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Jovo  
#1 Posted : Saturday, August 15, 2020 8:57:11 AM(UTC)
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Hello!

A fair number of people are using ASIO devices in their production workflow.

In my case, I'm using a Behringer X32 alongside vMix Call. 7 guests feed out via ASIO to the X32, processed and mixed down to a bus, then returned via ASIO to vMix.

With the forced buffer of 1024, using both input and output, this results in a *lot* of delay, even with Safe mode off.

The X-USB utility lets you set the buffer, but loading a show file overwrites the setting.

So, can we please have the ability to set it ourselves? Those of us with the hardware to support a smaller buffer would benefit greatly.

Thanks!
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stefan svanstrom  
#2 Posted : Wednesday, September 16, 2020 3:56:37 PM(UTC)
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+1 we have the same problem with our umc 1820 behringer, so it is poor behringer asio drivers.

So i hope vmix can sort this out togheter with behringer. Cause it would be great with low latency mode for alot if aplications, like sport commentary when the comentator is not in picture

Then you could use vst plugins to make the sound better, your cant do that with 1024 audio buffer.
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mr_z  
#3 Posted : Thursday, September 17, 2020 10:23:45 AM(UTC)
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Hello,

I'm having the same issue using a Focusrite Scarlett 18i20 Gen3. vMIX does not respect the buffer size as set in the "Focusrite Device Settings" application. Regardless of what is set on the Focusrite, vMIX is changing buffer size to 960, which is bizarre considering it's not even an option available in the Focusrite app.
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mavik  
#4 Posted : Thursday, September 17, 2020 4:59:08 PM(UTC)
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+1
using the behringer xr 18
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admin  
#5 Posted : Thursday, September 17, 2020 6:59:29 PM(UTC)
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Hi,

For important technical reasons this will not be possible. A minimum buffer of 960 is required to avoid dropouts.
Lower latency may be possible on systems only processing audio, but since live video processing is very taxing on
system resources, low latency would just cause audio dropouts and glitches constantly.

We don't do this to irritate users, honest! But is there for a very import reason.

Regards,

Martin
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thanks 3 users thanked admin for this useful post.
stefan svanstrom on 9/17/2020(UTC), eduardocfs on 9/19/2020(UTC), elvis55 on 10/23/2020(UTC)
stefan svanstrom  
#6 Posted : Thursday, September 17, 2020 7:31:28 PM(UTC)
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Originally Posted by: admin Go to Quoted Post
Hi, thanx for the answer martin, its difficult for a newbe to understand. I come from the audio side, so we have been spoiled with very low buffer settings for years

So well it have to be 2 computers then to cope with this, not a big deal just a practical issue, big thanks for reply

I am looking forward to vmix 24, i need 1 more replay channel for our ice hockey broadcasts.

For important technical reasons this will not be possible. A minimum buffer of 960 is required to avoid dropouts.
Lower latency may be possible on systems only processing audio, but since live video processing is very taxing on
system resources, low latency would just cause audio dropouts and glitches constantly.

We don't do this to irritate users, honest! But is there for a very import reason.

Regards,

Martin
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mr_z  
#7 Posted : Friday, September 18, 2020 5:52:59 AM(UTC)
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Originally Posted by: admin Go to Quoted Post
Hi,

For important technical reasons this will not be possible. A minimum buffer of 960 is required to avoid dropouts.
Lower latency may be possible on systems only processing audio, but since live video processing is very taxing on
system resources, low latency would just cause audio dropouts and glitches constantly.

We don't do this to irritate users, honest! But is there for a very import reason.

Regards,

Martin


Hey Martin,

I appreciate the answer! I think it would help if you could provide us with just a bit of technical information as to why. OBS/Streamlabs doesn't appear to have the same issue (at least during my testing), even running dual 4K UHD @ 60FPS, so helping us understand the fundamentals would go a long way to easing our minds.

Thank you
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paco3346  
#8 Posted : Tuesday, November 24, 2020 3:25:53 PM(UTC)
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Originally Posted by: admin Go to Quoted Post
Hi,

For important technical reasons this will not be possible. A minimum buffer of 960 is required to avoid dropouts.
Lower latency may be possible on systems only processing audio, but since live video processing is very taxing on
system resources, low latency would just cause audio dropouts and glitches constantly.

We don't do this to irritate users, honest! But is there for a very import reason.

Regards,

Martin


This perplexes me. Of all the advanced settings that we aren't protected from where we can shoot ourselves in the foot this is the one that we're prevented from changing?
I honestly don't get it.
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Steffoslovakien  
#9 Posted : Sunday, January 24, 2021 2:45:03 PM(UTC)
Steffoslovakien

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This perplexes me to. The main audio delay from my Focusrite force me to delay my SDI video inputs and that makes my computer working much harder.
A lower audio buffer size (in my world) should be more gentle to the system.

This issue is a big deal for the VMix team to solve. Don´t force us to leave you for another Video Production solution!
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JanDeWever  
#10 Posted : Tuesday, February 23, 2021 10:50:24 PM(UTC)
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This perplexes me too! We're trying to do the opposite, but same issue:

  • bring in audio (Ravenna audiostream, latency no issue there, we checked!) from remote location (over dark fiber) to Lawo Virtual Patch Bay
  • from VPB, bring this audiosource via asio into vMix (in vMix audio set to asio, etc ...)
  • mix/cut/... in vMix
  • vMix PGM out goes as asio back to VPB.
  • in VPB we stream that PGM audio again as Ravenna 'away'.
  • end-to-end latency is 170ms (time between microphone to headphones, with 2 times Ravenna, VPB and vMix (with asio) in between. Unusable off course. We're now looking into several workarounds with external mixers, but all of this could be solved if vMix behaved nicely, not introducing extra delay.


We're doing this on a very beefy machine, offloading to an RTX4000 GPU etc ... and now we stumble on vMix not able to handle low latency audio (sub 10ms delay for audioprocessing)
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tom@24bit.io  
#11 Posted : Friday, November 12, 2021 12:21:15 AM(UTC)
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I also need to be able to change this setting for troubleshooting purposes. I've had a client suffering from the kind of 'robotic' sound that is often caused by a sample rate mismatch, but when coming from an audio interface could also be caused (I think) by a buffer overflow.
The symptoms are described quite well here: https://obsproject.com/f...t-audio-interface.61400/
I'm using an A&H Qu-SB with vMix, but I've had this same problem with my Motu interface and various other software.
So I want to try changing the buffer size up to 2048 because latency isn't an issue for them. But no, I'm not allowed.
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