audio effects

Audio sound effects are split into two main categories. These are time-based, such as reverb and delay, and modulation-based, such as chorus and flanging. Modulation-based effects use LFO's to modulate the time difference between input and output signals in various ways, thereby creating their characteristic sweeping movement. The different effects use a wide variety of processes to transform an input signal and then mix it with the original signal. The sections below contain individual explanations and audio examples of these various types of effects.

[Click here to find out about effects techniques.]

The terms wet and dry are often used when discussing effects. Wet refers to a signal that has been processed to some degree by an effect. The term dry refers to an unprocessed signal, or more specifically, the part of an input signal that passes through an effect unit unaltered.


time-based effects

The main two time-based effects are reverb and delay...

Time-based effects all manipulate input signals in the time domain and use units of time as main parameter values, ie. milliseconds. The main two are reverb and delay (sometimes known as echo). There are a great number of variations of these effects available, including reverse reverbs and multi-tap delays, with filters built in to the feedback loops.

time-based effects  •  reverb

Probably the best known effect is reverb. It is a simulation of the actual acoustic phenomenon that occurs where ever there is sound. It basically describes the countless tiny echoes or reflections that bounce off all surrounding surfaces when there is a sound source present and build up as one sound to form a 'tail' to the sound source. The first group of reflections to be heard are called early reflections (ER) and can sometimes be heard separately from the reverb tail. Effects processors generally don't have enough processing power to accurately reproduce the billions of reflections needed for a convincing reverb effect, but often a simple reverb used sparingly will still enhance a signal.

Decay time determines the amount of time that the reverb level will take to fall from maximum to -60 dB...

Surroundings of different materials will affect acoustic reverb and so reverb effects often provide a colour, or damping, control to simulate this. Decay time determines the amount of time that the reverb level will take to fall from maximum level to a measurement of -60 dB, not to silence. A pre-delay time determines the length of time between the input signal and the start of the reverb. Some effects units also give control over early reflection levels and times for a more convincing effect. Click the listen button to hear an example of reverb, featuring a dry sound first, followed by the wet sound.

time-based effects  •  delay feeds a portion of the output signal back into the input and determines the number of repeats that a delay will produce...

(Digital) delay is an effect where an input signal is temporarily stored in memory and then played back after a set time. Controls are usually basic and include delay time and feedback, which feeds a portion of the output signal back into the input and basically determines the number of repeats, or taps, that a delay will produce, ie. a delay with no feedback will only produce one delay tap.

Occasionally, the number of delay circuits an effect has is referred to in terms of taps, so a four tap delay effect would in fact produce four separate taps with no feedback. Multi-tap effects such as these often allow separate control over each tap. Sometimes equalisers or filters are provided to alter the sound of the delay taps. The provided audio example is of a simple dry sound, followed by the same sound with an added stereo delay.


modulation-based effects

Modulation-based effects use LFO's to modulate various parameters...

Modulation-based effects work by the modulating one or more parameters of a sound, such as the amplitude, time delay, or pitch, depending on the type of effect. They generally use LFO's to modulate the various parameters, although the use of on-board envelopes is also common. There are several different types that are explained below and examples of each can be heard by clicking on the various listen buttons. While they all use very similar processes, they can sound distinctly different.

modulation-based effects  •  flangers

A flanger is one of the simplest and most effective of the modulation-based effects. Its effect is simply created by slightly modulating the (very short) delay time of a delay effect and so is actually also a time-based effect. This produces a sweeping comb filter effect with evenly spaced peaks and notches running along the frequency spectrum. The delay time can also be manually altered to change the sound further... shorter delays affect higher frequencies, while longer delay times affect lower frequencies and on some effects, can make the individual delay repeats audible.

...positive and negative feedback give considerably different results...

To enhance the effect, flangers have feedback controls, which feed a portion of the output signal back into the input to enhance, or boost the effect. Occasionally, these controls offer both positive and negative feedback, which give considerably different results... positive feedback can give a somewhat mettalic 'sheen' to a sound, while negative feedback can produce a hollowing of the sound. Click the listen button to hear a dry sequence, followed by the same (slightly longer) sequence, first with a positive feedback flange, then with a negative feedback flange and finally, one with a faster modulation rate.

modulation-based effects  •  chorus

The chorus effect is very similar to a flanger, but it has no feedback circuitry...

The chorus effect is very similar to a flanger, but it has no feedback circuitry and uses the modulation of pitch as well as time. Its purpose is to thicken a sound - it was created to emulate the appearance of several instruments (or vocalists) playing (or singing) in unison, from a signal from just one. Often the only controls supplied are for the LFO speed, the delay time and the chorus amount heard, but that is all that is really needed. The audio example features a basic string sound from a pair of detuned saw waveforms, first without and then with a stereo chorus effect.

modulation-based effects  •  phasers

The phasing effect is created by splitting an input signal into two, modulating the phase of one and then adding it back to the other.

The phasing effect is created by splitting an input signal into two, modulating the phase of one, using miniscule amounts of delay, and then adding it back to the other. This produces a series of moving peaks and notches throughout the frequency spectrum, depending on the number of delay circuits used. Flanging is a basic kind of phasing and so phasers also have feedback controls to enhance the effect.

Some phasers also provide control over the number of peaks and notches, or stages and how spread out they are across the frequency spectrum. To hear examples of some the different kinds of effects achievable with a phaser, click the listen button. First is a dry short sequence, followed by the same (slightly longer) sequence with a six stage phaser and then a two stage phaser.

modulation-based effects  •  auto pan

Auto pan creates the effect of movement across the stereo field...

Auto pan is a simple modulation effect, where the on-board LFO adjusts the amplitudes of the relative left and right inputs signals 180° out of phase. This creates the effect of movement across the stereo field, which can bring an added interest to sounds. Controls are usually just comprised of the LFO speed, shape and the effect amount, ie. the stereo width of the effect. As a general rule of thumb, to keep things sounding natural, the faster the LFO rate is, the narrower the stereo width should be, ie. slow, wide sweeps or fast, gentle shimmering, etc. Auto pan generally works best on background sounds, as main sounds should really be panned fairly centrally. This audio example is from one of the tracks on the Nord Modular G2 page, which uses auto pan on several of the sounds.



Distortion is the deformation of a sound's waveform, usually unwanted, but can be beneficial in synthesis and modern music production.

Distortion can be thought of as a dynamics and frequency based effect, although technically speaking, it is not really an effect, but instead a transformation process. Again, there are many different kinds, including overdrive, saturation and rectification and they all work in the same way by deforming the input signal, each according to their own transfer characteristics. These processes actually alter the shape of the input signal's waveform and so are also known as wave shaping. In most fields, distortion is usually unwanted, but for synthesis and modern music production, it can be beneficial and is commonly used.

The end result of adding a little distortion, whatever the type, is usually approximately the same... a sound that has become a little brighter, maybe fatter too and one that stands out a bit further in a mix. Another effect distortion can have, is to add some sustain to a sound, somewhat like the effect of a compressor. You can think of the different types as making subtly different dynamic and frequency adjustments to the input signal. Distortion appears far more evident when used on bass sounds than it does on higher frequency signals and at high levels it can sound horrible, but used well, can improve a sound dramatically.

The different distortion processes are described underneath and audio examples can be heard by clicking the listen button below. The first two feature a short note sequence with the amount of distortion increasing over time. The first uses saturation, while the second uses a 'fat' overdrive and the last examples are of a single undistorted sound, followed by the same sound with a variety of distortion types added. There is also a link to an excellent freeware VST wave shaping distortion plug-in on the about nitetime studio page... click here to go there.

distortion  •  overdrive

Overdrive adds harmonics to the input signal, increasing its high frequency content.

One of the most common types of distortion is overdrive. It was originally created by rock musicians by simply turning the gain on guitar amplifiers past the optimum position (overdriving the gain), causing a clipping of the audio signal. Nowadays, overdrive can be found on effect pedals, rack units and in computer plug-in form. Controls are usually simple, often only gain, but sometimes others are included to allow some tonal and/or frequency adjustments to be made. Generally, overdrive adds harmonics to the input signal, increasing its high frequency content. It can also produce a fatter sound, due to a compressor-like effect.

distortion  •  saturation

Saturation is a type of harmonic distortion that has the effect of introducing a 'warmth' to a signal...

Saturation is a type of harmonic distortion that was originally gained by recording onto old analogue multi track tape machines at high levels, due to inaccuracies of their components. This has the effect of introducing a 'warmth' to a signal and can be very useful to eradicate the bright and harsh artifacts of some digital sounds. It does so by adding harmonics, fattening the lower end of the frequency spectrum and slightly reducing the level of high end material, which often sounds more natural to the ear. Its pleasing effect has been noticed for a long time and as such, there are a wide array of saturation effects available.

distortion  •  clipping

Clipping graphs
clipping graphs - time plotted against amplitude

Power is finite, so there is a maximum level that a signal can have and once that level has been passed, clipping will occur. When a signal is pushed over this limit, the top and the bottom of the input signal waveform is simply 'clipped' off. This means that the portion of the signal that has been raised past the maximum level is replaced by a flat line along this level. Clipping will therefore instill some properties of a square wave onto the input signal, producing extra harmonics and generally, a brighter sound. The graphs above show this and represent one cycle of a sine wave before and after clipping at a moderate and then an extreme level.

Low clipping levels generally results in a brighter and more powerful signal...

At low clipping levels, this generally results in a brighter and more powerful signal, but can sound very harsh at higher levels. As such, clipping should usually be used sparingly. This effect can occur naturally by overdriving the gain on analogue equipment only and should never be tried with most digital equipment as digital clipping works differently and can sound very unpleasant. However, there are many hardware units and computer plug-ins that emulate this effect, so it is possible to clip audio in the digital domain.

distortion  •  rectification

Rectification graphs
rectification graphs - time plotted against amplitude

A rectifier converts alternating currents (AC), such as audio, into direct currents (DC), so that the signal becomes totally positive. Half wave rectification simply removes the negative part of the input signal, while full wave rectification 'reflects' the negative part onto the positive side. The graphs to the right represent one cycle of a sine wave before and after both half and full wave rectification.

...the result can easily sound dirty, so caution with rectification is advised.

Some rectifiers also offer signal inverters to allow the output to become completely positive or negative... there is only a very subtle difference in sound between the two. Half wave rectification produces a large boost in the presence range (the upper mid to high frequency range), while full wave also boosts low frequencies, sometimes over dramatically. Using either, the result can just as easily sound dirty as it can be a benefit, so caution with rectification is advised.

distortion  •  bit depth

Digital audio uses binary to store audio data. To cut a long story short, the more bits that are used to describe each tiny fragment of audio, the truer the dynamic representation of the original audio. For example, CD's use a fairly clear sounding 16 bits, while children's toys and novelty greetings cards with audio that plays when you open them usually use only a few, often resulting in robot-like and almost unintelligible audio. Click here to find out more about bits and bit rates.

Bit rate distortion is caused by there not being enough bits to accurately represent the input signal.

Bit rate/resolution/depth distortion is simply caused by there not being enough bits to accurately represent the input signal. As described in the audio greetings card example above, this should generally be avoided, but it can be a benefit in some instances. When samplers went from 12 bit to 16 bit capability, many producers found that certain sounds, such as drums, actually sounded better on their old machines. This was because a side effect of using lower bit depths is a kind of compression that works on both the dynamic and frequency range.

distortion  •  sample rate

Sampling rates define the undistorted upper frequency range of a signal.

In short, sampling rates define the undistorted upper frequency range of a signal... click here to find out more. Lower sampling rates lower the highest possible frequency that the system can reproduce and also the frequency of the aliasing artifacts that are produced in all digital systems, into the range of human hearing. On today's digital systems, this aliasing is way above our upper hearing limit, but on lower rate systems and sampling rate distortion units, manifests itself as unpleasant, high frequency noise, giving the input signal a metallic sounding quality.

When using a sampling frequency of around 10 kHz, this can have the effect of simply brightening an input signal , but using very low rates below 300 Hz or so, can almost completely destroy the signal. A nice effect can be produced by slowly fading from the lowest sampling frequency to full frequency resolution, where your sounds will slowly appear out of something that was totally unrecognisable as music at first.



...the order in which the effects are linked can make a big difference.

While all of these individual effects noted above can definitely improve sounds on their own, the real reward comes from using multi-effects - basically, the linking of one or more effects to form a more powerful, composite effect. While this statement may seem obvious, it should be noted that there is more to this idea than appears... this is because the order in which the effects are linked can make a big difference. All multi-effects units, whether hardware or software, will allow versatile routing capabilities for this reason.

A good example of this is the difference between using a delay before, or after a modulation effect. If the delay effect is routed first, then the sweep of the modulation effect will sound continuously, as it is processing both the original material and the delay's output, ie. each delay tap will have a different portion of the sweep on it. However, when the modulation effect is first in line, it only affects the original signal and not the output of the delay, so each delay tap will be the same as the last (but quieter), with only a portion of the sweep being repeated in each tap.

Sticking with this example, there is a further difference when the effects are set to 100% wet and used on an auxiliary mixer channel... click here for information about auxiliary effects. If the delay effect is routed first in this case, then the modulation effect will only appear in the delay taps and not on the original signal. Therefore, the modulation effect needs to be routed first in line, if it is to be heard on the original sound and not just the delay taps.


If you found that to be interesting or helpful, you might like to discover more about music production... to find out about synthesis click here, for modular synthesis click here, for information about music production equipment click here, or for general music production tips, click here.



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