Audio Device In Chop Latency

Hi, I am using Audio Device in Chop Operator.
And, I am thinking about latency.

Following official site wiki,…, this operator’s buffer length parameter will effect latency. Trying to minimum this buffer length, it seems to be this buffer length number limit is 0.0086sec.

Then, I am trying to reduce latency of Audio Device In with Audio InterFace UAC2-ZOOM. But I could not find right way to reduce latency.…/audio…/uac-2-usb-30-audio-converter

Do anybody know the apropriate way to reduce latency?? :unamused: :unamused: :unamused:

I overcome latency by using the methods described in the link above. Pretty much this: You never beat latency digitally, so you beat it with analog. What you need is a send on each channel, sending to a monitor mix. When you are recording , you listen to analog monitors. If the UAC2 has realtime headphones, send those headphones to your monitors while recording.

Thank you.

I have tested latency in several ways. I can show the result as follows.
This results is made by just 2 hours test, so if you could find another way to reduce latency, I would like to know it. This is just my case.

MAC OS MacBookPro
Device - Software - Latency
Zoom UAC2 - MAX7 - 0.00072 Sec
Zoom UAC2 - TouchDesigner - AudioDeviceIn 0.0087 Sec
Roland Octa Caputure - MAX7 - 0.00072Sec
Roland Octa Caputure - TouchDesigner - 0.0087 Sec

Windows OS
Device - Software - Latency
Zoom UAC2 - MAX7 - ?
Zoom UAC2 - TouchDesigner - ?
Roland Octa Caputure - MAX7 - 0.0058 Sec
Roland Octa Caputure - TouchDesigner - 0.0087Sec

Hi, I’m bumping this thread up instead of creating a new one in order to minimize duplicates on the forum.

I’ve had much trouble with AudioDeviceIn latency, and was even considering buying some audio-to-midi drummer hardware to try to make it work. But I think I found a solution that I never saw mentioned in the forums.

I never could find the right “buffer length” in milliseconds to lower the latency enough (before sound would just stop being detected). So I changed the unit to “F” (frames), and managed to go as low as 1 frame buffer length ! Which now makes my interactions quasi-perfect!

If you need the audio to go to an out, you might want to go to 2 frames, because at 1, I get some crackling noises… but since I only use it for controlling visuals, it works perfectly for me. (and only using the built-in shitty soundcard of my laptop… I will test with a Scarlett this August).

I hope this post helps some people in the future, because this solution really did wonders for my project.



Nice find, I’m going to try this with my older projects that I discontinued because of the latency.

Let me know if it worked, seems too easy to be true, but it really seems to work for me (although I haven’t been able to test it in proper “live show” conditions yet).