20 Dec
2006
20 Dec
'06
1:53 p.m.
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.