RTSP Latency

Sebastian Dröge sebastian at centricular.com
Mon Apr 18 07:07:10 UTC 2016


On Do, 2016-04-14 at 08:26 -0700, cowprod wrote:
> Hi Sebastian,
> 
> I used
>     gst_debug_set_default_threshold(GST_LEVEL_LOG ); //6
> instead of 
>     gst_debug_set_threshold_for_name("gst-player", GST_LEVEL_TRACE);
> and the result is really really big :) I tried to find witch decoder is used by default but without success (I certainly missed it !)
> 
> can I filter with gst_debug_set_threshold_for_name ? on witch name ?
> 
> If you have a moment to look at the following log file you may find what we are looking for

It uses vtdec, which is not perfect and not really controllable about
latency. (Also don't set a latency of 0 on rtpsrc/rtpjitterbufer!)

Your pipeline is also configured with a latency of 0, so in theory
there shouldn't be any latency and nothing should work at all as 0 is
clearly wrong for anything going over the network.

-- 
Sebastian Dröge, Centricular Ltd · http://www.centricular.com

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 949 bytes
Desc: This is a digitally signed message part
URL: <https://lists.freedesktop.org/archives/gstreamer-devel/attachments/20160418/43e30a17/attachment-0001.sig>


More information about the gstreamer-devel mailing list