/********** 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 an AMR audio file (as defined in RFC 3267) // and streams it using RTP // main program #include "liveMedia.hh" #include "BasicUsageEnvironment.hh" #include "GroupsockHelper.hh" UsageEnvironment* env; char const* inputFileName = "test.amr"; AMRAudioFileSource* audioSource; RTPSink* audioSink; 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: struct in_addr destinationAddress; destinationAddress.s_addr = chooseRandomIPv4SSMAddress(*env); // Note: This is a multicast address. If you wish instead to stream // using unicast, then you should use the "testOnDemandRTSPServer" // test program - not this test program - as a model. const unsigned short rtpPortNum = 16666; const unsigned short rtcpPortNum = rtpPortNum+1; const unsigned char ttl = 255; const Port rtpPort(rtpPortNum); const Port rtcpPort(rtcpPortNum); Groupsock rtpGroupsock(*env, destinationAddress, rtpPort, ttl); rtpGroupsock.multicastSendOnly(); // we're a SSM source Groupsock rtcpGroupsock(*env, destinationAddress, rtcpPort, ttl); rtcpGroupsock.multicastSendOnly(); // we're a SSM source // Create a 'AMR Audio RTP' sink from the RTP 'groupsock': audioSink = AMRAudioRTPSink::createNew(*env, &rtpGroupsock, 96); // Create (and start) a 'RTCP instance' for this RTP sink: const unsigned estimatedSessionBandwidth = 10; // 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* rtcp = RTCPInstance::createNew(*env, &rtcpGroupsock, estimatedSessionBandwidth, CNAME, audioSink, NULL /* we're a server */, True /* we're a SSM source */); // Note: This starts RTCP running automatically // Create and start a RTSP server to serve this stream. RTSPServer* rtspServer = RTSPServer::createNew(*env, 8554); if (rtspServer == NULL) { *env << "Failed to create RTSP server: " << env->getResultMsg() << "\n"; exit(1); } ServerMediaSession* sms = ServerMediaSession::createNew(*env, "testStream", inputFileName, "Session streamed by \"testAMRAudioStreamer\"", True /*SSM*/); sms->addSubsession(PassiveServerMediaSubsession::createNew(*audioSink, rtcp)); rtspServer->addServerMediaSession(sms); char* url = rtspServer->rtspURL(sms); *env << "Play this stream using the URL \"" << url << "\"\n"; delete[] url; // 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(audioSource); // Note that this also closes the input file that this source read from. play(); } void play() { // Open the input file as an 'AMR audio file source': AMRAudioFileSource* audioSource = AMRAudioFileSource::createNew(*env, inputFileName); if (audioSource == NULL) { *env << "Unable to open file \"" << inputFileName << "\" as an AMR audio file source: " << env->getResultMsg() << "\n"; exit(1); } // Finally, start playing: *env << "Beginning to read from file...\n"; audioSink->startPlaying(*audioSource, afterPlaying, audioSink); }