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.
