sallen key filter ..please

As topic says …would be fantastic.

PIcture shows a resonant saw wave trough a sallen key filter ( ms 20 ) , someone stated that the redux k35 is based on a korg filter , but it probably aint based on the ms 20 .

Kraken can you give some more info ?


Maybe you are talking about me? :slight_smile:
I could find any info about k35. I know this filter from Caustic and nothing more…So i thought that it can be ms10 / ms20 / monotron filter. It isnt exactly same but you can get something very similar (if you add DRIVE). The biggest difference is in lowend but i am ok with it :-).
BUT I would love to hear something more about filters.

Maybe you are talking about me? :slight_smile:
I could find any info about k35. I know this filter from Caustic and nothing more…So i thought that it can be ms10 / ms20 / monotron filter. It isnt exactly same but you can get something very similar (if you add DRIVE). The biggest difference is in lowend but i am ok with it :-).
BUT I would love to hear something more about filters.

Ta’s te point , you don’t get anyting similiar wen you ad drive .
I like te sound of the filter , but I wouldn’t call it similar to a sallen key filter
What all the shown examples ( synths ) have in common (except the redux k filter ); is a resonance that can go into self oscilation.
When the input gain is then increased you get those typical resonant shapes (see osciloscope).
I tried increasing the sample gain in redux before going into the k filter , still not able to mimic that behaviour .
THe redux ‘k filter’ shows the same behaviour as ‘cytomic the drop’ ms filter when it’s resonance is set to safe mode( preventing from self oscilation.) ,or any other shown example when resonance is below self oscilation .

The most info i can get about K35 filter is from this forum.

This is essential.
"1) The point of the K35 design, and all of my latest designs, has involved simulating either the actual or the conceptual signal flow diagrams of filters. Typically, this is not the same as simulating the analog block-diagram. As you point out, theoretically, a second order filter would only require 2 integrators/reactive components. If you only analyze the analog transfer function H(s) for a 2nd order filter directly, you will arrive at either the SVF or Tow-Thomas bi-quad versions (see VanValkenberg’s book for excellent derivations). If that were the case, then why do analog filter books have dozens or sometimes even scores of other block-diagrams/circuits that produce the exact same frequency response, often times with much more complex structures? The answer there has to do with signal flow and sensitivity to circuit component variations, and as I point out in the synth book the “problem at hand” for the analog designer, i.e. the end application. Of course, we don’t worry about component variation. My curiosity lies more in high-level analog signal flow rather than lower level block diagram forms. For me, minimizing components or integrators is not as interesting as coming up with different/unique filters that have never been realized before.

In the case of the K35, I noticed that (1) The SVF self self-oscillates only when the inner feedback is -1.0 and if you use the equations for this, your knob/slider would need to go from 0 to infinity, or you would need a switch that turned the inner feedback to -1.0 after some point (Q>1,000,000?). The K35 is built around the sallen-key architecture which has a simple, continuous Q control that moves from no resonance to self-oscillation linearly as K goes from ~0 to 2.0. Thus the SVF structure isn’t going to have the same “knob feel” as the K35. (2) The SVF would never distort the same way as the K35 with the saturator in the integrator as in Zavalishin. The K35 signal-flow+saturation reveals how the distortion occurs.

Since I have an MS-20 for comparison, I spent some time turning knobs with my eyes closed (on the MS-20 vs on a MIDI controller mapped to RAFX parameters). The idea here is to simulate the user-experience with tactile control. The SVF doesn’t have the same tactile response, or the same distortion on the output, as the K35. If you wanted to, you could replace the DF1 blocks with TPT blocks, but you would still have 3 delay elements. This reveals the difference between signal flow charts and block-diagrams based off solving the differential equation for the filter. Solving the differential equation gives you one structure. But, many different signal flow diagrams can implement the same analog bock diagram structure. This is what interests me.

(2) Early on, I had several students as well as book readers who emailed me, who were mis-informed about the delay-less loop resolution. They were under the impression that the delay-less loop resolution could only be done using the TPT structure. The reality is, as pointed out the Harma paper, that the structure only needs to be in a linear form with current + stored outputs. I have even used FIR filters in the structures since they can also be fashioned into this format. After doing the modified Harma derivation, the very first thing I did was to implement the TPT-Moog ladder filter along with the DF1 (biquad) version in the Harma paper, and modulate their cutoffs with a sawtooth LFO that has a discontinuity, then use RAFX’s “Process into Wave File” to process and capture both outputs with the same stimulus. I observed several things. (a) I could not hear any difference in the two. I could not hear “artifacts” or anything else artificial in the DF1 version. How would you even quantify “artifact” here? How would you detect and measure these artifacts? (b) the two filters have identical frequency responses. © the two filters DO have different time-responses to modulation. I found this by subtracting the two filtered outputs and finding the residual. Sure enough, there were differences right at the discontinuity in the LFO. I would not call these “artifacts,” just the difference in the two outputs. The residual’s RMS value was about -45dB, so these differences were quite low in amplitude. You would really need to do comparisons against actual analog circuits to make meaningful statements about artifacts or other distortions/etc and so far, I have not seen any work done on that, but would enjoy seeing it if anyone wants to do that! Right off the bat, that sets up many problems. Making comparisons with analog circuits and doing meaningful analysis is difficult due to the analog vs. digital nature of the two.

In the case of the Novel Filters paper, there are no analog structures to compare the filters with. Again, how do you quantify “artifacts” here with nothing to compare with? Ultimately, this has become my new focus - generating new filters/signal processors that have never existed before. And, yes, none of the filters in the Novel Filters paper use the TPT structure; this was on purpose to show that you don’t need these structures to synthesize interesting filters with delay-free loops.

Please note that I am not downplaying the significance of the TPT structure - after all, every single filter in the Synth Book uses the TPT version! But after doing a lot of my own experiments and listening tests, and after going back to Lindquist’s analog book and re-deriving the signal flow of his various filter “classes” and working on my own new variations, I decided to let others worry about the TPT “artifact” argument (and there is a lot of arguing going on, just peruse the music.dsp lists…) and move forward and design new filters no one has seen before.

I hope that sheds light on where I am coming from as well as where I am moving towards.

All the best,


Bumping this one ( to kraken/gore )

COuld it be possible to make the k filter sound more like an ms filter ?