Subject: Re: [linux-audio-user] jaming over the internet?
From: Juhan Leemet (juhan_AT_logicognosis.com)
Date: Wed Oct 01 2003 - 21:52:12 EEST
GuyCLO~ wrote:
>I agree. I found that larger latencies (100 < latency < 200) are usable for
>having fun. I mean I have used softsynths on a computer without tuning
>latency and without beeing root.
Hmm. How are you measuring latency? I'm not sure how to do it (sufficiently
accurately). I'm running SuSE 8.2, which some have suggested includes the
low-latency patch, but I don't think so. I've been too busy (lazy?) to check
or implement. I find even when playing some .ogg file to jam along with, that
hiccups are "disturbing" or distracting. Maybe I've got problems (as a
musician wannabe) with my timing? I've got my .ogg files on a server, but I
believe xmms pre-buffers (I recall setting it to 1/2 second at one point?)
its compressed audio stream, so I think I'm only hearing jitter from variable
interrupt response (and temporarily blocked interrupts?)?
Your other comment (in another post) about "real life lost packets" (UDP
comparison) is interesting. You would have to transmit a time code in each
packet, so the player can continually "re-sync" the audio. I don't think the
current audio streams do that? I think they "assume" (Benny Hill?) that the
audio stream is continuous, and therefore you can derive the timing?
-- Juhan Leemet Logicognosis, Inc.
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