I build a well know Karlsen 24 db ladder filter in PD, and I really like the sound of it. It's in a stabil, working state, but it uses around 9-10 % of my older Macbook. I've tried a few things for optimisation but I am kind of stuck atm, I can't optimise it anymore, with the knowledge I have(hobby coder/Pd user).

I was wondering if anyone have suggestions for optimisation?

Besides that, merry christmas to everyone

Jakob!

The filter can be found here:
https://github.com/jaffasplaffa/Pure-data-patches/tree/master/Karlsen-Ladder-24-db

spoiler.pd
!filter gets unstable at higher frequencies

also, do you notice any difference in sound with oversampling + anti aliasing filter..?
maybe make it optional and put the anti aliasing filter in a subpatch with [switch~]

Merry christmas to you too!

why fexpr~ for a simple onepole filter.. when your patch already runs with blocksize 1? you can replace [fexpr~] with [lop~] or [rpole~] or some delays + math like in the original code.
I couldn't resit building the thing following this example: https://www.musicdsp.org/en/latest/Filters/240-karlsen-fast-ladder.html

spoiler.pd
!filter gets unstable at higher frequencies

also, do you notice any difference in sound with oversampling + anti aliasing filter..?
maybe make it optional and put the anti aliasing filter in a subpatch with [switch~]

Merry christmas to you too!

I tried removing the block~ 1 a while ago, for the reason you mention, that fexpr~ is already processing every sample, but I did not get the same result, So kind of scratched it, but I will look into it again.

I am still beginner with filters, but the reason I used fexpr~ was because of the "processes every sample", so I was kind og thinking this is what I need.

for the lop~ are you sure that these filters are made in the same way, the lop~ and the karlsen filter? There are a few filters that are made feom one pole filters on musicdsp, which are all a bit different. I am thinking the difference makes up different result?

For the oversampling usually oversampling makes sure that it doesn't blow up at higher frequencies. For example bob~, if you don't oversample it blows up too. With oversampling, it's much more stabil.

Ahh yes I should probably do the switch thing for the oversampling/butterworth filter. For now I just set the oversampling to 1, which is no oversampling, but yeah the filter is still running, you are right.

Anyway, thanks I will check your version and you gave me some food for thought here. Some things to check out

edit: (btw I recommend using some pd where signal and control in/outlets are distinguishable from each other, either Purr-Data or my mod https://github.com/sebshader/pdnext/releases)

edit2: maybe the 2 [*~ ]s to raise to the 4th power in the 4x1-pole subpatch should be replaced by [expr~ $v1*$v2*$v2*$v2*$v2], I never know which will be more efficient..

edit3: seems like all these changes make it a ~3-4 1/10 ms/s "cputime" more efficient here

edit4: I updated the patch so cutoff stays roughly the same when changing the upsampling factor

Solipps versions seems a bit unstable, but I'll try oversampling it and see how that works out

Seb's version is by listening pretty much the same as my version, except it's "cheaper", mine uses around 8-9% cpu, Seb's uses only 4-5%. Thats a noticeable performance gain.

I was not really sure how those pole filters works, but will check them out in the future, instead of fexpr~.

For the oversampling and the filter/butterworth, I think the only thing it actually does in this filter is expanding the frequency range. And also affect the hipass filter. It actually sounds nice and does not blow up even at standard sample rate, within the range I've set for each param. Which is nice.

The plan was to use it for a bass like synth so I don't really need those high frequencies anyway

Anyway thanks for the tips

edit: changing the blocksize to account for the upsampling didn't work well when I tried, idk what to do but just be aware that the feedback time/cutoff are dependent on samplerate, not sure exactly how to fix it atm

I tried removing the block~ 1 a while ago, for the reason you mention, that fexpr~ is already processing every sample, but I did not get the same result, So kind of scratched it, but I will look into it again.

you need the block~ 1 for 1 sample feedback delay in a resonant filter

I am still beginner with filters, but the reason I used fexpr~ was because of the "processes every sample", so I was kind og thinking this is what I need.

for the lop~ are you sure that these filters are made in the same way, the lop~ and the karlsen filter?

yes, they are exactly the same. you can just stack 4 lop~ an feed back the output with one sample delay. or, if you want signal input for cutoff use rpole~ like this:

rpole is like [fexpr~ $x1 + $x2 * $y1] (see help) with the additional math it is what the fexpr~ in your patch does. same as lop~..

For the oversampling usually oversampling makes sure that it doesn't blow up at higher frequencies.

[clip~ 0.0001 0.8] prevents your patch from blowing up.

Thanks for the info, more food for thought

Either way, a great sounding filter as is and a definite keeper.

you might want to account for how many more times the feedback happens too, seems like at higher samplerate the resonance gets lower
edit: never mind I'm wrong the resonance is fine

Checking your message again. I think I need to try the lop~ version too. So many options here, always great to learn new things

I always stayed away from filter design cause it seemed a bit overwhelming, as a hobbyist

Try Sebs version, it's the lightest one for now, I think

Thanks for the info. Was actually wondering why/how the oscillators got the shapes it has on some analog synths. Need to look into that too then.

I have been working on a Model-D clone, not measuring anything, just copying the signal path, but might as well get as close as possible

Sure, you can use it for Automatism

Thinking about maybe getting a band pass filter out of it too and play around with the even harmonics saturation.

So maybe it will be updated, but it works pretty well as it is now, so if you can use it as it is now, go ahead

Looks fine

Here is an image of the synth project I am working on. Need to fix a bunch of stuff, though. The oscillators I use screws up completely when modulating them in some cases. And it uses WAY to much cpu, so also working on that.

Anyway, here it is:

Yes and in some cases, if you use one of the oscillators as lfo and send the output back to itself(kind of fm feedback) it acts weird. I have seen it work in other projects, so I think I just need to try some other bandwidth limited oscs.

Thanks. I did that in some other filters I made for VCV, from the same site, it did make the filter sound the same on all oversample rates

Thanks man that's really cool. Those looks nice

Will try it out tommorow, when I have more time.

For now, MERRY Christmas!