/********** 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 .) 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 // main program #include "liveMedia.hh" #include "BasicUsageEnvironment.hh" #include "GroupsockHelper.hh" UsageEnvironment* env; char const* inputFileName = "test.mpg"; char const* inputFileNamePcm = "test.wav"; WAVAudioFileSource* pcmSource; ByteStreamFileSource* fileVideoSource; FramedSource* audioSource; FramedSource* videoSource; RTPSink* audioSink; RTPSink* videoSink; MPEG1or2VideoStreamFramer* MpegSource; char* mimeType; unsigned char payloadFormatCode; Boolean iFramesOnly = False; void play(); // forward void start(void *); // 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) // To stream *only* MPEG "I" frames (e.g., to reduce network bandwidth), // change the following "False" to "True": int main(int argc, char** argv) { // Begin by setting up our usage environment: TaskScheduler* scheduler = BasicTaskScheduler::createNew(); env = BasicUsageEnvironment::createNew(*scheduler); pcmSource = WAVAudioFileSource::createNew(*env, inputFileNamePcm); if (pcmSource == NULL) { *env << "Unable to open file \"" << inputFileNamePcm << "\" as a pcm audio file source: " << env->getResultMsg() << "\n"; exit(1); } // Get attributes of the audio source: unsigned char const bitsPerSample = pcmSource->bitsPerSample(); if (bitsPerSample != 16) { *env << "The input file contains " << bitsPerSample << " bit-per-sample audio, which we don't handle\n"; exit(1); } unsigned const samplingFrequency = pcmSource->samplingFrequency(); unsigned char const numChannels = pcmSource->numChannels(); unsigned bitsPerSecond = samplingFrequency*bitsPerSample*numChannels; *env << "Audio source parameters:\n\t" << samplingFrequency << " Hz, "; *env << bitsPerSample << " bits-per-sample, "; *env << numChannels << " channels => "; *env << bitsPerSecond << " bits-per-second\n"; mimeType = "L16"; if (samplingFrequency == 44100 && numChannels == 2) { payloadFormatCode = 10; // a static RTP payload type } else if (samplingFrequency == 44100 && numChannels == 1) { payloadFormatCode = 11; // a static RTP payload type } else { payloadFormatCode = 96; // a dynamic RTP payload type } *env << "Converting to network byte order for streaming\n"; if ( samplingFrequency != 48000 || numChannels != 2 ) { *env << "BIG WARNING: The receiver client is hardcoded to PCM 48KHz 2 Channels\n"; exit(1); } // Open the input file as a 'byte-stream file source': fileVideoSource = ByteStreamFileSource::createNew(*env, inputFileName); if (fileVideoSource == NULL) { *env << "Unable to open file mpeg video ES\"" << inputFileName << "\" as a byte-stream file source\n"; exit(1); } // 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 const unsigned char rtpPayloadType = 96; 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); audioSink = SimpleRTPSink::createNew(*env, &rtpGroupsockAudio, payloadFormatCode, samplingFrequency, "audio", mimeType, numChannels); // Create (and start) a 'RTCP instance' for this RTP sink: const unsigned estimatedSessionBandwidthAudio = bitsPerSecond/1000; // 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 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); // Note: This also closes the input file that this source read from. fileVideoSource = ByteStreamFileSource::createNew(*env, inputFileName); pcmSource = WAVAudioFileSource::createNew(*env, inputFileNamePcm); // Start playing once again: play(); } void play() { // We must demultiplex Audio and Video Elementary Streams // from the input source: FramedSource* videoES = fileVideoSource; // Create a framer for each Elementary Stream: MpegSource = MPEG1or2VideoStreamFramer::createNew(*env, videoES, iFramesOnly); audioSource = EndianSwap16::createNew(*env, pcmSource); /*******************************************************************/ /* Finally, start with some delay (3 seconds) */ /* Ouch...This will create a GAP between audio and video !!!!! */ /* Change 3000000 to 0 and no GAP */ /*******************************************************************/ env->taskScheduler().scheduleDelayedTask(3000000, start, NULL); } void start(void *) { *env << "Beginning to read from file...\n"; //MpegSource->flushInput(); // !!!!! Remove comment to solve delay problem audioSink->startPlaying(*audioSource, afterPlaying, audioSink); videoSink->startPlaying(*MpegSource, afterPlaying, videoSink); }