Subject: [linux-audio-dev] questions about real-time for audio
From: John Regehr (jdr8d_AT_cs.virginia.edu)
Date: Thu Apr 13 2000 - 01:36:41 EEST
Hi all,
I've been lurking on this list for a while in order to learn about audio
applications, but the time has come for me to ask a few questions - I'd
really appreciate it if someone could take time to answer them. Audio is
one of the real-time application domains I'm learning about as part of a
paper I'm writing.
It looks like Linux dispatch latencies have gotten down into the small
millisecond range. Is this low enough for just about every audio app, or
is there still motivation to put some audio driver and/or DSP code into
RTLinux tasks?
How much of the processing workload during real-time synthesis would be
characterized as periodic? That is, would it be useful if Linux
guaranteed your application that 2ms out of every 10ms of CPU time would
be available to it (or 20ms out of 100ms, or whatever)? If audio
processing is not periodic, what are the events that trigger processing
and how often do they occur?
Do most of the latency requirements come from the fact that a person is in
the loop, or do they come from the threat of losing data or creating sound
glitches due to small hardware buffers?
When there is a glitch (audible or otherwise), is it usually because of OS
latency or are processors not fast enough to do the DSP in time?
How many glitches would you tolerate from your software sound system
before you started looking at other solutions? For example, 1 per minute,
1 per hour, 1 per century...
Thanks!
John
-- John Regehr | regehr_AT_virginia.edu | http://www.cs.virginia.edu/~jdr8d/ grad student | Department of Computer Science | University of Virginia
This archive was generated by hypermail 2b28 : Thu Apr 13 2000 - 02:23:41 EEST