I just did the ASIO test, does this look good?Ĭode: Select all #0001. Yes, I've tried higher buffer sizes, but going past 96 gives a lot more cracks and pops for some reason, no clue how, it's what I'm trying to figure out. Have you joined the VB-Audio Discord server? Maybe there you can find people with a similar setup and try to discuss your issues. I am trying to verify if Audacity uses WASAPI Exclusive mode but apparently (in my setup) it doesn't at least for the Playback device. I would like to try to reproduce your latency measurement setup.
#SWAM ENGINE BUFFER SIZE TOO SMALL DRIVER#
There is also an ASIO driver tester utility that you can use to "test" your interface. I know you have tested almost all the available settings but I am not sure that you have found at least one stable setup. I understand your final objective is to improve latency but at this moment it is important to verify that your setup works as intended.
May I suggest to start with a baseline setup using an ASIO buffer size of at least 256 samples (or even 512) to verify stability. Unfortunately I don't own a USB audio interface to try to replicate your setup. I've also tried reinstalling everything (it's mentioned in the sticky), but no difference. I've also checked out the sticky thread on pops and cracks, but none of those made an improvement. If they do work perfectly (without pops and cracks) the latency ends up being several times higher than just using my DAC bare. Just out of experimentation, I tried playing with different smp latency settings, but they all perform the same, except for the very, very low settings (less than 1536) wherein they start to distort more and give more cracks and pops.Īnyone got any ideas? I tried the other driver modes in A1, but their much worse. So I'm thinking maybe the smp latency is too low in the virtual cable control panel, so I go ahead and crank it up all the way to 24576, however, no difference. it's too random and the difference is too miniscule. I then try the buffer size 96 setting, and I'm honestly not sure if it improves it or not because of how random these pops and cracks are, if I had to pick, I think no? I go ahead and play with different buffer size settings, ranging from 32 to 48, to 64, and 96, but the results are all roughly the same, if I had to pick though, I think maybe 64 is best, but I've had moments where 32 seemed better, sometimes 96 was better, etc. So I'm thinking maybe the buffer size is too small? So I increase it to 128 from 64 and I immediately get massive pops and cracks and distortions, like with the 16 setting. Now it sounds fine, no problem, however sometimes it seemingly randomly does small pops and cracks sometimes it goes through phases that make it do it more often than before, then it calms down, and then a few minutes later starts to do it more often for 10 seconds or so, then calms down, etc. I'm thinking the buffer size is too big, so I put it to 64 from 16.
I then go ahead and set my voicemeter as my default playback device and start to listen to some music I have and immediately I get massive pops, cracks, and distortions. So I go ahead and open up the VB virtual cable control panel for voicemeter, the smp latency is set to 7168, ok that's fine for now. For reference, my focusrite's buffer size by default is set to 16. In the system settings I leave everything at default (don't touch), engine mode on normal and WDM on NO.
#SWAM ENGINE BUFFER SIZE TOO SMALL DRIVERS#
I am able to use ASIO drivers for my A1 with the focusrite, so I select this. So far I've installed voicemeter and hooked up my focusrite to it. I use audacity to measure the latency by generating a ping* sound, then putting my headphones close to my microphone and record it then I calculate the delay. If I just use the DAC by itself without any software, I get about 70ms audio latency. What I essentially want is to reduce the overall system audio latency (windows). I have a focusrite scarlett 2i2 3rd gen, a USB DAC.