In non-tunable analog filters, opamp (or transistor follower) clipping might be the main nonlinearity. That would often be rather hard clipping, just depending on the component. Some opamps would run out of oomph nearing the rails, and some opamps couldn't get all the way to the rails. Some opamps and perhaps voltage follower transistors had assymetrical drive-- Some could source more current than they could sink or vice-versa, so on large signals perhaps they could charge a capacitor harder than they could discharge it, or vice-versa..niarn wrote:Do you mean inserting non-linearities into the svf itself? like some saturation on the states? because this is what is happening in the analog svf?
In tunable filters directly controlled by potentiometers, such as parametric equalizers, or tunable speaker active crossovers or benchtop filter test sets, perhaps about the same conditions as above, unless someone can think of special cases.
In sweepable filters such as synthesizer filters, or auto-wah effects, or some flavors of de-esser, etc-- The job of the variable resistor had to be done by some form of "voltage controlled resistor" or "current controlled resistor". To my knowledge, at least in the old days, there was no ideal well-behaved "voltage controlled resistor". All options had various flaws and nonlinearities so far as I know.
Common methods included modulation of transistor transconductance, or diode transconductance, or OTA (operational transconductance amp), or CDS cell optoisolator.
Each voltage controlled resistor had its own characteristic nonlinearities. In a second order state variable filter there would be two of them sitting in the circuit waiting for an opportunity to distort.
Also, there were at least a couple of obvious ways to wire up a synth filter. The "clean well-behaved" wiring would have constant gain on the loudest part of the resonance peak. Not likely to blatantly long-term overload if you crank up the resonance. But this kind of filter didn't have much "balls". If you turn up the resonance of such a low-pass filter, it would turn down the gain of the passband, to keep the gain of the resonant peak constant. The higher you crank the resonance, the lower the gain in the passband. To some musicians it felt like turning up the resonance caused the sound to "lose power".
Another way of wiring, you always keep the passband gain the same, and when you crank up the resonance, the resonant peak gets loud as hell. Which if the filter did not distort somehow, and you sweep down thru a ramp wave at high resonance, the volume jumps when the resonance peak passes thru loud harmonics might hurt the ears or blow speakers. So some "sonorous" soft clipping was beneficial in getting a fat ballsy sound with a high-resonance filter without hurting people's ears.
**** There was one method that perhaps in theory would be fairly linear, a switched-capacitor scheme. Use a fast CMOS or FET switch in place of the tuning resistor, and drive it with an ultrasonic pulse train. Modulate the width or density of pulses to modulate the charging of the capacitor. But using old off the shelf parts, that was kinda a nasty circuit to build and get it to behave correctly. You normally wouldn't want high-amplitude ultrasonic digital signals sitting right beside an audio signal on a circuit board. Nowadays that is how digital power amplifiers work, and so maybe some of the digital power amp chips could somehow be re-purposed for use in tunable analog filters, dunno.
