VST Hosts That Create Latency?

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Noticed on Windows 11 certain DAWs (Ableton 12, Serato Studio) are limiting your ASIO soundcard sample buffer from being set any lower than 32.

Any others?
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Just so that I understand you: The mentioned DAWs don't allow something which you can set in your audio interfaces ASIO settings?

Also, do you understand the point of a sample buffer?

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If any software is not accepting asio buffers less than 32 samples, I'd say it's for good reasons and the right thing to do.

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I also wonder why you'd need a buffer size lower than 32. That's nuts.

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https://www.sweetwater.com/sweetcare/ar ... in-my-daw/

Good article :// You can usually raise the buffer size up to 256 samples without detecting much latency in the signal.//

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chk071 wrote: Mon Mar 04, 2024 1:24 pm I also wonder why you'd need a buffer size lower than 32. That's nuts.
FX Increase RTL. You want as low stable buffer as possible to start
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electro wrote: Wed Mar 06, 2024 10:04 pm
chk071 wrote: Mon Mar 04, 2024 1:24 pm I also wonder why you'd need a buffer size lower than 32. That's nuts.
FX Increase RTL. You want as low stable buffer as possible to start
Most daws have ability to turn off pdc while recording, to only have hardware interface limitations for RTL.

There is really no need to run a full mix to record further as far as I discovered.
- just the ones needed for the right feel

StudioOne always add an extra asio buffer no matter what you do.

Cubase has this Asio Guard where you can decide which plugins are to alter behavior.

Reaper add asio buffers as needed as soon as any plugin has any latency. One as standard, then if just a single sample on a plugin, extra buffers are added for monitoring.
- using Reaper for a bit 10 years ago I had to go from 128 to 64 samples because of this

Samplitude has a full table of options that need a masters degree almost to pick the right options for monitoring while recording.

Using Sonar I never actually set buffer in program, just in RME panel outside any software.
- Sonar just use what is there

And the usuall often forgotten feature
- use a daw with smooth freeze options
- Cakewalk/Sonar is the best IMO

If some plugin in general add a lot of latency, just freeze that track for the purpose of recording again.
- in Sonar this is one click to freeze and one click to unfreeze

A year ago I got a new computer and tested some new interfaces over usb.
- tried going down below my standard 64 samples too
- cpu doubled each step going down to 32 or 16 samples
- so settled for 64 samples which is no problem at all

Just a few ideas....

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When you want to use the DAW on realtime incoming audio like a analog board with virtual inserts and sends, the latency through plugins needs to be super low
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electro wrote: Thu Mar 07, 2024 3:31 pm When you want to use the DAW on realtime incoming audio like a analog board with virtual inserts and sends, the latency through plugins needs to be super low
If RTL is below 10ms don't worry about it.

I looked at the eDrumkit training a while ago and the gated practice you can do
- if keeping everything at +/- 26 ms at 90 bpm you are considered a Pro
- so keeping all hits within that from click, your hit is not muted

Just to put things in perspective a bit.
- selecting a daw over that going below 32 samples is over the top
- pick any but StudioOne or Reaper and you are fine

With 64 samples 48k asio buffer I am at 5 ms RTL.
- then add 256 samples plugins to that and you are still fine

Note that having 256 on every track does not make pdc more than 256.
- I think it is only pitch correcting plugins that go over that
Last edited by lfm on Thu Mar 07, 2024 5:34 pm, edited 1 time in total.

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electro wrote: Thu Mar 07, 2024 3:31 pm When you want to use the DAW on realtime incoming audio like a analog board with virtual inserts and sends, the latency through plugins needs to be super low
How is that different to audio coming from in the box? Isn't it the output you should be worried about?

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chk071 wrote: Thu Mar 07, 2024 5:18 pm
electro wrote: Thu Mar 07, 2024 3:31 pm When you want to use the DAW on realtime incoming audio like a analog board with virtual inserts and sends, the latency through plugins needs to be super low
How is that different to audio coming from in the box? Isn't it the output you should be worried about?
Not when you want realtime incoming audio in sync with a slaved DAW stepsequencer
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