I'm seeking suggestions for a low latency usb (or PCIE) sound
interface (or card). Or, maybe some good news that USB sound
devices are generally reasonably low in latency.
My target latency would be around 25-50msec per component. This
is a thin-client setup, so playback sound would go through three
components. Two-channel playback is the only essential
requirement.
Anyway, what I'm thinking is to attach a real PCIE or USB sound
card or interface to the main machine (well, two of them); point
the .asoundrc files at one of them, feed analog signal into
another USB device's input, into a Raspberry PI, over ethernet
via NetJACK or maybe just netcat, and to the thin-client machine.
That way, the real hardware sound card would throttle the samples
at the front end of the chain, which should be massively better
than the current situation.
The detail, lest I not provide enough background:
I have one main machine (because I insist on ECC RAM) and a
couple of thin-client machines. The main machine currently runs
Mageia 2 on a Xeon W3680 (so CPU horsepower is ample). I have
several loopback sound cards (one per user) using snd_aloop. The
users' .asoundrc files play into a loopback/snd_aloop card,
NetJACK sends sound from there to the thin-client, and it goes
out over the thin-client machine's hardware (HDMI and green
jack). The thin-client runs a remastered TinyCore Linux, on an
i3 540 CPU, booting via PXE. There's a private gigabit LAN
between the main machine and the thin-client machines. Only one
thin-client is generally powered on at a time. All machines run
straight ALSA (and NetJACK); Pulse is _NOT_ being used. (I'd
uninstall pulse, but I don't like RPM complaining about massive
numbers of missing dependencies.) Neither CPU nor network are
significant limiting factors.
The above setup initially used Mageia 1 on the main machine, and
life was good. Sound was essentially perfect. However, when I
switched the main machine to Mageia 2, kernel
3.4.52-server-1.mga2, sound became hideously chopped. Somewhere
around 25% or more of the sound samples are being lost due to
xruns. Research into the problem showed snd_aloop was _NOT_
keeping time but would accept samples as rapidly as the playback
client would send them. That resulted in massive xruns. The
problem can be demonstrated with nothing more than aplay,
snd_aloop, and arecord in Mageia 2 on the main machine. I
believe it's a bug in snd_aloop, but I don't have the expertise
to write a convincing bug report for something in kernelspace.
Anyway, are there USB sound interfaces with latencies no more
than 25-50msec? (Or, are they around 500msec latencies like I
hear at work on Doze-based laptops over Lync?)
Thanks,
Robert Riches
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