Hi,
We are working on application, which is based on IOS platform, to send Video
and Audio to Wowza streaming server.
Video & Audio data are captured and encoded at IOS library APIs in H.264 & AAC
format then encoded Video and Audio bitstream data are passed to Wowza
streaming server using Darwin-injector class.
Please find below code snippet of implementation.
We are facing issue of Video stuck on Wowza streaming server when Video and
Audio both are enabled for streaming but if Video is only enabled then issue
isn't observed.
Please provide your input if miss any settings.
//Function to create Client session
{
exitEventLoop = 0;
//CREATE SCHEDULER
BasicTaskScheduler* scheduler = BasicTaskScheduler::createNew();
uenv = BasicUsageEnvironment::createNew(*scheduler);
injector = DarwinInjector::createNew(*uenv, "live");
struct in_addr dummyDestAddress;
dummyDestAddress.s_addr = 0;
//set Groupsock for video
Groupsock rtpGroupsockVideo(*uenv, dummyDestAddress, 0, 0);
Groupsock rtcpGroupsockVideo(*uenv, dummyDestAddress, 0, 0);
//set Groupsock for audio
Groupsock rtpGroupsockAudio(*uenv, dummyDestAddress, 0, 0);
Groupsock rtcpGroupsockAudio(*uenv, dummyDestAddress, 0, 0);
//set buffer size for High resolutin.
OutPacketBuffer::maxSize = 600000;
//For Video Sink
rtpVideoSink = H264VideoRTPSink::createNew(*uenv, &rtpGroupsockVideo, 96,
spsByte, [sps length], ppsByte, [pps length]);
//For Audio Sink
rtpAudioSink = MPEG4GenericRTPSink::createNew(*uenv, &rtpGroupsockAudio,
97,44100,"audio","AAC-hbr", ConfigStr, 2);
if( rtpVideoSink == NULL )
{
return;
}
if(rtpAudioSink == NULL )
{
return;
}
// Create (and start) a 'RTCP instance' for this RTP sink:
const unsigned estimatedSessionBandwidthVideo = 4500; // 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
const unsigned estimatedSessionBandwidthAudio = 160;
//Crate RTSP client instance.
RTCPInstance* videoRTCP =RTCPInstance::createNew(*uenv,
&rtcpGroupsockVideo, estimatedSessionBandwidthVideo, CNAME, rtpVideoSink, NULL
/* we're a server */);
RTCPInstance* audioRTCP= RTCPInstance::createNew(*uenv,
&rtcpGroupsockAudio, estimatedSessionBandwidthAudio, CNAME, rtpAudioSink,NULL
/* we're a server */);
// Note: This starts RTCP running automatically
// Add audio and video stream to 'Darwin injector':
injector->addStream(rtpVideoSink, videoRTCP);
injector->addStream(rtpAudioSink, audioRTCP);
//Try to connect with server with given ip ann username password.
if(!injector->setDestination(ipaddress, appStreamName, "live","", port,
username, password))
{
printf("Connection failed !!!!\n");
}
else
{
printf("Connection Done !!!!\n");
}
play();
*uenv << "Play this stream (from the Darwin Streaming Server) using the
URL:\n"
<< "\trtsp://" << "localhost" << "/" << "test.sdp" << "\n";
env->taskScheduler().doEventLoop(&exitEventLoop); // does not return
return 0; // only to prevent compiler warning }
//Start playing to send data to server
void play()
{
//VIDEO FRAMED SOURCE
videoFramedSource = LiveSourceWithx264::createNew(*uenv);
if(videoFileSource == NULL)
{
return;
}
FramedSource* videoES = videoFramedSource;
videoSource = H264VideoStreamDiscreteFramer::createNew(*uenv, videoES);
// Finally, start playing:
rtpVideoSink->startPlaying(*videoSource, afterPlaying, rtpVideoSink);
//AUDIO FRAMED SOURCE
audioSource = AACFramedSource::createNew(*uenv);
if (audioSource == NULL)
{
return;
}
// Finally, start playing:
rtpAudioSink->startPlaying(*audioSource, afterPlaying, rtpAudioSink);
}
Here VideoFrameSource and AudioFrameSource class are derived class of
FramedSource class.
Regards,
Rachit
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