Reducing jitter and latency when re-encoding SRT

[Olivier Crête]

Hi,

On Fri, 2021-04-16 at 10:33 +0200, Jan Willamowius wrote:
> I'm having a pipeline to re-encode an SRT feed from OBS.
> 
> gst-launch-1.0 srtsrc latency=200 blocksize=1400 uri=srt://:5000 !
> tsdemux ! faad ! audioconvert ! audio/x-raw ! opusenc audio-
> type=voice ! rtpopuspay ! queue ! multiudpsink
> clients=192.168.0.111:5000
> 
> It introduces quite a bit of jitter and even without the re-encoding
> it
> adds about 1.7 sec of latency.
> 
> Is there a way to add a jitter buffer and reduce the latency ?

You can reduce the latency by reducing the latency property on tsdemux,
but be aware that yo need a sender that puts the PCR timstamps in the
TS stream often enough (700ms is the worse case allowed by the MPEG-TS
spec).

The tsdemux should already absorb the jitter, its output should have
accurate timestamps (those from the incoming mpeg-ts stream).


-- 
Olivier Crête
olivier.crete at collabora.com

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