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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) 1996-2000, Live Networks, Inc. All rights reserved
// A test program that streams a MP3 file via RTP/RTCP
// main program
#include "liveMedia.hh"
#include "GroupsockHelper.hh"
#include "BasicUsageEnvironment.hh"
// To stream using 'ADUs' rather than raw MP3 frames, uncomment the following:
//#define STREAM_USING_ADUS 1
// To also reorder ADUs before streaming, uncomment the following:
//#define INTERLEAVE_ADUS 1
// (For more information about ADUs and interleaving,
// see <http://www.live.com/rtp-mp3/>)
// 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)
#ifdef IMPLEMENT_RTSP_SERVER
RTSPServer* rtspServer;
#endif
UsageEnvironment* env;
// A structure to hold the state of the current session.
// It is used in the "afterPlaying()" function to clean up the session.
struct sessionState_t {
FramedSource* source;
RTPSink* sink;
RTCPInstance* rtcpInstance;
Groupsock* rtpGroupsock;
Groupsock* rtcpGroupsock;
} sessionState;
char const* inputFileName = "test.mp3";
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 = 6666;
const unsigned short rtcpPortNum = rtpPortNum+1;
const unsigned char ttl = 1; // 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);
sessionState.rtpGroupsock
= new Groupsock(*env, destinationAddress, rtpPort, ttl);
sessionState.rtcpGroupsock
= new Groupsock(*env, destinationAddress, rtcpPort, ttl);
#ifdef USE_SSM
sessionState.rtpGroupsock->multicastSendOnly();
sessionState.rtcpGroupsock->multicastSendOnly();
#endif
// Create a 'MP3 RTP' sink from the RTP 'groupsock':
#ifdef STREAM_USING_ADUS
unsigned char rtpPayloadFormat = 96; // A dynamic payload format code
sessionState.sink
= MP3ADURTPSink::createNew(*env, sessionState.rtpGroupsock,
rtpPayloadFormat);
#else
sessionState.sink
= MPEG1or2AudioRTPSink::createNew(*env, sessionState.rtpGroupsock);
#endif
// Create (and start) a 'RTCP instance' for this RTP sink:
const unsigned estimatedSessionBandwidth = 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
sessionState.rtcpInstance
= RTCPInstance::createNew(*env, sessionState.rtcpGroupsock,
estimatedSessionBandwidth, CNAME,
sessionState.sink, NULL /* we're a server */,
isSSM);
// Note: This starts RTCP running automatically
#ifdef IMPLEMENT_RTSP_SERVER
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 \"testMP3Streamer\"", isSSM);
sms->addSubsession(PassiveServerMediaSubsession::createNew(*sessionState.sink, sessionState.rtcpInstance));
rtspServer->addServerMediaSession(sms);
char* url = rtspServer->rtspURL(sms);
*env << "Play this stream using the URL \"" << url << "\"\n";
delete[] url;
#endif
play();
env->taskScheduler().doEventLoop(); // does not return
return 0; // only to prevent compiler warning
}
void afterPlaying(void* clientData); // forward
void play() {
// Open the file as a 'MP3 file source':
sessionState.source = MP3FileSource::createNew(*env, inputFileName);
if (sessionState.source == NULL) {
*env << "Unable to open file \"" << inputFileName
<< "\" as a MP3 file source\n";
exit(1);
}
#ifdef STREAM_USING_ADUS
// Add a filter that converts the source MP3s to ADUs:
sessionState.source
= ADUFromMP3Source::createNew(*env, sessionState.source);
if (sessionState.source == NULL) {
*env << "Unable to create a MP3->ADU filter for the source\n";
exit(1);
}
#ifdef INTERLEAVE_ADUS
// Add another filter that interleaves the ADUs before packetizing them:
unsigned char interleaveCycle[] = {0,2,1,3}; // or choose your own order...
unsigned const interleaveCycleSize
= (sizeof interleaveCycle)/(sizeof (unsigned char));
Interleaving interleaving(interleaveCycleSize, interleaveCycle);
sessionState.source
= MP3ADUinterleaver::createNew(*env, interleaving, sessionState.source);
if (sessionState.source == NULL) {
*env << "Unable to create an ADU interleaving filter for the source\n";
exit(1);
}
#endif
#endif
// Finally, start the streaming:
*env << "Beginning streaming...\n";
sessionState.sink->startPlaying(*sessionState.source, afterPlaying, NULL);
}
void afterPlaying(void* /*clientData*/) {
*env << "...done streaming\n";
// End this loop by closing the current source:
Medium::close(sessionState.source);
// And start another loop:
play();
}
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