Hi all !
Just curious.
=> is there a VST3 / AU “Oversampling” Plugin that you can put at the end of a chain to apply, say, 2x or 4x or 8x or 16x Oversampling in Real Time to the entire Rack Content ?
Thanks.
Alexi
I think if you place your plugins inside the DDMF Metaplugin, then yes you can.
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How would you use this?
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Thanks. Am aware of DDMF - not what I am thingking about.
I’m thinking / envisaging if (?) a normal VST3 / AU “Oversampling” Plugin exists that can go at the end of a Rackspace - just like a Reverb - which can - in real-time - apply a selected Oversampling Rate [ 2x, 4x, 8x, 16x ] to all audio being fed into and through it ?
Somnething like this => Voxengo Overtone GEQ - Features & Compatibility
I’ll try it out
- it has zero latecny processing
- and adjustable oversampling up to 8x
So am thinking run the EQ and all other settings FLAT and just crank up the Oversampling ?
I’ll report back 
I that that’s not going to work. A plugin can internally up-sample and then do some processing, but before the samples leave the plugin the buffer has to be down-sampled: the plugin host is expecting the configured sample rate. That’s why the ddmf- meta plugin works for all the plugins that running inside of it: it up-samples, then put the data through the embedded plugin chain (these plugins are running at the higher sample-rate, because these are hosted by ddmf-meta) and at the end the output is down-sampled.
But if you want to use higher sample rates: the easiest way to do that is cranking up the sample rate for the audio device.
Edit:
As @dhj already asked: what do you expect from oversampling? As far as I can tell, it only can do good if applied at the beginning of the plugin chain. Then plugins sensitive to aliasing can benefit from it. For oversampling at the end of the chain I doubt if it’s beneficial.
Anyway: keep in mind that using higher sample rates also means higher cpu utilization. If you double the sample rate, it will double the cpu usage.
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As @dhj asked, how (and why) would you do this?
To reduce aliasing from a non‑linear plugin, the oversampling has to happen inside or around that specific process so the harmonics are generated at the higher rate and filtered before they fold into the audible band. As @Frank1119 said, if you only oversample right at the very end, all the aliasing is already baked into the signal at your original sample rate, so it just gets “zoomed in” and back out again; it doesn’t magically become removable with a filter.
However, to answer your question, there are a number providers that offer chain/”container” oversampling, like DDMF, or in some cases even per-plugin oversampling in a chainer, like Blue Cat Patchwork. Some DAWs offer similar solutions, like Reaper (has per‑plugin and per‑FX‑chain oversampling options, effectively acting like a built‑in oversampling container around third‑party plugins) or Cakewalk (Offer track-/plugin‑level higher‑rate processing to similar effect). These are just a few potential solutions if you need something like this.
If/when you determine you actually need and can benefit from oversampling for a specific problem, you can identify the solution for it.
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