28 May
2004
28 May
'04
4:04 p.m.
Hmm, so just for my own understanding of this, if let's say 2 soundcards A
and B lack sync between themselves, yet are being fed in appropriate
intervals small buffers of audio data from JACK, what is preventing them
from staying in sync?
For instance, the way I see it is that if one card even spits out the audio
at a fraction faster than other, due to sample buffer size being small
enough, such inconsistency imho should not be even noticed unless obviously
there is something really screwed up and the speed of output is seriously
off in which case such card is obviously deemed inadequate for any serious
audio work anyhow.
Plase correct my assumption if I am [most likely] wrong. Also a verbose
explanation would be highly appreciated.
Thanks!
Ivica Ico Bukvic, composer & multimedia sculptor
http://meowing.ccm.uc.edu/~ico/
-----Original Message-----
From: linux-audio-user-bounces(a)music.columbia.edu [mailto:linux-audio-
user-bounces(a)music.columbia.edu] On Behalf Of Jack O'Quin
Sent: Friday, May 28, 2004 1:46 PM
To: The Linux Audio Developers' Mailing List
Cc: 'A list for linux audio users'
Subject: Re: [linux-audio-dev] re: [linux-audio-user] A bit of good news--
paper now available for your viewing pleasure and/or comments
"Ivica Ico Bukvic" <ico(a)fuse.net> writes:
Hasn't there been some success stories in the
past regarding this? I
might be obviously very wrong about this but I thought that if one
designed a meta-device in the asoundrc making two soundcards one
multichannel soundcard and then invoking JACK on top of it, that it
should work?
Many attempts have been reported. I don't recall any success stories,
but maybe there were some. Most people want to do this with two cheap
consumer "multimedia" cards. That won't work, because they can't be
synchronized.
--
joq