On Thu, 2001-10-04 at 07:39, [EMAIL PROTECTED] wrote: > I mention this because although the obsd server seems to work > fine, clients experience some glitches (due to VBR?):
Its probably not VBR -- just a guess. If the player stops playing for a second and has to buffer up, then maybe its a VBR problem. However, I suspect that you're getting the famous 'gurgling' sounds that an mp3 decoder makes when a portion of its stream is missing. Try this: Set up icecast and listen to the streams using winamp/freeamp/xmms, whatever. Do the problems go away? If so, then you have a multicast problem. Multicast is carried over UDP, which is not error corrected. When there is a collision on the network or for some reason one of the mp3 packets gets lost, the decoder is missing crucial information, and its decoder gets screwed up and outputs gurgling garbage. Also, does the problem get worse with a higher load on your network? > Question: may this problems be due to VBR encoding? may the use of > reencoding solve the matter? Is reencoding integrated on the 0.4.0 branch? No. Reencoding will only make matters worse. Trust me. :-) > Secondly, I notice that there are very audible delay's/mismatch on > the music listened from different machines. Is this normal? It would be > great to have all be in synch, so that users near to others listening (at > moderate volumes) music from their own speakers hear the same, dunno if > this is impossible though. This one is tough, if not impossible to solve. The fundamental problem is that most sound cards to not take exactly one second to play one second of audio. Some cards do it in 1.000001 seconds and some do it in .9999999 seconds. Two identical cards from one manufacturer will even drift differently. So, one computer may be playing a bit faster than another computer. To the human the sound will be right because we cannot detect such minute changes. But two computers next to each other will drift apart from another over time, and there is little you can do. You can lower the input buffer size a bit (in freeamp anyways), which will drop the delay between the start of streaming and the start of playback, and initially the players may be more in synch, but over time they will still drift. Plus, the smaller buffer size will make the audio stream more susceptible to high network traffic loads. I hope this all makes sense. P.S. I'd love to see a Spanish version of Obs. I'll do the German translation. :-) -- --ruaok Freezerburn! All else is only icing. -- Soul Coughing Robert Kaye -- [EMAIL PROTECTED] -- http://www.mayhem-chaos.net _______________________________________________ Obs-dev mailing list [EMAIL PROTECTED] http://www.freeamp.org/mailman/listinfo/obs-dev
