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) _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
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 |
Thank you a lot. By adjusting buffer-time and latency-time properties of alsasink, I've been able to get a delay of ~90ms.2017-01-03 18:50 GMT+01:00 Tim Müller <[hidden email]>: On Tue, 2017-01-03 at 14:55 +0100, Kévin Aupée wrote: _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
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