Hello sir,
i made an streamer using your libraries which streams the live video from
camera
int initLm555Settings(void)
{
scheduler = BasicTaskScheduler::createNew();
env = BasicUsageEnvironment::createNew(*scheduler);
destinationAddressStr
#ifdef USE_SSM
= "232.255.42.42";
#else
= StreamingIp;
#endif
const unsigned short rtpPortNum = 18888;
const unsigned short rtcpPortNum = rtpPortNum+1;
const unsigned char ttl = 7;
struct in_addr destinationAddress;
destinationAddress.s_addr = our_inet_addr(destinationAddressStr);
const Port rtpPort(rtpPortNum);
const Port rtcpPort(rtcpPortNum);
Groupsock rtpGroupsock(*env, destinationAddress, rtpPort, ttl);
Groupsock rtcpGroupsock(*env, destinationAddress, rtcpPort, ttl);
#ifdef USE_SSM
rtpGroupsock.multicastSendOnly();
rtcpGroupsock.multicastSendOnly();
#endif
g_ExitEventLoop = 0;
ideoSink =
SimpleRTPSink::createNew(*env, &rtpGroupsock, 33, 90000, "video",
"MP2T",
1, True, False /*no 'M' bit*/);
setSendBufferTo(*env, rtpGroupsock.socketNum(), 1024 * 1024);
// Create (and start) a 'RTCP instance' for this RTP sink:
const unsigned estimatedSessionBandwidth = 5000; // 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,
videoSink, NULL /* we're a server */, isSSM);
UserAuthenticationDatabase* authDB = NULL;
portNumBits rtspServerPortNum = 554;
unsigned reclamationTestSeconds=65U;
rtspServer = RTSPServer::createNew(*env,rtspServerPortNum, authDB,
reclamationTestSeconds);
if (rtspServer == NULL)
{
*env << "Failed to create RTSP server: " <<env->getResultMsg()<<"\n";
rtspServerPortNum = 8554;
rtspServer = RTSPServer::createNew(*env,rtspServerPortNum);
if (rtspServer == NULL)
{
return 0;
}
else
{
*env << "Created RTSP server.."<<"\n.";
}
}
else
{
*env << "Created RTSP server.."<<"\n.";
Boolean const inputStreamIsRawUDP = False;
char const* descriptionString={"Session streamed by \"testOnDemandRT\""};
sms= ServerMediaSession::createNew(*env, streamName,
streamName,descriptionString);
sms->addSubsession(MPEG2TransportUDPServerMediaSubsession::createNew(*env,destinationAddressStr,rtpPortNum1,inputStreamIsRawUDP));
rtspServer->addServerMediaSession(sms);
char* url = rtspServer->rtspURL(sms);
*env << "Play this stream using the URL \"" << url << "\"\n";
delete[] url;
if (rtspServer->setUpTunnelingOverHTTP(sport) ||
rtspServer->setUpTunnelingOverHTTP(sport) ||
rtspServer->setUpTunnelingOverHTTP(sport))
{
cout<<"\n\n\n(We use port "<<rtspServer->httpServerPortNum()<<" for
optional RTSP-over-HTTP tunneling.)\n";
}
else
{
cout<<"\n\n\n(RTSP-over-HTTP tunneling is not available.)";
}
play();
env->taskScheduler().doEventLoop(&g_ExitEventLoop);
if(rtspServer)
Medium::close(rtspServer);
if(rtcp)
Medium::close(rtcp);
if(videoSink)
Medium::close(videoSink);
if(fileSource)
Medium::close(fileSource);
rtpGroupsock.removeAllDestinations();
rtcpGroupsock.removeAllDestinations();
env->reclaim();
delete scheduler;
return 0; // only to prevent compiler warning
}
void afterPlaying(void* /*clientData*/) {
*env << "...done reading from file\n";
videoSink->stopPlaying();
// Note that this also closes the input file that this source read from.
Medium::close(videoSource);
// Start playing once again:
play();
}
//================================================================
// play(): Play the input source.
//=================================================================
void play() {
// Open the input file as a 'byte-stream file source':
fi_params.nFICardFrameSize = TRANSPORT_PACKETS_PER_NETWORK_PACKET *
TRANSPORT_PACKET_SIZE;
fi_params.pfnGetRTPPayload = GetRTPPayload;
fi_params.socketNum = videoSink->groupsockBeingUsed().socketNum();
DeviceParameters temp;
fileSource = DeviceSourceFICard::createNew(*env, fi_params, temp);
if (fileSource == NULL) {
*env << "Unable to open Foresight card as a byte-stream file source\n";
exit(1);
}
FramedSource* videoES = fileSource;
// Create a framer for the Video Elementary Stream:
videoSource = MPEG1or2VideoStreamDiscreteFramer::createNew(*env,
videoES);//original
// Finally, start playing:
*env << "Beginning to read from file...\n";
videoSink->startPlaying(*videoSource, afterPlaying, videoSink);
// env->taskScheduler().scheduleDelayedTask(uSecsToDelay,
(TaskFunc*)periodicbrMeasurement1, videoSink);
}
void StartRTPProcess(void)
{
g_hRtpComThread = CreateThread((LPSECURITY_ATTRIBUTES) NULL, 0,
(LPTHREAD_START_ROUTINE)initLm555Settings, 0, 0,
&g_dwRtpComThreadID);
if(g_hRtpComThread) SetThreadPriority(g_hRtpComThread,
THREAD_PRIORITY_LOWEST/*THREAD_PRIORITY_NORMAL*/);
}
int StopRTProcess(void)
{
try{
if( videoSource )
videoSource->stopGettingFrames();
*env <<"in StopRTProcess\n";
Sleep(500);
Medium::close(rtspServer);
g_ExitEventLoop = 1;
g_ExitEventLoop = 0;
g_hRtpComThread = 0;
g_dwRtpComThreadID = 0;
return 0;
}
The streaming done by above streamer is catched by proxy server to which i
give the url given by streamer at the line "char* url".
this streaming is then seen by the client by using the proxy server
URL,when the client say to stop stream the streamer calls its method
StopRTProcess(void) but i got stuck in the line "
Medium::close(rtspServer);",can you please tell why?
I know i have modified your code but still need some of your help
Thanks
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