Reducing delay in RTP streaming

[Kévin Aupée]

Thank you a lot.

By adjusting buffer-time and latency-time properties of alsasink, I've been
able to get a delay of ~90ms.
The delay is slightly higher with sync=false (100-110ms). Why should I keep
it ? Better sound quality ?


2017-01-03 18:50 GMT+01:00 Tim Müller <tim at centricular.com>:

> On Tue, 2017-01-03 at 14:55 +0100, Kévin Aupée wrote:
>
> Hi,
>
> > I'm using Gstreamer for RTP streaming with this pipeline :
> >
> > > gst-launch-1.0 udpsrc port=5000 caps=application/x-rtp !
> > > rtpjitterbuffer latency=50 drop-on-latency=true ! rtpopusdepay !
> > > opusdec ! alsasink sync=false
> >
> > The delay is about 300ms. It is not so bad but I want to reduce it as
> > much as I can.
> > I don't really understand all the properties of the elements and I
> > would like to know which ones can help reduce the latency (besides
> > the ones I already use).
>
> Maybe also have a look at the various alsasink properties that affect
> the audio sink internal latency and buffer size. You should probably
> drop the sync=false though.
>
> Cheers
>  -Tim
>
> --
> Tim Müller, Centricular Ltd - http://www.centricular.com
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