Reducing delay in RTP streaming

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Reducing delay in RTP streaming

kecalace
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).

Thanks :)


(sorry if there is some grammar mistakes)

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Re: Reducing delay in RTP streaming

Tim Müller
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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Re: Reducing delay in RTP streaming

kecalace
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 <[hidden email]>:
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
_______________________________________________
gstreamer-devel mailing list
[hidden email]
https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel


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