Distortion (clipping)

Seems I can't stop working on my synth these days. There are two major parts missing from my synth voice cards: Digital playback and pre-filter distortion. I am not sure about the need for the last one, so I decided to build that first.

Googling distortion circuits I came across a resource that I have read long ago and forgotten about: Design your own distortion.

This page gives a great step by step introduction to two types of distortion: Hard and soft clipping. It also has some on filtering and tone control but it contains some errors (50nF cap in distortion does not give a lower frequency of 31Hz but 3.1kHz, the shown circuit will not give any distortion and indeed very little gain for frequences lower than 3.1kHz).

I decided to try simulating the circuit in LTSpice (with +/-15v supply lines) before breadboarding it, to get a feel of what it is doing.

Hard clipping

The hard clipping part of the circuit is easy. Two parallel diodes in reverse order from signal to ground after a buffer or similar that disconnects it from the input.

Hard clipping at around 750mV

What is going on? When the input signal, and thus the voltage across the diodes reaches the forward voltage drop of one of the diodes, it switches on, sinking the current to ground. This effectively cuts off the peaks of every wave cleanly. A fairly normal diode voltage drop is around 0.75V, meaning the signal is clipped at 0.75V. The amount of clipping is adjusted by attenuating the input (assuming a +/-5v input) to where you want the clipping to occur.

If one wants a higher clipping point it is also possible to connect diodes in series, going from 0.75V to 1.5V to 2.25V etc.

Hard clipping with series diodes, clips at around 1.5V

Soft clipping

This is slightly more involved. Soft clipping means that instead of brutally chopping off the wave tops, they are rounded off. Soft clipping can be achieved by placing two diodes in parallel / reverse order in the negative feedback loop of an op amp, parallel to the feedback resistor.

I first simulated inverting amplification instead of the non inverting "tube screamer" version on the web page because I could not get that one working due to the HP cutoff error in the article.

Here is what I think is going on, and why this is different to hard clipping:

When the voltage across the feedback resistor Rf is low, the diode will be turned off. Current only flows through Rf. Increasing the voltage (and thus current) gradually switches on the diode. Some of the current starts flowing through the diode while some still flows through the resistor.

Soft clipping currents - the original input current (red) is equal to the sum of the current through the resistor (blue) and diode (green). The diode only starts conducting after a while, but when it does almost all current goes through it.

Abiding to Ohms law, the opamp output voltage is still the current flowing through the resistor times the resistance, but as some of the current required to keep the two opamp inputs equal now flows through the diode, the output voltage is less than the input voltage (given a unity gain configuration where Rin = Rf).

As the input current increases, more and more current flows through the diode, and the amount of current flowing through the resistor flats out, cutting off the top of the input waves.



Another interesting thing to note is at what voltage the cut happens at is affected by the resistance of Rf. With a larger resistance, current starts flowing through the diode earlier (as its "resistance" is lower). Thus, the voltage across the resistor when it starts clipping is also lower, meaning the output is clipped more than with a lower value resistor.

Example values for a +/-1v input are clipped peaks at:

550mV for 1k,
450mV for 10k and
355mV for 100k.


A +/-5v input has peaks at 650mV and +/-15v peaks at 712mV (at 100k?)

Soft clipping with 1k resistor in feedback loop

Soft clipping with 10k resistor in feedback loop

Soft clipping with 100k resistor in feedback loop

Currents with 1k resistor, total current is 1mA

Currents with 10k resistor, total current is 100uA. A proportionally larger part of the input current goes through the diode (the diode "resistance" is still the same so it's easier for the current to go this way) and the output current is clipped earlier.

Currents with 100k resistor, total current is 10uA, an even larger part of the current goes through the diode

The non-inverting version probably works in the same way.

I first had issues getting the non-inverting circuit from Design your own distortion to work because of the HP cutoff error, but finally I found and was able to simulate a similar non-inverting circuit, this one:

https://electronics.stackexchange.com/questions/473989/need-help-designing-and-implementing-an-op-amp-based-distortion-circuit

Non inverting circuit, increased value of filter cap, meaning a lower high pass filter point distorts the wave.

The same circuit as above but without filter cap. The clipping part is more visible here.

The slight skewing is because of the cap in the feedback loop. Gain is (10k + 220) / 220 = 46

It also explains a very important thing: The guitar signal amplitude is 40mV, nothing near the 1-5V I've been experimenting with so far. Taking this into account I was able to simulate a +/-15v version of the original circuit, but WITHOUT the 50nF capacitor in series with the 1k input resistor on the positive terminal of the op amp, an the resistor connected to GND instead of the negative supply (in the original circuit it is connected to ground, but ground is the negative supply as this is a 9V circuit). See the filter section below for an explanation of why removing the 50nF cap was necessary (the cap value is simply wrong due to a calculation error in the article).

Further more, Rf must be much larger than 1k to get distortion, without this all the current flows through Rf and none through the diodes so no clipping happens. I've been working with 30k.

Working soft and hard clipping. The input (green) is multiplied by 31 to see how it matches (the non inverting opamp configuration here amplifies the input 31 times).

I have also simulated various types of diodes for the soft clipping circuit, more about this in a separate post.

Filtering

The webpage on top also talks a lot about filtering. The original article is about distortion for guitars and says that one should aim for keeping frequencies from 40Hz to 30kHz (NB: There is a serious error in the HP filter calculations in the article, see the end of this section). It may be different for a synth but that's a good starting point.

The article discusses high pass and low pass filtering in a non-inverting amplifier. The chosen circuit is often referred to as shelving filters elsewhere, it is not the most common circuit in examples on the web.

Shelving filters actually have two parts, the "normal" high or lowpass filter, and then a second part where it flattens out again at a lower or higher frequency with lower gain, but where the gain stays at this level instead of continuing to drop off, letting all frequencies below (for HP) or above (for LP) pass. - see http://www.linkwitzlab.com/filters.htm#5.

Here are the sub-circuits and formulas needed for calculating the 3dB points:

The full circuit (NB: Other component names than the original):

The second parts of the shelving filter: Both these have gain = 1. To find the frequencies, replace R1 in the HP formula with (R1 + R2), and R2 in the LP formula with (R1*R2) / (R1 + R2).

For example:

Given the values
R1 = 1kOhm
R2 = 10kOhm
C1 = 50nF
C2 = 470pF

For the HP filter:
HP filter frequency: 3.1kHz (Gain = 11)
Lower frequency: 290Hz (Gain = 1 for all freqs below this)

For the LP filter:
LP filter frequency: 33.9kHz  (Gain = 11)
higher frequency: 373kHz (Gain = 1 for all freqs above this)

PS: In the "Design your own distortion" article HP frequency capacitor selection calculation is wrong. 1 / (2 * PI * 1000 * 40) is not 0.039uF, it is 3.98uF. This explains my inability to get the distortion working without removing the cap during my simulation - I simulated using a 400Hz sine wave, this is well below the 3.1kHz HP cutoff, meaning it has unity gain and is not at all distorted.

In general, there is a lot of info about filtering missing from the "Design your own distortion" article. The Tube Screamer analysis at https://www.electrosmash.com/tube-screamer-analysis has a much better description, talking about how the filter frequencies are affected by changes in the gain of the distortion and how bass frequencies are passed undistorted (but also un-amplified).


When deciding on component values, one first chooses the gain, then calculate the necessary capacitors.

See also

https://www.premierguitar.com/articles/diode-stew-1
https://sound-au.com/articles/soft-clip.htm
https://anasounds.com/od-disto-fuzz-differences/
https://www.electrosmash.com/tube-screamer-analysis

Noise research

I intend for the XM8 to have noise as a waveform for both oscillators, and also to be able to switch between various 'colors' of noise. At the very least, white and pink noise should be present, possibly even red. But what exactly does this mean?

White noise is noise where the signal has equal intensity at all frequencies. In the synth world, it is commonly generated by using a transistor with one leg disconnected.

Pink noise is a signal where each octave carries the same amount of noise energy. But how is this achieved in practice?

According to Wikipedia, pink noise falls off at 3dB per octave. To get pink noise one filters white noise through a filter with 3dB/octave drop off.

Problem is, most basic active low pass filter has a 6dB drop off (which would actually give us red noise if used). So how may this be solved?

This page shows one method - use multiple filter sections to approximate a 3dB filter with a flat response. The more sections the better, but even four sections is pretty good for a 20-20 000kHz signal.

As a side note - the same page mentions NP capacitors, bipolar electrolytics, and says that film capacitors cannot replace them - this is interesting information as I've stumbled across NP in other circuits.

A similar approach seems to be in use on this page, which is a modification for the Sequential Circuits Pro One. It uses fewer sections (two?) and has an additional cap (C3).

But how does one calculate the frequency of each section?

In a normal active low pass filter (6dB), the frequency is 1/(2*PI*R2*C) and the gain is -R2/R1 where R2 is the resistor in the feedback loop.

It seems that the same holds true for each section in the multi section filter. For example:

1/6.28*100nF*1MOhm) = 1.59Hz
1/6.28*33nF*330kOhm) = 14.6Hz
...
which matches the frequencies next to the sections.

This would mean that the lower section of the Pro one filter is 338.8Hz, but the rest - the 270k and 3.3n combined gives us a 268.1Hz filter which seems a bit strange - however, I'm not sure this is the way to calculate the combined frequencies.

As for the gain, if the same formula as before is correct, it would be -270k/15k = -18.

The pink noise filter in the pro one matches the inverting shelving low pass filter found on this page. However, the only formula, found in the gif, is missing the lone cap in the feedback circuit.

Funny enough, the same filter topology is found in the BOSS CE-2 pedal's de-emphasis filter :). The de-emphasis filter reduces treble, which of course means it is a low pass filter of sorts. :)

On that page, the circuit is fully explained. When calculating the frequencies, the lone cap is omitted. It says that it is an LP of some sort but it is not essential when calculating the shelving frequencies.

The filter right after the transistor in the pro one circuit is a simple non-inverting HP filter. C2 and U1b forms a shelving HP filter like this.

The first filter in the Ray Wilson Noise Cornucopia is a simplified non-inverting amplifier filter as shown here and here. Its gain is 1 + R10/R9 (=48), the frequency is 1/(2*PI*R10*C4), or approx 34kHz.

ERROR: det er et lowpass non inverting shelving filter.