Distortion Explained

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.

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

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”. This 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. This 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

Transistor clipping stages, on the other hand, 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, this 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.

So that is basically how Tube and Solid state amps create distortion. Let’s talk about some different types of distortion sounds as found in effects pedals.

Distortion Pedals

Overdrive Distortion

Overdrive distortion is a well-known 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 to the Boss DS-1 Distortion Pedal]

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 to the Boss OD-3 Overdrive Pedal]

Fuzz

Fuzz was originally intended to recreate the classic 1960’s tone of an overdriven tube amp combined with torn speaker cones. Oldschool 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 to the Electro-Harmonix Big Muff Fuzz Pedal]

Hi-Gain

Hi-Gain is the sound most used in Heavy metal. 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 to the Line-6 Uber Metal Distortion Pedal]

Power-tube pedal

Power tube distortion is 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 to the Ibanez Tube King Distortion Pedal]

Moving away from guitars there are a couple more ways to distort something.

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 plugin.

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.

Why Is This Important?

Alright, so why did I just explain those different types of distortion sounds? Because 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 the samples except the last two were copied from various internet sources, mostly manufacturer websites.

Compression

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. Thats 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 4dB of signal above the threshold 1dB 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.

Example:

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 to this example]

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.

Example:

In this example there is an ambient room mic for a drumkit. 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 to this example]

Limiter?

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 brickwall limiter aka Maximizer, is a mastering tool used to increase the volume of a song as much as possible. These brickwall 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 -0.02dB and then crank the threshold to crush everything and make it sound really loud and obnoxious (like Death Magnetic). The misuse of the brickwall limiter is often associated with the loudness war and with compression in general.

Multi-band?

Another common mastering tool is the Multi-band compressor.

A Multi-band Compressor is actually 4 compressors in one. The frequency range is split up into 4 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.

De-Esser

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 Plugins

Simple compressor: Massey CT4

Advanced compressor: Digidesign Smack!

Master Limiter: Massey L2007

Multi-band Compressor: Wave Arts MultiDynamics 5

De-Esser: Massey De-esser

A noise gate is a form of dynamics processing used to increase dynamic range by lowering the noise floor. A gate 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.

Parameters

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 – aka Decay, how long until the gate is fully closed again
Range – how much the gated signal will be attenuated
Sidechain – the sidechain is 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 setup a gate properly, start with the the attack, hold, and release as fast as possible. Set the range to maximum, and the threshold to 0dB.

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.

Listen to me explain this

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.

Delay Processors

In it’s 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 70s and used solid state electronics as an alternative to the tape delay.

Digital Delay

In the late 70s, 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 plugins 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 thats it for the history of delay processors, now lets move onto 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 [double-track.mp3]

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

Here it is on some guitar [slapback.mp3]

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.

[ping-pong.mp3]

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

Some tips on using delays

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.

The last tip I have is be careful with that feedback control, things can get very loud, very quickly.

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.

Parameters

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

High Pass Filter clip

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

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

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

High Shelf clip

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

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 plugin equalizers will be parametric.

EQ Usage Tips

Favorite EQ Plugins

Some of my favorite Equalizer plugins are:

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. This 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 plugins.

Analog reverbs come in 3 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′x8′ 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 [plateverb.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. [roomverb.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 and a half 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’s 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 reverb was a popular effect in the 80s, but it’s sounding pretty cheesy these days. [gated.mp3]

Reversed reverb 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 are the reflections 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% wet.

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

Hi Freq. Roll Off – Helps control the decay of high Frequencies (as it’s 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.

1. Tremolo
2. Vibrato
3. Flanging
4. Phase Shifting or Phasing
5. Chorus

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

All of these effects are built around a Low Frequency Oscillator more commonly referred to as just an LFO. An LFO is an audio signal usually less than 20Hz 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 the effects listed use Sine wave as the waveshape 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.

Listen to an example of Flanging

Phase shifting or phasing is a similar effect to flanging, but is accomplished in a much different way. Phasers split the signal, one part goes through an allpass filter then into an lfo then recombined with the original sound. An allpass 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 2 main parameters, Sweep Depth: how far the notches sweep up and down the frequency range. Speed or Rate: 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 20-30ms compared to flanging which is 1-10ms. 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 Digidesign ModDelay2.

Now that you understand what these effects are doing, put that knowledge to use!