Full List: Audio Effects & Processes For Mixing/Production

Audio effects and processors have become critical parts in music and audio production. Whether it’s crafting the perfect mix or the perfect tone for your instrument, audio effects and processes will help you get the sound the results you’re after.

In this article, we’ll have a look at the audio effects and processes that are used by musicians, producers and engineers to garner a better understanding of the tools at our disposal. I’ll list out examples of effects units/processors when possible, covering rack-mounted units, Eurorack modules, effects pedals and audio plugins/software.

An audio effect or processor is considered a part of the main signal chain and so I won’t be discussing instruments that process audio files (granular synthesizers, resynthesis engines, samplers, vocoders, etc.) though these units certainly process and affect audio.

This is going to be a rather long resource, so I’ve separated effects and processes into sub categories to keep similar tools grouped together. Each of the following listed items will have a link to skip ahead to a more in-depth explanation of the effect or processes. I’ll also include links to more focused articles when applicable.

Here is the full list of audio effects and processes for mixing/production:

Dynamic Effects

Modulation Effects

Sound Manipulation Processes

Spectral Processes

Time-Based Effects

Without further ado, let’s discuss each of these effects!

Dynamic Effects

Dynamic effects alter the dynamics of an audio signal. That is, the changes is amplitude over time.

By altering the amplitude of the signal, dynamic effects will also alter the shape of the signal waveform which is, by definition, signal distortion.

Any distortion in a signal will have an effect on the frequency content of the audio, making dynamic effects useful for modifying both frequency and amplitude characteristics of the sound.

And so, dynamic effects/processes include the following:


What is audio compression? Dynamic range compression is the process of reducing the dynamic range of an audio signal by attenuating the amplitude of the signal above a set threshold. A compressor can be software or hardware and performs compression on an audio signal (digital or analog).

Compression effectively attenuates the “loudest” parts of an audio signal, thereby reducing the overall dynamic range. Make-up gain is often applied in order to bring the signal back up to its peak level, making it sound louder (as the “quietest” parts are brought up).

It is sometimes easy to visualize the effect of compression in the following way. Imaging having your finger on a track fader and every time that track’s audio signal surpasses a set amplitude threshold, you duck the fader before bringing it back to the starting position as the amplitude comes back down. The distance you move the fader (the amount of attenuation) is proportional to the amount the signal surpasses the threshold.

The threshold of the compressor refers to the input amplitude at which the compressor will be triggered. Below the threshold, the compressor will not compress/attenuate the signal. Above the threshold, the compressor will.

Once the threshold is exceeded, there is a ratio at which the signal will be attenuated. This ratio is defined as:

[input signal dB above the threshold] : [output signal dB above the threshold]

So a ratio of 4:1 means that for every 4 dB the input signal surpasses the compressor’s threshold, the output signal will only be 1 dB above the threshold.

Threshold and ratio are the two main parameters we need to know to understand the basics of compression. Here are a few illustrations to help us visualize compression:

To learn more about compressor threshold and ratio, check out the following My New Microphone articles:
Dynamic Range Compression: What Is The Threshold Control?
Dynamic Range Compression: What Is The Ratio Control?

In addition to threshold and ratio controls, a compressor may also have attack and release controls. They are defined as follows:

  • Attack: the amount of time it takes for the compressor to react once the threshold is surpassed.
  • Release: the amount of time it takes for the compressor to disengage once the input signal drops back below the threshold.

To learn more information on compressor attack and release controls, check out my article Dynamic Range Compression: Attack & Release Controls.

Related articles:
The Complete Guide To Audio Compression & Compressors
Top Best 11 Compressor Pedals For Guitar & Bass
What Are Compressor Pedals (Guitar/Bass) & How Do They Work?

Let’s have a look at some examples of audio dynamic range compression.

FET Compression

What is an FET compressor? An FET compressor is an analog compressor that utilizes a field effect transistor at the core of the circuit. These compressors are fast-acting and offer greater transient preservation than other types of compressors.

For more information on FET compressors, check out the following My New Microphone articles:
What Is A FET Compressor & How Does It Work?
Top 11 Best FET Compressor Emulation Plugins For Your DAW

Multi-band Compression

What is multi-band compression? Multi-band dynamic range compression is a type of processing that splits the frequency spectrum into different bands and compresses each band by its own unique compression settings.

A multi-band compressor can be thought of as several compressors in one with each compressor acting on its own defined band of frequencies. Each band will generally have its own set of parameters including threshold, ratio, attack, release and make-up gain.

Multi-band compression is a fantastic tools and helps to avoid over-compression in signals that are particularly well-represented in certain frequency bands (bass instruments, for example).

For more information on multiband compression, check out the following My New Microphone articles:
What Is Multiband Compression & How Do MB Compressors Work?
Top 10 Best Multiband Compressor Plugins For Your DAW

Optical Compression

What is an optical compressor? An optical compressor is an analog compressor that uses a light element and optical cell to alter the dynamics of an audio signal. As the amplitude of the audio signal increases, the light element emits more light and causes the optical cell to attenuate the amplitude of the output signal.

For more information on optical compressors, check out the following My New Microphone articles:
What Is An Optical Compressor & How Does It Work?
Top 10 Best Optical Compressor Emulation Plugins

Parallel/Manhattan Compression

What is parallel compression? Parallel compression (also known as New York or Manhattan compression) is a technique where one audio track (or several) is sent to a bus and that bus is heavily compressed. Both versions of the audio are then mixed together to achieve a punchy sound without losing the dynamic of the dry signal(s).

The easiest way to set up parallel compression is to send a track (or a group of tracks) to a bus in a mixer (whether hardware or software). Insert a compressor or limiter on the bus and hit the signal hard (low threshold, high ratio).

Once the parallel bus is sufficiently squashed, mix it appropriately with the dry signal from the original track(s). The result should have the dry signal well-represented with the additional punch of the over-compressed bus signal.

Parallel compression is a particularly effective process on drums.

Sidechain Compression

What is a sidechain compression? Sidechain compression is a type of compression where the compressor is acting to compress a track but is controlled by another track. As the controlling signal surpasses the set threshold, the compressor attenuates the sidechained signal.

Not all compressors will allow side-chaining. For those hardware compressors that do, there will be a sidechain insert that will allow a sidechain source to control the compression applied to the input signal. In software compressors, there will generally be a sidechain dropdown menu that will allow you to choose the sidechain source.

Sidechaining is a popular mix technique and has been made famous/infamous by the aggressive sidechaining to the kick of various elements in certain electronic music genres/mixes.

To learn more about sidechain compression, check out my article The Complete Guide To Sidechain Compression In Audio.

Variable-Mu Compression

What is a variable-mu compressor? A variable-mu (variable gain) compressor is an analog compressor centred around a vacuum tube. As the input signal increases, the current sent to the grid of the tube decreases, which results in a reduction in the overall level.

For more information on variable-mu compressors, check out the following My New Microphone articles:
What Is A Variable-Mu (Tube) Compressor & How Does It Work?
Top 11 Best Variable-Mu Compressor Emulation Plugins

VCA Compression

What is a VCA compressor? A VCA (voltage-controlled amplifier) compressor is an analog compressor that utilizes a VCA control to apply compression. The input signal is split through an integrated circuit into a detector path (to control the VCA compression) and an output path.

For more information on VCA compressors, check out the following My New Microphone articles:
What Is A VCA Compressor & How Does It Work?
Top 11 Best VCA Compressor Emulation Plugins For Your DAW


What is an audio de-esser? De-essing is the process of attenuating sibilance and/or harshness in a vocal/voice audio signal. This can be achieved using a dynamic EQ, multi-band compressor, sidechain compressor with automation in a mix, or manually.

Sibilance can be quickly defined as the hissing sound. In English, sibilance happens on the consonant sounds of S, Z, Sh, and Zh (as is “leisure” – lei-zh-ure). Though a necessary part of speech intelligibility, sibilance can often be overly harsh in a vocal track and may require attention to smooth out.

Sibilance is typically in the frequency range of 5 kHz to 8 kHz (though it may occur below or above that range).

A de-esser is designed to reduce the harshness of sibilance by attenuating the sibilant frequencies when they reach a certain amplitude.

De-essing is often achieved via a multi-band compressor that compresses the sibilant frequency band more than the other bands. It can also be achieved via a dynamic EQ set to attenuate sibilant frequencies as they surpass a set threshold.


What is audio distortion? Audio distortion, technically speaking, is any change/deformation in an output waveform relative to its input causes by any non-linear behaviour of the signal path. There are many different types of distortion. Some are unwanted but many are used as audio effects.

Distortion can be a wanted or unwanted effect in our audio signals as they travel through their respective signal chains. There are many factors that may introduce distortion to the signal and some factors are more controllable than others.

We won’t be getting into great detail about distortion in this article. Rather, we’ll touch briefly on the various distortion effects that are used in music and audio with a focus on controllable, wanted distortion effects/processes.


What is audio bitcrushing? Bitcrushing is an audio distortion effect that causes distortion by reducing the resolution (bit depth) or bandwidth (sample rate) of a digital audio signal. By reducing the resolution and/or bandwidth, the digital signal becomes less accurate with more quantization noise and distortion.

Bitcrushing, then, is a digital distortion effect. Digital audio can be thought of as a digital representation of an analog audio signal. A digital audio signal will have a sample rate (the amount of samples per second that are taken of the signal) and a bit-depth (the number of possible amplitudes that may be assigned to each sample).

The higher the same rate and bit-depth, the most accurate the representation.

Bitcrushing reduces the sample rate and/or the bit-depth of the digital signal, thereby causing digital distortion to the waveform.

The best way to describe bitcrushing, without hearing the effect, is by visualizing the effect. Let’s have a look at two signals. Each is a digital representation of a 1 kHz Sine wave. One full cycle has a period of second.

Here is a visual representation of the 1 kHz sine wave represented digitally with a 48 kHz sample rate and a 24-bit bit-depth. Notice how the digital representation is close but not perfectly representative of the analog signal (dotted line):

Now let’s look at this same analog waveform represented digitally with a sample rate of 12 kHz and a bit-depth of 8-bits. Notice how much more distorted the resulting digital signal is:


What is audio clipping? Clipping is a form of distortion where the amplitude of the waveform tries to exceed the maximum possible amplitude and is thereby “clipped” at its max. This is possible in analog amplifiers (both tube/valve and solid-state/transistor-based) and in digital signal processing.

Hard clipping, where the audio signal is completely flattened at the maximum amplitude, can be achieved via hard limiting; attempting to exceed the maximum digital binary amplitude, or with a specialized clipping effect. Fuzz and distortion effects utilize hard clipping.

Here’s a visual representation of clipping where the blue signal is pre-amplification/compression and the red signal is hard clipped:

There is also such thing as “soft clipping”, where the signal waveform is not completely flattened at the top. This type of clipping is found in typical compression, overdrive and saturation effects.

Here is a visual representation of soft clipping. Notice how similar it is to the aforementioned photo representing compression.


Confusingly enough, there’s a distortion effect in audio called “distortion”.

What is the audio distortion effect? The “distortion” effect in audio is produced by hard-clipping the input signal through transistor-based or tube-based circuits. Distortion is the most common on electric guitars but also suits many other instruments.

With the distortion effect, one or multiple clipping stages (often, but not always, transistor-based) are used to drive the signal into hard-clipping. The resulting signal is compressed and full of added harmonics, particularly odd-ordered harmonics.

Distortion will sound more aggressive and edgy than typical overdrive, saturation or compression. The signal’s dynamic range will shrink and its harmonic character will change significantly.

Related articles:
Guitar Pedals: Boost Vs. Overdrive Vs. Distortion Vs. Fuzz
Top 11 Best Distortion Pedals For Guitar & Bass

Top 11 Best Distortion Plugins For Your DAW


What is the audio fuzz effect? The Fuzz effect is caused by hard-clipping a signal so much that it nearly turns into a square wave. It’s perhaps the most extreme example of distortion as an audio effect and completely changes the sound of the audio signal.

Fuzz is an effect that drives the audio signal so far into hard clipping that the output becomes a square wave (or at least a waveform very close to a square wave).

We can visualize this with the following photo:

The fuzz effect pretty much kills all dynamics in the signal and alters the harmonic content of the signal significantly by producing the typical odd-order harmonics of a square wave.

Related articles:
Guitar Pedals: Boost Vs. Overdrive Vs. Distortion Vs. Fuzz
Top 12 Best Fuzz Pedals For Guitar & Bass


What is audio overdrive? Overdrive is a distortion effect that is caused by (or aims to emulate) pushing a tube amplifier just past its amplitude limits. The signal is compressed and “soft-clipped,” resulting in warm saturation in the signal.

The overdrive effect can be achieved by simply pushing a tube amp (or even many solid-state amps). However, there are dedicated overdrive effects that will simulate the effect with greater precision without the need to crank up the volume of an amp.

The effect is that of soft-clipping, as we discussed earlier. This causes some amount of compression and saturation in the signal. Overdrive offers a bit of extra grit without having too much of an effect on the dynamic range of the signal.

Related articles:
Guitar Pedals: Boost Vs. Overdrive Vs. Distortion Vs. Fuzz
Top 10 Best Overdrive Pedals For Guitar & Bass

Tape Saturation

What is tape saturation? Tape saturation is a type of distortion that happens when the voltage sent to an analog tape exceeds the tape’s ability to record it. In playback, this causes a non-linear saturation of the original source.

Tape saturation can be achieved by using actual tape. There are plenty of effects units (hardware and software) that aim to emulate this effect as well.

Valve Saturation

What is valve saturation? Valve saturation is a type of distortion that happens when a valve (vacuum tube) amplifier is overloaded at its output. If the amplification attempts to exceed the tube’s maximum output, clipping distortion/saturation will occur in which the output becomes distorted.

Valve saturation is what the overdrive effects aim to emulate.

Related article: Top 11 Best Saturation Plugins For Your DAW


What is an audio exciter? An audio/aural exciter is a type of parallel saturation/filter combo effect in which a signal is saturated only in the top-end (often above 3 kHz) in order to enhance or “excite” the sound. Think of it as duplicating the signal, high-passing and saturating the copy and mixing the two back together.

Audio “exciting” is a type of parallel process that can be achieved with or without a dedicated exciter unit/plugin.


What is audio expanding? Expanding can be thought of as the opposite of compression. It aims to increase the dynamic range of the signal. An expander will reduce the amplitude of the signal if it drops below the set threshold, thereby “expanding” the signal’s dynamic range.

As explained above, expansion can be thought of as the opposite of compression and increases the dynamic range of a signal.


What does level mean in terms of audio? The term level refers to the strength of an audio signal. In mixing/production, we generally control the relative levels of tracks with faders (analog or digital) in order to get the balance we want from the various elements in the mix. Automating levels over time is a common mixing technique.

Level can be changed by various means:

Related articles:
Top 7 Best Volume Pedals For Guitar & Bass
Top 9 Best Boost/Preamp Pedals For Guitar & Bass


What is audio limiting? Limiting is a type of hard compression whereby the signal is not allowed above a certain threshold. Rather than attenuating the signal (above the threshold) by a ratio, the limiter will simply cut off the signal at the threshold. We can think of a limiter as a compressor with an infinite ratio.

Related article: Top 10 Best Limiter Plugins For Your DAW

Noise Gating

What is audio noise gating? Noise gating is an effect that kills the output signal if the input signal drops below a set threshold. This helps to gate or remove noise from the signal when an instrument (or other sound source) is not playing. Noise gates are especially useful in noisy rigs, which often include vintage gear.

A noise gate to an expander is what a limiter is to a compressor. By effectively muting the output if an input signal drops below a certain threshold, a noise gate will gate out any noise when an appropriate amount of signal is present.

Noise Reduction

What is audio noise reduction? Noise reduction is the process of removing, or at least attenuating, noise from an audio signal. Various noise reduction technologies exist and can be applied during recording; as noise happens, and/or during playback.

Noise reduction can be a fairly complex process that attempts to isolate noise from the intended signal and to eliminate this noise from the output.

Related article: Top 7 Best Noise Reduction Plugins For Your DAW

Transient Shaper

What is an audio transient shaper? A transient shaper is a dedicated envelope control for manipulating the attack (and often the decay and release) of individual transients in an audio signal. These transients are typically percussive hits or notes. Transient shapers can either harden or soften the transient attacks within a signal.

A transient shaper can be thought of as a combination of a compressor and expander. It can be set to expand or compress certain parts of the transient. Notably, a transient shaper can expand (increase the amplitude) of the initial transient and compress (decrease the amplitude) of the transient tail, thereby sharpening the transient of the signal.

Compressors trigger when their input signal passes a threshold. Transient shapers, on the other hand, trigger based on the rate at which their input signal level increases.

Modulation Effects

Modulation effects, loosely defined, are effects that modify the source audio signal with another signal (generally an oscillator). Because of this loose definition, there are plenty of effects that fall into the modulation category.

The usual culprits are chorus, flanger and phaser, though the modulation-type effects span much further than this.

By modulating an audio signal with another signal, we can achieve all sorts of cool effects. By our definition, sidechain compression is even a type of modulation effect, though we’ve already spoken about this in the Dynamic Effects section.

Modulation dynamic effects/processes include the following:

For an in-depth resource on modulation effects, please check my article The Complete Guide To Audio Modulation Effects (With Examples).


What is the chorus effect in audio? Chorus is an effect that produces copies of a signal (the original signal and each of its copies has its own “voice”) and detunes each voice to produce a widening and thickening of the sound. Each voice interacts with the other voices to produce slight modulation and an overall larger-than-life sound.

The chorus effect is named after the use of chorus in music. That is, a group of people singing or playing the same note in unison. Naturally, there will be some slight pitch variation in the voices that make up the chorus. This slight detuning varies across time.

The chorus effect mimics this by running the input signal through a delay circuit and modulating the delay time with an LFO (low-frequency oscillator).

The delay time is generally in the range of 18-24 milliseconds. By modulating the delay time, the chorus causes pitch variation in the delayed signal. As the delay time is shortened, the waveform is slightly compressed, causing an increase in pitch/frequency. As the delay time is lengthened, the waveform is slightly stretched, causing a decrease in pitch/frequency.

Mixing these two signals together yields the chorus effect.

Though only one copy of the delayed signal is necessary to achieve the effect, some chorus units utilize multiple.

Also note that chorus is different than the unison effect found in some synthesizers, which simply adds detuned oscillators to the output but does not modulate a delayed copy of the original oscillator(s).

Related articles:
Top 11 Best Chorus Modulation Plugins For Your DAW
Top 11 Best Chorus Pedals For Guitar & Bass
What Are Chorus Pedals (Guitar/Bass FX) & How Do They Work?
Complete Guide To The Chorus Audio Modulation Effect?


What is the flanger effect in audio? Flanger is a modulation audio effect whereby a signal is duplicated and the phase of one copy is continuouly being shifted. This changing phase causes a sweeping comb filter effect where peaks and notches are produced in the frequency spectrum or the signal’s EQ.

The flanger effect was first heard by playing two identical tapes in parallel and pressing down on the flange of one of the tapes to cause a comb filter sweep across the combined output as one tape fell out of sync (became more and more delayed relative to the other).

Flanger units achieve this effect modulating the delay time of a delay line (with an LFO) and feeding the delayed copy of the signal back into itself. Mixing the dry and wet/delayed signals together produces the famed “jet whoosh” comb filter sweep of the flanger effect.

The delay time of a flanger’s delay line should not exceed 20 milliseconds in order to maintain a tight enough phase shift.

Related articles:
Top 9 Best Flanger Modulation Plugins For Your DAW
Complete Guide The Flanger Audio Modulation Effect?
Top 11 Best Flanger Pedals For Guitar & Bass
What Are Flanger Pedals (Guitar/Bass FX) & How Do They Work?


What is the phaser effect in audio? Phaser is a modulation audio effect whereby a series of peaks and troughs are produced across the frequency spectrum of the signal’s EQ. These peaks and troughs vary over time, typically controlled by an LFO (low frequency oscillator), to create a sweeping effect known as phaser.

The phaser effect causes a series of notch filters to sweep across the frequency response, creating a unique effect.

A phaser will work with a series of all-pass filters that do nothing to alter the frequency of the signal but do alter the phase of the signal. For every 2 APFs, there will be 1 notch in the resulting frequency response of the output as certain frequencies fall out-of-phase.

The locations of these notch filters along the resulting output frequency response are then modulated via an LFO.

Related articles:
Top 9 Best Phaser Modulation Plugins For Your DAW
Complete Guide To The Phaser Audio Modulation Effect
Top 11 Best Phaser Pedals For Guitar & Bass
What Are Phaser Pedals (Guitar/Bass FX) & How Do They Work?

Ring Modulation

What is the ring modulation effect in audio? Ring modulation is an amplitude modulation effect where two signals (an input/modulator signal and a carrier signal) are summed together to create two brand new frequencies which are the sum and difference of the input and carrier signals. The carrier is typically a simple wave selected by the effects unit while the modulator signal is the input signal.

A ring modulator’s input signal will have certain frequency content. Each of these frequencies is modulated by the carrier signal and the unit outputs a new frequency profile that is the sum and difference of each of the frequencies.

This is easier to visualize with a sine wave (single frequency) carrier and a simple modulator. Let’s have a look at a 100 Hz triangle wave with only 4 harmonics (odd-order at 300, 500, 700 and 900 Hz) as the modulator and a 50 Hz sine wave as the carrier.

Here’s a frequency graph of the 100 Hz triangle wave:

The sidebands of the triangle and sine wave would be produced at each harmonic:

  • 100 Hz fundamental would become:
    • 100 – 50 = 50 Hz
    • 100 + 50 = 150 Hz
  • 300 Hz first harmonic would become:
    • 300 – 50 = 250 Hz
    • 300 + 50 = 350 Hz
  • 500 Hz second harmonic would become:
    • 500 – 50 = 450 Hz
    • 500 + 50 = 550 Hz
  • 700 Hz third harmonic would become:
    • 700 – 50 = 650 Hz
    • 700 + 50 = 750 Hz
  • 900 Hz fourth harmonic would become:
    • 900 – 50 = 850 Hz
    • 900 + 50 = 950 Hz

The ring modulator output, in this example, would resemble the following:

Related article:
Top 8 Best Ring Modulation Pedals For Guitar & Bass
What Are Ring Modulation Effects Pedals & How Do They Work?
Complete Guide To The Ring Modulation Audio Effect

Rotary Effect

What is the rotary effect in audio? The rotary effect (aka Leslie effect) was initially produced by the famous Leslie speaker, a unit with a rotating speaker. As the speaker rotates, three separate effects are produced in the [stationary] listener’s ears. Those effects are tremolo, the Doppler effect (vibrato) and Phasing.

The rotary effect can be produced by a rotating speaker or with an effects unit.

To achieve the rotary effect is to combine tremolo, vibrato and phasing (a phaser) into a single chain or effect.


What is the tremolo effect in audio? Tremolo is a fast variation in amplitude. Tremolo is similar to vibrato, except that it acts on amplitude/level rather than pitch.

Tremolo is actually an outlier in some ways. Though it’s often classified in the modulation category of effects, it actually does not rely on a modulated delay circuit or filter circuit whatsoever. In many ways, tremolo fits better as a dynamic effect as it’s simply the amplitude of the signal that’s modulated over time.

What’s more, is the confusion that often arises when comparing tremolo and vibrato, especially among guitarists. The “tremolo arm” (aka whammy bar) on a guitar actually designed to provide an effect similar to vibrato (called a glissando, since it’s smooth and is not linked to time), not tremolo.

Tremolo is achieved by modulating the output level of a signal with an LFO.

Related articles:
Top 11 Best Tremolo Modulation Plugins For Your DAW
Complete Guide To The Tremolo Audio Modulation Effect?
What Are Tremolo Guitar Effects Pedals & How Do They Work?
Top 11 Best Tremolo Pedals For Guitar & Bass


What is the vibrato effect in audio? Vibrato is a fast but slight up-and-down, variation in pitch. Vibrato is used in signing and in instruments to add character and improve tone.

Vibrato is achieved by running a signal through a delay line and modulating the delay time with an LFO. Only the wet/delayed signal is outputted.

As the delay time is shortened, the waveform is slightly compressed, causing an increase in pitch/frequency. As the delay time is lengthened, the waveform is slightly stretched, causing a decrease in pitch/frequency.

Related articles:
Top 8 Best Vibrato Pedals For Guitar & Bass
What Are Vibrato Guitar Effects Pedals & How Do They Work?

Sound Manipulation Processes

The sound manipulation category is a rather vague since all effects/processes with “manipulate” the audio signal in one form or another.

Sound manipulation, in the case of this article, is any process that will alter the sound of the audio without acting directly upon the dynamic or spectral content of the signal.

These processes can be achieved within samplers, some synthesizers, digital audio workstations and other audio tools. There’s aren’t any dedicated effects units for these processes but they’re worth mentioning in this article anyway.

The sound manipulation processes we’ll discuss in this article are as follows:


What is the reverse effect in audio? The reverse audio effect is any effect that reverses an audio signal for playback. This can be reversing an audio clip in a DAW; playing a tape backward, or even capturing audio in a delay circuit and playing it back in reverse after a set period of time.

Time Compression

What is time compression in audio? Audio time compression is the process of decreasing the length (in time) of an audio file. Depending on the processor, time compression may or may not increase the pitch of the audio.

Time Expansion

What is time expansion in audio? Audio time expansion is the process of increasing the length (in time) of an audio file. Depending on the processor, time expansion may or may not decrease the pitch of the audio.

Spectral Processes

Spectral processes refer to the frequency spectrum of audio and, sometimes, to the panoramic spectrum of an audio mix.

These effects/processes, then, have to do with altering the frequency information of an audio file or the position of the audio files in a stereo or multi-channel mix.

Spectral processes include the following:


What is audio equalization? Audio equalization (EQ) is the process of altering the amplitude of certain frequencies/frequency bands in an audio signal. This includes filtering out sound below or above a certain cut-off frequency (high-pass and low-pass filtering, respectively). It also refers to shelving, notching, boosting and cutting frequencies.

In other words, EQ (equalization) is the process of adjusting the balance between frequencies within an audio signal. EQ typically works on the audible frequency range between 20 Hz and 20,000 Hz though some EQ units are capable of affecting frequencies beyond this audible range.

By adjusting the EQ of a signal, we can mix it into the general context of the mix and improve the character of the audio by reducing problem/competing frequencies and boosting characteristic frequencies.

EQ can cut out noise and low-end rumble; alter the timbre of an audio source and even make source appear closer or further away in the mix.

We can think of EQ as a frequency-specific volume/gain control. We can turn up some frequencies while turning others down.

EQ, along with compression, is a critical tool in the mixing engineer’s arsenal.

Related articles:
Complete Guide To Audio Equalization & EQ Hardware/Software
Top 12 Best EQ Pedals For Guitar & Bass
What Are EQ Pedals (Guitar/Bass) & How Do They Work?

Dynamic EQ

What is dynamic audio equalization? Dynamic EQ is a type of equalization where the EQ of certain frequencies is triggered dynamically as those frequencies surpass a set amplitude threshold in the audio signal. Dynamic EQ, like a compressor, will have threshold, attack and release settings to alter the EQ of a signal dynamically.

Dynamic EQ is actually quite similar to multi-band compression in the way that it attenuates specific bands of frequencies as these frequencies surpass a set threshold.

Unlike a multi-band compressor, a Dynamic EQ doesn’t split the signal into bands to begin with. Rather, each EQ band (regardless of filter type) is only engaged as the frequencies within that band exceed the set threshold of that band. Dynamic EQs can also boost frequencies as the band exceeds the threshold, whereas a multi-band compressor would require expansion capabilities to achieve the same thing.

For more information on dynamic EQ, check out the following My New Microphone articles:
The Complete Guide To Dynamic Equalization/EQ
Top 10 Best Dynamic EQ Plugins For Your DAW

Graphic EQ

What is graphic audio equalization? Graphic equalization is a style of EQ where predetermined bands, centred around set frequencies with set Q values, can be either boosted or cut. The name comes from the fact that the EQ settings of a graphic EQ unit typically look very obvious and “graphic”.

Graphic EQ is easy to understand but limited in its flexibility to EQ audio. It gives us a set number of bands centred around set frequencies. We chose whether to boost or cut at any of these bands.

For more information on graphic EQ, check out the following My New Microphone articles:
The Complete Guide To Graphic Equalization/EQ
Top 8 Best Graphic EQ Plugins For Your DAW

Parametric EQ

What is parametric audio equalization? Parametric EQ offers full customization of the frequency bands including the choice of filter type; centre frequency; Q value and relative gain (boost/cut).

A parametric EQ gives us a great deal of control over the individual bands we’re able to affect.

We’re able to sweep the frequency of a parametric EQ and set it exactly where we need it to be. We can also control the Q parameter and, of course, the amount of gain.

A parametric EQ will often featured a high-pass filter and low-shelf option as well as a low-pass filter and high shelf-option, each with adjustable cutoff/set points.

Some software parametric EQs allow us to change the typical bell-type bands in the centre into notch or band-pass filters as well.

For more information on parametric EQ, check out the following My New Microphone articles:
The Complete Guide To Parametric Equalization/EQ
Top 10 Best Digital Parametric EQ Plugins For Your DAW
Top 10 Best Parametric EQ Emulation Plugins For DAWs

Semi-Parametric EQ

What is semi-parametric audio equalization? Semi-parametric EQ offers some, but not all, of the customization of a parametric EQ. The customization of the frequency bands could include the choice of filter type; centre frequency; Q value and relative gain (boost/cut).

For more information on semi-parametric EQ, check out my article What Is Semi-Parametric Equalization/EQ In Audio?

Shelving EQ

What is shelving audio equalization? Shelving equalization is the boosting or cutting of frequencies by a certain decibel level above (high shelf) or below (low shelf) a certain frequency. Shelf filters will nearly always have a slope across a few frequencies as the EQ is boosted or cut.

To learn more about shelving filters, check out my article Audio Shelving EQ: What Are Low Shelf & High Shelf Filters?


Band-pass Filter

What is a band-pass filter in audio? A band-pass filter “passes” a band of frequencies (a defined range above a low cutoff and below a high cutoff) while progressively attenuating frequencies below the low cutoff and above the high cutoff.

  • Notable band-pass filter Eurorack module: York Modular rBPF2

To learn more about band-pass filters, check out my article Audio EQ: What Is A Band-Pass Filter & How Do BPFs Work?

Bell Curve Filter

What is a bell curve filter in audio? A bell curve filter is a second-order (or higher) filter capable of producing resonance (boost in EQ) or anti-resonance (cut in EQ) around a specified frequency. These filters are defined by a central frequency, Q factor (width of the boost/cut) and relative gain.

Envelope Filter

What is a envelope filter in audio? Envelope filtering is the filtering triggered by the envelope or transients of a signal. These filters, therefore, act according to the dynamic rise and fall of a signal and are most often used on bass, guitar and synthesizer instruments.

Envelope filter can utilize band-pass, high-pass or low-pass filters and these filters can be swept upward or downward for a variety of effects.

Related articles:
Top 13 Best Envelope Filter Pedals For Guitar & Bass
What Are Envelope Filter Effects Pedals & How Do They Work?
What Is The Auto-Wah/Envelope Filter Modulation Effect?

High-pass Filter

What is a high-pass filter in audio? A high-pass filter (HPF) “passes” the high-frequencies above their cutoff frequency while progressively attenuating frequencies below the cut-off frequency. In other words, high-pass filters remove low-frequency content from an audio signal below a defined cut-off point.

Notable high-pass filter Eurorack module: Aion Modular 904B

Related articles:
What Is A Microphone High-Pass Filter And Why Use One?
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High Shelf Filter

What is a high shelf filter in audio? A high shelf filter is a filter that either boosts (increases amplitude) or cuts (decreases amplitude) frequencies above a certain frequency set point. These first-order filters can have a slope of up to 6 dB per octave in the transition region.

To learn more about high-shelf filters, check out my article Audio Shelving EQ: What Are Low Shelf & High Shelf Filters?

Low-pass Filter

What is a low-pass filter in audio? A low-pass filter (LPF) “passes” the low-frequencies below their cutoff while progressively attenuating frequencies above their cutoff. In other words, low-pass filters remove high-frequency content from an audio signal above a defined cut-off point.

Notable low-pass filter Eurorack module: Aion Modular 904A

To learn more about low-pass filters, check out my article Audio EQ: What Is A Low-Pass Filter & How Do LPFs Work?

Low Shelf Filter

What is a low shelf filter in audio? A low shelf filter is a filter that either boosts (increases amplitude) or cuts (decreases amplitude) frequencies below a certain frequency set point. These first-order filters can have a slope of up to 6 dB per octave in the transition region.

To learn more about low-shelf filters, check out my article Audio Shelving EQ: What Are Low Shelf & High Shelf Filters?

Notch Filter

What is a notch filter in audio? A notch filter (aka band-reject filter) works by removing frequencies in a specified band within the overall frequency spectrum. It allows frequencies below the low cutoff point to pass along with frequencies above the high cutoff point.

To learn more about notch/band-reject/band-stop filters, check out my article Audio EQ: What Is A Band-Stop Filter & How Do BSFs Work?


What is the audio wah effect? Wah (or Wah-Wah) is a filtering effect that is common on guitars and keyboard instruments. Wah is achieved by sweeping one or more boosts in EQ up and down in frequency, thereby mimicking the human vowel sound of “wah”.

Wah effects aim to achieve the same spectral glide as the human voice saying “wah” forward and backward. The modulation of the EQ peaks caused by the effect is designed to resembled the movement of formants in the natural response of the human voice.

Related articles:
Top 14 Best Wah Pedals For Guitar & Bass
What Are Wah-Wah Guitar Effects Pedals & How Do They Work?


What is audio imaging? Imaging, as an audio production technique, is any effect that involves the perceived spatial location of the sound source(s) in a mix, both laterally (stereo/surround direction) and in depth. Panning, level, time-based effects, recording position and mic techniques all play a role in imaging.

Related article: Top 9 Best Stereo Imaging Plugins For Your DAW


What is audio panning? Audio panning is the placement of a sound signal (mono or otherwise) into a stereo or multi-channel (surround sound, for example) sound field format. Panning refers to the “panorama” of stereo/multi-channel formats and gives a sound a sense of direction in the audio image.

Pitch Correction

What is pitch correction in audio? Pitch correction is the process of identifying and altering the intonation of an audio signal in order to alter the pitch of notes without (ideally) altering any other aspects of the sound.

Auto-Tune is perhaps the most famous pitch correction tool.

Pitch Shifting

What is pitch shifting in audio? Pitch-shifting, as the name suggests, is any audio effect that shifts the pitch of the input signal.

Related articles:
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Time-Based Effects

Time-based effects include all processes where some form of time manipulation occurs to the signal. The most obvious examples of time-based effects are delay and reverb.

Technically speaking, there are plenty of effects that utilize time manipulation but they generally belong to a superseding category. For example, chorus and flanger use time-modulated delay while phasers use time-modulated phase shifting. Even pitch shifting and time expansion/compression can be considered time-based effects.

However, for the sake of main effect categories, the main time-based effects are:


What is the delay effect in terms of audio? Delay is a time-based effect where an input signal is recorded, for a relatively short amount of time, and is played back after a set period of time after the initial recording. There are many ways to achieve delay and different style/types of the effect.

There are plenty of delay effects out there. They will generally have the following parameters:

  • Effect level: the relative amplitude of the delayed signal, often in relation to the dry/input signal.
  • Delay time: the amount of time between the initial signal and the delayed signal.
  • Repeats: the number of times the signal is repeated before trailing off.
  • Feedback: how much, if any, of the delayed signal is fed back into the delay circuit.

Depending on the delay effect unit, the processed/delayed signal can be played back once or multiple times. The delayed signal level can often be adjusted along with the mix relative to the original dry signal. It may also be fed back into the delay processing circuit to produce further delay.

If we compare the audio delay effect to the world of acoustics, it would be analogous to the initial reflections we’d hear after a sound bounces off a reflective surface.

In the real world, we can often hear “delay” as the sound is reflected around us. However, the delay is all too often drown out, when it’s noticeable at all, by the reverberation of the environment.

Related articles:
Top 13 Best Delay Pedals For Guitar & Bass
What Are Delay Pedals (Guitar Effects) & How Do They Work?

Analog (BBD) Delay

What is the analog (BBD) delay audio effect? Analog delay is a delay effect that utilizes bucket-brigade devices (BBDs). The BBD is a network of capacitors connected in cascade that, when controlled by a clock, will effectively delay the signal by a set time. The quality of each delay is degraded, giving a warm sound.

Related article: Top 10 Best Analog Delay Emulation Plugins For Your DAW

Digital Delay

What is the digital delay audio effect? Digital delay is a delay effect that utilizes digital signal processing to record an incoming signal and playback a delayed version of the signal. Parameters such as delay time, level, panning and feedback can be adjusted. Digital delay can also act to emulate/combine different types of delay.

Related article: Top 10 Best Digital Delay Plugins For Your DAW

Doubling Echo

What is the doubling echo audio effect? The doubling echo effect is a short single delay that mimics the effect of double-tracking or unison performance.

Haas Effect

What is the Haas effect? The Haas effect (aka precedence effect) is a psychoacoustic phenomenon that states that when one sound is followed by a delayed sound below 40 ms, the two will be perceived as a single sound. The Hass effect can be used to enhance the directionality or to increase the stereo width of a sound.

Ping Pong Delay

What is the ping pong delay audio effect? Ping pong delay is an effect where each subsequent delay of a signal alternates between the left and right stereo channels. This “back-and-forth” between the two stereo channels is reminiscent of the sound of a ping pong match, hence the name.

Reverse Delay

What is the reverse delay audio effect? Reverse delay is a type of delay where the delayed signal information is processed to reverse the audio before it is outputted. Reverse delay effect will have inherent lag as reversing a signal cannot be done in real-time during playback.

Shimmer Delay

What is the shimmer delay audio effect? Shimmer delay is a type of delay where the delayed signal is pitch-shifted upward (typically be an octave) to give a direct signal with a “shimmering” delay tail.

Slapback Delay

What is the slapback delay audio effect? Slapback delay is a delay with only a single repeat and a relatively short delay time (typically in the range of in the range of 40-120 milliseconds).

Tape Delay

What is the tape delay audio effect? Tape delay is a delay effect achieved by routing the audio signal to a second tape recorder. The audio would be fed to the secondary tape’s record head and be played back just milliseconds later on the playback head. This delayed playback signal would be sent back to the main recording.

Related article: Top 10 Best Tape Delay Emulation Plugins For Your DAW


What is looping in terms of audio? Looping is the process of recording a period of audio and having it repeat. Once a loop is established, it is common to stack other audio recordings on top of the loop (or take other audio recordings out of the loop).


What is the reverb effect in terms of audio? The reverb effect recreates the natural effect of Reverberation, which happens when a sound wave hits a surface (or multiple surfaces) and reflects back to the listener at varying times and amplitudes. This creates a complex echo that holds information about the physical space.

Reverb effects aim to recreate the natural sense of reverberation for the real world. Reverberation, in acoustics and psychoacoustics, is defined as the persistence of sound in the environment after the sound has been produced by its source.

Reverberation happens naturally as sound waves reflect off surfaces in the environment and reach our ears after we’ve heard the original source. As the sound waves bounce off a variety of surfaces and reach our ears at different times, the reverb is heard as an indistinct sound rather than a distinct repeat of the sound.

This is way over-simplified but it gets the general point across.

Let’s have a look at an illustration of a reverberant sound to further our understanding:

Let’s quickly define the four distinct sections of the reverb effect:

  • Source: the sound that is being affected by the reverb. With pedals, this is typically the guitar/instrument signal.
  • Early reflections: the starting of the reverb effect that is typically delayed 1 – 50 milliseconds after the source signal.
  • Reverb body: the reverb effect when all the reflections/delayed signals interact with each other to produce the “reverb effect” This typically starts around 25 and will last as long as the decay time control allows it to.
  • Reverb tail: the reverb tail is defined as the end of the reverb once the effect drops 60 dB below the source signal level.

Related articles:
12 Best Reverb Plugins (Spring, Plate, Algorithmic, Convolution)
Top 13 Best Reverb Pedals For Guitar & Bass
What Are Reverb Pedals (Guitar Effects) & How Do They Work?

Acoustic Emulation Reverb

What is the acoustic emulation reverb audio effect? Acoustic emulation reverb is a digital reverb affect that aims to emulate the reverb of a physical space. Common acoustic emulation reverb types include room, chamber, hall and cathedral reverbs.

Bloom Reverb

What is the bloom reverb audio effect? Bloom reverb is an unnatural digital reverb that “blooms” into effect as the reverb evolves. It has very quiet initial reflections that continuously increase in amplitude until a certain point where the reverb tail is kicked in.

Convolution Reverb

What is the convolution reverb audio effect? Convolution reverb is achievable by recording a physical acoustic space, sampling it and analyzing it thoroughly. Profiles are made for the initial reflections, decay time and frequency damping. An algorithm is then created to simulate that physical space.

Gated Reverb

What is the gated reverb audio effect? Gated reverb takes any reverb type (often room or chamber) and cuts off the decay once the reverb body drops below a certain amplitude.

Plate Reverb

What is the plate reverb audio effect? Plate reverb emulates the sound of reverb by vibrating a plate and recording the results. The plate is vibrated by an input transducer (converts an audio signal into vibrations). As the plate vibrates, an output transducer will convert these vibration to audio.

Reverse Reverb

What is the reverse reverb audio effect? Reverse reverb is a special effect in which the reverb tail of a sound is reversed for playback.

Shimmer Reverb

What is the shimmer reverb audio effect? Shimmer reverb combines pitch-shifting with reverb. The wet/effected reverb signal is pitched up (typically by an octave) relative to the dry/direct signal.

Spring Reverb

What is the spring reverb audio effect? Spring reverb emulates the sound of reverb by vibrating a spring and recording the results. One end of the spring is vibrated by an input transducer (converts an audio signal into vibrations). The other end of the spring connects to an output transducer (converts the spring vibrations into audio).


Arthur is the owner of Fox Media Tech and author of My New Microphone. He's an audio engineer by trade and works on contract in his home country of Canada. When not blogging on MNM, he's likely hiking outdoors and blogging at Hikers' Movement (hikersmovement.com) or composing music for media. Check out his Pond5 and AudioJungle accounts.

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