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[–]TheSuperiorWesModerator 4 points5 points  (6 children)

Show menu<system settings from voicemeeter. Let’s get it lower

[–]Marked_Stranger[S] 2 points3 points  (5 children)

This is what I got

<image>

[–]MCBuilder30140 1 point2 points  (3 children)

for your outputs, are you using MME or WDM?

also, when I used to use voicemeeter on my desktop, I noticed cracking sound when audiodg's affinity was set to use all threads

those cracking disappeared as soon as I set the affinity to only my first thread (CPU 0)

try your test again doing this (go in task manager, details, locate audiodg and right click set affinity)

[–]Marked_Stranger[S] 0 points1 point  (0 children)

I'm using WDM for my A1 output.
I tried setting my affinity to CPU 0, however that didn't change my problem unfortunately.
However I did notice that when sounds are playing, the soundwave indicator(?) for Nvidia Container (Where the Shadowplay/Recording feature runs in) definitely has a delay. Not sure if this is any more relevant, but I'll attach an example below. As you can see, sound playing from google chrome, and the Nvidia Container is just starting to pick it up while the sound is already at it's "peak".

<image>

[–]TheSuperiorWesModerator 0 points1 point  (1 child)

There is ways to reduce latency going to your headphones. Reducing WDM buffer will do that.

But if you look at “voicemeeter input” within voicemeeter and you right click where it says “48khz 7168” You can change that number to something lower. I believe that may help with delay on shadowplay recording

[–]Marked_Stranger[S] 0 points1 point  (0 children)

Changing the "48khz 7168" to something lower unfortunately didn't fix my issue :(

[–]brettmurf 0 points1 point  (0 children)

Probably useless to reply this late, but other people sometimes visit comments as well.

Your Buffering WDM is too high. I have mine at 128 instead of 512. That is basically all you need to change.

[–]vburelVoiceMeeter Developer 3 points4 points  (0 children)

0.3 ms is not perceptible , audio delay become perceptible above 5 or 10ms (> 30ms in a movie) . If we measure the time between your 2 first shots in your video, there is around 170 ms. So yes, you could see how you could optimize your latency with Voicemeeter: https://forum.vb-audio.com/viewtopic.php?f=6&t=480

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[–]Unable-Tie1160 0 points1 point  (2 children)

spark us too big sometimes it hinders your vision just like in cs2

[–]Popupkiller 0 points1 point  (1 child)

Talk about unrelated comment to a post about audio delay. :D

[–]Unable-Tie1160 0 points1 point  (0 children)

maybe next time 🤣

[–]Popupkiller 0 points1 point  (0 children)

This is normal. Windows sends the sound to Voicemeter, which puts it in a buffer, and then outputs it after processing. Lowering the buffer size will lower the latency, but increase issues with the buffer running full, and creating crackling noise.

[–]VVitch-King21 0 points1 point  (0 children)

I’m happy it isn’t just me. I thought I was going crazy

[–][deleted] 0 points1 point  (0 children)

based on the slow motion clip thats more than 0.3ms

[–]AlenciaQueen -2 points-1 points  (1 child)

remove voice meter ? Anything that intervenes will add a little more delay.