testProgs/testAMRAudioStreamer.cpp

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/**********
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This library is distributed in the hope that it will be useful, but WITHOUT
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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
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**********/
// Copyright (c) 1996-2013, 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";

  audioSink->stopPlaying();
  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);
}

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