notes
ATTN Admin: You should remove the old version named "Karlsen" on your website, and rather
code
 1 2 3 4 5 6 7 8 9 10 11 12 // An updated version of "Karlsen 24dB Filter" // This time, the fastest incarnation possible. // The very best greetings, Arif Ove Karlsen. // arifovekarlsen->hotmail.com b_rscl = b_buf4; if (b_rscl > 1) {b_rscl = 1;} b_in = (-b_rscl * b_rez) + b_in; b_buf1 = ((-b_buf1 + b_in1) * b_cut) + b_buf1; b_buf2 = ((-b_buf2 + b_buf1) * b_cut) + b_buf2; b_buf3 = ((-b_buf3 + b_buf2) * b_cut) + b_buf3; b_buf4 = ((-b_buf4 + b_buf3) * b_cut) + b_buf4; b_lpout = b_buf4;

Where are the coefficients? How do I set the cutoff frequency?
The parameters are:
b_cut - cutoff freq
b_rez - resonance
b_in1 - input

Cutoff is normalized frequency in rads (2*pi*cutoff/samplerate). Stability limit for b_cut is around 0.7-0.8.

There's a typo, the input is sometimes b_in, sometimes b_in1. Anyways why do you use a b_ prefix for all your variables? Wouldn't it be more easy to read like this:

resoclip = buf4; if (resoclip > 1) resoclip = 1;
in = in - (resoclip * res);
buf1 = ((in - buf1) * cut) + buf1;
buf2 = ((buf1 - buf2) * cut) + buf2;
buf3 = ((buf2 - buf3) * cut) + buf3;
buf4 = ((buf3 - buf4) * cut) + buf4;
lpout = buf4;

Also note that asymmetrical clipping gives you DC offset (at least that's what I get), so symmetrical clipping is better (and gives a much smoother sound).

-- peter schoffhauzer
Tee b_prefix is simply a procedure I began using when I started programming C. Influenced by the BEOS operating system. However it seemed to also make my code more readable, atleast to me. So I started using various prefixes for various things, making the variables easily reckognizable. Peter, everyone, I am now reachable on www.str8dsp.com - Do also check out the plugin offers there!
Here's even another filter, I will probably never get around to making any product with this one so here it is, pseudo-vintage diode ladder.

// limit resonance, rzl, tweak smearing with fltw, 0.3230 seems to be a good vintage sound.
in = in - rzl;
in = in + ((-in +kbuf1) * cutoff);
kbuf1 = in + ((-in + kbuf1) * fltw);
in = in + ((-in +kbuf2) * cutoff);
kbuf2 = in + ((-in + kbuf2) * fltw);
etc..
"Cutoff is normalized frequency in rads (2*pi*cutoff/samplerate):

This seems to be valid for very low ( < 200 Hz ) frequencies - higher sample rates seem to be "Closer"

thanks
I also did a 9th order gaussian filter (minimal phase), using only 5 orders, for my limiter, which is released under the GPL LICENCE. http://www.paradoxuncreated.com
• Date: 2012-11-14 08:15:06
• By: Generalized perfect digital “ladder” filter, with the desired aspects of analog.
Hi, I have now generalized the ladder filter, into fast code, and with the desired aspects of analog, but retaining digital perfectness.

Peace Be With You.
I have also moved domains now, and consolidated the information on this ultimate digital filter, with "analog sound", here:

http://ovekarlsen.com/Blog/abdullah-filter/

Peace Be With You!
• Date: 2016-02-14 01:31:32
• By: ove hy karlsen @ facebook.com
Karlsen Fast Ladder III - inspired by "transistors set to work as diode" type Roland filters. The best fast and non-nonsensical approximation of popular analog filter sound, as in for instance Roland SH-5, and the smaller TB-303.

//          // for nice low sat, or sharper type low deemphasis saturation, one can use a onepole shelf before the filter.
//          b_lf = b_lf + ((-b_lf + b_v) * b_lfcut); // b_lfcut 0..1
//          double b_lfhp = b_v - b_lf;
//          b_v = b_lf + (b_lf1hp * ((b_lfgain*0.5)+1));

double b_rez = b_aflt4 - b_v; // no attenuation with rez, makes a stabler filter.
b_v = b_v - (b_rez*b_fres); // b_fres = resonance amount. 0..4 typical "to selfoscillation", 0.6 covers a more saturated range.

double b_vnc = b_v; // clip, and adding back some nonclipped, to get a dynamic like analog.
if (b_v > 1) {b_v = 1;} else if (b_v < -1) {b_v = -1;}
b_v = b_vnc + ((-b_vnc + b_v) * 0.9840);

b_aflt1 = b_aflt1 + ((-b_aflt1 + b_v) * b_fenv); // straightforward 4 pole filter, (4 normalized feedback paths in series)
b_aflt2 = b_aflt2 + ((-b_aflt2 + b_aflt1) * b_fenv);
b_aflt3 = b_aflt3 + ((-b_aflt3 + b_aflt2) * b_fenv);
b_aflt4 = b_aflt4 + ((-b_aflt4 + b_aflt3) * b_fenv);
b_v = b_aflt4;

// Behave.
// Ove Hy Karlsen.
Hey Ove

I am wondering about the last filter the  Fast ladder diode III. Where is the input supposed to go?

Sorry, I am still learning and thanks for some great filters, btw :)

Thanks, Jakob