Add audio and record RTP stream

Sebastian Dröge sebastian at centricular.com
Tue Jul 12 06:29:47 UTC 2016


On Fr, 2016-07-08 at 10:53 -0400, Andrew Borntrager wrote:
> Hello again. 
> I *think* I'm sending audio along with video. The pipeline is running
> (I changed to ports 5000 and 5002) with no errors.  Now I'm trying
> this on my windows laptop.(doesn't work):
> 
> gst-launch-1.0  udpsrc caps="application/x-rtp, media=(string)video,
> clock-rate=(int) 90000, encoding-name=(string)H264,
> sampling=(string)YCbCr-4:4:4, depth=(string)8, width=(string)320,
> height=(string)240, payload=(int)96, clock-base=(uint)4068866987,
> seqnum-base=(uint)24582" port=5000 ! rtph264depay ! decodebin !queue!
> autovideosink ! udpsrc caps=$AUDIO_CAPS port=5002 !
> gstrtpjitterbuffer! rtpopusdepay ! opusdec plc=true ! alsasink

There is no alsasink on Windows, try using directsoundsink or just
autoaudiosink if you want to use the same code on all platforms. Also
you should insert videoconvert ! videoscale before autovideosink, and
audioconvert ! audioresample before the audio sink.

If that doesn't help, how does it not work? You might also want to set
a bigger value on the buffer-size property on the udpsrcs, especially
for raw video.

-- 

Sebastian Dröge, Centricular Ltd · http://www.centricular.com
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