Your Preferred Sampling Frequency

What ist your preferred sampling frequency and why?

48 kHz, 24bit, 256 byte buffer. I A/B’d the 44.1 and 48 sampling rates and prefer the 48kHz one, sound slightly more there to me, somewhat less squished dynamically speaking. I also experimented with 96 but that added nothing to my (not young) ears so 48 it has been ever since :wink:

The minimum sampling frequency I use is 48KHz, which matches my interface. In Rack V1 I ran some performance tests, and it seemed 48 was a bit more CPU efficient than 44, presumably because the Audio module did not need to resample. I haven’t seen that effect in V2, but I still fall back to 48 if I have a CPU hungry patch.

In general I run as fast as my machine will allow, depending on the complexity of the patch. This has nothing to do with inherent frequencies to the patch, as my 60 year old ears cannot hear above 12KHz. But a great many patches manipulate the audio in a way that introduces ultrasonic frequencies that get aliased in an inharmonic way into audio territory, and typically I do not want that effect. Running at high sampling rates significantly reduces or eliminates this effect - most if not all the aliasing remains in the ultrasonic territory.

My machine can run most of my patches at 96KHz, so that is generally where I leave it. But some patches with extreme FM or wave folding, or wave splicing, or … may have significant aliasing issues at 96KHz. Running at 192 or even 384KHz can make a world of difference - as long as the patch is not too complex so my machine can handle those high sample rates.

48 kHz, because Dan Worral said it’s usually the best choice on Youtube.

In Rack we usually don’t have oversampled modules, so higher samplerates might be better, especially with modules that tend to alias - the next step up would use twice the cpu, though.

Maybe it would be a nice feature if we could oversample specific modules individually.


48kHz, 24 bit, 128 samples

128 samples buffer size to lower latency if trying to monitor while doing vocals or guitar.

24 bit to give a little overhead if I am recording quietly.

48kHz because I’m in my fifties and anything higher would be wasted on my ears. But mainly in practical terms my three ADA8200 micpres are linked to my interface via ADAT and they don’t go higher than 48kHz. Even ones that do, like my Focusrite, you then have to trade odd number of channels e.g. only 4 ins and outs at 96kHz as opposed to 8 ins and outs at 48kHz,

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Ah, the age old controversy! Ignoring the question of whether one can hear over 20k (not that argument again!), there are other reasons higher rates can sound better. One is that it’s a very crude way or reducing aliasing from poorly designed plugins. The other is that filter responses can get warped at high frequencies, so filters either need to adapt to that (very rare and difficult), or you have to not hear it (my solution), or you can go up to 96k or higher.

I think someone else posted this nice video before. It talks about using high sample rate to fight aliasing (and why it’s a bad idea). Also mentions at the end that in the real world it’s difficult to hear the difference: Samplerates: the higher the better, right? - YouTube

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literally two posts up from yours :slight_smile:


Many of mine are/were. More would be in VCV if people cared, I think.


Hmm is ∞Hz too much? :laughing:

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