Add audio and record RTP stream.

[Nicolas Dufresne]

Le jeudi 07 juillet 2016 à 08:31 -0400, Andrew Borntrager a écrit :
> Hi all! I have a working pipeline running on a Raspberry Pi:
> 
> gst-launch-1.0 -v vl2src device=/dev/video0 ! queue ! video/x-h264,
> width=1280, height=720, framerate=15/1 ! queue ! h264parse ! queue !
> rtph264pay pt=127 config-interval=4 ! udpsink
> host=***********.ddns.net port=5000
> 
> I have a windows laptop with this:
> 
> 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

You should use rtpjitterbuffer and set it's latency property, right
after each udpsrc. It will remove the burst effect and ensure sync
between audio and video.

> 
> This works very well. However, I would like to add audio (Im using a
> Logitech C920 webcam). Also, i would like to record incoming stream
> directly to laptop (possibly the "tee" command??). Any latency
> optimizations would be greatly appreciated, even if it means
> momentary degradation of video. I'm kind of a newbie, so copy/paste
> and inserting into my pipeline is also greatly appreciated! 

For the audio part, it's pretty much the same method. It depends on
your raspery pi setup. There is 2 possible audio source, alsasrc, for
which you need to set the device property based on the output of
arecord -L, or pulsesrc which also have a device property and the list
of sources can be opbained using "pactl list short sources". Most
people uses OPUS to encode audio, the clock rate is always 48000.

cheers,
Nicolas
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