStreamFileSource* fileSource
= ByteStreamFileSource::createNew(*env, inputFileName, inputDataChunkSize);
if (fileSource == NULL) {
*env << "Unable to open file \"" << inputFileName
<< "\" as a byte-stream file source\n";
exit(1);
}
// Create a 'framer' for the input source (to give us proper inter-packet gaps):
videoSource = MPEG2TransportStreamFramer::createNew(*env, fileSource);
// Finally, start playing:
*env << "Beginning to read from file...\n";
videoSink->startPlaying(*videoSource, afterPlaying, videoSink);
}
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