Audio Device In Chop Latency

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

Following official site wiki, derivative.ca/wiki099/index.php…, 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.
zoom.co.jp/…/audio…/uac-2-usb-30-audio-converter

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

youtube.com/watch?v=4l0BsPKpi-Y

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.

Cheers!
Damien

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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).

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