<html><head></head><body><div>Hi,</div><div><br></div><div>On Fri, 2021-04-16 at 10:33 +0200, Jan Willamowius wrote:</div><blockquote type="cite" style="margin:0 0 0 .8ex; border-left:2px #729fcf solid;padding-left:1ex"><div>I'm having a pipeline to re-encode an SRT feed from OBS.</div><div><br></div><div>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<br></div><div><br></div><div>It introduces quite a bit of jitter and even without the re-encoding it<br></div><div>adds about 1.7 sec of latency.<br></div><div><br></div><div>Is there a way to add a jitter buffer and reduce the latency ?<br></div></blockquote><div><br></div><div>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).</div><div><br></div><div>The tsdemux should already absorb the jitter, its output should have accurate timestamps (those from the incoming mpeg-ts stream).</div><blockquote type="cite" style="margin:0 0 0 .8ex; border-left:2px #729fcf solid;padding-left:1ex"></blockquote><div><br></div><div><span><pre>-- <br></pre><div data-evo-paragraph="" class="">Olivier Crête</div><div data-evo-paragraph="" class=""><a href="mailto:olivier.crete@collabora.com">olivier.crete@collabora.com</a></div><div data-evo-paragraph="" class=""><br></div></span></div></body></html>