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