[Jack-Devel] alsa_in/out increasing time between record and play on second card

mdrslmr mdrslmr at online.de
Tue Apr 12 14:29:51 CEST 2016


thanks for the quick and sound response.

On Tue, 12 Apr 2016, Jörn Nettingsmeier wrote:

> On 04/12/2016 10:42 AM, mdrslmr wrote:

>>  We use a script starting:
>>  aplay -q -D $channel Test-Ton.wav &
>>  arecord -q -D $channel -f S16_LE -r 48000 -c 1 -d 8 $result/recorded.wav
> Why are you using alsa tools when you are otherwise working with JACK? This 
> will make the problem analysis more complicated...
Just because I'm, familiar with it. I'm not sure what to replace it
with, jack.play, jack.record?
>>  The resulting wav file shows that the time between the start of the
>>  recording
>>  and the start of the actual sound continuously increases.
>>  In the beginning (after starting the system up) the time difference is
>>  just about 3ms. It increases
>>  constantly to almost one second now
>>  after 8 weeks of uptime.
> If audioadapter works by just providing a big buffer (without any 
> resampling), the reason would be clock drift. Such a minuscule clock drift 
> between two consumer cards however would be sensationally good.

Yes I think it's just clocks drift.

I did jack_unload/jack_load and got the channels back (almost) in synch.

> First of all, consider if you can actually sync the cards. If both have an 
> SPDIF i/o, you can use that. Just connect the secondary card's SPDIF in to 
> the master's SPDIF out (no matter whether you use toslink or RCA), and you 
> can be sure your cards will be locked.

Our cards M-Audio Delta44 don't have SPDIF ports.

> Secondly, try zita-ajbridge. If you can sync the cards in hardware, there is 
> an option for zita-ajbridge to skip the resampling, which will conserve cpu 
> power.

I'll check it out.


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