Compression types explained

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When I first started out (over 18 years ago now...) I found that the hardest notions to grasp were the differences between compressor types. Feedforward vs feedback, peak vs "RMS"/integrated, VCA vs FET vs Vari-Mu vs optical, etc. (well, that's it really). Not to mention compressor families (1176 vs LA-2A vs Fairchild etc.). I wish to make things easier for beginners who might find themselves overwhelmed. While I won't go deep into specific compressors, because at the end of the day what counts the most is the general design, I will mention them in passing. My articles here will be mostly dictated by my experience, and I might make mistakes or inaccuracies, so you're free to correct me whenever needed.

Let's start out from feedforward vs feedback.

Actually, let's take a step back and explain how a compressor works.

A compressor is generally divided into two main circuits: the gain computer and an actual (variable gain) amplifier, controlled by the former. In essence, the signal goes through both circuits; the gain computer senses the signal, processes it (usually rectification + smoothing) and produces a control signal, according to parameters such as threshold, attack, release (and hold, if present), which looks very much like an ADSR curve on a synth. The amplifier, on the other hand, with no control signal present will pass the signal through unaltered, or with a fixed boost¹. Whenever the signal goes over the threshold (which can be fixed, eg. 1176, or variable, eg. LA-2A) the gain computer will produce a control signal that will cause the amplifier to vary its gain, attenuating the signal correspondingly to that "ADSR" curve (in reality, just attack + release usually) I mentioned earlier.

A large part, if not most of the character of a compressor derives from its attack times (of course, ratio and release also have an effect).

When attack times are short, the resulting sound will have less "attack" (meaning, that "click" or "pop" you hear at the start of a drum hit or a guitar strum) than it would uncompressed, because the compressor will act promptly and squash the signal down as soon as the signal goes over the threshold. And that is useful to chop off transients, the "starting portion" of a signal (eg. the beater striking the head of a kick drum, a pick strumming a string, etc.), which can be significantly louder than the rest of the body of the signal (or rather the single event: the single drum hit, the single guitar strum, etc.). When do you want to chop off transients? When you have very "clicky" tracks, with too much energy right at the start of the drum hit or whatever, overly poking out of the mix while the rest of the hit sounds weak or even just right (so that you can't turn the volume down). Or, traditionally (eg. recording on tape), when the source you're trying to record has too much dynamic range and staging the gain for the transient (so that it doesn't clip) makes the rest of the body of the sound too weak and plagued with noise. Or as a special effect in extreme compression, for instance for those extremely squashed lo-fi acoustic drums where everything sounds LOUD and THICK, with the harmonics of the drums ringing out as loudly as the hit itself, and the cymbals "squashing" instead of "pinging" or "crashing".

What about long attack times? Well, that will cause the opposite effect. It will actually let more of that transient through, before squashing down on the signal. And that can be useful for times you might otherwise reach for a transient designer², for instance, to add more aggression and dynamics to overly soft and syrupy tracks (for example, a bass guitar or a snare that was played too soft), or when you do want the track to pop out of the mix a little bit. But usually, attack times are lengthened just enough to avoid chopping off the transient entirely and making the track sound soulless and flat.

...but what is a short attack time, in practice? It really depends on the type of compressor. And, of course, on the source. A finger-plucked bass note has a totally different envelope compared to a slap bass note, for instance, and what's short for the former might be extremely long for the latter.

So: what about feedforward vs feedback compression?

Feedforward compression places the gain computer *before* the amplifier. It takes the signal right from the source, processes it and controls the amplifier basically at the same time the signal passes through the amplifier. Feedback compression instead places the gain computer *after* the amplifier. So the signal passes through the amplifier, it is fed into the gain computer which then processes the signal and controls the amplifier itself.

In theory and ideally, there should be no difference. An ideal amplifier and gain computer react instantaneously, and their order should not matter much. In practice, due to physical speed limitations (of tubes and transistors, or sometimes digital circuits), phase shifts (due to filtering, for instance) and a number of other factors, feedback compressors (and in general feedback circuits) are slower and less "precise" than feedforward compressors. They are spongier in feel, with longer actual attack times, and with smoother curves (as opposed to abrupt) compared to their counterparts.

So why use or design feedback compressors at all? Because they are much simpler to design, (usually) require fewer parts, and work quite well after all despite, or maybe because of their quirks.
A feedforward compressor, for instance, needs special care to ensure that it doesn't overcompress (attenuating the transient so much that it dips *below* the body of the sound). Feedback compressors tend to self-correct spontaneously (even though all compressors, even the best ones, can overcompress with the right signal and the right threshold and the right attack and release times).
In general, a feedback compressor will sound "bigger" than a feedforward compressor, which instead is technically "better at doing its job" (when designed well). A lot of digital compressors tend to default to feedforward operation, because in the digital realm it's almost free, both in terms of CPU cycles and design issues. Feedforward compression doesn't have to be clinical, though; there are lots of feedforward compressors with a recognisable sound signature.

As a rule of thumb, a 5ms attack time is rather medium-fast on a feedforward compressor, but tends to be slow for a feedback compressor. A useful rule of thumb is that a feedforward compressor has
*similar* attack times when set to 2-3x the attack time of a feedback compressor.

That's it for now.

1: it depends on technology (eg. FETs require very low signal levels to function distortion-free) or general operation (downward compression, that is, a compressor that attenuates everything over the threshold, or upward compression, which boosts everything until the signal hits the threshold, and then boosts less or not at all; the difference between the two is less significant than it seems, and you can achieve similar results with both).

2: a transient designer is very similar to a compressor, though it acts differently (it has basically two gain computers, responding based on both threshold and time; the attack computer engages when the signal goes over the threshold, and is in control for a set amount of time, eg. 30ms, then control passes to the sustain computer until a new threshold crossing occurs).

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Thanks.
Please do not forget about diode-bridge (Neve). Also, sidechain, ring-mod, ducking, upward vs downward compressors...
And if you mentioned transient designers, then a short intro into other general purpose dynamics plugins, and how they differ from compressors, might make sense IMHO: limiters, expanders, clippers, gates...

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It's nice you wanna help to newbies,but too much words will confuse them :)
Most important question is WHY we use compressors?
Because they make sound fit to other instruments in the mix like to put another item in already full bag.
Because they make final master sounds pro.
As simple as that.
The type of compressor depend of material and preference of fast,medium or slow compression - combination of two or three is even better.
For example 1176+la2a for vocal.
Most important is man to know the character of the comp.
1176 will make your kick distinguish.
La2a grey will add character to whatever you put it on.
la2a tube will saturate super gentle way the slow instruments like pads.
So on and so on...
The right compressor settings will make huge differentiate with your mix.
From chaotic mess it will become really tight and suberb.
There is a lot of videos in youtube man can learn important things.
Eq,reverb,compressor and saturator are the most used and important fx in modern production.
Cheers :)

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VELLTONE MUSIC wrote: Wed Jul 01, 2026 12:09 pm It's nice you wanna help to newbies,but too much words will confuse them :)
Most important question is WHY we use compressors?
Because they make sound fit to other instruments in the mix like to put another item in already full bag.
Because they make final master sounds pro.
As simple as that.
The type of compressor depend of material and preference of fast,medium or slow compression - combination of two or three is even better.
For example 1176+la2a for vocal.
Most important is man to know the character of the comp.
1176 will make your kick distinguish.
La2a grey will add character to whatever you put it on.
la2a tube will saturate super gentle way the slow instruments like pads.
So on and so on...
The right compressor settings will make huge differentiate with your mix.
From chaotic mess it will become really tight and suberb.
There is a lot of videos in youtube man can learn important things.
Eq,reverb,compressor and saturator are the most used and important fx in modern production.
Cheers :)
I don't like this way of explaining compression (or anything) to people. People need to understand why something works well in a certain situation. Why do you need both an 1176 and an LA-2A for vocals? (For instance, I don't care much for that combo, at least using UAD emus. Too much saturation for my taste. I usually prefer to use a clean feedforward compressor in place of the 1176, with 3-6dB gain reduction on peaks only).

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ampetrosillo wrote: Wed Jul 01, 2026 7:13 pm
VELLTONE MUSIC wrote: Wed Jul 01, 2026 12:09 pm It's nice you wanna help to newbies,but too much words will confuse them :)
Most important question is WHY we use compressors?
Because they make sound fit to other instruments in the mix like to put another item in already full bag.
Because they make final master sounds pro.
As simple as that.
The type of compressor depend of material and preference of fast,medium or slow compression - combination of two or three is even better.
For example 1176+la2a for vocal.
Most important is man to know the character of the comp.
1176 will make your kick distinguish.
La2a grey will add character to whatever you put it on.
la2a tube will saturate super gentle way the slow instruments like pads.
So on and so on...
The right compressor settings will make huge differentiate with your mix.
From chaotic mess it will become really tight and suberb.
There is a lot of videos in youtube man can learn important things.
Eq,reverb,compressor and saturator are the most used and important fx in modern production.
Cheers :)
I don't like this way of explaining compression (or anything) to people. People need to understand why something works well in a certain situation. Why do you need both an 1176 and an LA-2A for vocals? (For instance, I don't care much for that combo, at least using UAD emus. Too much saturation for my taste. I usually prefer to use a clean feedforward compressor in place of the 1176, with 3-6dB gain reduction on peaks only).
Agree,if you wanna go deep,but for the newbies in the beginning is total mess of too much terms and theory.
I don't need 1176+la2a for vocals,just people with way more advanced skills ,than mine, found it useful,so i just use,what pro mixers share.
I like experiment with different fx in a chain,not always the compressor do best job - saturation sometimes is better choice.
Cheers :)

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Once you learn corrective compression, it is much easier to know when and how to use artistically enhancing compression to taste.

Learning corrective compression is going to most likely involve you using a microphone to actually record a performance.
REAPER + Davinci Resolve Pro on Manjaro KDE. Neve 88m. Focusrite 18i20 2nd gen. Neumann NDH30 headphones. Mics: Telefunken TF39, AT4050, Miktek C7e, EV RE-15. VSTs: u-he Hive 2, F'em, Renoise Redux, Apisonic Speedrum 2.

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Best compression is no compression.

Second best compression is actually saturation.

Third best and if first two options dont apply, is specialized compression.

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good rundown. the piece worth adding for anyone learning: the circuit type (VCA/FET/opto/vari-mu) matters less than the attack/release behaviour and the detection path it implies. an opto is "slow and program-dependent" not because it's optical in some magic way but because the light-cell's release is frequency- and level-dependent, so it self-adjusts, that's the "musical" reputation. FET is fast attack + aggressive, vari-mu softens as it pushes. so when you hear "character," you're mostly hearing a timing + detection fingerprint, which is why two plugins modelling the same circuit can sound different if their envelopes differ. pick by the envelope behaviour you want, not the badge.

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Korneff is going to be writing about compressors in his next few Monday emails, including today's.
REAPER + Davinci Resolve Pro on Manjaro KDE. Neve 88m. Focusrite 18i20 2nd gen. Neumann NDH30 headphones. Mics: Telefunken TF39, AT4050, Miktek C7e, EV RE-15. VSTs: u-he Hive 2, F'em, Renoise Redux, Apisonic Speedrum 2.

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ampetrosillo wrote: Wed Jul 01, 2026 7:13 pm I don't like this way of explaining compression (or anything) to people. People need to understand why something works well in a certain situation.
No they really don't. FYI most people don't give a hoot why it works, only that it does, as that's all they need and are interested in. This is true of most things, really.

For example, I much prefer something like this, which gives a lot of info too, but has the summary/"TLDR" info at the bottom for those of us not interested in the lengthy particulars:

https://www.masteringbox.com/learn/audi ... cuit-types

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@mixguy I cannot agree with you. I found the article very informative and idc it, I expect, will be helpful. As you have no interest then just skip this thread. Some people like to understand things they use others are replaceable by trained monkeys.

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You don't have to find it personally useful, as long as it helps one person, I think it merits existing.
Also because there are *tons* of compression articles that treat the topic more like some of you prefer.

Anyway, here's my second entry.

So, I've talked about the fundamental difference between feedforward and feedback compression. The TL;DR is that basically well-designed feedforward compression is more controlled, precise and clinical, but it's comparatively harder to do right, whereas feedback compression tends to be slower (for the same parameters), more "ripple-y" and less transparent, but it's quite easy to design as it is mostly self-correcting.

The other big elephant in the room is peak vs integrated compression. Or, more accurately, peak vs integrated gain control.

As said before, a compressor is divided into two main circuits: the amplifier and the gain computer (which is sometimes referred to as the sidechain).

Peak compression is the simplest; basically, the sidechain takes the signal intended to control the compression (usually, the same signal as the input), processes it so that it can be used as a control signal (it rectifies it, so that it is one-sided, ie. always positive or negative), then smooths it out according to attack and release parameters. Therefore, the attack and release act based on a signal that shares a lot of properties with the input signal - namely, whatever is a peak in the input signal is also a peak in the control signal. And of course, this is perfectly sensible. One would expect compressors to act based on the peaks of the signal, because what we want to do, usually, is to clamp down on those peaks - which often carry a lot of energy that we don't actually hear - so that it is all more consistent. And attack and release parameters basically set the compressor so that it takes <attack setting> time to clamp down on the peaks over the threshold, and <release setting> time to stop compressing¹. If you want to preserve some of that energy in the transients, you set the attack so that it takes a little while to squash the signal - for example, 10-15ms on a feedforward compressor, maybe 3-5ms on a feedback compressor - and the compressor will be slightly less effective at controlling the peaks but in a way that usually translates to smooth levelling anyway. (As always, it depends on the dynamic envelope of the input signal; what's true for a snare hit may not be true for bass, and usually isn't true for vocals). And you set the release, usually, short enough not to hear the compression overtly working (set it shorter, and you'll hear that classic "pumping" effect, where the signal dips and swells² - which might be exactly what you're looking for!) and not long enough that all the compressor is doing is turning down the volume on the whole track.

So why come up with another way to compress, namely, integrated compression?

Integrated compression (often called "RMS" compression, even though this is a misnomer) passes the sidechain signal through an integrator - basically, something that "adds up" the energy of the signal over a set amount of time and outputs the energy within that amount of time.

You know how compressors sometimes have high-pass and (more rarely) low-pass filters?
Now, high-pass filters attenuate the low-end components of the signal, so that they impact less on the final compression - very useful when you have bass elements that trigger the compression far more readily than mid and treble elements. For example, a drum bus, where the kick may be stronger energetically than the snare, but we don't hear it as strong, we don't want it to be too compressed, and we don't want it to dictate how the rest of the drum bus should be compressed. So we filter away whatever is, say, below 120Hz or so, so that kick and snare, usually, cause a similar amount of gain reduction.
Low-pass filters do the opposite: they cause treble elements to impact less on the final compression. Which, per se, isn't very useful for compression. The energy in the treble is far lower, in an average track, than the energy in the bass region. But... treble components do cause the waveform to have steeper slopes: compare a sine wave, for instance (just a single frequency component), to a square wave (which has a lot of harmonics going up to infinity, in theory), and take an oscilloscope or waveform viewer, and gradually low-pass the square wave - you'll see how the slope between the high and low levels of the square wave will finally end up being shallower.

An integrator is a sort of low-pass filter. Without being too technical: it's something that, as mentioned before, responds over a set amount of time, let's say 10 or 20ms. This is something that tends to smooth out sharp transitions (which may take maybe 1ms, or 1 microsecond, or shorter still!). If you want to imagine what an integrator does, just compare it to a blur effect in graphics: when you blur an image, the borders (sharp transitions) become smoother, and sometimes are completely lost.

So, an integrator in the sidechain would take its time to reach its highest level - in a sense, an integrator has a preset minimum attack time *over* the one you can set with a knob or parameter. So, why would something like this be useful?
Integrated compression tends to respond to average power rather than peak energy. Imagine two drum hits, one with a strong peak but a fast decay, and one with a lower peak but which dies out more slowly. With peak compression, only the first might be affected. With integrated compression, both could be affected similarly, or maybe the first wouldn't be at all while the second would, so that the end effect is that you no longer compress for peaks but you compress for actual signal power (which correlates with perceived volume better than peaks). Integrated compression is less for transients (up to a certain point) and more for the actual body of the sound. In fact, you can set a very low attack time on an integrated compressor and it won't completely squash your transients as you would expect with a peak compressor.

What does this translate to, in practical terms?

Peak compression works best for, well, yes, peak control. It is very useful while tracking, or on tracks, because this way you can ensure that the source will not overly distort your preamps, or that your final signal will tend to be within certain limits in order not to mess up your gain staging. And also, it controls those stray hits that are far too peaky compared to the rest. And they can provide a certain character because they can be faster and they follow your input signal quite closely. In a sense, peak compression works best to shape hits "individually".

Integrated compression works best when you want the track or bus not just to be, but also to sound consistent - which is why they tend to live on subgroups or mixbuses (but bass is often tracked with "RMS" compressors too). They take those already massaged tracks and they smooth them out a bit more - they make them feel closer to an actual "record".

But can't you just achieve the same with a peak compressor and a longer attack time? Well, yes and no. Peak compressors still respond to peaks, rather than signal power. With lots of signals, there is strong correlation between "peakiness" and power, so yes, you can sometimes just set a peak compressor to a long-ish attack and it will perform similarly. But integrated compressors, by their very nature, tend to track what we actually hear instead of what's printed on tape or file.

¹ : remember! Release time is the time the compressor takes to stop compressing once the signal has gone below the threshold. Not the time passed after the peak occurred! So release happens, and be heard, once the signal has gone quieter than the threshold.

² : another side effect of low release times is distortion, which is inescapable anyway in compression by its very nature. But this can also be used as an effect. Low attack and release times have often been used precisely to grunge up tracks that were otherwise too pristine.

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