Understanding Compression and Clipping

When I first started recording, I got an inaccurate understanding of compression stuck deep in my mind. I only really started to exorcise it a few years ago, and I still think the way compression is often explained is confusing. (Maybe I’ll post a follow-up on The Compression Conspiracy, or maybe it will come out here.) I’ll attempt to explain compression clearly and simply, touching on limiting and clipping along the way.

Compression is automatic volume reduction, or “variable attenuation” if you prefer. Compression is not soft clipping or peak reduction*.

* Peaks will be reduced in the course of automatic volume reduction processing, but since the whole signal is being reduced it’s confusing to call that result “peak reduction”

Compression is like having a gremlin enthralled to your instructions, whose hand is on a volume knob. You can tell the gremlin how fast to turn the volume down when your input signal gets loud (compressor “attack” time), how fast to turn the volume back up when the input signal gets quieter again (compressor “release” time), how loud is “loud” to you (compressor “threshold”), and whether you want your gremlin to turn down the volume a little, or a lot (compressor “ratio”).

The gremlin’s name is the “control circuit” or “sidechain,” a version of the input signal that is processed to detect volume, and therefore when to turn the signal volume down and back up.

Automatic volume reduction
or “variable attenuation”
Clipping
Base versionCompressorSoft clipper
VariationLimiter
(Theoretically a compressor
with an infinite ratio. In practice,
a compressor with a ratio of
about 10:1 or greater)
Hard clipper
Effect on
input signal
Change in wave
amplitude (volume)
Change in wave shape (timbre)
and peak level
Reaction timeVaries; usually a few milliseconds
to a few hundred milliseconds
Instantaneous

The results of compressors are often visualized incorrectly, so let’s start with an article that does it right; “Understanding Compressor Attack And Release Times” by Pro Tools Expert. They created the excellent image below, which I’ve scribbled on.

Notice how the attack time begins when the input signal initially crosses the threshold, and the first cycle of the signal passes through the compressor with no volume reduction (red circles). Also notice how the output signal retains its wave shape (nice and pointy at the peak); only the amplitude (i.e. volume, i.e. distance from the gray center line to the peak in the image) is reduced.

What has happened here, musically? The compressor has emphasized the initial transient (red-shaded “attack time” section) relative to the sustain or “body” of the sound (orange-shaded section). We might describe the result as having more “punch.” We might perceive the compressed sound as quieter than the input (notice the decreased amplitude of the wave in the orange-shaded section) and decide to turn up the volume of the signal across the board, after the compressor in the signal path, to compensate.

Here is the incorrect – but shockingly common – way to visualize the effect of compression. What is shown here is actually soft clipping. (If you go to the trouble of image-searching this you’ll see the author acknowledges such later in the article; kudos to them for that. Many people don’t!)

The wave labeled “Compressed” would be more accurately called “Soft-clipped”

Why is this misleading? If you compare this to the first image, you’ll see that this image doesn’t show anything about attack or release time, and it changes the shape of the signal peak, which compressors don’t do! Extremely unhelpful. This is how compression was first explained to me, resulting in many years of confusion and frustration. If this post can help one person avoid that, I’ll be happy. (HI CORY)

Here’s a slightly better version (scribble in orange), though it still doesn’t explain anything about attack and release.

Note the wave shape hasn’t changed; only it’s amplitude – the distance from the center line to the wave peak – is different

Getting from compression to limiting is a matter of ratios. Think about our gremlin again. For gentle compression we might tell them “For every two clicks louder than the threshold the signal gets, turn the volume down one click.” That’s a 2:1 compression ratio. Or we could tell them four clicks in, one click out; a 4:1 ratio.

Now imagine we tell our gremlin “Whatever happens, DO NOT UNDER ANY CIRCUMSTANCES let the sound get louder than the threshold!” That would be an infinite ratio; no matter how loud the input signal, our gremlin turns down the volume knob as much as necessary to make sure the output does not exceed the threshold. (It still takes the gremlin a few milliseconds to actually turn the knob down and up, so attack and release times are still in play, and the peak levels of the output signal might exceed the threshold in those milliseconds while the gremlin is turning the volume down.)

In practice, once we get to a ratio of 10:1 or 20:1 it’s roughly equivalent to an infinite ratio to human ears. We call compressors with high ratios and fast attack and release times “limiters,” but it’s all automatic (gremlin-controlled) volume reduction and there’s not a sharp line between what gets called a compressor and what gets called a limiter.

Clipping

Clipping is changing the peak shape of a signal. (Image credit.)

“Normal” means input signal

As you can see the peak level is also changed as a result of the shape change – they’re two sides of the same coin – but it’s the change in shape that changes the timbre (i.e. tone, or quality) of the sound. Clipping is distortion of the input signal in a literal sense, and it’s also the sound of distortion in a subjective sense. The sound you imagine when I wrote “distorted guitar,” that’s the guitar’s signal being clipped.

Clipping is instantaneous; just like you see in the image above, it occurs on a cycle-by-cycle basis.

I want to stress the differences between clipping and compression, since they’re often mixed up. But one similarity between soft clipping and lower-ratio compression is that the output signal remains correlated with the input signal above the threshold level. Likewise, hard clipping is similar to infinite ratio limiting in that above the threshold, increases in input signal do not result in corresponding increases in output (with the previously noted exception for limiters’ attack times).

Another difference between clipping and compression is where the “threshold” sits in the processing. In clipping the threshold is in the signal path; it changes the sound signal itself. In compression and limiting the threshold is in the control signal path (i.e. it is a rule given to the volume knob gremlin). A compressor’s threshold setting does not change the signal itself directly, but rather controls the automatic volume reduction of the audio signal in combination with the attack and release times and compression ratio.

Long story short:

  • Compression and clipping are very different audio processing effects. Compression is like a gremlin-controlled volume knob. Clipping is distortion (which can range from subtle or planet-destroying)
  • Limiting is a type of compression (not a type of clipping)
  • There is a lot of confusing information about compression online; think critically!

Understanding Modulation Effects

The world of guitar (and synth, and bass…) modulation effects overwhelmed me at first, but I’ve learned that most pedals and plugins are based around just a couple of ideas.

Time-based effectsFilter-based effects
Base versionChorusPhaser
VariationsFlange (chorus with feedback)

Vibrato (chorus with no dry signal)
Uni-Vibe or “Vibe” (phaser
with mis-matched filters)

Time-based effects are created by 1) copying the input signal, 2) delaying the copy by a small amount (a few milliseconds), 3) automatically changing the delay time (say, from 27 ms to 33 ms and back), and then 4) mixing the input (or “dry,” or “not delayed) and delayed signals together (typically at a 50:50 ratio). The resulting sound is a pseudo-doubling effect with a bit of pitch going up and down, some frequencies of the input signal enhanced, and others diminished. Boom; basic chorus.

Flangers operate on the same principle, with shorter delay times and feedback in the delay signal path (just like a delay or echo pedal), resulting in a more pronounced effect. Flangers with knobs for feedback amount are often great modulation pedals because they can cover a lot of territory from subtle near-chorus to crazy flying saucer flange.

Vibrato – the signal’s pitch moving up and down – is just chorus without the dry signal. When the delay time is getting shorter, the pitch is going up, and vice versa. You can test this out on your own with any delay pedal; play a sustaining note or chord, then turn the delay time faster or slower. You’ll hear the pitch of the delayed signal go up or down accordingly.

Filter-based effects

Filter-based effects are created by 1) copying the input signal (noticing a pattern here?), 2) sending the copy through a series of all-pass filters, 3) automatically changing the center frequencies of the all-pass filters* (say, from 500 Hz to 2000 Hz and back), and then 4) mixing the input (or “dry,” or “not all-passed”) and filtered signals back together. The resulting sound has a series of mid frequency cuts that move up and down. We’re phasing!

* Optional sidebar – Filters are usually used to boost or cut certain frequencies in a signal. All-pass filters don’t boost or cut any frequency, but they delay the frequencies around the center frequency a tiny amount. If an all-pass filter’s center frequency is 1000 Hz the frequencies around 1000 Hz are delayed the most (still a tiny amount), the frequencies around 500 and 2000 Hz are delayed less, the frequencies around 250 and 4000 Hz are delayed less (almost none), and so on.

The all-pass filters in Uni-Vibe or “Vibe” effects are purposefully mis-matched. This was originally done to mimic the sound of a rotary speaker cabinet. Most people don’t consider it a particularly accurate re-creation of that sound, but it has a cool sound of its own (I adore the “Vibe”-type sounds of my Wilson Effects Haze! There’s even an unreleased Mars Lights song named after that pedal).

Other modulation effects

Amplitude (volume) modulation – tremolo – is the other main type of modulation effect. This is simply the volume of a signal going up and down regularly, like a couple of times per second.

Some Fender amps incorrectly call this type of built-in amplitude modulation “vibrato” instead of tremolo.

Real rotary speaker cabinets and re-creations of them have much more complex modulation going on than a chorus or phaser. There’s pitch modulation, phase modulation, amp distortion, speaker breakup, and more, and these changes interact with each other in complex ways. Rotary speaker cabinets are amazing.

Vibrato and “Vibe”-style effects are often and understandably confused for each other due to the similar names. However, they are generated by different types of circuits and have different sounds.

Wah and auto-wah effects are also filter-based, but a different type of filter (resonant bandpass or lowpass) than phasers (all-pass). Wah effects are not usually considered modulation because the player controls the change in sound, rather than it being done automatically by the circuit.