# Distortion

How are you using distortion in or with the 301?

I like to play with the limiter a lot. Have them sometimes all over the placeâŚ

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CoolâŚ do you tweak them at all?

Itâs kindaa a shame that the moduleâs latency seems to reshape feedback, because total control over the eq sounds like itâs be really helpful with noise

Before delving into this, I wondered if anyone knew / could explain how traditional analog distortion pedals or modules work or are designedâŚ I already know that distortion chops the tops off of wave forms, either hard or soft â reducing the signal when it goes above a certain level but allowing it to increase above that level. But I wondered, if you went to a distortion exhibition and people were discussing their distortion products, how those different distortions would be described or defined.

Is there a language for talking about distortion â describing the attenuation slope or whatever?

Any time you filter or process a signal itâs a type of âdistortionâ, but people usually mean the kind of saturating distortion that you get from running a amplifier past its limits.

What that means from a signal point of view is that there are absolute voltage limits on the signal imposed by the ampâs electrical design and as you crank the gain on the input signal more and more it starts to get cut off when the amp canât track it any higher.

So as a very simple math equation, the most basic clipping distortion is like

distorted = max(min(limit, signal*gain), -limit)

If signal is a sine wave (which oscillates between +1 and -1 with the familiar wavy wave) and limit is 1, when gain is 1 or less the signal will pass through unaltered. When gain goes greater than 1, the top and bottom start to get flattened. When gain is really high, the distorted signal becomes a square wave.

Harmonically, this means that you go from a single harmonic value for the pure sine wave up to an infinite odd harmonic series for the square wave. This means you are adding increasing amounts of high frequency content as the wave gets clipped more and more.

In reality, there is a family of other functions besides straight clipping that gives different character to distortions. Also, the positive and negative limits donât have to be the same, and the clipping functions donât have to be the same for the positive and negative half. That actually happens in real amplifiers because of the details of the electrical design. And there are other effects of running amps at their limits, like compression, weird frequency dependent effects, and other kinds of details that make for a fascinating journey of discovery

You can use the sample scanner to implement various kinds of symmetric or asymmetric clipping functions if you are comfortable doing a little waveform math.

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Super useful passage of detail concerning distortion, thankyou. Harmonics are the âthingâ I go searching for in distortion patches that almost always involve a feedback path. And what fascinates me always is the idea that a sine wave, with its complete lack of harmonic character, is the source of much harmonic potential when clipped or folded or distorted any which way. Will definitely be taking another look at the sample scanner!

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The interesting thing about creating new harmonic content is that it can only happen from ânonlinearâ processes. Adding signals together, gain on a signal, âpureâ filtering (lowpass or highpass) can only increase or decrease whatâs already there, they canât create new content.

To add harmonic content you have to do something nonlinear: multiply signals together (ring mod/VCA), take absolute value (rectification), or clip/truncate/limit (distortion/bitcrush/fuzz). And the ear is sooooo sensitive to that added content, you can just barely limit or clip a signal and itâs instantly audible.

Then we get to pass it back through a cool LPF to get rid of that new content we added

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of course. why wouldnât you?

• i usually start with a round limiter (inv. sqrt) at the end of a given chain
and just turn the post gain between -6 to -2dB. i do this mostly to
save the output dynamics from excessive peaks.
• then i put a second (the distorting) limiter before the first and leave the post gain at -2dB
while messing (excessively) with the pre-gain.
• i also like to add a manually or cv controlled vca in front of those two limiters
so i can drive the input (of the distorting limiter) even more
• this is also good for a feedback path of a delay or probably for any other feedback path for that matterâŚ
• or in any combination with any other module

this alone gives you plenty of possibilities for saturation and âdistortionâ techniques. but wait until youâve tried
on these things when you search for itâŚ

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how do you do this in the 301???

cheers.

You can just use the âlimiterâ unit set to âhardâ. It will clip values outside the threshold range. The pre-gain control is the âgainâ I mention. The thresholds are set to +/- 1.

You can implement asymmetrical limiting using more than 1 limiter unit in a chain with an offset, and managing the pre- and post-gain carefully. The limiter unit also has different curves other than hard-clip which behave slightly differently.

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oh right, i get it

I think we should keep asking @bgribble questions.

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He seems to know something about distortion.

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confusing does this link to âsubtractive synthesisâ

I just meant that any âchangeâ in a signal can be thought of as âdistortionâ since thatâs sort of what the word âdistortionâ means â something distorted, or changed. So filtering, wave folding, anything you do to a signal is distorting it.

When people use the word âdistortionâ in a music or sound design context they usually mean saturating/clipping distortion like we have been discussing. But as modular nerds know, thatâs just one very narrow slice of all the ways you can interestingly mangle a signal!

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yeah, i get itâŚ cheers

When I produce my complex basedrum layers for my production or my own sample pool, I love to use mixer units including custom set eqÂ´s for high passing, shaping and cutting my bassdrum layers. Every layer has a limiter unit after the eqÂ´s for compressing/clipping. the way the limiter compress (in cubic type) and soft clip the signal sounds amazing if I do it right. Before I bought the 301 I have done this in my DAW and had to put my samples back in an Erica synths Pico drum or similar sampler, now I can translate my workflow in the 301. This is faster and more creative for my live performance or forthcoming productions and I have less to work on de DAW and can focus more on my rack.

Regards Dominik

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âsoft clip the signalâ thatâs what i meant - how to [infinitely] vary soft clipping along some recognized parameters

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