<div dir="ltr"><div><div>Thank you a lot.<br><br></div>By adjusting buffer-time and latency-time properties of alsasink, I've been able to get a delay of ~90ms.<br></div>The delay is slightly higher with sync=false (100-110ms). Why should I keep it ? Better sound quality ? <br><div><div><br></div></div></div><div class="gmail_extra"><br><div class="gmail_quote">2017-01-03 18:50 GMT+01:00 Tim Müller <span dir="ltr"><<a href="mailto:tim@centricular.com" target="_blank">tim@centricular.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><span class="">On Tue, 2017-01-03 at 14:55 +0100, Kévin Aupée wrote:<br>
<br>
Hi,<br>
<br>
> I'm using Gstreamer for RTP streaming with this pipeline :<br>
><br>
> > gst-launch-1.0 udpsrc port=5000 caps=application/x-rtp !<br>
> > rtpjitterbuffer latency=50 drop-on-latency=true ! rtpopusdepay !<br>
> > opusdec ! alsasink sync=false<br>
><br>
> The delay is about 300ms. It is not so bad but I want to reduce it as<br>
> much as I can.<br>
> I don't really understand all the properties of the elements and I<br>
> would like to know which ones can help reduce the latency (besides<br>
> the ones I already use).<br>
<br>
</span>Maybe also have a look at the various alsasink properties that affect<br>
the audio sink internal latency and buffer size. You should probably<br>
drop the sync=false though.<br>
<br>
Cheers<br>
<span class="HOEnZb"><font color="#888888"> -Tim<br>
<br>
--<br>
Tim Müller, Centricular Ltd - <a href="http://www.centricular.com" rel="noreferrer" target="_blank">http://www.centricular.com</a><br>
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