problem with synchronization between multiple audio clients

[Tim Müller]

On Mon, 2013-11-11 at 13:59 +0100, Javier Domingo wrote:

Hi Javier,

> First of all I must say I am a total newbie, and I don't fully
> understand some (many) stuff about gstreamer.
> 
> I am interested in continuing on the thread with this same title[1].
> Reading the thread, I have supposed you had this scheme (very
> simplified) with the following configuration:
> 
> Server: buffer at least as min-buffer + laggiest client
> playbin ! buffer ! rtpbin
> 
> Client: internal buffer at much as laggiest client
> rtpbin ! audiosink
> 
> I am still wondering how you introduce the clock synchronization there
> to allow the server know which is the client with more lag. Also,
> about the clock, are you just using the Server's clock and making a
> difference in the clients?
> 
> I would appreciate if you could provide me the pipeline you are using
> in both client and server,

The easiest way to do this might be to use a recent version of
gst-rtsp-server (from git, which depends on gstreamer 1.2.x, there
should hopefully be a release any day now). With that, you just set the
'time-provider' property of the GstRTSPMedia to TRUE, then it will send
some special headers to the client, which rtspsrc (>= 1.2.x) will then
pick up automatically (both base time and time provider to sync to), so
you don't have to configure/distribute these things manually anymore.

You can also check out https://github.com/thaytan/aurena for another
approach.

 Cheers
  -Tim

-- 
Tim Müller, Centricular Ltd - http://www.centricular.com