In The Studio: Audio Effects Explained (Includes Audio Samples)
Starting with a variety of modulation effects and moving along to a whole lot more...

August 11, 2014, by Jon Tidey

This article is provided by Audio Geek Zine.

A while ago I mentioned using modulation effects to help create movement within a mix. Here, I’ll explain the different types of modulation effects that we have available for mixing, and then move along to gates, compression, EQ, delay, reverb, de-essing, and a whole lot more.

The modulation effects I’ll be discussing include:

—Phasing or Phase Shifting

I’ll start with some easy ones then move on to the harder to explain—but more commonly used—effects.

All of them are built around a low-frequency oscillator, more commonly referred to as just an LFO. An LFO is an audio signal usually less than 20 Hz that creates a pulsating rhythm rather than an audible tone.

These are used for manipulate synthesizer tones, and as you will see, to create various modulation effects. All of the effects listed use sine wave as the wave shape for the LFO.


Tremolo is an effect where the LFO is modulating the volume of a signal. The signal attenuation amount is controlled by the depth and the rate adjusts the speed of the LFO cycles.

Listen to an example of Tremolo

Vibrato is an effect where the LFO is modulating the pitch of a signal. This is accomplished by delaying the incoming sound and changing the delay time continually. The effect usually not mixed in with the dry signal. The depth control adjusts the maximum delay time, and rate controls the lfo cycle.

Listen to an example of Vibrato

Flanging is created by mixing a signal with a slightly delayed copy of itself, where the length of the delay is constantly changing. Historically this was accomplished by recording the same sound to two tape machines, playing them back at the same time while pushing down lightly on one of the reels, slowing down one side. The edge of a reel of tape is called the flange, hence the name of the effect.

These days we accomplish the same effect in a much less mechanical way. Essentially the signal is split, one part gets delayed and a low frequency oscillator keeps the delay time constantly changing. Combining the delayed signal with the original signal results in comb filtering, notches in the frequency spectrum where the signal is out of phase.

We usually have depth and rate controls. The depth controls how much of the delayed signal is added to the original, and the rate controls how fast it will change.

Phasing (or phase shifting) is a similar effect to flanging, but is accomplished in a much different way. Phasers split the signal - one part goes through an all-pass filter then into an LFO, and is then recombined with the original sound.

An all-pass filter lets all frequencies through without attenuation, but inverts the phase of various frequencies. It actually is delaying the signal, but not all of it at the same time. This time the LFO changes which frequencies are effected.

Phase shifters have two main parameters: Sweep Depth, which is how far the notches sweep up and down the frequency range; and Speed or Rate, which is how many times the notches are swept up and down per second.

Listen to an example of Phasing

Chorus is created in nearly the same way as Flanging, the main difference is that Chorus uses a longer delay time, somewhere between 20-30 ms, compared to Flanging which is 1-10 ms. It doesn’t have the same sort of sweeping characteristic that Flanging has, instead is effects the pitch.

Again the LFO is controlling the delay time. The depth control affects how much the total delay time changes over time. Changing the delay time up and down results in slight pitch shifting.

Listen to an example of Chorus

You may have noticed that the majority of effects here involve delay. You can recreate most of the effects by using a digital delay with rate and depth controls, such as the Avid ModDelay2.


The different types and methods, and I’ll also explain the most important parameters. I’ll mostly be talking about the kinds you will be using when mixing and what is available as plugins.

Digital Reverb Technology
There are two ways of creating a reverb effect in the digital world, by using mathematical calculations to create a sense of space, which is called algorithmic. And, by creating an impulse response, a snapshot of a real space, and applying that to the sound, which is called convolution.

Reverb is essentially a series of delayed signals, and algorithmic reverbs work pretty well to recreate this. Most reverb plugins, stomp boxes, and racks are algorithmic style.

When you want really realistic reverb, then convolution can not be beat. To create an impulse response the creator goes into a room and records the sound of a starter pistol going off and the natural reverb of the room.

The recordings are then deconvolved in software which is removing the sound of the starter pistol from the recording, leaving only the reverb.

Sine wave sweeps can also be used for the impulse creation. This is a more accurate way of creating reverb because it also captures the character of the room, and the way different frequencies react in the room.

The same process can be used to create impulse responses of speaker cabinets, guitar amps, vintage rack gear or basically anything that can make a sound.

Analog Reverb Types
In the analog world there are a few other ways, most of which will not be available to the home studio musician, except for their recreations in plug-ins. Analog reverbs come in three flavors—plate, spring, and chamber.

Invented in 1957 by EMT of Germany, the plate reverb consist of a thin metal plate suspended in a 4-foot by 8-foot sound proofed enclosure. A transducer similar to the voice-coil of a cone loudspeaker is mounted on the plate to cause it to vibrate. Multiple reflections from the edges of the plate are picked up by two (for stereo) microphone-like transducers. Reverb time is varied by a damping pad which can be pressed against the plate thus absorbing its energy more quickly.

This is what a plate reverb sounds like: platereverb.mp3

A spring reverb system uses a transducer at one end of a spring and a pickup at the other, similar to those used in plate reverbs, to create and capture vibrations within a metal spring. You find these in many guitar amps, but they were also available as stand alone effect boxes. They were a lot smaller than plate reverbs and cost a lot less.

This is a spring reverb: springverb.mp3

The first reverb effects used a real physical space as a natural echo chamber. A loudspeaker would play the sound, and then a microphone would pick it up again, including the effects of reverb. Although this is still a common technique, it requires a dedicated soundproofed room, and varying the reverb time is difficult.

This is a chamber: Chamber.mp3

These three types of reverb are all available in digital form in addition to a few other styles simulating real spaces, and others not found in nature.

Natural Reverb Types

Room – A room is anything from a classroom to conference room. There is generally a short decay time of about 1 second: room.mp3

Hall – A hall is larger than a room, it could be from a small theatre with 1 second of decay up to a large concert hall with a decay time up to 2.5 seconds: hall.mp3

Church – The decay time of a church can vary between 1.5 seconds to 2.5 seconds: church.mp3

And Cathedral decay times can go above 3.5 seconds: cathedral.mp3

Remember, the sound of a room is not just the decay time. The materials it was built with make a huge impact on the character of the sound. Stone, wood, metal and tile all sound drastically different.

There are also a few other types of reverb that are not natural - these are Non Linear, Gated and Reversed.

Non-Linear has a decay that doesn’t obey the laws of physics: non-lin.mp4

Gated was a popular effect in the 1980s, but it’s sounding pretty cheesy these days: gated.mp3

Reversed sounds like this: reverse.mp3

Reverb Parameters

Reverb Type – What kind of reverb emulation it is. There are Halls, Rooms, Chambers, Plates, etc…

Size – What the physical size of the space is. This can range from small through large.

Diffusion – How far apart the reflections are from each other.

Pre-Delay – Sets a time delay between the direct signal and the start of the reverb

Decay Time – Also known as RT60, which is how long it takes for the signal to reduce in amplitude 60 decibels.

Mix (Wet/Dry) – Sets the balance between the dry signal and the effect signal. When you have the reverb effect on an insert you need to adjust the wet and dry ratio, when you are sharing the reverb in a send and return configuration you want the mix to be 100 percent wet.

Early Reflection Level – Controls the level of the first reflection you hear. Early reflections help determine the dimensions of the room.

High Frequency Roll Off – Helps control the decay of high frequencies (as it is found in natural reverb).

Tips For Using Reverb

—Using pre delay can help keep your vocals up front, while still giving them space.

—Try to keep decay times short for faster tempo music.

—Filter out low frequencies before the reverb to keep it from sounding muddy

—Try de-essing the reverb to reduce harsh sibilance.

EQ & Filtering

The terms EQ and filter seem to mean different things.

Filtering is generally what we say when we want to remove frequencies, and EQ is when we want to shape the sound by boosting and cutting.

The truth is, it’s all filtering.


Cutoff – Selects the frequency. This is measured in Hertz

Gain – How much boost or attenuation at the cutoff frequency. This is measured in decibels

Shape or Type – This chooses what kind of filter you will be using. The filter shapes are hi and low pass, band pass, peaking, notch, and shelf

Quality or Width – Usually just referred to as Q is the shape of the EQ curve and how much of the surrounding frequencies will be affected.

What does an equalizer actually do?
An equalizer adjusts the balance of frequencies across the audible range. EQ is an incredibly powerful tool for crafting a mix.

Filter Shapes

A low-pass filter also known as high-cut filter removes frequencies above the cutoff. A high-pass filter or low-cut filter does the opposite, it removes everything below the cutoff.

Low-Pass Filter Clip (click to play)
High-Pass Filter Clip (click to play)

When you use both these filter types at once, it’s called a band-pass filter, the top and bottom frequencies are removed. With these three filter shapes, Q effects the steepness of the filter.

Band-Pass Filter Clip (click to play)

A notch filter is the opposite of a band pass filter, it lets all frequencies through except for a narrow notch in the spectrum which is attenuated greatly. The Q effects the width of the notch.

Notch Filter Clip (click to play)

There are two filter shapes that allows you to control how much the frequency will be attenuated.

A low-shelf EQ will boost or cut anything below the cutoff, a high-shelf EQ gives you boost or cut above the cutoff. You can choose how steep the slope is with the Q control.

Low-Shelf Clip (click to play)
High-Shelf Clip (click to play)

A peaking filter is also known as a bell curve EQ; you can boost or cut any frequency with a peak at the cutoff and a slope on either side.

Peaking Filter Clip (click to play)

EQ Designs

There are two main types of EQ designs: graphic and parametric.

Graphic equalizers have multiple peaking filters at specific frequency bands and give you a few overlapping frequency bands with adjustable gain. These are most commonly seen on consumer music players, but in professional audio these are very useful for mixing live music.

Parametric equalizers give you the most flexibility you can choose the shape, cutoff, gain, and quality. This is the type you will be using when mixing. Nearly all plug-in equalizers will be parametric.

EQ Usage Tips

—Use high-pass filters to remove unnecessary low frequencies from your tracks

—Use notch filters to remove unwanted noises from a recording

—Get rid of the frequencies you don’t need before boosting the ones you do, although it may not be your first instinct when EQing, it works a lot better

—High Q values will cause ringing or oscillation when boosted, this is not usually something you want to happen

—Adjust the EQ so that the level remains constant whether engaged or bypasses, it’s too easy to be fooled by louder being better

Some of my favorite equalizer plug-ins:

Apulsoft ApQualizer: Very clean EQ with 64 bands, frequency analyzer and complete control.

Stillwell Audio Vibe-EQ: A vintage style EQ that has some nice coloration, I like it most on electric guitars.

Avid EQ III: Standard included Pro Tools plug-in does the job 99 percent of the time.


In its simplest form, a delay is made up of very few components.An audio input, a recording device, a playback device and an audio output.

Tape Delay
Early delay processors, such as the Echosonic, Echoplex and the Roland Space Echo, were based on analog tape technology. They used magnetic tape as the recording and playback medium.

Some of these devices adjusted delay time by adjusting the distance between the playback and record heads, and others used fixed heads and adjustable tape speed.

Analog Delay
Analog delay processors became available in the 1970s and used solid state electronics as an alternative to the tape delay.

Digital Delay
In the late 1970s, inexpensive digital technology led to the first digital delays. Digital delay systems function by sampling the input signal through an analog-to-digital converter, after which the signal is recorded into a storage buffer, and then plays back the stored audio based on parameters set by the user. The delayed (“wet”) output may be mixed with the unmodified (“dry”) at the output.

Software Delay
And these days you’ll most likely be using plug-ins for your delay processing, same principles, just without the moving parts, additionally, they can sound pretty close to any of the other styles or be totally unique like OhmBoyz below.


Effect Parameters
OK, so that’s it for the history of delay processors. Now let’s move on to the parameters.

Delay Time—How long before the sound is repeated

Tempo Sync—Each repeat of the delay will be in time with the song, ¼ notes or 1/8 notes

Tap Tempo—Tap this button along with the song to set the delay time

Feedback—Output is routed back to input for additional repeats

Mix/Wet-Dry—Mix of original signal with delayed signal

Rate—LFO rate to change delay time

Depth—Range of delay time change for LFO

Filter—Usually a high cut filter, each repeat gets darker sounding

Stereo delays often have separate Left and Right controls.

What Does It Do?
Now, what sort of sounds can we get with delay processors?

An automatic double tracking effect can be accomplished by taking a mono signal, run it into a stereo delay, have no processing on the left side, and a very short delay on the right side. Have a listen here.

A slap back or slap delay has a longer delay time from about 75 to 200 milliseconds. This is the sort of delay was a characteristic of the 50s rock n roll records. Listen to it on guitar here.

A ping pong delay uses two separate delay processors that feed into each other. First the dry signal is heard, the signal is sent to the left side, this delayed signal is sent to the right side, and the right side is sent back to the left.

Chorus, flanging and phasing can all be created with delays as well. Listen to The Home Recording Show #11 or read about it here for more on that.

Tips On Using Delay

—On vocals, try using a short delay instead of reverb, sometimes it works better.

—Set up a ping pong delay after a large reverb, so the reverb seems to get steadily wider.

—Be careful with that feedback control, things can get very loud, very quickly.

Gates, Comps, De-Essers

A noise gate is a form of dynamics processing used to increase dynamic range by lowering the noise floor, and it is an excellent tool for removing hum from an amp, cleaning up drum tracks between beats, background noise in dialog, and can even be used to reduce the amount of reverb in a recording.

The common parameters for a noise gate are:

Threshold – Sets the level that the gate will open, when the signal level drops below the threshold the gate closes and mutes the output.

Attack – How fast the gate opens.

Hold – How long before the gate starts to close.

Release – a.ka decay—how long until the gate is fully closed again.

Range – How much the gated signal will be attenuated.

Sidechain – For setting an alternate signal for the gate to be triggered from, sometimes called a Key.

Filters – The filters section allows you to fine tune the sidechain signal.

What’s It For?

The normal use for gating is for removing background noise. An essential tool for clean dialog recording. Some other uses for gates are gated reverb and using the sidechain to activate other effects.

How To Set A Noise Gate

To set up a gate properly, start with the the attack, hold, and release as fast as possible. Set the range to maximum, and the threshold to 0 dB.

Start lowering the threshold until the sound starts to get chopped up by the gate. Slow down the attack time to remove any unnatural popping. Adjust the hold and release times to get a more natural decay.

If you don’t want the background noise to be turned down as much then you can reduce the range control.

Other Uses

Gated reverb was a popular effect in the 80s, mostly because of Phil Collins records.

To set it up, take your drum tracks and send them to a stereo reverb with a large room preset. After the reverb, insert a stereo gate. Adjust the gate settings so that the reverb is cut off before then next hit.

In this example you’ll hear the unprocessed drums, then with reverb, then adding the gate. (Listen)

Favorite Gates

The classic Drawmer DS201 is a hardware noise gate that is hard to beat.

The gate on the Waves SSL E-Channel is good, simple and effective.

The free ReaGate VST is quite good as well.

Noise gates aren’t very much fun to talk about, but they are a powerful tool that you need to know how to use.


Compression is an effect that can take a while to understand because the results are not always as obvious as other effects. To explain it as simply as possible, when a signal goes into a compressor, it gets turned down. That’s it. How it does this, how fast, and smoothly is what makes each one unique.

Compressor Controls

Most compressors will have the same set of controls:

The Threshold control sets what level will start the gain reduction.

The Ratio sets how much gain reduction, with a 4:1 ratio for every 4 dB of signal above the threshold 1 dB will be allowed through.

The Attack control sets how fast the compressor reacts to peaks.

The Release control sets how fast the compressor reacts as the signal lowers

Makeup gain is used to bring up the overall level of the compressor after the peaks have been reduced.

Sometimes there is an auto makeup gain control, which will increase the output level to match the gain reduction.

Some compressors have a knee control that starts compressing at a lower ratio as the threshold is approached, this is very helpful for a more natural compression.

Compressors will usually have a few meters, an input level, gain reduction and output level. If there are only two meters there is usually a switch to change the output level to show gain reduction. Gain reduction meters go in the opposite direction of the level meters.

Setting A Compressor

This is my method for setting a compressor:

I choose a ratio depending on how aggressive I want the compression to be. The type of sound I’m using it on determines this, softer sounds like voice get lower ratios, bass gets a medium ratio and drums get a higher ratio.

I turn the attack and release controls to the fastest setting, and make sure the meter is showing gain reduction.

Then I lower the threshold level until I’m getting about 1 decibel of gain reduction on the peaks.

From there I’ll fine tune the attack and release for whatever sound most natural, and use the makeup gain to match the output with the input level.

If I want more compression, I’ll lower the threshold more.

Here’s an example of some electric guitar with and without compression. I’m using more compression than I normally would on this so that the effect will be easier to hear. It should be pretty obvious that the compressor has evened out the dynamics of the performance. (Listen)

Compression can bring out more details in a performance, but it will also bring up background noise especially at higher ratios, that’s not usually what you want.

A slow attack will let some of the transient through, you can use this when you want to increase the punch of drums. You want to compress the sustain of the drum, and use the make up gain to make the drums larger than life.

In this example there is an ambient room mic for a drum kit. First you will hear it without compression, then with (actually with a ton of compression), and I’ll increase (slow down) the attack time with each loop. Notice the increased bigness of the drums, and how the transients get through and keep it punchy. (Listen)


A limiter is a compressor that’s output stays at or below a specific level regardless of the input level. It only turns down remember. The compression ratio starts at 10:1 and can go up to infinity. Limiters need very fast attack and release to be effective.

A brick-wall limiter (aka Maximizer), is a mastering tool used to increase the volume of a song as much as possible. These brick-wall limiters have an infinite ratio and will not let anything past the threshold. This type of limiter has two main controls, one for threshold and one for the maximum output level.

With these you basically set the maximum output level, something like -0.02 dB and then crank the threshold to crush everything and make it sound really loud and obnoxious (like Death Magnetic). The misuse of the brick-wall limiter is often associated with the loudness war and with compression in general.



Another common mastering tool is the multi-band compressor.

A multi-band compressor is actually four compressors in one. The frequency range is split up into four bands like an equalizer, Low, low mid, high mid and high frequency bands. This can give you a much smoother compression with a lot more control.


There is one more type of dynamics processor, the de-esser. A de-esser is designed to reduce the harsh esss sounds in a voice. The compression works on a single frequency or frequency range rather than the entire input signal. These are generally used for voice processing but you might find some other uses for it.

Recommended Plug-Ins

Simple compressor: Massey CT4

Advanced compressor: Avid Smack!

Master limiter: Massey L2007

Multi-band compressor: Wave Arts MultiDynamics 5

De-esser: Massey De-esser


I find it hard to think about the electric guitar without thinking about distortion. There was a time when electric guitars were always clean. Hard to imagine now.

Traditionally distortion was an unwanted feature in amplifier design. Distortion only occurred when the amp was damaged or overdriven. Possibly the first intentional use of distortion was in the 1951 recording of “Rocket 88″ by Ike Turner and the Kings of Rhythm.

Chuck Berry liked to use small tube amps that were easy to overdrive for his trademark sound and other guitarists would intentionally damage their speakers by poking holes in them, causing them to distort.

Leo Fender then started designing amps with some light compression and slight overdrive and Jim Marshall started to design the first amps with significant overdrive. That sound caught on quickly and by the time Jimi Hendrix was using Roger Mayer’s effects pedals, distortion would forever be associated with the electric guitar.

Not Just For Guitars

When you’re recording and mixing, you can use a bit of distortion to give any sound more edge, grit, energy and excitement. Drums, vocals, bass, samples – they can all benefit from a touch of distortion at times. Understanding the different ways distortion can be created and how they sound can help you get better sounds and make better recordings.

So What Is Distortion?

The word distortion means any change in the amplified waveform from the input signal. In the context of musical distortion this means clipping the peaks off the waveform. Because both valves and transistors behave linearly within a certain voltage region, distortion circuits are finely tuned so that the average signal peak just barely pushes the circuit into the clipping region, resulting in the softest clip and the least harsh distortion.

Because of this, as the guitar strings are plucked harder, the amount of distortion and the resulting volume both increase, and lighter plucking cleans-up the sound. Distortion adds harmonics and makes a sound more exciting.

Amp Distortion—Tube & Solid State

Valve Overdrive. Before transistors, the traditional way to create distortion was with vacuum valves (also known as vacuum tubes). A vacuum tube has a maximum input voltage determined by its bias and a minimum input voltage determined by its supply voltage.

When any part of the input waveform approaches these limits, the valve’s amplification becomes less linear, meaning that smaller voltages get amplified more than the large ones. This causes the peaks of the output waveform to be compressed, resulting in a waveform that looks “squashed.”

It is known as “soft clipping”, and generates even-order harmonics that add to the warmth and richness of the guitar’s tone. If the valve is driven harder, the compression becomes more extreme and the peaks of the waveforms are clipped, which adds additional odd-order harmonics, creating a “dirty” or “gritty” tone.

Valve distortion is commonly referred to as overdrive, as it is achieved by driving the valves in an amplifier at a higher level than can be handled cleanly. Multiple stages of valve gain/clipping can be “cascaded” to produce a thicker and more complex distortion sound.

In some modern valve effects, the “dirty” or “gritty” tone is actually achieved not by high voltage, but by running the circuit at voltages that are too low for the circuit components, resulting in greater non-linearity and distortion. These designs are referred to as “starved plate” configurations.

Transistor Clipping. On the other hand, transistor clipping stages behave far more linearly within their operating regions, and faithfully amplify the instrument’s signal until the input voltage falls outside its operating region, at which point the signal is clipped without compression, this “hard clipping” or limiting. This type of distortion tends to produce more odd-order harmonics.

Electronically, it is usually achieved by either amplifying the signal to a point where it must be clipped to the supply rails, or by clipping the signal across diodes. Many solid state distortion devices attempt to emulate the sound of overdriven vacuum valves.

Distortion Pedals

Overdrive distortion. While the general purpose is to emulate classic “warm-tube” sounds, distortion pedals can be distinguished from overdrive pedals in that the intent is to provide players with instant access to the sound of a high-gain Marshall amplifier such as the JCM800 pushed past the point of tonal breakup and into the range of tonal distortion known to electric guitarists as “saturated gain.”

Some guitarists will use these pedals along with an already distorted amp or along with a milder overdrive effect to produce radically high-gain sounds. Although most distortion devices use solid-state circuitry, some “tube distortion” pedals are designed with preamplifier vacuum tubes. In some cases, tube distortion pedals use power tubes or a preamp tube used as a power tube driving a built-in “dummy load.”

The Boss DS-1 Distortion is a pedal with this design. This is what that sounds like: Listen

Overdrive/Crunch. Some distortion effects provide an “overdrive” effect. Either by using a vacuum tube, or by using simulated tube modeling techniques, the top of the wave form is compressed, giving a smoother distorted signal than regular distortion effects. When an overdrive effect is used at a high setting, the sound’s waveform can become clipped, which imparts a gritty or “dirty” tone, which sounds like a tube amplifier “driven” to its limit.

Used in conjunction with an amplifier, especially a tube amplifier, driven to the point of mild tonal breakup short of what would be generally considered distortion or overdrive, or along with another, stronger overdrive or distortion pedal, these can produce extremely thick distortion.

Today there is a huge variety of overdrive pedals, including the Boss OD-3 Overdrive: Listen

Fuzz. This was originally intended to recreate the classic 1960s tone of an overdriven tube amp combined with torn speaker cones. Old-school guitar players would use a screwdriver to poke several holes through the the guitar amp speaker to achieve a similar sound.

Since the original designs, more extreme fuzz pedals have been designed and produced, incorporating octave-up effects, oscillation, gating, and greater amounts of distortion.

The Electro-Harmonix Big Muff is a classic fuzz pedal: Listen

Hi-Gain. High gain in normal electric guitar playing simply references a thick sound produced by heavily overdriven amplifier tubes, a distortion pedal, or some combination of both – the essential component is the typically loud, thick, harmonically rich, and sustaining quality of the tone.

However, the hi-gain sound of modern pedals is somewhat distinct from, although descended from, this sound. The distortion often produces sounds not possible any other way. Many extreme distortions are either hi-gain or the descendants of such.

An example of a hi-gain pedal is the Line 6 Uber Metal: Listen

Power-Tube. A unique kind of saturation when tube amps output stages are overdriven, unfortunately, this kind of really powerful distortion only happens at high volumes.

A Power-Tube pedal contains a power tube and optional dummy load, or a preamp tube used as a power tube. This allows the device to produce power-tube distortion independently of volume.

An example of a tube-based distortion pedal is the Ibanez Tube King: Listen

Other Ways To Distort

Tape Saturation. One way is with magnetic tape. Magnetic tape has a natural compression and saturation when you send it a really hot signal. Even today, many artists of all genres prefer analog tape’s “musical,” “natural” and especially “warm” sound. Due to harmonic distortion, bass can thicken up, creating the illusion of a fuller-sounding mix.

In addition, high end can be slightly compressed, which is more natural to the human ear. It is common for artists to record to digital and re-record the tracks to analog reels for this effect of “natural” sound. While recording to analog tape is likely out of the home studio budget, there are tape saturation plugins that you can use while mixing that simulate the effect quite well.

Here’s a bass guitar with a bit of tape saturation from the Ferox VST plug-in: Listen

Digital Wave Shaping. The word clipping in recording is usually a bad thing. And generally it is, unless we’re trying to distort something on purpose. In the digital world we can use powerful wave shaping tools to drastically distort and manipulate a sound.

Rather than subject you to the technical explanation of how it works, just listen to Nine Inch Nails, they use this a lot. It’s perfect for really harsh, aggressive, unnatural and broken sounds.

Here’s some examples of Ohmforce Ohmicide on a drum loop: Listen

Why Is This Important?

Knowing those sounds can help you be a better musician, engineer and producer. It will help you make decisions on what gear to purchase and what is appropriate for a song.

What Else?

Besides guitar, what else is distortion good for? Well, pretty much anything, as long as it’s appropriate for the song.

—Slight distortion can make something sound more exciting, too much can sometimes make it really tiny sounding.

—When recording electric guitars, you can get a way bigger sound by using less gain and recording the same part multiple times, double or quad-tracking.

—Distortion can sound really cool on drums, but you may have to heavily gate the drums, the sustain can get out of control.

*Note: All audio samples except the last two were copied from various internet sources, mostly manufacturer websites.

Jon Tidey is a Producer/Engineer who runs his own studio, EPIC Sounds, and enjoys writing about audio on his blog To comment or ask questions about this article go here.


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In The Studio: Audio Effects Explained (Includes Audio Samples)