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Forums     Plugins     Impulse Modeler and Deconvolver Deconvolver

I have been using deconvolver, and successfully capturing my Lexicon reverbs.  Very exciting!

I tried to apply the same approach to capture a unique EQ in a preamp of mine that I really want to get because I can not replicate that in any software.

My efforts do not work.

I get a 1kb impulse file that has no effect on audio.

What do I do that is different for getting an eq impulse than a reverb impulse?

When capturing an EQ impulse you should add a fair amount of post silence (like you do with reverbs), or otherwise deconvolution won't capture the tail oscillation which actually creates the EQ effect.


I have success now

I am still understanding what the lack of captured dynamics has on an impulse compared to the original.

eg. the shimmering tail of my lexicon reverb is not captured.

And of course the valve distortion/compression before the EQ I am capturing ( therefore change in tonal balance and eq effect) does not dynamically effect the material processed with PristineSpace, like audio going through the hardware unit would

There is no PDF for 'Voxengo Deconvolver'?

What does "Minimum-phase transform option" mean?

I can see it creates a different looking impulse, but the effect sounds the same...?

Also I nearly rejected the PristineSpace for good, simply because despite claims of low latency it was not what I was experiencing.  It was only by accident I discovered the second menu.

Perhaps you may have this by default or make it clear?  I would not want you to lose sales because of this.  I am glad you will get mine.

Thanks for making great products

Each Voxengo product, including Deconvolver, has a user's manual in HTML form.  Minimum-phase transform removes a so called 'pre-ringing', preserves the frequency response, but totally transforms the phase response.

Pristine Space also has a user's manual which contains topic on global settings, including latency setting.  By default latency is set to 4096 samples - this mode is good as this latency is pretty low while CPU usage is efficient enough.  I.e. it is a good starting point.  Of course, you can lower the latency as you wish, but for a better performance you should also adjust audiocard buffersize in parallel.

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