What goes into modeling synth filters like minimoog in Monark?

Modular Synth design and releases (Reaktor, SynthEdit, Tassman, etc.)
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Last edited by claudedefaren on Thu Apr 15, 2021 7:33 pm, edited 1 time in total.


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claudedefaren wrote: Thu Feb 06, 2020 4:17 pm What's the difference between a "sterile" digital filter and an analog modeled digital filter, what creates that?
Distortion mostly.

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Last edited by claudedefaren on Thu Apr 15, 2021 7:32 pm, edited 1 time in total.

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Short answer : NON linearities
You could easily do that yourself by researching .
There are numerous topics about filters , non linearities , bi-quad filters , zdf design etc...
Eyeball exchanging
Soul calibrating ..frequencies

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claudedefaren wrote: Thu Feb 06, 2020 8:25 pm
dave dove wrote: Thu Feb 06, 2020 7:54 pm this:
https://www.native-instruments.com/file ... _2.1.0.pdf

Dave
Awesome. If you've read that, then maybe you can give me the two paragraph layman's breakdown, rather than a 500 page dissertation :)
I started reading that a while ago as well. TBH, I didn't understand a word. Definitely something for more advanced users or developers. :)
The prerequisites for the reader include familiarity with the basic DSP concepts, complex algebra and the basic ideas of mathematical analysis

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claudedefaren wrote: Thu Feb 06, 2020 8:25 pm
dave dove wrote: Thu Feb 06, 2020 7:54 pm this:
https://www.native-instruments.com/file ... _2.1.0.pdf

Dave
Awesome. If you've read that, then maybe you can give me the two paragraph layman's breakdown, rather than a 500 page dissertation :)
real electronics have subtle variations, irregularities and nonlinearities, ergo so do filters made with them. idealised digital algorithms based on abstractions of the filter architecture, rather than a model of an actual filter, dont.
our brains perceive those variations as more interesting. because they are.
my other modular synth is a bugbrand

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in short:
as well as the non-linearities of electronic circuits an analogue resonant filter works by injecting feedback from the output back into the circuit - a bit like a delay
in an analogue circuit this all occurs simultaneously (to all intents and purposes)
a simple digital filter has to add a single sample delay into the feedback loop because it can't do the maths without it
this means it is always playing catch up and the final result is not accurate
using zavalishin's maths one can solve for the feedback without introducing the single sample delay
there are a few ways of doing this, some more cpu efficient than others

hope that's better for you

Dave

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The 'Zero Delay Feedback' technique that dave described requires some tricky maths, plus adding non-linearities (distortion / saturation) into the feedback path makes the maths more complicated.

Vadim Zavalishin created the ZDF Toolkit in Reaktor Core and it really is a thing of beauty!

Using the ZDF Toolkit you can create your own ZDF filters with feedback and saturation as easily as hooking up guitar pedals, no maths or advanced DSP knowledge needed. It's really fun to just noodle around with it. You can make changes to your filter's design in real-time and hear the changes instantly. I expect these techniques were used to design and fine tune the Monark filter.

There's a brilliant video of Vadim demonstrating the toolkit here:

https://www.youtube.com/watch?v=e61n8qZAOBU&t=60s

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or you can go hardcore and do the maths in reaktor core...
if you want to begin to understand the maths this is the easiest way in:
https://www.adsrsounds.com/reaktor-tuto ... n-reaktor/
is how i got started and was made before the zdf toolkit (i think)

dave

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Thanks for sharing this video David!
Best
YY

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