It turns out I am doing something very wrong, and i *can* get much lower latency’s even
with the large buffersize.
I could not get jack_delay itself to work (it wouldn’t connect to any ports no matter what
i tried), but
i did use jack rather than my own alsa implementation and with a period size of 32 samples
@ 48KHz I measured a roundtrip delay of around 169 samples (3.5 ms), which is very
reasonable.
The method I used was perhaps a bit crude, so it could be off by a few samples. But it
seems accurate enough. (I used another soundcard on a different machine, and plugged one
output straight back in, and the other went through the converter /madi fx / jack and then
back in. The difference in timing should be the rtd).
So now i am a but confused how to get the same kind of latency in my own code (using alsa
directly rather than through jack)
Basically what happens is when i have a buffersize of 8192 and a period of 32 is that
snd_pcm_avail_update(capture_handle), will keep returning 0 until i have played back the
full 8192 samples. After that I will start receiving input samples, but obviously the
latency is now more than 8182 samples…
How can i make alsa not wait until the entire buffer is full?
best,
Fokke
On Feb 3, 2016, at 15:40 , Jörn Nettingsmeier
<nettings(a)stackingdwarves.net> wrote:
On 02/03/2016 02:54 PM, Fokke de Jong wrote:
Hi Adrian,
I could’ve guessed you were on this list as well :-)
I’m going to try this tomorrow when back in the studio, or maybe I can even do it remote
if I have left everything patched up correctly :-)
I will come back with the results.
Thanks for your input!
fokke
watching this with interest, keep us posted about your results and longer-term experience
with this card!
best,
jörn
--
Jörn Nettingsmeier
Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487
Meister für Veranstaltungstechnik (Bühne/Studio)
Tonmeister VDT
http://stackingdwarves.net
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