Looking for resources which explain how filters work
Hi!
So I'm looking for explanations on how filters work (one-pole/onezero and biquad). I'm coming across a lot of videos and documentations which explain it very mathematically. Can anyone give me some resources where they don't go too much into depth with the mathematics and rather explain what happens with sound and how it works?
explaing it without math is quite difficult because that´s what it is.
but of course if you read up on "FIR" or "biquad" in a book, the math shown there consists for 80% of the parameter calculation, while the "kernel" of these filters would be at least a much easier mathematical process.
i am convinced that you do not need to understand the z-transform to use a biquad, actually you do not even need to understand it to program your own.
maybe instead of "how does it work" we should start with "why does it work."
the music signal you are processing already contains all the mathematical values which the filter will change when processing the signal.
the simplest example for a frequency filter is interpolation ( [slide~ 2 2] object).
it performs something like "this sample value plus half of the next sample value plus half of the last sample value = output.)
if the input signal is a sinus of 50Hz, nothing relevant will happen if this calculation is applied. but if the input signal is a sinus tone of 22 kHz, its amplitude will be lowered for around 30%.
now if your input audio has both tones, 50 and 22k, the same happens.
this is more or less how frequency filters can change the amplitude spectrum of an input signal. higher frequencies means faster changes between adjecent samples. this is what the filter is looking for and develops its effect.
this series is brilliant:
^some of it is still difficult for me to understand but i feel that after repeat-readings they become more and more clear to me(sorry in advance if you've already seen em).
another way i think about how 'audio filters' work, in general, whether analog or digital, is to look at a slower/simpler moving style of filter, such as 'flanger', 'phaser', or 'comb' style effect(see, within Max menubar: "Help->Examples->gen...", many examples you can look at in visual patching style), whereby delaying a signal, and then adding the delayed signal back onto its not-so-delayed self, will color the sound in slow or fast moving ways. starting from there, it's easier to see how more specific algorithms of this type of mixing in 'feedback'/'feedforward' techniques, can create more specific filtering.
(@raja i wanted to send you something, but you left the line? maybe you can write me a mail.)
(@Roman - ya, i left Lines because it was getting too cloistered and cliquish, all the mods there are getting strict to the point of whittling away all the interesting artistic/eccentric responses that kept it fun to begin with... Discord communities and email seem the best places to interact these days... anyways, just sent you an email)