[Ardour-users] WAV files, xruns

Robert Jonsson robert.jonsson at dataductus.se
Sat Feb 21 05:35:21 PST 2004

lördagen den 21 februari 2004 12.28 skrev Anthony DiSante:
> Robert Jonsson wrote:
> >>OK, I'm a knucklehead.  I only started using the --realtime option when I
> >>started using the -p option (I didn't know about --realtime before that).
> >>When I use --realtime, I can use a period as low as 512 without xruns.
> >>(And jackd actually gives me an error and quits if I try to set it lower
> >>than that.)
> >
> > Atleast for SB-Live the alsa driver does not permit using buffers smaller
> > than 512 samples if you run full duplex (can be much smaller if you
> > specify only playback or only record) Presumably this limitation is there
> > in the Audigy driver too. (And it _is_ a driver limitiation, not
> > hardware)
> Ah.  OK, then it turns out that if I specify -C (capture only), jackd still
> won't let me specify a period below 512.  But if I specify -P (playback
> only), then it will let me use a period as low as 32.  

You might be right there, my bad,

> So for playback, I 
> _can_ get latency as low as <2ms (=2*32/48kHz), but for recording, the
> lowest I can go is 11ms (=512/48kHz).
> So what numbers does everyone else get?  Is 11ms "good enough" for
> recording, say, 8 tracks without problems? 

This is impossible to answer, technically the delay imposed does not matter. 
The recordings you make will not "wander", the 11ms are fixed. I record quite 
oftenly at 512*2, though 256*2 is more comfortable.

I'm not sure if ardour supports latency compensation (yet) during recording, 
if it does then I guess it's possible that subsequent recordings are not 
correctly aligned to the former?

What happens that might be a problem is if you are listening to your own 
"take" through ardour, while you are playing. Then what you are listening to 
will be delayed 11ms + 11ms which definitely is noticeable and might be a 
If you are playing a softsynth there is no way around this latency, you _have_ 
to listen to the generated sound. In cases where the soundsource is analogue 
there is oftenly other options for monitoring.

Actually, you can train the brain to not be so easily distracted by latency. 
I've heard the example of an church organ player mentioned on one of these 
lists before, they have to deal with 0.5 seconds(!) of latency, it probably 
affects the playing style _alot_, but still, it works. 


> (Not that my current hardware 
> supports that, but I do plan on getting something like a 1010LT sometime.)
> -Anthony
> http://nodivisions.com/
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