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@xxxxxxxxxxxxxxxxxx [mailto:linux-audio- > user-bounces@xxxxxxxxxxxxxxxxxx] 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@xxxxxxxx> 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
