Uses to reduce latency of a pipeline

[Nicolas Dufresne]

Le lundi 28 janvier 2019 à 13:04 +0100, Jack a écrit :
> Hello,
> 
> I would like to know if you have recommandations on how to reduce
> latency on a running pipeline using gstwebrtc ?
> 
> For instance, a pipeline 1 like (using Python) :
> 
> webrtcbin name=sendrecv bundle-policy=max-bundle

You can configured the jitterbuffer latency using child proxy
properties. But default it's 200ms (rtpbin latency property is in ms),
this example would be 20ms:

  webrtcbin rtpbin::latency=20

> filesrc location=my_stereo_sound.wav ! wavparse ! volume name="vol0"
> volume=0.5 ! opusenc frame-size=10 ! rtpopuspay pt=96 !
> application/x-rtp,media=audio,encoding-name=OPUS,payload=96,clock-rate=48000
> ! sendrecv.
> 
> and a pipeline 2 like (always with Python) :
> 
> webrtcbin name=sendrecv bundle-policy=max-bundle
> filesrc location=my_stereo_sound.wav ! wavparse ! deinterleave name=d
> d.src_0 ! queue ! volume name="vol0" volume=0.5 ! i.sink_0 d.src_1 !
> queue ! volume name="vol1" volume=0.5 ! i.sink_1 interleave name=i !
> audiopanorama ! opusenc frame-size=10 ! rtpopuspay pt=96 !
> application/x-rtp,media=audio,encoding-name=OPUS,payload=96,clock-rate=48000
> ! sendrecv.
> 
> If I change dynamically the value on "vol0" on the first pipeline (with
> set_property()), I can hear the modification 1 second later (for me it
> is acceptable). But if I do the same on the pipeline 2, the modification
> is heard 2 seconds later ! Huge difference ! It is normal regarding the
> "complexity" of the second pipeline (compare to the first) ? What do you
> advise to reduce latency on the second pipeline ?
> ++
> 
> Jack
> 
> _______________________________________________
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> gstreamer-devel at lists.freedesktop.org
> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
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