[linux-audio-user] MOre realtime!

Lee Revell rlrevell at joe-job.com
Wed Dec 20 12:53:15 EST 2006


On Wed, 2006-12-20 at 14:27 +0000, J M Needham wrote:
> Ok, thanks Lars. I guess I'll be treating myself to an audiophile for
> Christmas unless anyone's got a better idea.
> 

These hardware constraints are defined in max/min bytes per period
across all supported formats.  So, to get lower latency, increase the
channel count or sample rate.

64 frames at 44100Hz is ~1.45ms each way or 2.9ms (plus any hardware
latency) round trip.  At 48000Hz this would give 2.66ms, and at 96000Hz
1.33ms.

Lee

> 
> On Wed, 20 Dec 2006, Lars Luthman wrote:
> 
> > On Wed, 2006-12-20 at 10:34 +0000, J M Needham wrote:
> > > So I've just installed Ubuntu 6.09, and followed the instructions on
> > > http://fort2.xdas.com/~kor/oss2jack/install.html to install the
> > > realtime-lsm module and I've added
> > > @audio - rtprio 80
> > > @audio - memlock 500000
> > >
> > > to /etc/security/limits.conf and set the realtime mode on Jack. Seems ok,
> > > the only thing is that I can't get below 5.8 ms latency. Jack's behaving
> > > nicely with very few xruns, but won't do any better. The output of the
> > > messages with 32 frames for capture, for example, is:
> > >
> > > 10:41:35.039 Startup script...
> > > ...
> > > configuring for 44100Hz, period = 32 frames, buffer = 2 periods
> > > ALSA: cannot set period size to 32 frames for capture
> > > ALSA: cannot configure capture channel
> > > cannot load driver module alsa
> >
> > It looks like your hardware simply can't handle buffer sizes that small.
> >
> >
> 
> 




More information about the Linux-audio-user mailing list