Understanding Audio Compression – Get started now!

At some point of your music or audio engineering career, you are bound to come across various audio signal processors. These are equipment that professionals in the industry constantly rely on, in order to achieve that perfect mix. Today, we will discuss about the compressor, and help you work on understanding audio compression!

The compressor is used in various industries related to audio. In this article, we will focus on the basic design and the different controls and functions of the compressor. I will stick to the important bits of information that are needed for the beginner to start experimenting with this processor (wait, you are new to this right?). Okay, so let us begin!

So, what is it exactly?

Tube Compressor

Photo by Andyzweb / CC BY-SA 3.0

Also referred to as “dynamic range compression” (DRC) or simply as “compression”, it is basically an electronic signal process that functions to “compress” an audio signal’s dynamic range in order to either reduce the volume of louder sounds, or further amplifies quieter sounds. Other than in live sound production and broadcasting, you can also find many instrument amplifiers (usually bass amps) that uses compressors.

Compression of audio allows the reduction of loud sounds that are above a certain threshold (set by the user), while not affecting the quiet sounds. Starting in the early 2000s, compressors have become widely available in audio software for recording.

The dedicated electronic hardware unit or audio software used to apply compression is called a compressor. In studio recording and live music production, the compressor’s parameters are constantly adjusted to achieve different effects. You may also come across the term “limiting”, which is also identical in process but different in degree and perceived effect.

How does it work?

To fully explain all the electronic functions of the compressor will probably take hours, days, weeks, months (okay, you get the point). Basically folks, what I am trying to say is that we will keep this section pretty brief, just so you can have a basic understanding of the process.

Compressor Design

Photo by Izhaki / CC BY 3.0

When an audio signal enters the input of a compressor, it is split into two, one is sent to a “variable-gain amplifier” and the other to a “side-chain”, where a circuit controlled by the signal level applies the required gain to an amplifier stage. This is also known as the “feed-forward” design, which is used in most compressors today. Most earlier designs implemented the “feedback” layout, where the signal feeding the control circuit was taken after the amplifier.

Various technologies are used for the “variable gain amplification” and each of them have different advantages and disadvantages. Some of these technologies uses “vacuum tubes”, “voltage controlled amplifiers” (VCA) and “light dependent resistors” (LDR). Different technologies will have different effects on the overall compression of the signal and ultimately affects the way it sounds.

When it comes to digital audio, digital signal processing techniques are used to apply compression by using digital audio editors or dedicated workstations. Often times, the algorithms used, are designed to emulate analog compressor technologies.

Types of compression

Downward Compression

Downward Compression / Photo by Iainf / CC BY 2.5

There are two main types of compression and they are:

  • Downward Compression (reduces loud sounds over a certain threshold while quiet sounds remain unaffected)
  • Upward Compression (increases the loudness of sounds below a certain threshold while leaving louder sounds unaffected)

Both types of compression have the same function of reducing the dynamic range of an audio signal.

Controls and parameters

The moment you laid eyes on your brand new compressor unit, you saw a few knobs and buttons and then you thought to yourself, “hmm, maybe I shouldn’t have bought this in the first place”. Hey, don’t worry, because in this section, we will touch on the “nuts and bolts” of compressors.

Upward Compression

Upward Compression / Photo by Iain Fergusson / CC BY-SA 3.0

There are quite a few of them to go through and they are:

  • Threshold
  • Ratio
  • Attack and Release
  • Soft and Hard Knees
  • Peak vs RMS Sensing
  • Stereo Linking
  • Make-up Gain
  • Look-ahead


A compressor is designed to start reducing the level of an audio signal, only if its amplitude exceeds a certain threshold (which is set by you). The common unit of measurement used would be in decibels “dB”, where if you set a lower threshold (e.g. -60 dB), that means a larger portion of the signal will be treated, and if you set it to a higher threshold (e.g. −5 dB), a smaller portion of the signal will be treated.


Ratio determines the amount of gain reduced. If you have set a ratio of 4:1, that means if the input level is 4 dB over the threshold, the output signal level will be 1 dB over the threshold. Hence, the gain level has been reduced by 3 dB. Here’s an example:

Threshold = −20 dBCompression Ratio
Input = −16 dB (4 dB above the threshold)
Output = −19 dB (1 dB above the threshold)

“Limiting” is often known as having the the highest ratio of “∞:1” . It is normally achieved using a ratio of 60:1, and effectively causes any signal that is above the threshold to be brought down to the threshold level (except for a brief period of time, where there is a sudden increase in input loudness, known as an “attack”).

Attack and Release

Most compressors allow you to have a degree of control over how quickly it acts. This is where you will need to adjust the “attack phase”, which is the period when the compressor is reducing gain to reach the level that is determined by the ratio. On the other hand, “release phase” is the period when the compressor is increasing gain to the level determined by the ratio, or to zero dB, once the signal level falls below the threshold.


Photo by Iainf / CC BY-SA 3.0

The length of each period is determined by the rate of change and the required change in gain. To make the operation more intuitive, a compressor’s attack and release controls are normally labelled as a unit of time (often milliseconds). This represents the amount of time it will take for the gain to change a set amount of dB (decided by the manufacturer), usually 10 dB. For instance, if the compressor’s time constants are referenced to 10 dB, and the attack time is set to 1 ms, it will take 1 ms for the gain to decrease by 10 dB, and 2 ms to decrease by 20 dB.

Soft and Hard KneesHard knee, Soft knee

You might also come across compressors that offer a “hard/soft knee” parameter. This determines whether the bend in the response curve between “below threshold” and “above threshold” is steep (hard) or gradual (soft). A soft knee allows for a slow increase of the compression ratio as the signal level increases and eventually reaches the compression ratio that you have set. A soft knee also helps to reduce the audible change from uncompressed to compressed, which is even more noticeable for higher ratios with a hard knee setting.

Peak vs RMS Sensing

A compressor with “peak sensing”, will respond to sudden increases in level of the input signal. While this may be useful in some cases, peak sensing might apply very quick changes in gain reduction, more noticeable compression or sometimes even cause distortion. You should look for compressors that have an averaging function (commonly RMS) on the input signal before its level is compared to the threshold. This will result in a more relaxed compression which relates to the human perception of loudness.

Compressor Plugin

Software Plugin / Photo by Klaus Göttling / CC BY-SA 3.0

Stereo Linking 

In stereo linking mode, the same amount of gain will be applied to both the left and right channels. This is to prevent “image shifting” from happening, if each channel is compressed individually. You can easily hear this effect when a loud audio element, panned to the edge of the stereo field (for example, left channel), raises the level of the overall mix to the compressor’s threshold, which in turn causes its image to shift toward the centre of the stereo field.

Stereo linking is achieved in two ways:

1) Either the compressor sums to mono the left and right channel at the input, leaving only the left channel controls functional; or

2) the compressor will calculate the required amount of gain that needs to be reduced for each channel and then applies the highest amount of gain reduction to both (in this case, it would make sense to dial different settings on left and right channels as you might want to have less compression for either channels).

Makeup Gain

Since the compressor works on reducing the gain (or level) of the signal, the function of “make-up gain” (ability to add a fixed amount of gain) at the output is usually provided so that an optimum level can be used.

Waveform Compressed

Photo by Alexander J Turner / CC BY 3.0


This function is designed to rectify the issue of having to compromise between slow “attack” rates that will produce smooth-sounding gain changes, and fast “attack” rates that are capable of catching transients. “Look-ahead” is actually a misnomer in that the future is not actually observed.

The way this works, is by having the input signal split, and one side delayed. The non-delayed signal will drive the compression of the delayed signal, which then appears at the output. This method produces a smooth-sounding slower “attack” rate that can be used to catch transients. The only cost of this solution is that the signal will be delayed.

That’s all for this article. The information provided here should be enough for people who are relatively new to this, to start experimenting with compressors. I might do a follow-up article covering other aspects of compression, so stay tuned!

I hope you enjoyed reading and don’t forget to leave a comment below!



When I'm not rocking out to great music, I'd prefer to be sleeping on a field on a windy day =)


  1. Hi there. I am lost honestly on how compressor functions. When you say audio signal, it means the sound wave right? Or it’s a different thing.

    Also, technologies like vacuum tubes, VCA and LDR are some of the modern features compressors have these days?

    As for decibels, is there such thing as positive? Like 10 dB? Or it’s always a negative?

    • Hello Tar, thanks for reading my article.

      A “sound wave” is what travels to your ears from the sound source (say, a speaker), and therefore, your brain will perceive that as “sound” that you are hearing.

      “Audio Signal” is a representation of sound, and it is often in the form of an “electrical voltage”. It is basically the signal that originates from a transducer (like a microphone), and travels through all your electrical audio equipment, until it reaches the speakers (which then converts this electrical audio signal into “sound”).

      As for technologies, all the terms that I have mentioned are various elements that are incorporated into different technologies, used in different compressors (This refers to the “variable gain amp” part).

      As for decibels, yes there is a positive. Basically, anything below 0db (unity gain), would be negative.

      Hope that helps =)

  2. Might is say great work this type of thing is right up my street so I think I’m on pretty much the same wave length. Also I love all the information that you have its great to see people giving out audio education because I believe its a great subject to learn great work.

    • Hello S A H, thanks for the kind comments. I am glad you find it useful and informative =)

  3. Hi Farhan,

    Very interesting! I was immediately intrigued when I read the words “dynamic range compression”. As an audio enthusiast, I’m not super technical on this matter. I’m no engineer whatsoever. But I do appreciate good quality sound recordings.

    One of my hobbies is constantly searching for the best versions of my favorite music. Whether this is analog or digital 16/44, 24/96, etc … is not my primary concern. I always look at how it is mastered. This is also where the dynamic range comes into play.

    I use quality speakers and headphones and it hurts my ears (especially with headphones) to listen to music that is so compressed that it – for example – has a DR value of 4. Especially on music where real instruments are being used. It sounds so congested when several instruments come into play together that I personally cannot enjoy such recordings.

    I would love to hear your take on this Farhan.


    • Hello Jurgen, thank you for your comments!

      I am afraid that the music you were listening to, have fallen victim to the “Loudness war”, as they call it.

      “Loudness war” is the term given to most modern music these days that are seeing an increasing trend of high audio levels in their mix, since the 90s. When CDs became popular, more producers are keen to make their music even louder, as they had the impression that people preferred louder sounding CDs.

      Well, I am pretty sure you already know that this trend comes with a price. With heavier compression and equalization in mixes, music have become less “dynamic” so to speak.

      I am afraid there is nothing you can do to an already overly compressed mix, however, what I normally do is that I will just reduce the overall volume to a comfortable level, and use my music player’s graphic equalizer, to bring down certain frequencies that are overwhelming to my ears (usually mid range frequencies from 300hz to 600hz).

      Heavy mid-range frequencies does cause fatigue to your ears after a while. So after you think you have gotten the appropriate frequencies reduced (this could be subjective, so do experiment on your own), you can then gradually increase the overall volume to make up for the loss of “gain”.

      Hope that helps =)

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