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123 lines
4.3 KiB
C++
123 lines
4.3 KiB
C++
/**********
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This library is free software; you can redistribute it and/or modify it under
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the terms of the GNU Lesser General Public License as published by the
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Free Software Foundation; either version 3 of the License, or (at your
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option) any later version. (See <http://www.gnu.org/copyleft/lesser.html>.)
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This library is distributed in the hope that it will be useful, but WITHOUT
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ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS
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FOR A PARTICULAR PURPOSE. See the GNU Lesser General Public License for
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more details.
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You should have received a copy of the GNU Lesser General Public License
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along with this library; if not, write to the Free Software Foundation, Inc.,
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51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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**********/
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// Copyright (c) 1996-2025, Live Networks, Inc. All rights reserved
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// A test program that reads an AMR audio file (as defined in RFC 3267)
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// and streams it using RTP
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// main program
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#include "liveMedia.hh"
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#include "BasicUsageEnvironment.hh"
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#include "announceURL.hh"
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#include "GroupsockHelper.hh"
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UsageEnvironment* env;
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char const* inputFileName = "test.amr";
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AMRAudioFileSource* audioSource;
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RTPSink* audioSink;
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void play(); // forward
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int main(int argc, char** argv) {
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// Begin by setting up our usage environment:
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TaskScheduler* scheduler = BasicTaskScheduler::createNew();
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env = BasicUsageEnvironment::createNew(*scheduler);
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// Create 'groupsocks' for RTP and RTCP:
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struct sockaddr_storage destinationAddress;
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destinationAddress.ss_family = AF_INET;
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((struct sockaddr_in&)destinationAddress).sin_addr.s_addr = chooseRandomIPv4SSMAddress(*env);
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// Note: This is a multicast address. If you wish instead to stream
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// using unicast, then you should use the "testOnDemandRTSPServer"
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// test program - not this test program - as a model.
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const unsigned short rtpPortNum = 16666;
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const unsigned short rtcpPortNum = rtpPortNum+1;
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const unsigned char ttl = 255;
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const Port rtpPort(rtpPortNum);
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const Port rtcpPort(rtcpPortNum);
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Groupsock rtpGroupsock(*env, destinationAddress, rtpPort, ttl);
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rtpGroupsock.multicastSendOnly(); // we're a SSM source
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Groupsock rtcpGroupsock(*env, destinationAddress, rtcpPort, ttl);
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rtcpGroupsock.multicastSendOnly(); // we're a SSM source
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// Create a 'AMR Audio RTP' sink from the RTP 'groupsock':
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audioSink = AMRAudioRTPSink::createNew(*env, &rtpGroupsock, 96);
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// Create (and start) a 'RTCP instance' for this RTP sink:
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const unsigned estimatedSessionBandwidth = 10; // in kbps; for RTCP b/w share
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const unsigned maxCNAMElen = 100;
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unsigned char CNAME[maxCNAMElen+1];
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gethostname((char*)CNAME, maxCNAMElen);
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CNAME[maxCNAMElen] = '\0'; // just in case
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RTCPInstance* rtcp
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= RTCPInstance::createNew(*env, &rtcpGroupsock,
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estimatedSessionBandwidth, CNAME,
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audioSink, NULL /* we're a server */,
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True /* we're a SSM source */);
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// Note: This starts RTCP running automatically
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// Create and start a RTSP server to serve this stream.
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RTSPServer* rtspServer = RTSPServer::createNew(*env, 8554);
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if (rtspServer == NULL) {
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*env << "Failed to create RTSP server: " << env->getResultMsg() << "\n";
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exit(1);
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}
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ServerMediaSession* sms
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= ServerMediaSession::createNew(*env, "testStream", inputFileName,
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"Session streamed by \"testAMRAudioStreamer\"",
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True /*SSM*/);
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sms->addSubsession(PassiveServerMediaSubsession::createNew(*audioSink, rtcp));
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rtspServer->addServerMediaSession(sms);
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announceURL(rtspServer, sms);
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// Start the streaming:
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*env << "Beginning streaming...\n";
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play();
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env->taskScheduler().doEventLoop(); // does not return
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return 0; // only to prevent compiler warning
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}
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void afterPlaying(void* /*clientData*/) {
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*env << "...done reading from file\n";
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audioSink->stopPlaying();
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Medium::close(audioSource);
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// Note that this also closes the input file that this source read from.
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play();
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}
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void play() {
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// Open the input file as an 'AMR audio file source':
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AMRAudioFileSource* audioSource
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= AMRAudioFileSource::createNew(*env, inputFileName);
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if (audioSource == NULL) {
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*env << "Unable to open file \"" << inputFileName
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<< "\" as an AMR audio file source: "
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<< env->getResultMsg() << "\n";
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exit(1);
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}
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// Finally, start playing:
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*env << "Beginning to read from file...\n";
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audioSink->startPlaying(*audioSource, afterPlaying, audioSink);
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}
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