[PD] Guitar distortion (analogue) was: (Chebyshev?)

Pierre Massat pimassat at gmail.com
Sat Oct 23 17:34:12 CEST 2010

This is very interesting. Although i must confess that i have no idea how to
model your feedback loop in Pd.
As for the lp filter, i have never used an upsampled filter. Is it enough if
i use the filter built in Miller's J07 Upsampling example? It's a 3rd order
butterworth lp and the subpatch is upsampled 16 times.
More generally, what is the best way to split a signal into low and high
frequencies? I thought of using fft, with, say, a gain equal to 1 for
frequencies between 0 and X Hz, and equal to 0 above. Would this work? This
would be the sharpest lp filter, wouldn't it?


2010/10/14 - <fallen_devil at gmx.de>

> Am 12.09.2010 19:24, schrieb Martin Peach:
> > On 2010-09-12 12:05, Mathieu Bouchard wrote:
> >> On Sun, 12 Sep 2010, Martin Peach wrote:
> >>
> >>> It's not the capacitors, it's the amplifier losing gain when it
> >>> approaches the power supply.
> >>
> >> Yeah, but it seems to be a pattern similar to the one found in
> >> capacitors, because capacitor theory has exp(-x) all over it, and the
> >> only way that capacitors behave like [hip~] is when the signal is much
> >> below the capacity rating (?F)... otherwise they lose gain... when they
> >> don't, it's because exp(-x) can be well approximated by x.
> >>
> >
> > I guess it's similar since capacitors charge at a rate proportional to a
> > voltage difference, while transistors can supply charge carriers at a
> > rate proportional to a voltage difference, so caps charge fastest when
> > they are nearly empty and transistors have the best gain with small
> > signal inputs.
> >
> > The whole universe has exp written all over it in fact...
> >
> >> And then, exp is very close to tanh in several different ways, one of
> >> them being this (use gnuplot) :
> >>
> >> plot [-2:2] [-1:1] exp(x*sqrt(2))-1, 1-exp(-x*sqrt(2)), tanh(x), x
> >>
> >> I put the plain 'x' at the end to show what I mean above (though you
> >> already know that)
> >>
> >
> > Of course, all the hyperbolic trig functions are made from exp, by
> > definition.
> >
> > http://en.wikipedia.org/wiki/Hyperbolic_function
> >
> > Another use of exp is the sigmoid function used in biology, that can be
> > used to make a soft transition from one state to another as in 'fuzzy
> > logic'.
> >
> > Martin
> Sorry I don't know (jet) how to answer a mail from the archives.
> The soft clipping in a normal analogue guitar distortion (like the tube
> screamer) comes from the diodes (tubes are warmer). Which have log
> written over them (their resistance depending on V). (Are even used some
> time as a simple analogue log function)
> A analogue guitar distortion is build via a feedback loop like this:
> |--var resistor (gain)
> |----  ->diode -|
> |----  <-diode -|
> |----cap as lp -|
> |               |
> |------ (-)     |
>       op-amp---|---out
> input-- (+)
> The resistor defines the max gain
> The cap is used as a lp (short cuts high frequency's to a gain of 1)
> An op-amp tries to have always the same voltage at both inlets.
> Which means if you have 1v at + it increases its output till you have 1v
> at - as well. 1+(-1)==0
> And finally the diodes:
> At negative voltages they blockade completely
> At low voltages they have a high resistance so the gain of the resistor
> counts. At around .7V their resistance drops leading to a lower gain.
> With a tanh(sig) you are simulating the diodes in a simple way.
> What usually is forgotten in the digital domain is the lp filtering. You
> may want to split your signal into a high and low part. Run the low part
> through the tanh and sum it later up with the high part.
> It gets really funny if you want to try to model the whole feedback loop
> in something like pd. There is a mess of phase changes and whatnot which
> surely is part of the interesting analogue sound.
> An important thing to mention is that, as someone else noted, you need
> to (should) oversample to avoid aliasing with high order functions. Tanh
> is one of the worst because it generates very high harmonics.
> Where in the analogue field a natural lp filtering happens all the time
> (and no aliasing can occur). As an Example: An opamp has a frequency
> response up to GHz. But only at a gain of 1. The higher the gain the
> lower the frequency response.
> I'm bad at explaining things. But if you want to i can search my
> bookmarks for the analogue or digital sources i tried to explain here.
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