testProgs/testAMRAudioStreamer.cpp

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00001 /**********
00002 This library is free software; you can redistribute it and/or modify it under
00003 the terms of the GNU Lesser General Public License as published by the
00004 Free Software Foundation; either version 2.1 of the License, or (at your
00005 option) any later version. (See <http://www.gnu.org/copyleft/lesser.html>.)
00006 
00007 This library is distributed in the hope that it will be useful, but WITHOUT
00008 ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS
00009 FOR A PARTICULAR PURPOSE.  See the GNU Lesser General Public License for
00010 more details.
00011 
00012 You should have received a copy of the GNU Lesser General Public License
00013 along with this library; if not, write to the Free Software Foundation, Inc.,
00014 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301  USA
00015 **********/
00016 // Copyright (c) 1996-2012, Live Networks, Inc.  All rights reserved
00017 // A test program that reads an AMR audio file (as defined in RFC 3267)
00018 // and streams it using RTP
00019 // main program
00020 
00021 #include "liveMedia.hh"
00022 #include "BasicUsageEnvironment.hh"
00023 #include "GroupsockHelper.hh"
00024 
00025 UsageEnvironment* env;
00026 char const* inputFileName = "test.amr";
00027 AMRAudioFileSource* audioSource;
00028 RTPSink* audioSink;
00029 
00030 void play(); // forward
00031 
00032 int main(int argc, char** argv) {
00033   // Begin by setting up our usage environment:
00034   TaskScheduler* scheduler = BasicTaskScheduler::createNew();
00035   env = BasicUsageEnvironment::createNew(*scheduler);
00036 
00037   // Create 'groupsocks' for RTP and RTCP:
00038   struct in_addr destinationAddress;
00039   destinationAddress.s_addr = chooseRandomIPv4SSMAddress(*env);
00040   // Note: This is a multicast address.  If you wish instead to stream
00041   // using unicast, then you should use the "testOnDemandRTSPServer"
00042   // test program - not this test program - as a model.
00043 
00044   const unsigned short rtpPortNum = 16666;
00045   const unsigned short rtcpPortNum = rtpPortNum+1;
00046   const unsigned char ttl = 255;
00047 
00048   const Port rtpPort(rtpPortNum);
00049   const Port rtcpPort(rtcpPortNum);
00050 
00051   Groupsock rtpGroupsock(*env, destinationAddress, rtpPort, ttl);
00052   rtpGroupsock.multicastSendOnly(); // we're a SSM source
00053   Groupsock rtcpGroupsock(*env, destinationAddress, rtcpPort, ttl);
00054   rtcpGroupsock.multicastSendOnly(); // we're a SSM source
00055 
00056   // Create a 'AMR Audio RTP' sink from the RTP 'groupsock':
00057   audioSink = AMRAudioRTPSink::createNew(*env, &rtpGroupsock, 96);
00058 
00059   // Create (and start) a 'RTCP instance' for this RTP sink:
00060   const unsigned estimatedSessionBandwidth = 10; // in kbps; for RTCP b/w share
00061   const unsigned maxCNAMElen = 100;
00062   unsigned char CNAME[maxCNAMElen+1];
00063   gethostname((char*)CNAME, maxCNAMElen);
00064   CNAME[maxCNAMElen] = '\0'; // just in case
00065   RTCPInstance* rtcp
00066     = RTCPInstance::createNew(*env, &rtcpGroupsock,
00067                               estimatedSessionBandwidth, CNAME,
00068                               audioSink, NULL /* we're a server */,
00069                               True /* we're a SSM source */);
00070   // Note: This starts RTCP running automatically
00071 
00072   // Create and start a RTSP server to serve this stream.
00073   RTSPServer* rtspServer = RTSPServer::createNew(*env, 8554);
00074   if (rtspServer == NULL) {
00075     *env << "Failed to create RTSP server: " << env->getResultMsg() << "\n";
00076     exit(1);
00077   }
00078   ServerMediaSession* sms
00079     = ServerMediaSession::createNew(*env, "testStream", inputFileName,
00080                    "Session streamed by \"testAMRAudioStreamer\"",
00081                                            True /*SSM*/);
00082   sms->addSubsession(PassiveServerMediaSubsession::createNew(*audioSink, rtcp));
00083   rtspServer->addServerMediaSession(sms);
00084 
00085   char* url = rtspServer->rtspURL(sms);
00086   *env << "Play this stream using the URL \"" << url << "\"\n";
00087   delete[] url;
00088 
00089   // Start the streaming:
00090   *env << "Beginning streaming...\n";
00091   play();
00092 
00093   env->taskScheduler().doEventLoop(); // does not return
00094 
00095   return 0; // only to prevent compiler warning
00096 }
00097 
00098 void afterPlaying(void* /*clientData*/) {
00099   *env << "...done reading from file\n";
00100 
00101   audioSink->stopPlaying();
00102   Medium::close(audioSource);
00103   // Note that this also closes the input file that this source read from.
00104 
00105   play();
00106 }
00107 
00108 void play() {
00109   // Open the input file as an 'AMR audio file source':
00110   AMRAudioFileSource* audioSource
00111     = AMRAudioFileSource::createNew(*env, inputFileName);
00112   if (audioSource == NULL) {
00113     *env << "Unable to open file \"" << inputFileName
00114          << "\" as an AMR audio file source: "
00115          << env->getResultMsg() << "\n";
00116     exit(1);
00117   }
00118 
00119   // Finally, start playing:
00120   *env << "Beginning to read from file...\n";
00121   audioSink->startPlaying(*audioSource, afterPlaying, audioSink);
00122 }

Generated on Thu May 17 07:11:47 2012 for live by  doxygen 1.5.2