| /********** |
| 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 3 of the License, or (at your |
| option) any later version. (See <http://www.gnu.org/copyleft/lesser.html>.) |
| |
| 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., |
| 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
| **********/ |
| // Copyright (c) 1996-2020, Live Networks, Inc. All rights reserved |
| // A test program that streams GSM audio via RTP/RTCP |
| // main program |
| |
| // NOTE: This program assumes the existence of a (currently nonexistent) |
| // function called "createNewGSMAudioSource()". |
| |
| #include "liveMedia.hh" |
| #include "GroupsockHelper.hh" |
| |
| #include "BasicUsageEnvironment.hh" |
| |
| ////////// Main program ////////// |
| |
| // 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) |
| |
| #ifdef IMPLEMENT_RTSP_SERVER |
| RTSPServer* rtspServer; |
| #endif |
| |
| UsageEnvironment* env; |
| |
| void afterPlaying(void* clientData); // forward |
| |
| // A structure to hold the state of the current session. |
| // It is used in the "afterPlaying()" function to clean up the session. |
| struct sessionState_t { |
| FramedSource* source; |
| RTPSink* sink; |
| RTCPInstance* rtcpInstance; |
| Groupsock* rtpGroupsock; |
| Groupsock* rtcpGroupsock; |
| } sessionState; |
| |
| 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: |
| 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 rtpPortNum = 6666; |
| const unsigned short rtcpPortNum = rtpPortNum+1; |
| const unsigned char ttl = 1; // low, in case routers don't admin scope |
| |
| struct in_addr destinationAddress; |
| destinationAddress.s_addr = our_inet_addr(destinationAddressStr); |
| const Port rtpPort(rtpPortNum); |
| const Port rtcpPort(rtcpPortNum); |
| |
| sessionState.rtpGroupsock |
| = new Groupsock(*env, destinationAddress, rtpPort, ttl); |
| sessionState.rtcpGroupsock |
| = new Groupsock(*env, destinationAddress, rtcpPort, ttl); |
| #ifdef USE_SSM |
| sessionState.rtpGroupsock->multicastSendOnly(); |
| sessionState.rtcpGroupsock->multicastSendOnly(); |
| #endif |
| |
| // Create a 'GSM RTP' sink from the RTP 'groupsock': |
| sessionState.sink |
| = GSMAudioRTPSink::createNew(*env, sessionState.rtpGroupsock); |
| |
| // Create (and start) a 'RTCP instance' for this RTP sink: |
| const unsigned estimatedSessionBandwidth = 160; // 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 |
| sessionState.rtcpInstance |
| = RTCPInstance::createNew(*env, sessionState.rtcpGroupsock, |
| estimatedSessionBandwidth, CNAME, |
| sessionState.sink, NULL /* we're a server */, |
| isSSM); |
| // Note: This starts RTCP running automatically |
| |
| #ifdef IMPLEMENT_RTSP_SERVER |
| rtspServer = RTSPServer::createNew(*env, 8554); |
| if (rtspServer == NULL) { |
| *env << "Failed to create RTSP server: " << env->getResultMsg() << "%s\n"; |
| exit(1); |
| } |
| ServerMediaSession* sms |
| = ServerMediaSession::createNew(*env, "testStream", "GSM input", |
| "Session streamed by \"testGSMStreamer\"", isSSM); |
| sms->addSubsession(PassiveServerMediaSubsession::createNew(*sessionState.sink, sessionState.rtcpInstance)); |
| rtspServer->addServerMediaSession(sms); |
| |
| char* url = rtspServer->rtspURL(sms); |
| *env << "Play this stream using the URL \"" << url << "\"\n"; |
| delete[] url; |
| #endif |
| |
| play(); |
| |
| env->taskScheduler().doEventLoop(); // does not return |
| return 0; // only to prevent compiler warning |
| } |
| |
| void play() { |
| // Open the input source: |
| extern FramedSource* createNewGSMAudioSource(UsageEnvironment&); |
| sessionState.source = createNewGSMAudioSource(*env); |
| if (sessionState.source == NULL) { |
| *env << "Failed to create GSM source\n"; |
| exit(1); |
| } |
| |
| // Finally, start the streaming: |
| *env << "Beginning streaming...\n"; |
| sessionState.sink->startPlaying(*sessionState.source, afterPlaying, NULL); |
| } |
| |
| |
| void afterPlaying(void* /*clientData*/) { |
| *env << "...done streaming\n"; |
| |
| sessionState.sink->stopPlaying(); |
| |
| // End this loop by closing the media: |
| #ifdef IMPLEMENT_RTSP_SERVER |
| Medium::close(rtspServer); |
| #endif |
| Medium::close(sessionState.rtcpInstance); |
| Medium::close(sessionState.sink); |
| delete sessionState.rtpGroupsock; |
| Medium::close(sessionState.source); |
| delete sessionState.rtcpGroupsock; |
| |
| // And start another loop: |
| play(); |
| } |