Regarding pipeline latency and how to achieve a low-latency pipeline

[danny.smith]

Hi!

I am working on a networked scenario in which one sender will transmit audio
to multiple receivers whom will render the audio synchronized, following the
description in this presentation:

https://gstreamer.freedesktop.org/data/events/gstreamer-conference/2015/Sebastian%20Dr%C3%B6ge%20-%20Synchronised%20multi-room%20media%20playback%20and%20distributed%20live%20media%20processing%20and%20mixing%20with%20GStreamer.pdf

and have some questions:

1) Why is setting the total pipeline latency necessary if the receivers are
identical?

2) Which element will handle the additional buffering required by setting
the total pipeline latency higher than the minimum latency?

3) what is the explanation for min/max latency for a pipeline?

4) For the recievers; If I set a jitterbuffer latency of 100ms and alsasink
buffer-time of 100ms, will this result in a total delay of 200ms?

5) I am using the gstreamer netclock to achieve synchronized playback,
sometimes during periods of high load on my network I get simultaneous skew
on all my recievers (etime drifts roughly the same amount in the same
direction). When the network load decreases this drift disappears. Is there
a explanation for this and maybe a solution?

6) What would be the best approach to achieve a <100ms latency "best effort"
pipeline?

Would be grateful is someone could shed some light on the above topics :)

Regards,
Danny





--
View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Regarding-pipeline-latency-and-how-to-achieve-a-low-latency-pipeline-tp4682305.html
Sent from the GStreamer-devel mailing list archive at Nabble.com.