[LAU] "Skewed" Audio with JACK

[Len Ovens]

On Sat, 3 Dec 2016, David Klann wrote:

> Long-time Linux user, and relatively new JACK user here. I have built
> some audio workstations for the community radio station where I
> volunteer (WDRT, Viroqua, WI, US). I recently switched one of the
> workstations to use JACK, along with PulseAudio. We use Audacity to edit
> audio and we have noticed that the "left" and "right" channels are out
> of sync with each other. We have witnessed the "skew" to be as few as
> two samples (which is unnoticeable to the ear) to as many as a couple
> hundred samples (which sounds a lot like a phase error).

Pulse is pretty much guarenteed to loose samples here and there. Audacity 
uses portaudio (I think) to talk to jack, I don't know how good that is or 
not.

> We performed a lot of troubleshooting, including swapping PCI audio
> cards (ESI Juli@, Digigram VX222), disabling JACK, running Audacity on
> the same hardware booted from a USB stick and a completely different
> Debian environment. I am not *completely* confident, but the likely
> culprit seems to be JACK.

I do not know about the juli, but the digigram does rather a lot of 
processing including the posibility of sample rate conversion. One of it's 
features is a continuously variable sample rate which may be a SRC derived 
process. There have in the past been audio interfaces (SB 94ish?) that 
have a one speed codec that runs through a HW SRC unit to provide the 
computer with the requested sample rate. In that case they even did 
48k->48k SRC with the result that any long recording had a different sized 
file for left and right. As src is common in broadcast audio interfaces 
(particularily on aes3 inputs) I would want to make sure that is turned 
off.

Have you tried mhWaveEdit instead of Audacity? It may not provide all the 
features Audacity does plugin wise, but other than that it seems quite 
solid.

--
Len Ovens
www.ovenwerks.net