[pulseaudio-discuss] rtp question

Lennart Poettering lennart at poettering.net
Wed Mar 26 12:22:08 PDT 2008

On Tue, 19.02.08 22:10, Paul Fox (pgf at foxharp.boston.ma.us) wrote:

> hi -- i'm new to using pulse.  i'm slowly replacing a system based
> on NAS (network audio system -- not network accessible storage) that
> i use for home audio distribution.
> i'm curious about the rtp capability -- is there anything in the
> protocol to guarantee synchronization all the way to the sound hardware?
> i understand that since the stream is multicast, all network receivers
> will get it at the same time, but different servers may have different
> latencies.  i suppose these latencies will tend to be constant, and
> not drift, but i'm worried that if the sound hardware is driving systems
> that can be heard from one another (e.g., one in the kitchen, one in
> the livingroom two rooms away) that the delay may be annoying.
> any experience with this?

The RTP code in PA doesn't handle sampling rate deviations. So the
distance between the playback positions will grow larger and larger
(first it's just a huge stereo effect, and then it will become an
echo) and eventually you'll either get skipping audio or a short
dropout and then the the sender and the receiver are in sync again and
everything starts from the beginning.

We have a pretty elaborate mechanism to do adpative sampling rate
detection now. We also have a good resampler. It's just a matter of
hooking things up properly. Patches welcome.


Lennart Poettering                        Red Hat, Inc.
lennart [at] poettering [dot] net         ICQ# 11060553
http://0pointer.net/lennart/           GnuPG 0x1A015CC4

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