Audio Terminology Compiled from the Audio Dictionary Third

Edition by Glenn D. White and Gary J. Louie and the Sound on

Sound Glossary of technical terms.

Active Loudspeaker or Monitor-A loudspeaker system in which the

input signal is passed to a line-level crossover, the suitably filtered
outputs of which feed two (or more) power amplifiers, each connected
directly to its own drive unit. The line-level crossover and amplifiers
are usually (but not always) built in to the loudspeaker cabinet.

Algorithm-A sequence of instructions describing how to perform a

specific task. Algorithms are often implemented in a computer
language and compiled into a computer program. In the context of
effects units, algorithms usually describe a software building block
designed to create a specific effect or combination of effects.

Ambience-The acoustical qualities of a space

Amplitude-The strength of a signal

Analog- An Audio signal that is an electrical replica for the waveform of

the sound it represents.

Analog to Digital Converter(ADC)-In digital Audio systems the audio

signal(Analog) must be converted to digital form before it can be further
processed.

Anechoic-Literally without echo

Antinode-A place of minimum sound pressure level in a standing way

Attack Time-the time it takes for a compressor or limiter to reduce its

gain after a strong signal has been applied to it.

Attenuation-Reduction in amplitude or level of a signal usually

expressed in decibels(dB)

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Audio — Literally, “I hear” in Latin. The term refers to any signal that
can be heard.

Aural Exciter — a device that typically adds even order harmonic

distortion to a signal to make the signal a little brighter or crisper.
Perhaps most often used to make a vocal standout

Auxiliary Sends (Auxes) — a separate output signal derived from

an input channel on a mixing console, usually with the option to select
a pre- or post-fader source and to adjust the level. Corresponding
auxiliary sends from all channels are bussed together before being
made available to feed an internal signal processor or external
physical output. Sometimes also called effects or cue sends.

Aux Return — Dedicated mixer inputs used to add effects to the mix.
Aux return channels usually have fewer facilities than normal mixer
inputs, such as no EQ and access to fewer aux sends. (cf. Effects
Return)

A-weighting — An equalization curve applied to level meters in an

attempt to make their measurements correspond better to perceived
loudness. It decreases the sensitivity of the meter to frequencies below
1000 hertz.

Balance — this word has several meanings in recording. It may refer

to the relative levels of the left and right channels of a stereo
recording (eg. Balance Control), or it may be used to describe the
relative levels of the various instruments and voices within a mix (ie.
Mix balance).

Balanced — Refers to Audio lines in which the signal is not carried by

the shield for the principla purpose of noise suppression.

Bandpass filter — A filter which has a bandwidth. The filter may be

broad or narrow. They may be fixed or variable in frequency or
bandwidth.

Bandwidth — the bandwidth of a bandpass filter is the upper cutoff

frequency minus the lower cutoff frequency.

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Bass — is the portion of the Frequency that encompasses lower pitches.
Often considered to be from 20 Hz to 200 Hz

Bass Intermodulation(BIM) — refers to distortion caused by subsonic

noise or audio content.

Bit rate — The time rate at which bits are transmitted in a digital audio
system measured in bits per second.

Bits — Binary digits using the numbers 0 and 1 to encode digital audio

Bounce — to combine multiple tracks of Audio in a mix down to a

mono, stereo or multichannel(surround) format.

Broadband — means encompassing or consisting of a wide range of

frequencies.

Buffer — A temporary location in a computer or computer based digital

audio workstation(DAW)

Bus — point where many connections or signals may be combined

Butt Splice — an edit in which there is no fading or crossfading

between the two segments

BWF (Broadcast Wave Format) — A digital audio file format

introduced by the EBU (European Broadcast Union) in 1996 to facilitate
professional editing and file exchange represented in a DAW by the
suffix .wav

Cents — 1/100th of a semi tone

Channel — an independently recorded signal

C-Weighting — A form of electrical filter which is designed to mimic

the relative sensitivity of the human ear to different frequencies at
high sound pressure levels (notionally 100 Phons or about 87dBA
SPL). Essentially, the filter rolls-off the low frequencies below about
20Hz and the highs above about 10kHz. This filtering is often used

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when making measurements of high-level sounds, such as when
calibrating loudspeaker reference levels.

Clipping — When an audio signal is allowed to overload the system

conveying it, clipping is said to have occurred and severe distortion
results. The ‘clipping point’ is reached when the audio system can no
longer accommodate the signal amplitude–either because an
analogue signal voltage nears or exceeds the circuitry’s power supply
voltage, or because a digital sample amplitude exceeds the quantile’s
number range. In both cases, the result is that the signal peaks are
‘clipped’ because the system can’t support the peak excursions—a
sinewave source signal becomes more like a squarewave. In an
analogue system clipping produces strong harmonic distortion
artefacts at frequencies above the fundamental. In a digital system
those high frequency harmonics cause aliasing which results in
enharmonic distortion where the distortion artefacts reproduce at
frequencies below the source fundamental. This is why digital clipping
sounds so unlike analogue clipping, and is far more unpleasant and
less musical.

Clocking — The process of controlling the sample rate of one digital

device with an external clock signal derived from another device. In a
conventional digital system, there must be only one master clock
device, with everything else ‘clocked’ or ‘slaved’ from that master.

Comb-Filter — a series of deep filter notches created when a signal

is combined with a delayed version of itself. The delay time (typically
less than 10ms) determines the lowest frequency at which the filter
notches start.

Comping — Short for ‘compilation.’ The process of recording the

same performance (e.g. a lead vocal) several times on multiple tracks
to allow the subsequent selection of the best sections and
assembling them to create a ‘compilation’ performance which would
be constructed on a final track.

Compressor — A device (analogue or digital) which is designed to

reduce the overall dynamic range of an audio signal either by
attenuating the signal if it exceeds a set threshold level according, or

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by increasing the level of quiet signals below a threshold. The amount
of attenuation is defined by a set ratio, while the speed of response
(attack) and recovery (release) can usually also be controlled.

Console — An alternative term for mixer (See also Desk).

Converter — A device which transcodes audio signals between the

analogue and digital domains. An analogue-to-digital (A-D) converter
accepts an analogue signal and converts it to a digital format, while a
digital-to-analogue (D-A) converter does the reverse. The sample rate
and wordlength of the digital format is often adjustable, as is the
relative amplitude of analogue signal for a given digital level.

DAW — (Digital Audio Workstation): A term first used in the 1980s to

describe early ‘tapeless’ recording/sampling machines like the
Fairlight and Synclavier. Nowadays, DAW is more commonly used to
describe Audio+MIDI ‘virtual studio’ software programs such as
Cubase, Logic Pro, Digital Performer, Sonar and such-like.
Essentially elaborate software running on a bespoke or generic
computer platform which is designed to replicate the processes
involved in recording, replaying, mixing and processing real or virtual
audio signals. Many modern DAWs incorporate MIDI sequencing
facilities as well as audio manipulation, a range of effects and sound
generation.

dB-The decibel is a method of expressing the ratio between two

quantities in a logarithmic fashion. Used when describing audio signal
amplitudes because the logarithmic nature matches the logarithmic
character of the human sense of hearing. The dB is used when
comparing one signal level against another (such as the input and
output levels of an amplifier or filter). When the two signal amplitudes
are the same, the decibel value is 0dB. If one signal has twice the
amplitude of the other the decibel value is +6dB, and if half the size it
is -6dB.

dB/Octave — A means of measuring the slope or steepness of a

filter. The gentlest audio filter is typically 6dB/Octave (also called a
first-order slope). Higher values indicate sharper filter slopes.

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24dB/octave (fourth order) is the steepest normally found in analogue
audio applications.

De-esser — A device for reducing the effect of sibilance in vocal

signals.

Decay — The progressive reduction in amplitude of a sound or

electrical signal over time, eg. The reverb decay of a room. In the
context of an ADSR envelope shaper, the Decay phase starts as
soon as the Attack phase has reached its maximum level.

Dither — A system whereby low-level noise equivalent to one

quantizing level is combined with a digitized audio signal in such a
way as to perfectly linearize the digital system. Dither must be
employed whenever the wordlength is reduced, otherwise quantizing
distortion errors will manifest.

DSP — Digital Signal Processor. A powerful microchip used to

process digital signals.

Dry (cf. Wet) — A signal that has had no effects added.

Ducking — A system for controlling the level of one audio signal with
another. For example, in a broadcast radio context a music track can
be made to 'duck' or reduce in volume whenever there's a voice over.

Dynamic Range — The amplitude range, usually expressed in

decibels, between the loudest signal that can be handled by a piece
of equipment and the level at which small signals disappear into the
noise floor.

Dynamics — A way of describing the relative levels within a piece of

music.

Effect — A treatment applied to an audio signal in order to change or

enhance it in some creative way. Effects often involve the use of
delays, and include such treatments as reverb and echo.

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Effects Loop — An interface system, usually involving separate send
and receive connections, which allows an external signal processor to
be connected into the audio chain. (cf. Insert Point)

Effects Return — An additional dedicated mixer input channel,

usually with minimal facilities, designed to accommodate the output
from an effects unit. (cf. Aux Return)

Equalizer (cf. Filter) — A device which allows the user to adjust the
tonality of a sound source by boosting or attenuating a specific range
of frequencies. Equalizers are available in the form of shelf
equalizers, parametric equalizers and graphic equalizers — or as a
combination of these basic forms.

Exciter (cf. Enhancer) — An audio processor that works by

synthesizing new high frequency harmonics.

Expander — A device designed to increase the dynamic range,

typically by reducing the volume of low level signals (below a set
threshold), or to increase the volume of high level signals (above a
threshold). (See Compressor.)

Fader — A sliding potentiometer control used in mixers and other

processors.

FET — Field Effect Transistor. A solid-state semiconductor device in

which the current flowing between source and drain terminals is
controlled by the voltage on the gate terminal. The FET is a very high
impedance device, which makes it highly suited for use in impedance
converter stages in capacitor and electret microphones.

FET-Compressor — A form of audio compressor in which an FET is

used to provide variable signal attenuation. FET compressors are
fast-acting in comparison to opto-compressors.

Filter (cf. Equalizer) — An electronic circuit designed to attenuate a

specific range of frequencies. (See low-pass, high-pass and band-
pass.)

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Filter Frequency — The ‘turnover’ or ‘corner’ frequency of a high- or
low-pass filter. Technically, the frequency at which the signal
amplitude has been attenuated by 3dB.

Flanging — An effect which combines a modulated delay with the

original signal, using feedback to create a dramatic, sweeping sound.

Formant — The frequency components or resonances of an

instrument or voice sound that doesn't change with the pitch of the
note being played or sung. For example, the body resonance of an
acoustic guitar remains constant, regardless of the note being played.

Frequency — The number of complete cycles of a repetitive

waveform that occur in 1 second. A waveform which repeats once per
second has a frequency of 1Hz (Hertz).

Frequency Response — The variation in amplitude relative to the

signal frequency. A measurement of the frequency range that can be
handled by a specific piece of electrical equipment or loudspeaker.
(Also see Bandwidth)

Fundamental — The lowest frequency component in a harmonically

complex sound. (Also see harmonic and partial.)

FX — Shorthand term for Effects.

Gain — The amount by which a circuit amplifies a signal, normally

denoted in decibels.

Gain Staging — The act of optimizing the signal level through each
audio device in a signal chain, or through each section of a mixing
console, to maintain an appropriate amount of headroom and keep
the signal well above the system noise floor.

Gate — An electronic device (analogue or digital) designed to mute

low level signals so as to improve noise performance during pauses
in the wanted material. (Also see Expander.)

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Graphic Equalizer — A form of equalizer whereby multiple narrow
segments of the audio spectrum are controlled by individual cut/boost
faders. The name comes about because the fader positions provide a
graphic representation of the EQ curve.

Harmonic — High frequency components of a complex waveform,

where the harmonic frequency is an integer multiple of the
fundamental.

Harmonic Distortion — The addition of harmonics that were not

present in the original signal caused by non-linearities in an electronic
circuit or audio transducer.

Headroom — The available ‘safety margin’ in audio equipment

required to accommodate unexpected loud audio transient signals. It
is defined as the region between the nominal operating level (0VU)
and the clipping point. Typically, a high quality analogue audio mixer
or processor will have a nominal operating level of +4dBu and a
clipping point of +24dBu — providing 20dB of headroom. Analogue
meters, by convention, don’t show the headroom margin at all; but in
contrast, digital systems normally do — hence the need to try to
restrict signal levels to average around -20dBFS when tracking and
mixing with digital systems to maintain a sensible headroom margin.
Fully post-produced signals no longer require headroom as the peak
signal level is known and controlled. For this reason, it has become
normal to create CDs with zero headroom.

Hertz (Hz) — The standard measurement of frequency. 10Hz means

ten complete cycles of a repeating waveform per second.

High-Pass Filter (HPF) — A filter which passes frequencies above

its cut-off frequency, but attenuates lower frequencies.

High-range (highs) — The upper portion of the audible frequency

spectrum, typically denoting frequencies above about 1kHz.

IC — An abbreviation of Integrated Circuit, a collection of miniaturized

transistors and other components on a single silicon wafer, designed
to perform a specific function.

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Impedance — The ‘resistance’ or opposition of a medium to a
change of state, often encountered in the context of electrical
connections (and the way signals of different frequencies are
treated), or acoustic treatment (denoting the resistance it presents to
air flow). Although measured in Ohms, the impedance of a ‘reactive’
device such as a loudspeaker drive unit will usually vary with signal
frequency and will be higher than the resistance when measured with
a static DC voltage. Signal sources have an output impedance and
destinations have an input impedance. In analogue audio systems,
the usually arrangement is to source from a very low impedance and
feed a destination of a much higher (typically 10 times) impedance.
This is called a ‘voltage matching’ interface. In digital and video
systems, it is more normal to find ‘matched impedance’ interfacing
where the source, destination and cable all have the same
impedance (eg. 75 Ohms in the case of S/PDIF).
Microphones have a very low impedance (150 Ohms or so) while
microphone preamps provide an input impedance of 1,500 Ohms or
more. Line inputs typically have an impedance of 10,000 Ohms and
DI boxes may provide an input impedance of as much as 1,000,000
Ohms to suit the relatively high output impedance of typical guitar
pickups.

Initialize — Resetting a device to its 'start-up' state. Sometimes used

to mean restoring a piece of equipment to its factory default settings.

Insert Points — The provision on a mixing console or ‘channel strip’

processor of a facility to break into the signal path through the unit to
insert an external processor. Budget devices generally use a single
connection (usually a TRS socket) with unbalanced send and return
signals on separate contacts, requiring a splitter or Y-cable to provide
separate send (input to the external device) and return (output from
external device) connections . High end units tend to provide
separate balanced send and return connections. (cf. Effects Loop)

Input Impedance — The input impedance of an electrical network is

the ‘load’ into which a power source delivers energy. In modern audio
systems, the input impedance is normally about ten times higher than
the source impedance — so a typical microphone preamp has an

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input impedance of between 1500 and 2500 Ohms, and a line input is
usually between 10 and 50k Ohms.

Interface — A device that acts as an intermediary to two or more

other pieces of equipment. For example, a MIDI interface enables a
computer to communicate with MIDI instruments and keyboards.

Intermittent — Something that happens occasionally and

unpredictably, typically a fault condition.

Intermodulation Distortion — A form of non-linear distortion that

introduces frequencies not present in and musically unrelated to the
original signal. These are invariably based on the sum and difference
products of the original frequencies.

I/O — The input/output connections of a system.

IPS — Inches Per Second. Used to describe tape speed. Also, the
Institute of Professional Sound (www.ips.org.uk)

K-Metering — An audio level metering format developed by

mastering engineer Bob Katz which must be used with a monitoring
system set up to a calibrated acoustic reference level. Three VU-like
meter scales are provided, differing only in the displayed headroom
margin. The K-20 scale is used for source recording and wide
dynamic-range mixing/mastering, and affords a 20dB headroom
margin. The K-14 scale allows 14dB of headroom and is intended for
most pop music mixing/mastering, while the K-12 scale is intended
for material with a more heavily restricted dynamic-range, such as for
broadcasting. In all cases, the meter's zero mark is aligned with the
acoustic reference level.

K-Weighting — A form of electrical filter which is designed to mimic

the relative sensitivity of the human ear to different frequencies in
terms of perceived loudness. It is broadly similar to the A-Weighting
curve, except that it adds a shelf boost above 2kHz. This filter is an
integral element of the ITU-R BS.1770 loudness measurement
protocol. (See also A-Weighting and C-Weighting)

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Latency (cf. Delay) — The time delay experienced between a sound
or control signal being generated and it being auditioned or taking
effect, measured in seconds.

Limiter — An automatic gain-control device used to restrict the

dynamic range of an audio signal. A Limiter is a form of compressor
optimized to control brief, high level transients with a ratio greater
than 10:1.

Linear — A device where the output is a direct multiple of the input

with no unwanted distortions.

Line-level — A nominal signal level which is around -10dBV for semi-

pro equipment and +4dBu for professional equipment.

LKFS — see LUFS

Loop — The process of defining a portion of audio within a DAW,

and configuring the system to replay that portion repeatedly. Also, a
circuit condition where the output is connected back to the input.

Low Frequency Oscillator (LFO) — An oscillator used as a

modulation source, usually operating with frequencies below 20Hz.
The most common LFO waveshape is the sine wave, though there is
often a choice of sine, square, triangular and sawtooth waveforms.

Low-Pass Filter (LPF) — A filter which passes frequencies below its

cut-off frequency, but attenuates higher frequencies.

Loudspeaker (also Monitor and Speaker) — A device used to

convert an electrical audio signal into an acoustic sound wave. An
accurate loudspeaker intended for critical sound auditioning
purposes.

Loudness — The perceived volume of an audio signal.

Low-range (low, lows) — The lower portion of the audible frequency

spectrum, typically denoting frequencies below about 1kHz

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LUFS — The standard measurement of loudness, as used on
Loudness Meters corresponding to the ITU-TR BS1770 specification.
the acronym stands for 'Loudness Units (relative to) Full Scale.
Earlier versions of the specification used LKFS instead, and this label
remains in use in America. The K refers to the 'K-Weighting' filter
used in the signal measurement process.

Master — A device which controls slave devices. Often used to refer

to synchronized recorders, or digital clocking devices.

Maximum SPL — The loudest sound pressure level that a device

can generate or tolerate.

Memory — A computer's memory (RAM) used to store programs and

data. This data is lost when the computer is switched off and so must
be stored to disk or other suitable archive media.

Metering — A display intended to indicate the level of a sound signal.

It could indicate peak levels (eg. PPMs or digital sample meters),
average levels (VU or RMS meters), or perceived loudness (LUFS
meters).

Mic Level — The nominal signal level generated by a microphone.

Typically around -50dBu. Mic level signals must be amplified to raise
them to line-level.

Mid-range (mid, mids) — The middle portion of the audible frequency

spectrum, typically denoting frequencies between about 300Hz and
3kHz.

MIDI — Musical Instrument Digital Interface. A defined interface

format that enables electronic musical instruments and computers to
communicate instructional data and synchronize timing. MIDI sends
musical information between compatible devices, including the pitch,
volume and duration of individual notes, along with many other
aspects of the instruments that lend themselves to electronic control.
MIDI can also carry timing information in the form of MIDI Clock or
MIDI Time Code for system synchronization

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Modelling — A process of analyzing a system and using a different
technology to replicate its critical, desired characteristics. For
example, a popular but rare vintage signal processor such as an
equalizer can be analyzed and its properties modelled by digital
algorithms to allow its emulation within the digital domain.

Monitor (also Loudspeaker ) — A device used to convert an

electrical audio signal into an acoustic sound wave. An accurate
loudspeaker intended for critical sound auditioning purposes. Also
used to refer to a computer display screen (VDU), or the act of
auditioning a mix or a specific audio signal.

Monitor Controller — A line-level audio signal control device used to

select and condition input signals for auditioning on one or more sets
of monitor loudspeakers. Some monitor controllers also incorporate
facilities for studio talkback and artist cue mixes.

Mono — A single channel of audio.

M-S (Mid-Side) – A specialist form of coincident microphone array

which, when decoded to left-right stereo, creates an equivalent XY
configuration. In the MS array one microphone is pointed directly
forward (Mid) while the second is arranged at 90 degrees to point
sideways (Side). The Mid microphone can employ any desired polar
pattern, the choice strongly influencing the decoded stereo
acceptance angle. The Side microphone must have a figure-eight
response and be aligned such that the lobe with the same polarity as
the Mid microphone faces towards the left of the sound stage.
Adjusting the relative sensitivity of the Mid and Side microphones
affects the decoded stereo acceptance angle and the polar patterns
of the equivalent XY microphones.

Near Field — The acoustic zone close to a sound source or

microphone. Often used to describes a loudspeaker system designed
to be used close to the listener – although some people prefer the
term 'close field'. The advantage is that the listener hears more of the
direct sound from the speakers and less of the reflected sound from
the room.

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Noise-shaping — A system using spectrally-shaped dither to
improve the perceived signal-to-noise performance of a digital audio
system.

Non-linear Recording — A term which describes digital recording

systems that allow any parts of the recording to be played back in any
order with no gaps. Conventional tape is referred to as linear,
because the material can only play back in the order in which it was
recorded.

Nyquist Theorem — The rule which states that a digital sampling

system must have a sample rate at least twice as high as that of the
highest audio frequency being sampled in order to avoid aliasing and
thus reproduce the wanted audio perfectly. Because anti-aliasing
filters aren't perfect, the sampling frequency has usually to be made
slightly more than twice that of the maximum input frequency —
which is why the standard audio rate of 44.1kHz was chosen for a
nominally 20kHz audio bandwidth.

Opto-electronic Device – A device where some electrical parameter

changes in response to a variation in light intensity. For example,
variable photo-resistors are sometimes used as gain control elements
in compressors where the side-chain signal modulates the light
intensity.

Overdrive — The intentional use of overloaded analogue circuitry as

a musical effect.

Overload — To exceed the maximum acceptable signal amplitude of

an electronic or electrical circuit. Overloading a device results in a
noticeable increase in distortion but this may be deemed musically
beneficial and desirable, or completely unacceptable and
inappropriate, depending on context and intent. Overloading an
analogue device typically results in the waveform peaks becoming
flattened (so tending towards a square wave) and a consequent rapid
increase in odd-order harmonic distortion where the distortion
products appear at higher frequencies than the source signal
fundamentals, but remain musically related to them. In contrast,
overloading a digital system inherently contravenes the Nyquist

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Theorem, since he generated harmonic distortion products generally
extend far above half the sampling frequency, and so become aliased
and actually appear at lower frequencies than the source
fundamentals with a non-musical relationship. This is why digital
overloads sound so obvious and unpleasant in comparison to
analogue overloads.

Pad — A resistive circuit for reducing signal level.

Pan-pot — A control found on mixers to move the signal to any point

in the stereo soundstage by varying the relative levels fed to the left
and right stereo outputs.

Parallel — A means of connecting two or more circuits together so

that their inputs are connected together, and their outputs are all
connected together.

Parameter — A variable value that affects some aspect of a device's

performance.

Parametric EQ — An equalizer with separate controls for frequency,

bandwidth and cut/boost.

PCI Card — Peripheral Component Interconnect: an internal

computer bus format used to integrate hardware devices such as
sound cards. The PCI Local Bus has superseded earlier internal bus
systems such as ISA and VESA, and although still very common on
contemporary motherboards has, itself, now been superseded by
faster interfaces such as PCI-X and PCI Express.

PCM — Pulse Code Modulation — the technique used by most digital

audio systems to encode audio as binary data.

Peak — The maximum instantaneous level of a signal.

PFL — Pre-Fade Listen. A system used within a mixing console to

allow the operator to audition a selected signal, regardless of the
position of the fader controlling that signal.

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Phase — The relative position of a point within a cyclical signal,
expressed in degrees where 360 degrees corresponds to one full
cycle. (Also see Polarity)

Phaser — An effect which combines a signal with a phase-shifted

version of itself to produce creative comb-filtering effects. Most
phasers are controlled by means of an LFO.

Phantom Power — A means of powering capacitor and electret

microphones, as well as some dynamic microphones with built-in
active impedance converters. Phantom power (P48) provides 48V
(DC) to the microphone as a common-mode signal (both signal wires
carry 48V while the cable screen carries the return current). The
audio signal from the microphone is carried as a differential signal
and the mic preamp ignores common-mode signals so doesn’t see
the common-mode power supply (hence the ghostly name, phantom).
This system only works with a balanced three-pin mic cables. Two
alternative phantom power specifications also exist, with P12 (12V)
and P24 (24V) options, although they are relatively rare.

Pink Noise — A random signal with a power spectral density which is

inversely proportional to the frequency. Each octave carries an equal
amount of noise power. Pink noise sounds natural, and resembles
the sound of a waterfall. (cf. White Noise)

Pitch — The musical interpretation of an audio frequency.

Pitch-bend — A special control message specifically designed to

produce a change in pitch in response to the movement of a pitch
bend wheel or lever. Pitch bend data can be recorded and edited, just
like any other MIDI controller data, even though it isn't part of the
Controller message group.

Pitch-shifter — A device for changing the pitch of an audio signal

without changing its duration.

Plug-in — A self-contained software signal processor, such as an

Equalizer or Compressor, which can be ‘inserted’ into the notional
signal path of a DAW. Plug-ins are available in a myriad of different

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forms and functions, and produced by the DAW manufacturers or
third-party developers. Most plug-ins run natively on the computer’s
processor, but some require bespoke DSP hardware. The VST
format is the most common cross-platform plug-in format, although
there are several others.

Polarity — This refers to a signal's voltage above or below the

median line. Inverting the polarity of a signal swaps the positive
voltage to negative voltage and vice versa. This condition is often
referred to (incorrectly) as 'out-of-phase'.

Potentiometer (Pot) — A form of electrical potential divider in which

the ratio of the upper and lower resistances can be changed either
with a rotary control or slider (eg. a fader).

Power Amplifier — A device which accepts a standard line-level

input signal and amplifies it to a condition in which it can drive a
loudspeaker drive unit. The strength of amplification is denoted in
terms of Watts of power.

Power supply — A unit designed to convert mains electricity to the

DC voltages necessary to power an electronic circuit or device.

Powered Loudspeaker or Monitor — A powered speaker is a

conventional passive loudspeaker but with a single power amplifier
built in or integrated with the cabinet in some way. The amplifier
drives a passive crossover, the outputs of which connect to the
appropriate drive units.

Post-fade — A signal derived from the channel path of a mixer after

the channel fader. A post-fade aux send level follows any channel
fader changes. Normally used for feeding effects devices.

PPM — Peak Program Meter. A meter designed to register the

approximate peak amplitude of a signal, rather than the average level
indicated by, for example, a VU meter. However, PPMs have a
defined integration time (typically 10ms) which means that they
actually under-read on the fastest transient peaks. (cf. VU Meter)

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Pre-amp — Short for ‘pre-amplification’ : an active gain stage used to
raise the signal level of a source to a nominal line level. For example,
a microphone pre-amp.

Pre-fade — A signal derived from the channel path of a mixer before

the channel fader. A pre-fade aux send level is unaffected by channel
fader changes. Normally used for creating Foldback or Cue mixes.

Proximity Effect — Also known as ‘Bass tip-up’. The proximity effect

dramatically increases a microphone’s sensitivity to low frequencies
when placed very close to a sound source. It only affects directional
microphones — omnidirectional microphones are immune.

Q — The ‘quality-factor’ of a filter which defines its bandwidth and

indicates a filter’s resonant properties. The higher the Q, the more
resonant the filter and the narrower the range of frequencies that are
allowed to pass.

Quantization — Part of the process of digitizing an analogue signal.

Quantization is the process of describing or measuring the amplitude
of the analogue signal captured in each sample, and is defined by the
word length used to describe the audio signal — eg. 16 bits.

Quantize — A means of moving notes recorded in a MIDI sequencer

so that they line up with user defined subdivisions of a musical bar,
for example, 16s. The facility may be used to correct timing errors,
but over-quantization can remove the human feel from a
performance.

RAM — An abbreviation for Random Access Memory. This is a type

of memory used by computers for the temporary storage of programs
and data, and all data is lost when the power is turned off. For that
reason, work needs to be saved to disk if it is not to be lost.

Real-time — An audio process that can be carried out as the signal

is being recorded or played back. The opposite is off-line, where the
signal is processed in non-real time.

Reflection — The way in which sound waves bounce off surfaces.

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Release — The time taken for a signal level or processor gain to
return to normal. Often used to describe the rate at which a
synthesized sound reduces in level after a key has been released.
Also used to describe the time taken for a compressor top restore
unity gain after a signal has fallen below the threshold. Also known as
‘recovery time .‘

Resistance — Opposition to the flow of electrical current. Measured

in Ohms.

Resonance — The characteristic of a filter that allows it to selectively

pass a narrow range of frequencies. See Q.

Reverb — Short for Reverberation. The dense collection of echoes

which bounce off acoustically reflective surfaces in response to direct
sound arriving from a signal source. Reverberation can also be
created artificially using various analogue or, more commonly, digital
techniques. Reverberation occurs a short while after the source
signal because of the finite time taken for the sound to reach a
reflective surface and return — the overall delay being representative
of the size of the acoustic environment. The reverberation signal can
be broadly defined as having two main components, a group of
distinct ‘early reflections’ followed by a noise-like tail of dense
reflections.

Reverberation Time — The time taken for sound waves reflecting

within a space to lose energy and become inaudible. A standard
measurement is ‘RT60’ which is the time taken for the sound
reflections to decay by 60dB.

RMS — Root Mean Square. A statistical measure of the magnitude of

a varying quantity. Its name comes from its definition as the square
root of the mean of the squares of the values of the signal.

Roll-off — The rate at which a filter or equalizer attenuates a signal

once it has passed the turnover frequency.

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Sample — Either a defined short piece of audio which can be
replayed under MIDI control; or a single discrete time element
forming part of a digital audio signal.

Sample rate — The number of times an A/D converter samples the

incoming waveform each second.

Sibilance — A high-frequency whistling or lisping sound that affects

vocal recordings, due either to poor mic technique or excessive HF
equalization.

Side-chain — A part of an audio circuit that splits off a proportion of

the main signal to be processed in some way. Compressors use a
side-chain process to derive a control signals to adjust the main path
attenuation.

Signal — An electrical representation of an audio event.

Signal Chain — The route taken by a signal from the input of a

system to the output.

Signal-to-noise Ratio — The ratio of nominal or maximum signal

level to the residual noise floor, expressed in decibels.

Sine Wave — The waveform of a pure sinusoidal tone with no

harmonics.

SPL — Sound Pressure Level. A measure of the intensity of an

acoustic sound wave. Normally specified in terms of Pascals for an
absolute value, or relative to the typical sensitivity of human hearing.
One Pascal is 94dB SPL, or to relate it to atmospheric pressures,
0.00001 Bar or 0.000145psi!

Standing Waves — Resonant low frequency sound waves bouncing

between opposite surfaces such that each reflected wave aligns
perfectly with previous waves to create static areas of maximum and
minimum sound pressure within the room. (See also Modes and
Modal Frequencies)

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Stems — When mixing complex audio material it is often useful to
divide the tracks into related sections and mix those sections
separately before combining the whole. In mixing film soundtracks,
the material would often be grouped as a dialogue stem, a music
stem, and an effects stem. Each stem might be mono, stereo or
multichannel, as appropriate to the situation. In music mixing, stems
might be used for the rhythm section, backline instruments, frontline
instruments, backing vocals, lead vocals and effects — or any other
combination that suited the particular project.

Sub-bass — Frequencies below the range of typical monitor

loudspeakers. Some define sub-bass as frequencies that can be felt
rather than heard.

Surround — The use of multiple loudspeakers placed around the

listening position with the aim of reproducing a sense of envelopment
within a soundstage. Numerous surround formats exist, but the most
common currently is the 5.1 configuration in which three
loudspeakers are placed in front of the listener (at ±30degrees and
straight ahead), with two behind (at ±120 degrees or thereabouts),
supplemented with a separate subwoofer.

Sweet Spot — The optimum position for a microphone, or for a

listener relative to monitor loudspeakers.

Tempo — The rate of the 'beat' of a piece of music measured in

beats per minute.

Timbre — The tonal 'color' of a sound.

Track — The term dates back to multitrack tape where the tracks are
physical stripes of recorded material, located side by side along the
length of the tape.

Tracking — The process of recording individual tracks to a

multichannel recorder. Tracking is also often discussed in the context

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of MIDI guitar synthesizers or controllers where the MIDI output
attempts to track the pitch of the guitar strings.

Transformer — An electrical device in which two or more separate

and electrically isolated coils of wire are wound around a common
ferromagnetic core. Alternating Current passing through one coil
creates a varying magnetic field which induces a corresponding
current in the other coil(s). In audio applications transformers are
often used to convey a signal without a direct electrical connection,
thus providing 'galvanic isolation' between the source and destination.
Winding a transformer with different numbers of turns for each coil
allows the output voltage to be increased or decreased in direct
proportion – a feature widely employed in mains power-supply
transformers to reduce the mains voltage to something more
appropriate for the circuitry, for example, or in microphone preamp
step-up transformers.

Transients — An element of a sound where the spectral content

changes abruptly. Most natural sounds start with a transient element
before settling into something more steady-state, and it is often that
transient element that provides most of the recognizable character of
the sound source.

Transparency — A subjective term used to describe audio quality

where the high frequency detail is clear and individual sounds are
easy to identify and separate.

Tremolo — A form of modulation of the amplitude of a sound using

an LFO. (cf. vibrato)

Transducer — A device for converting one form of energy to

another. Microphones and Loudspeakers are good examples of
transducer converts between mechanical and electrical energy.

Transpose — To shift a musical signal by a fixed number of

semitones.

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True Peak Meter – A form of digital audio meter which is capable of
determining the absolute amplitude value of a digital signal by using
oversampling to fully reconstruct the waveform.

Unbalanced — A 2-wire electrical signal connection where the signal

conductor is surrounded by a screen which provides a 0V reference
and also guards against electrical interference.

Unison — To play the same melody using two or more different

instruments or voices.

Unity Gain — A condition where the output signal is the same

amplitude as the input signal; the overall system gain is then x1 or
unity.

Valve — Also known as a ‘tube’ in America. A thermionic device in

which the current flowing between its anode and cathode terminals is
controlled by the voltage applied to one or more control grid(s).
Valves can be used as the active elements in amplifiers, and because
the input impedance to the grid is extremely high they are ideal for
use as an impedance converter in capacitor microphones. The
modern solid-state equivalent is the Field Effect Transistor or FET.

Vari-Mu Compressor — An audio compressor that employs a valve

(tube) as the variable audio attenuator. Mu is an engineering term for
gain, so this is a variable-gain compressor. In essence, the side-chain
signal continuously adjusts the bias o the valve to alter its gain
appropriately. Vari-Mu compressors are fast and smooth, with low
distortion.

VCA — Voltage Controlled Amplifier. An amplifier in which the gain

(or attenuation) is controlled by an external DC voltage. VCA's are
used in a wide range of audio and musical equipment, such as fader-
automation systems in large format mixing consoles, audio
compressors, and synthesizers.

VCA Compressor — See VCA. VCA compressors tend to be fast-

acting (at least in comparison to opto-compressors), a wide dynamic
range, and low distortion.

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Vibrato — Pitch modulation using an LFO to modulate a VCO. (cf.
Tremolo)

VU Meter — An audio meter designed to interpret signal levels in

roughly the same way as the human ear, which responds more
closely to the average levels of sounds rather than to the peak levels.
(cf. PPM)

Warmth — A subjective term used to describe sound where the bass

and low mid frequencies have depth and where the high frequencies
are smooth sounding rather than being aggressive or fatiguing. Warm
sounding tube equipment may also exhibit some of the aspects of
compression.

Waveform — A graphic representation of the way in which a sound

wave or electrical wave varies with time.

Wet — A signal that has effects added. (cf. Dry)

White Noise — A random signal with a flat (constant) power

spectrum density, ie. equal power within any frequency band of fixed
width. White noise sounds very bright (cf. Pink Noise).

Zero Crossing Point — The point at which a signal waveform

crosses from being positive to negative or vice versa.

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