Thanks Steve for your insight! 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? Please let me know so that I can make appropriate changes. Ivica Ico Bukvic, composer & multimedia sculptor http://meowing.ccm.uc.edu/~ico/

"Ivica Ico Bukvic" <ico(a)fuse.net> writes: JACK expects there to be *no* drift and does *nothing* to manage this problem. It can only work with high-end cards using word-clock or other sample-accurate synchronization. Probably someone has succeeded with that (though I can't recall a specific report). -- joq

"Ivica Ico Bukvic" <ico(a)fuse.net> writes: 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

On Fri, 2004-05-28 at 10:55, Ivica Ico Bukvic wrote: The slower card will have to be fed something other than the source material from time to time to be able to catch up to the faster one (the "source" is coming at only one speed from only one place and going to two different places that need to be fed at slightly different rates). The "something to be fed" will be probably silence, that is, a click :-) The size of the buffer and the amount of drift between cards will determine how often you get a click (if the software would support doing this at all, of course). -- Fernando

Yes, of course that is doable. I don't think you need to profile them. Just measure the drift between the hardware pointers for both cards as you play. A long time ago (98/98 or so) I _have_ used two sb128 soundcards without sample sync to play back four channel pieces under linux. If the cards are of the same model and manufacturing batch the drift will be small. Small enough to play a 10 or 15 minute 4 channel piece without clicks[*] :-) Guilty :-) -- Fernando [*] but either the front or back speakers will slowly drift away from you as the piece progresses, not a real problem :-)

On Friday 28 May 2004 15:19, Ivica Ico Bukvic wrote: I strongly suspect that you'd find your results to be non-repeatable. Many factors can subtly influence the output frequency of even crystal-locked SRGs: ambient temperature, power supply voltage variation, even component aging. This issue affects many more applications than just audio. *Any* system that requires precise replication of clock (as, for example, most any digital telecommunication scheme does) faces this dilemma. In the end, some form *locking*, slave clock to master, is needed. A variety of methodologies, such as PLL (phase lock loop) exist to do this, but the bottom line remains that some sort of hardware support will be necessary. Cheers! |-------------------------------------------------------------------------| | Frederick F. Gleason, Jr. | Director of Broadcast Software Development | | | Salem Radio Labs | |-------------------------------------------------------------------------| | I know the answer! The answer lies within the heart of all mankind! | | The answer is twelve? I think I'm in the wrong building. | | -- Charles Schulz | |-------------------------------------------------------------------------|

On 2004-05-28 19:49:08 +0000 Fred Gleason <fredg(a)salemradiolabs.com> wrote: what if you use a buffer {maybe something that would interactively adjust the relative latencies and serve to each card separately.You would have to set time somewhere else. {unless you set your time with one card and fed stuff to the other relative to that.}