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Zal1 - Słownik Akustyczny

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A-Type Plug, A-gauge plug

A-Weighting

AC

Accent Mic
Acoustic Foam

Acoustic Treatment

Active

Active Loudspeaker or Monitor

A/D [A-D] Converter


ADAT Lightpipe

Additive Synthesis

ADSR

Active Sensing

AES

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AES3

AES10

AES11

AES17
AES42

AES59

AFL

Aftertouch

Algorithm

Aliasing

Ambience

Amp (Ampere)
Amp/Amplifier

Amplitude
Analogue (cf. Digital)

Analogue Synthesis
Anti-alias Filter

Application (App)
Arming

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Arpeggiator

ASCII

Attack

Attenuate
Audio Data Reduction

Audio Frequency
Audio Interface

Audio Random Access (ARA)

Audio Scrubbing
Autolocator

Auxiliary Sends (Auxes)

Aux Return

Azimuth

B-Type Plug, B-gauge plug

Backup

Back Electret

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Balance

Balanced Wiring

Band-pass Filter (BPF)

Bandwidth

Bank

Bass Response

Bass Tip-up
Bass Trap
Bantam Plug

Beta Version
Bias

Binary

BIOS

Bit
Bit Rate (see also Sample Rate)

Bi-Timbral
Blumlein Array

BNC

Boom

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Boost/Cut Control

Booth
Bouncing
Boundary

Boundary Layer Microphone

BPM
Breath Controller
Buffer

Buffer Memory

Bug
Bus

Byte
C-Weighting

Cabinet

Cabinet Resonance

Capacitor

Capacitor Microphone

Capsule
Carbon Microphone

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Cardioid

CD-R

CD-R Burner
Channel

Chase

Chip
Chord
Chorus

Chromatic
Click Track
Clipping

Clocking

Clone
Close-Miking

Cloud

Coincident

Colouration
Comb-Filter

Common Mode Rejection

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Compact Cassette

Compander

Comping

Compressor

Computer

Condenser Microphone
Conductor
Cone

Console
Contact Cleaner

Continuous Controller

Control Voltage

Converter

Convolution

Convolution Reverb

Copy Protection
CPU – Central Processing Unit
Crash

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Crossover

Crossover frequency

Cut and Paste Editing


Cut-off Frequency
Cycle

CV
Daisy Chain

Damping

DANTE

DAT

Data
DAW - Digital Audio Workstation

dB

dB/Octave

DC

DCA

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DCA Group
DCC

DBX

DCO
DDL
DDP

De-emphasis
De-esser
De-Oxidising Compound
Decay

Decca Tree

Decibel
Decoupler (also isolator)

Defragment

Delay

Desk
Detent

DI

Diaphragm

DI Box

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Digital (cf. Analogue)

Digital Delay
Digital Reverberator
DIN Connector

Diode-Bridge Compressor

Direct Coupling
Dither

Disc
Disk
DMA

Dolby Noise-Reduction

Dolby Surround-Sound

Dolby HX

DOS
Dome

Double-ended Noise Reduction

Double-lapped Screen

DSP

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Drive unit

Driver

Dropout

Drum Pad
Drum Booth
Dry (cf. Wet)
Dubbing

Ducking

Dump

DVS

Dynamic Microphone

Dynamic Range

Dynamics
eSATA
Early Reflections

Effect

Effects Loop

Effects Return

Electret Microphone
Encode/Decode

Enhancer (cf. Exciter)

Envelope
Envelope generator

E-PROM

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Equaliser (cf. Filter)

Equivalent Input Noise

Erase
EuCon

Eurorack

Event

Exciter (cf. Enhancer)


Expander

Expander Module
Fader
Ferric
FET

FET-Compressor

Fidelity
Figure of Eight

File

Filter (cf. Equaliser)

Filter Frequency

FireWire

Flanging

Flash Drive (see 'solid-state drive')

Floppy Disk
Flutter

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Flutter Echoes

Foldback

Formant

Format

Fragmentation (cf. defragment)

Frequency

Frequency Response

FSK

Fukada Tree

Fundamental
FX
Gain
Gain Staging

Galvanic Isolation

Gate (CV)

Gate

General MIDI

Glitch

GM Reset

Gooseneck
Graphic Equaliser

Ground

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Ground Loop / Ground Loop Hum

Group

GS
GUI

Hard Disk Drive (cf. Solid-state Drive)

Harmonic

Harmonic Distortion

Head

Headroom

Hertz (Hz)
High-Pass Filter (HPF)
High-range (highs)
High Resolution

Hiss
Hub

Hum

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Hysteresis

Hz
IC

IEM

Impedance

Impulse Response

Inductor
Initialise

Insert Points

Input Impedance

Insulator
Instrument Level

Interface

Intermittent
Intermodulation Distortion

I/O
IPS
IRQ

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Isolation Room

Isolator (also decoupler)

Isopropyl Alcohol
Jackfield

Jack Plug

Jargon
Jog Wheel

K-Metering

K-Weighting

Latency (cf. Delay)

Lay Length

LED
LCD
LFO
LSB

Lightpipe
Limiter

Linear
Line-level
LKFS
Load

Local On/Off

Logic
Loom

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Loop

Low Frequency Oscillator (LFO)

Low-Pass Filter (LPF)


Loudspeaker (also Monitor and Speaker)

Loudness
Loudness-Normalisation

Loudness Wars

Low-range (low, lows)


LUFS

MADI

Magnetic Shielding

Master
Mastering

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Matrix

Maximum SPL
MB

Machine Head
MDM

Memory

Menu
Metering

Mic Level

Microphone
Microprocessor

Mid-range (mid, mids)

MIDI

MIDI Analyser
MIDI Bank Change

MIDI Controller

MIDI Control Change

MIDI File

MIDI Implementation Chart

MIDI In
MIDI Merge
MIDI Module
MIDI Multitimbral Module

MIDI Mode

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MIDI Note Number

MIDI Note On
MIDI Note Off
MIDI Out
MIDI Port

MIDI Program Change

MIDI Splitter
MIDI Sync
MIDI Thru
MIDI Thru Box

Mineral Wool

Mirror Points

Mixer

Modal Distribution
Modelling

Modes (room)
Monitor (also Loudspeaker )

Monitor Controller

Mono
Monophonic
Mono-synth
Motherboard
Moving Coil Microphone

M-S (Mid-Side)

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MTC - MIDI Time code


Mult

Multi-sample

Multi-timbral

Multitrack

Mutual Angle

Near-coincident

Near Field

Noise Reduction

Noise-shaping

Non-registered parameter Number

Non-linear Recording

Normalise

NOS

Nyquist Theorum

Nut

Octave
Off-line

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Off/On-axis

Ohm
Omnidirectional

Open Circuit
Open Reel
Open Sound Control
Operating System
Optimisation (of computer)

Opto-electronic Device

ORTF

OSC
Oscillator
Out-of-Phase
Output Impedance

Output Sensitivity

Overdubbing

Overdrive
Overload

Overtone

Pad
Pan-pot

Parallel

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Parameter
Parameteric EQ
Paraphonic

Partials
Passive
Passive Loudspeaker or Monitor

Patch

Patch Bay

Patch Cord
PCI Card

PCM
Peak
Peak-Normalisation

PFL

Phase

Phaser

Phantom Power

Phono plug (RCA-phono)

Pickup

Pink Noise

Pitch
Pitch-bend

Pitch-shifter

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Plug-in

Plug-in Power

Polar Pattern
Polarity

Polyphony

Poly-mode

Poly-Synth

Pop Shield

Port
Portamento

Post-production
Potentiometer (Pot)

Power Amplifier

Power supply
Powered Loudspeaker or Monitor

Post-fade

PPM

PPQN
PQ Coding

Pre-amp

Pre-emphasis

Pre-fade

Preset
Pressure
Print-through

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Processor

Program Change
Project Studio
Proximity Effect

Pulse Wave

Pulse-width Modulation

Punch-in

Punch-out

PWM Compression

PZM (Pressure Zone Microphone)


Q

Quantisation

Quantiser

Rack Mount

RAM

R-DAT
Real-time

Red Book CD

Reflection
Release

Resistance
Resonance
Reverb

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Reverberation Time

RF
RF Interference
RF Capacitor Microphone

Ribbon Microphone

Rider

Ring Modulator

RMS

Roll-off
ROM

Room Modes

Rotary Encoder

Safety Copy
Sample

Sample rate
Sample and Hold (S&H)

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SATA

Sawtooth Wave
Scrape Flutter

Scrubbing

SCSI

Session Tape
Sequencer

Shockmount

Short-Circuit

Sibilance

Side-chain

Signal
Signal Chain
Signal-to-noise Ratio
Sine Wave
Single-ended Noise Reduction

Slate

Slave

SMPS
SMPTE

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S/MUX
Snake

Solid-state Drive (cf. Hard Disk Drive)

Sound Card

Sound On Sound

Soundproofing

Spaced Array

S/PDIF

Speaker (also Loudspeaker and Monitor)


Spill
SPL

SPP
Square Wave
SRA
SSD
Standard Midi File
Standing Waves

Stage Box

Stems

Step Time
Stereo

Stereo Recording Angle

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Sticky Shed Syndrome

Stripe

Sub-bass

Subcode

Subgroup
Subtractive Synthesis
Subwoofer

Surge
Surround

Sustain

Swan Neck
Sweet Spot
Switching Power Supply

Sync
Synthesis
Synthesiser
SysEx

Talkback
Tape Head

Tempo
Test Tone

THD

Thru
Thunderbolt

Timbre
Timbral
TOSlink

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Track

Tracking

Transformer

Transients

Transmission-Line

Transparency

Tremolo
Transducer

Transpose
Triangle Wave

TRS

True Peak Meter

Truss Rod

TT Plug
Tube
Tweeter

Unbalanced

Unison

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Unity Gain

USB

USB-C

Valve

Vari-Mu Compressor

VCA

VCA Compressor

VCA Group

VDU
Velocity

Vocoder

Vocal Booth
Voice

Vibrato
VU Meter

Wah Pedal
Watt (W)
Warmth

Waveform
Way (as in, 2-way, 3-way)

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Wet
White Noise

Word Clock

Wow & Flutter

Wrap

Write
XG

XLR

X-Y

Y-Lead

Zenith

Zero Crossing Point


Zipper Noise

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A domestic and semi-pro form of jack plug, also known as TS or TRS and widely used for electric
instruments, headphones and line-level connections on semi-pro equipment. (cf. B-Type Plug)
A form of electrical filter which is designed to mimic the relative sensitivity of the human ear to different
frequencies at low sound pressure levels (notionally 40 Phons or about 30dBA SPL). Essentially, the filter
rolls-off the low frequencies below about 700Hz and the highs above about 10kHz. This filtering is often
used when making measurements of low-level sounds, like the noise floor of a device. (See also C-
Weighting and K-Weighting)
Alternating Current (cf. DC). Audio signals are represented in the electrical domain as currents flowing
alternately forward and back in the circuits as an analogue of the compression and rarefaction of acoustic
air
seepressure.
Close Miking
A specific type of open-celled expanded polyurethane foam that allows sound waves to enter and flow
through the foam, absorbing their energy and preventing them being reflected. The density and depth of
the foam affects the frequency range over which it is effective as an absorber.
A generic term embracing a range of products or constructions intended to absorb, diffuse or reflect
sound waves in a controlled manner, with the intention of bestowing a room with an acceptable
reverberation time and overall sound character.
Describes a circuit containing transistors, ICs, tubes and other devices that require power to operate, and
which are capable of amplification.
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.
A device which converts an analogue audio signal into a digital representation.
A widely used eight-channel optical digital audio interface developed by Alesis as a bespoke interface for
the company's digital eight-track tape machines in the early 1990s (Alesis Digital Audio Tape). The
interface transfers up to eight channels of 24-bit digital audio at base sample rates (44.1 or 48kHz) via a
single fibre-optic cable. This 'lightpipe' is physically identical to that used for the TOSlink optical S/PDIF
stereo interface found on many digital consumer hi-fi devices, but while the fibre itself can be used
interchangeably for either format, the S/PDIF and ADAT interfaces are not compatible in any other way.
The interface incorporates embedded clocking, and padding zeros are introduced automatically if the word
length is less than 24 bits. Although not supported by all ADAT interfaces, most modern devices employ
the S/MUX (Sample Multiplexing) protocol (licensed from Sonorus) which allows higher sample rates to be
employed at the cost of fewer channels of audio. The S/MUX2 format operates at double sample rates
(88.2 and 96kHz) but carries only four channels, while S/MUX4 operates at quad rates (176.4 and
192kHz) with two channels. S/MUX uses a clever technique that divides the high sample rate data across
the nominal channels in such a way that accidental level changes or dithering applied identically to each
channel in the data stream will not destroy the wanted demultiplexed signal.

A system for generating audio waveforms or sounds by combining basic waveforms or sampled sounds
prior to further processing with filters and envelope shapers. The Hammond tonewheel organ was one of
the first additive synthesizers.
When creating artificial waveforms in a synthesizer, changes in the signal amplitude over time are
controlled by an ‘envelope generator’ which typically has controls to adjust the Attack, Sustain, Decay and
Release times, controlled by the pressing and subsequent release of a key on the keyboard. The Attack
phase determines the time taken for the signal to grow to its maximum amplitude, triggered by the
pressing of a key. The envelope then immediately enters the Decay phase during which time the signal
level reduces until it reaches the Sustain level set by the user. The signal remains at this level until the key
is released, at which point the Release phase is entered and the signal level reduces back to zero.

A system used to verify that a MIDI connection is working. It involves the sending device sending frequent
short messages to the receiving device to reassure it that all is well. If these active sensing messages stop
for any reason the receiving device will recognise a fault condition and switch off all notes. Not all MIDI
devices support active sensing.
Acronym for Audio Engineering Society, one of the industry's professional audio associations.

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A digital audio interface which passes two digital audio channels, plus embedded clocking data, with up
to 24 bits per sample and sample rates up to 384kHz. Developed by the Audio Engineering Society and
the European Broadcasting Union, it is often known as the AES-EBU interface. Standard AES3 is
connected using 3-pin XLRs with a balanced cable of nominal 110 Ohm impedance and with a signal
voltage of up to 7V pk-pk. The related AES3-id format uses BNC connectors with unbalanced 75 Ohm
coaxial cables and a 1V pk-pk signal. In both cases the datastream is structured identically to S/PDIF,
although some of the Channel status codes are used differently.

An AES standard which defines the MADI interface (serial Multichannel Audio Digital Interface). MADi can
convey either 56 or 64 channels via single coaxial or optical connections.
An AES standard that defines the use of a specific form of AES3 signal for clocking purposes. Also
known as DARS (Digital Audio Reference Signal).
An AES standard that defines a method of evaluating the dynamic range performance of A-D and D-A
An AES standard which defines the connectivity, powering, remote control and audio format of ‘digital
microphones.’ The audio information is conveyed as AES3 data, while a bespoke modulated 10V
phantom power supply conveys remote control and clocking information.
An AES standard which defines the use and pin-outs of 25-pin D-sub connectors for eight-channel
balanced analogue audio and bi-directional eight-channel digital interfacing. It conforms fully with the
established Tascam interface standard.
After Fade listen. A system used within mixing consoles to allow specific signals to be monitored at the
level set by their fader. Aux sends are generally monitored AFL rather than PFL (see PFL).
A means of generating a control signal in a synthesizer based on how much pressure is applied to the
keys of a MIDI keyboard. Most instruments that support this do not have independent pressure sensing for
all keys, but rather detect the overall pressure by means of a sensing strip running beneath the keys.
Aftertouch may be used to control such functions as vibrato depth, filter brightness, loudness and so on.
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.
When an analogue signal is sampled for conversion into a digital data stream, the sampling frequency
must be at least twice that of the highest frequency component of the input signal. If this rule is disobeyed
the sampling process becomes ambiguous as there are insufficient points to define each cycle of the
waveform, resulting in unwanted enharmonic frequencies being added to the audible signal.
The result of sound reflections in a confined space being added to the original sound. Ambience may also
be created electronically by some digital reverb units. The main difference between ambience and
reverberation is that ambience doesn't have the characteristic long delay time of reverberation; the
reflections mainly give the sound a sense of space.
Unit of electrical current (A).
An Amplifier is an electrical device that typically increases the voltage or power of an electrical signal. The
amount of amplification can be specified as a multiplication factor (eg. x10) or in decibels (eg. 20dB).
The waveform signal level. It can refer to acoustic sound levels or electrical signal levels.
The origin of the term is that the electrical audio signal inside a piece of equipment can be thought of as
being ‘analogous’ to the original acoustic signal. Analogue circuitry uses a continually changing voltage or
current to represent the audio signal.
A system for synthesizing sounds by means of analogue circuitry, usually by filtering simple repeating
A very steep low-pass filter used to limit the frequency range of an analogue signal prior to A/D
conversion so that the maximum frequency does not exceed half the sampling rate.
Alternative term for computer program.
Arming a track or channel on a recording device places it in a condition where it is ready to record audio
when the system is placed in record mode. Unarmed tracks won’t record audio even if the system is in
record mode. When a track is armed the system monitoring usually auditions the input signal throughout
the recording, whereas unarmed tracks usually replay any previously recorded audio.

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A device (or software) that allows a MIDI instrument to sequence around any notes currently being
played. Most arpeggiators also allows the sound to be sequenced over several octaves, so that holding
down a simple chord can result in an impressive repeating sequence of notes.
American Standard Code for Information Interchange. An internationally recognised code used to
represent computer keyboard characters.
The time taken for a sound to achieve its maximum amplitude. Drums have a fast attack, whereas bowed
strings have a slow attack. In compressors and gates, the attack time equates to how quickly the
processor can reduce the signal level.
To reduce the signal amplitude or level.
A system used to reduce the amount of data needed to represent some information such as an audio
signal. Lossless audio data reduction systems, (eg. FLAC and ALAC) can fully and precisely reconstruct
the original audio data with bit-accuracy, but the amount of data reduction is rarely much more than 2:1.
Lossy data audio reduction systems (eg. MPeg. AAC, AC3 and others) permanently discard audio
information that is deemed to have been 'masked' by more prominent sounds. The original data can never
be retrieved, but the reduction in total data can be considerable (12:1 is common).

Signals in the range of human audio audibility. Nominally 20Hz to 20kHz.


A device which acts as the physical bridge between the computer’s workstation software and the
recording environment. An audio interface may be connected to the computer (via FireWire, USB,
Thunderbolt, Dante, AVB or other current communication protocols) to pass audio and MIDI data to and
from the computer. Audio Interfaces are available with a wide variety of different facilities including
microphone preamps, DI inputs, analogue line inputs, ADAT or S/PDIF digital inputs, analogue line and
digital outputs, headphone outputs, and so on. The smallest audio interfaces provide just two channels in
and out, while the largest may offer 30 or more.
Developed by Celemony and PreSonus in 2011, ARA relates to a data-exchange extension for DAW
plug‑ins like AU, VST and RTAS, to pass information relating to an entire track, rather than just about a
specific moment in time such as pitch, tempo, rhythm etc. ARA v2 was announced in 2018 with an
extended feature set including integrated undo synchronisation. Most of the popular DAWs are now
compatible with ARA v2, but the only plug-ins that use it at present are Melodyne and Revoice Pro.

see Scrubbing
A common facility on tape machines or other recording devices that enables specific time points to be
stored and recalled. For example, you may store the start of a verse as a locate point so that you can get
the tape machine or DAW to automatically relocate the start of the verse after you've recorded an
overdub.
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.
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)
the alignment of a tape head which references the head gap to the true vertical relative to the tape path.
(cf. Wrap and Zenith).
A professional form of jack plug derived from the telecommunications industry and also known as the
PO316. Widely used for balanced mic and line-level connections on professional patch bays. (cf. A-Type
Plug)
A safety copy of software or other digital data. A popular saying is that unless data exists in three
physically separate locations at the same time, it hasn’t been backed up properly!
A form of electrostatic or capacitor microphone. Instead of creating an electrostatic charge within the
capacitor capsule with an external DC voltage, an electret microphone employs a special dielectric
material which permanently stores a static-electric charge. A PTFE film is normally used, and where this is
attached to the back plate of the capsule the device is called a ‘back electret’. Some very early electret
microphones used the dielectric film as the diaphragm but these sounded very poor, which is why later
and better designs which used the back electret configuration were specifically denoted as such. Designs
which attach the PTFE film to the diaphragm are known as Front Electrets. Modern electret capsules
compare directly in quality with traditional DC-biased capacitor capsules, and are available in the same
range of configurations large, medium and small diaphragm sizes, single and dual membrane, fixed or
multi-pattern, and so on.

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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).
Where protection from electromagnetic interference and freedom from earth references are required, a
balanced interface is used. The term ‘balanced’ refers to identical (balanced) impedances to ground from
each of two signal carrying conductors which are enclosed, again, within an all-embracing overall screen.
This screen is grounded (to catch and remove unwanted RFI), but plays no part in passing the audio
signal or providing its voltage reference. Instead, the two signal wires provide the reference voltage for
each other Signals conveyed over the balanced interface may appear as equal half-level voltages with
opposite polarities on each signal wire. the most commonly described technique. However, modern
systems are increasingly using a single-sided approach where one wire carries the entire signal voltage
and the other a ground reference for it. Some advantages of this technique include less complicated
balanced driver stages, and connection to an unbalanced destination still provides the correct signal level,
yet the interference rejection properties are unaffected. Effective interference rejection requires both the
sending and receiving devices to have balanced output and input stages respectively.
A filter that removes or attenuates frequencies above and below the centre frequency at which it is set,
and only passes a specific range of frequencies. Band-pass filters are often used in synthesizers as tone
shaping
The range elements.
of frequencies passed by an electronic circuit such as an amplifier, mixer or filter. The
frequency range is usually measured at the points where the level drops by 3dB relative to the maximum.
(See also Q)
A specific configuration of sounds or other parameters stored in memory and accessed manually or via
MIDI commands.
The frequency response of a loudspeaker system at the lower end of the spectrum. The physical size and
design of a loudspeaker cabinet and the bass driver (woofer) determine the low frequency extension (the
lowest frequency the speaker can reproduce at normal level) and the how quickly the signal level falls
below that frequency.
see Proximity Effect.
A special type of acoustic absorber which is optimised to absorb low frequency sound waves.
Also known as TT or Tiny Telephone Plugs. A professional form of miniature jack plug derived from the
telecommunications industry and widely used for balanced mic and line-level connections on professional
patch bays. (cf. B-Type Plug)
Software which is not fully tested and may include bugs.
A high-frequency signal used in analogue recording to improve the accuracy of the recorded signal and to
drive the erase head. Bias is generated by a bias oscillator.
A counting system based on only two states: 1s and 0s. It is ideal for electronic equipment where it can
be represented as high and low voltages, light on/off, N-S or S-N magnetic domains, etc.
Part of a computer operating system (basic input-output system) held on ROM rather than on disk. This
handles basic routines such as accessing the disk drive.
A contraction of Binary digit, which may either be 1 or 0.
The number of data bits replayed or transferred in a given period of time (normally one second). Normally
expressed in terms of kb/s (kilo bits per second) or Mb/s (mega bits per second). For example, the bit rate
of a standard CD is (2 channels x 16 bits per sample x 44.1 thousand samples per second) = 1411.2
kilobits/second. Popular MP3 file format bit rates range from 128kb/s to 320kb/s, while the Dolby Digital
5.1 surround soundtrack on a DVD-Video typically ranges between 384 and 448kb/s.

A synthesizer than can generate two different sounds simultaneously (see multi-timbral).
A stereo coincident microphone technique devices by Alan Blumlein in the early 1930s, employing a pair
of microphones with figure-eight polar patterns, mounted at 90 degrees to each other with the two
diaphragms vertically aligned.
A type of bayonet-locking, two-terminal connector used for professional video and digital audio
connections. (See AES3-id)
A mechanical means of supporting a microphone above a sound source. Many microphone stands are
supplied with a ‘boom arm’ affixed to the top of the stand’s main vertical mast. The term may also be
applied to larger, remotely controlled microphone supports used in film and TV studios, or even to the
handheld ‘fishpoles’ used by film and TV sound recordists.

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A single gain control which allows the range of frequencies passing through a filter to be either amplified
or attenuated. The centre position is usually the 'flat' or 'no effect' position.
See Isolation Room
The process of mixing two or more recorded tracks together and re-recording these onto another track.
A physical obstruction to sound waves, such as a wall, or a large solid object. When sound waves reach a
boundary they create a high pressure area at the surface.
A specialised microphone where the diaphragm is placed very close to a boundary (eg. wall, floor or
ceiling). In this position the direct and reflected sound adds constructively, giving a 6dB increase in
sensitivity. It also avoids the comb-filtering that can occur when a conventionally placed microphone
captures the direct sound along with strong first reflections from nearby boundaries. Also known as PZM
or Pressure Zone Microphone.
Beats Per Minute.
A device that converts breath pressure into MIDI controller data.
An electronic circuit designed to isolate the output of a source device from loading effects due to the input
impedance of destination devices.
A buffer is essentially a short term data storage facility used to accommodate variable data read or write
periods, temporarily storing data in sequence until it can be processed or transferred by or to some other
part
Slangof the
termsystem.
for a software fault or equipment design problem.
(Also sometimes referred to as a buss) An electrical signal path along which multiple signals may travel. A
typical audio mixer contains several (mix) busses which carry the stereo mix, subgroups, the PFL signal,
the aux sends, and so on. Power supplies are also fed along busses.
A collection of digital data comprising eight bits.
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 when making measurements of high-level sounds, such as when calibrating loudspeaker
reference levels. (See also A-Weighting and K-Weighting)

The physical construction which encloses and supports the loudspeaker drive units. Usually built of wood
or wood composites (although other materials are often used including metal alloys and mineral
composites). Cabinets can be ‘sealed’ or ‘vented’ in various ways, the precise design influencing the bass
and time-domain characteristics.
Any box-like construction will resonate at one or more frequencies. In the case of a loudspeaker, such
resonances are likely to be undesirable as they may obscure or interfere with the wanted sound from the
drive units. Cabinets are usually braced and damped internally to minimise resonances.
A passive, two-terminal electrical component which stores energy in the form of an electrostatic field. The
terminals are attached to conductive ‘plates’ which are separated by a non-conductive dielectric.
Capacitance is measured in Farads. If a voltage is applied across the terminals of a capacitor a static
electric field develops across the dielectric, with positive charge collecting on one plate and negative
charge on the other. Where the applied voltage is an alternating signal, a capacitor can be thought of as a
form of AC resistance that reduces with increasing signal frequency. The old-fashioned term is a
‘condensor’.
Also known as a 'condenser microphone'. This is a specific form of electrostatic microphone which
operates on the principle of measuring the change in electrical voltage across a capacitor. The capacitor is
formed from two metal electrodes, one fixed (the back-plate) and the other a thin conductive membrane
that flexes in response to sound pressure. (See also Back Electret, and RF Capacitor Microphone.)
An alternative term for a transducer which converts acoustic sound waves into an electrical signal.
(Also known as a Carbon Button Microphone). An obsolete form of microphone in which carbon granules
are contained between two metal contact plates, one of which acts as the diaphragm and moves in
response to sound waves. The microphone has to be biased with a DC voltage which causes a current to
pass from one metal contact plate, through the carbon granules, to the other metal contact plate. The
varying pressure exerted on the carbon granules by the moving diaphgram causes a varying resistance
and thus a varying current which is analogous to the sound waves. Carbon Button Microphones were
used in the very early days of sound recording and broadcasting, as well as in domestic telephones up
until the 1980s when electret capsules became more commonplace.

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A specific form of polar response of a unidirectional microphone or loudspeaker. It is an inverted heart-


shape which has very low sensitivity at the back (180 degrees), but only slightly reduced sensitivity
(typically between 3 and 6dB) at the sides (90/270 degrees).
A recordable type of Compact Disc that can only be recorded once - it cannot be erased and reused. The
CD-R’s technical characteristics are defined in the ‘Orange Book’
A device capable of recording data onto blank CD-R discs.
A path carrying for audio or data. In the context of a mixing console a channel is a single strip of controls
relating to one input. In the context of MIDI, Channel refers to one of 16 possible data channels over
which MIDI data may be sent. The organisation of data by channels means that up to 16 different MIDI
instruments or parts may be addressed using a single cable.
A term describing the process whereby a slave device attempts to synchronise itself with a master device.
In the context of a MIDI sequence, Chase may also involve chasing events looking back to earlier
positions in the song to see if there are any program change or other events that need to be acted upon.
A slang term for an Integrated Circuit or IC.
Three or more different musical notes played at the same time.
An effect created by doubling a signal and adding delay and pitch modulation, intended to make a single
source sound more like an ensemble.
A scale of pitches rising or falling in semitone steps.
An audible metronome pulse which assists musicians in playing in time.
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 quantiser’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 anharmonic 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.

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.
An exact duplicate. Often refers to digital copies of digital tapes.
A mic technique which involves placing a microphone very close to a sound source, normally with the
intention of maximising the wanted sound and minimising any unwanted sound from other nearby sound
sources or the room acoustics. In classic music circles the technique is more often known as 'Accent
Miking'.
Essentially an internet communications network (either a Wide Area Network [WAN] or a private network)
in which a data-centre performs a range of services such as data storage (cloud storage) or remote apps
and programs (cloud computing). The term comes from the way network engineers used to draw system
diagrams with a cloud symbol to simplify a very complex (and irrelevant) network of routers, switches,
drives and cables into something that just showed the relevant external connection points.

A means of arranging two or more directional microphone capsules such that they receive sound waves
from all directions at exactly the same time. The varying sensitivity to sound arriving from different
directions due to the directional polar patterns means that information about the directions of sound
sources is captured in the form of level differences between the capsule outputs. Specific forms of
coincident microphones include ‘XY’ and ‘MS’ configurations, as well as B-format and Ambisonic arrays.
Coincident arrays are entirely mono-compatible because there are no timing differences between
channels.
A distortion of the natural timbre or frequency response of sound, usually but not always unwanted.
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.
A measure of how well a balanced circuit rejects an interference signal that is common to both sides of
the balanced connection.

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Originally conceived as a recording format for dictation machines in the early 1960s, it became a
mainstream music release format in the form of the Musicassette. A plastic shell protected 3.81mm wide
(1/8-inch) recording tape which ran at 4.75cm/s. A stereo track was recorded in one direction, and the
tape could be turned over to play a second stereo track recorded in the opposite direction.
An encode-decode device typically employed to pass a wide dynamic range signal over a channel with a
lower dynamic range capability. The source signal is compressed in the encoder to reduce the dynamic
range, and subsequently expanded by the decoder to restore the original dynamics. The Dolby noise
reduction codecs are examples of companders.
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.
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 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.
A device which can be instructed (or programmed) to carry out arithmetic or logical operations. Although
mechanical 'analogue' computers do exist, most are now electronic and digital, and process digital data.
see Capacitor Microphone
A material that provides a low resistance path for electrical current.
A specific shape of drive unit diaphragm intended to push and pull the air to create acoustic sound
waves. Most bass drivers use cone-shaped diaphragms, where the electromagnetic motor of the drive unit
is connected to the point of the cone, and its outer diameter is supported by some form of flexible
membrane.
An alternative term for mixer (See also Desk).
A compound designed to increase the conductivity of electrical contacts such as plugs, sockets and edge
connectors. (cf. De-Oxidising Compound)
Type of MIDI message used to translate continuous parameter changes, such as from a pedal, wheel or
breath control device.
A variable voltage signal typically used to control the pitch of an oscillator or filter frequency in an
analogue synthesizer. Most analogue synthesizers follow a one volt per octave convention, though there
are exceptions. To use a pre-MIDI analogue synthesizer under MIDI control, a MIDI to CV converter is
required.
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.
A mathematical process whereby the characteristics of one function (or signal) are imposed upon a
second function (signal) , to produce a third function which combines both of their characteristics. (See
Convolution Reverb). This process is performed in the digital domain by multiplying each individual sample
value of a source signal with the impulse response of the wanted effect signal, so that the characteristics
of the latter are imposed on the former.

A computationally demanding, but very accurate method, of creating artificial reverberation derived from
real acoustic spaces with all their natural complexity. In essence, a reverberant space is measured to
obtain its unique impulse response. That impulse response is then convolved with a 'dry' (reverberant-
free) source signal to create an output signal which contains the original (dry) source signal with the
desired room's reverberation characteristics imposed upon it. The convolution process is performed in the
digital domain and involves multiplying each individual sample of the source signal with the impulse
response of the convolving signal. The same technique can be employed to impose the characteristics of
any other audio signal processing or signal-shaping device onto a source signal, such as the
characteristics of legacy equalisers, compressors, tape recorders and so forth.

A method used by software manufacturers to prevent unauthorised copying.


The number-crunching heart of a computer or other data processor.
Slang term relating to malfunction of computer program.

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A set of audio filters designed to restrict and control the range of input signal frequencies which are
passed to each loudspeaker drive unit. A typical two-way speaker will employ three filters: a high-pass
filter allowing only the higher frequencies to feed the tweeter, a low pass filter that allows only the lower
frequencies to feed the woofer, and a second high-pass filter that prevents subsonic signals from
damaging the woofer.
The frequency at which one driver ceases to produce most of the sound and a second driver takes over.
In the case of a two-way speaker the crossover frequency is usually between 1 and 3kHz.
The ability to copy or move sections of a recording to new locations.
The frequency above or below which attenuation begins in a filter circuit.
One complete vibration (from maximum peak, through the negative peak, and back to the maximum
again) of a sound source or its electrical equivalent. One cycle per second is expressed as 1 Hertz (Hz).
see Control Voltage
An arrangement of sharing a common data signal between multiple devices. A ‘daisy chain’ is created by
connecting the appropriate output (or through) port of one device to the input of the next. This
configuration is often used for connecting multiple MIDI instruments together: the MIDI Out of the master
device is connected to the MIDI In of the first slave, then the MIDI Thru of the first slave is connected to
the MIDI In of the second slave, and so on... A similar arrangement is often used to share a master word
clock sample synchronising signal between digital devices.

The control of a resonant device. In the context of reverberation, damping refers to the rate at which the
reverberant energy is absorbed by the various surfaces in the environment. In the context of a
loudspeaker it relates to the cabinet design and internal acoustic absorbers.
A form of audio-over-IP (layer 3) created by Australian company Audinate in 2006. DANTE is an
abbreviation of 'Digital Audio Network Through Ethernet'. The format provides low-latency multichannel
audio over standard ethernet intrastructures. it has been widely adopted in the broadcast, music studio,
and live sound sectors.
An abbreviation of Digital Audio Tape, but often used to refer to DAT recorders (more correctly known as
R-DAT because they use a rotating head similar to a video recorder). Digital recorders using fixed or
stationary heads (such as DCC) are known as S-DAT machines.
Information stored and used by a computer.
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.
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. When one signal is being compared to a standard reference level the
term is supplemented with a suffix letter representing the specific reference. 0dBu implies a reference
voltage of 0.775V rms, while 0dBV relates a reference voltage of 1.0V rms. The two most common
standard audio level references are +4dBu (1.223V rms) and -10dBV (0.316V rms). The actual level
difference between these is close to 12dB. The term dBm is also sometimes encountered, and this relates
to an amount of power rather than a voltage, specifically 1mW dissipated into 600 Ohms (which happens
to generate a voltage of 0.775V rms). When discussing acoustic sound levels, 0dB SPL (sound pressure
level) is the typical threshold of human hearing at 1kHz.

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. 24dB/octave (fourth order) is
the steepest normally found in analogue audio applications.
Direct Current. The form of electrical current supplied by batteries and the power supplies inside electrical
equipment. The current flows in one direction only.
Digitally Controlled Amplifier. The digital equivalent of a VCA often found in digital synthesisers and
mixing consoles.

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See VCA Group


A stationary-head digital recorder format developed by Philips, using a bespoke cassette medium similar
in size and format to Compact Cassettes. It used an MPEG data reduction system to reduce the amount
of data that needs to be stored.
A manufacturer of audio processing equipment, most notably compressors and tape noise reduction
systems. The DBX NR systems were commercial encode/decode analogue noise-reduction processors
intended for consumer and semi-pro tape recording. Different models varied in complexity, but essentially
DBX compressed the audio signals during recording and expanded them by an identical amount on
playback.
Digitally Controlled Oscillator. Used in digitally-controlled synthesizers.
An abbreviation of Digital Delay Line, used to create simple delay-based audio effects.
Disc Description Protocol. A data description format used for specifying the content of optical discs
including CD, and used almost universally now for the delivery of disc masters to duplication houses. A
DDP file contains four elements: the Audio image (.DAT); the DDP identifier (DDPID), the DDP Stream
Descriptor (DDPMS); and a subcode descriptor (PQDESCR). Often an extra text file is also included with
track titles and timing data. Many DAWs and audio editing programs can now create DDP files.

A system which restores the spectral balance to correct for pre-emphasis.


A device for reducing the effect of sibilance in vocal signals.
A substance formulated to remove oxides from electrical contacts. (cf. Contact Cleaner)
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.
A form of ‘spaced microphone’ arrangement in which three microphone capsules (usually, but not always,
with omnidirectional polar patterns) are placed in a large triangular array roughly two metres wide, with the
central microphone one metre further forward. Sounds approaching from different directions arrive at each
capsule at different times and with slightly different levels, and these timing and level differences are used
to convey the directional information in the recording. The timing differences between channels can result
in unwanted colouration if they are combined to produce a mono mix.

see dB
A device intended to prevent the transmission of physical vibration over a specific frequency range, such
as a rubber or foam block.
The process of rearranging the files on a hard disk so that all the files are as contiguous as possible, and
that the remaining free space is also contiguous.
The time between a sound or control signal being generated and it auditioned or taking effect, measured
in seconds. Often referred to as latency in the context of computer audio interfaces.
An alternative term for mixer (See also console).
One or more physical click-stops which can be felt when a rotary control is moved. Typically used to
identify the centre of a control such as a pan or EQ cut/boost knob, or to give the impression of preset
positions on a gain control.
An abbreviation for ‘Direct Instrument’ or ‘Direct Inject’ The two terms being used interchangeably. Used
when an electrical sound source (eg electric guitar, bass or keyboard) is connected directly into an audio
chain, rather than captured with a microphone in front of a amp/loudspeaker.
the movable membrane in a microphone capsule which responds mechanically to variations in the
pressure or pressure gradient of sound waves. The mechanical diaphragm vibrations are converted into
an electrical signal usually through electromagnetic or electrostatic techniques such as ribbon, moving
coil, capacitor or electret devices.
Direct Injection, or Direct Instrument Box. A device which accepts the signal input from a guitar, bass, or
keyboard and conditions it to conform to the requirements of a microphone signal at the output. The
output is a mic-level, balanced signal with a low source impedance, capable of driving long mic cables.
There is usually a facility to break the ground continuity between mic cable and source to avoid unwanted
ground loop noises. Both active and passive versions are available, the former requiring power from
internal batteries or phantom power via the mic cable. Active DI boxes generally have higher input
impedances than passive types and are generally considered to sound better.

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A means of representing information (eg audio or video signals) in the form of binary codes comprising
strings of 1s and 0s, or their electrical or physical equivalents. Digital audio circuitry uses discrete voltages
or currents to represent the audio signal at specific moments in time (samples). A properly engineered
digital system has infinite resolution, the same as an analogue system, but the audio bandwidth is
restricted by the sample rate, and the signal-noise ratio (or dynamic range) is restricted by the word-
length.
A digital processor that generates delay and echo effects.
A digital processor which simulates acoustic reverberation.
A consumer multi-pin connection format used for vintage microphones, some consumer audio equipment,
and MIDI cabling. Various pin configurations are available.
A form of audio compressor which uses a diode-bridge (sometimes known as a diode-ring) arrangement
as the variable gain-reducing element. The design was popular in the 1960s as it provided faster
responses than typical opto-compressors, and less distortion than many FET designs. However, noise can
be an issue as the audio signal has to be attenuated heavily before the diode-bridge, and considerable
(~40dB) gain added subsequently. The diodes also need to be closely matched to maintain low distortion.

A means of connecting two electrical circuits so that both AC and DC signals may be passed between
A system whereby low-level noise equivalent to one quantising level is combined with a digitised audio
signal in such a way as to perfectly linearise the digital system. Dither must be employed whenever the
wordlength is reduced, otherwise quantising distortion errors will manifest.
Used to describe vinyl discs, CDs and MiniDiscs.
An abbreviation of Diskette, but now used to describe computer floppy, hard and removable data storage
Direct Memory Access. Part of a computer operating system that allows peripheral devices to
communicate directly with the computer memory without going via the central processor or CPU.
A manufacturer of analogue and digital audio equipment in the fields of tape noise-reduction systems and
cinema and domestic surround-sound equipment. Dolby’s noise-reduction systems included types B, C
and S for domestic and semi-professional machines, and types A and SR for professional machines.
Recordings made using one of these systems must also be replayed via the same system. These systems
varied in complexity and effectiveness, but essentially they all employed the principlals of spectral noise-
masking in ever-more complex ways using multiband encode/decode processing to raise low-level signals
during recording, while reversing the process during playback.

Dolby’s surround sound systems started with an analogue 4:2:4 phase-matrix system with a very
elaborate active-steering decoder called ProLogic, before moving into the digital realm with Dolby Digital,
Dolby Digital Plus, Dolby True HD, and others.
Invented by B&O and licensed to Dolby. HX (often marketed as HX Pro) stands for 'Headroom eXtension'
and it is a record-processing system used on some analogue open-reel and cassette tape-recorders
(there is no complementary replay processing). Dolby HX Pro varies the amount of recording bias
depending on the strength of high-frequencies in the audio signal itself to avoid magnetic saturation and
over-biasing. This makes it possible to record with a higher overall fluxivity on the tape.

Disk Operating System. Part of the operating system of PC and PC compatible computers
A specific shape of drive unit diaphragm intended to push and pull the air to create acoustic sound
waves. Most tweeters use dome-shaped diaphragms which are driven around the circumference by the
drive unit’s motor system. ‘Soft-domes’ are made of a fabric - often silk - while metal domes are
constructed from a light metal like aluminium, or some form of metal alloy.
A method for removing or attenuating the noise component of a recording or transmission system, in
which the signal is pre-conditioned in a specific way which is reversed on playback. Most analogue noise-
reduction systems are of the double-ended type, such as the Dolby and DBX systems.
Also known as a Reussen screen. The signal-carrying wires in a microphone cable are protected from
external electrostatic and RF interference by a ‘screen’ which is a surrounding conductor connected to
earth or ground. The Reussen screen is a specific form of cable screen, comprising two overlapping and
counter-wound layers which are unlikely to ‘open up’ if the cable is bent, yet remain highly flexible
Digital Signal Processor. A powerful microchip used to process digital signals.

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A physical device designed to generate an acoustic sound wave in response to an electrical input signal.
Drive units can be designed to reproduce almost the full audio spectrum, but most are optimised to
reproduce a restricted portion, such as a bass unit (woofer) or high-frequency unit (tweeter). A range of
technologies are employed, with most being moving-coil units, but ribbon and electrostatic drive units also
exist, each with a different balance of advantages and disadvantages. Also known as a ‘driver’.

A piece of software that handles communications between the main program and a hardware peripheral,
such as a soundcard, printer or scanner. Also a term used to refer to a physical loudspeaker drive unit - eg
bass driver.brief loss of signal level. In analogue recording tape this is often caused by a defect in the oxide
A sudden
layer caused by damage (eg. creasing) or by a temporary clogging of the replay head, and typically affects
the high-frequencies most. The effect is less likely and less noticable with wider and faster tape formats.
A synthetic playing surface which produces electronic trigger signals in response to being hit with drum
See Isolation Room
A signal that has had no effects added.
The practice of transferring material from one medium to another, or of adding further material to an
existing recording (cf. Over-Dub).
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.
To transfer digital data from one device to another. A SysEx dump is a means of transmitting information
about a particular instrument or module over MIDI, and may be used to store sound patches, parameter
settings and so
'Digital Vinyl on. is a way of controlling and manipulating the playback of digital audio files in a
System'
computer by using conventional analogue turntables as the interface, thus maintaining a traditional 'disk-
jockey' style of hands-on 'turntablism' control, including scratching. The system is based upon using
special vinyl discs which carry standard timecode signals. The output of the vinyl replay chain is routed
into the computer via an audio interface in the usual way, and the software is then able to analyse the
timecode to determine changes in playback speed, direction, and position of the pickup on the vinyl disc,
and apply that timing and speed information to the reproduction of a digital audio file. Some latency is
inevitable in such a system, but it is usually very short.
A type of microphone that works on the electric generator principle, such as moving Coil and ribbon mics.
An acoustical sound waves impact the microphone diaphragm which then moves an electrical conductor
within a magnetic field to generate a current, the amplitude and polarity of which reflects the acoustic
signal.
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. (See AES17)
A way of describing the relative levels within a piece of music.
see SATA
The initial sound reflections from walls, floors and ceilings following a sound created in an acoustically
reflective environment.
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.
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)
An additional dedicated mixer input channel, usually with minimal facilities, designed to accommodate the
output from an effects unit. (cf. Aux Return)
see Back Electret
A system that modifies a signal prior to recording or transmission, and subsequently restores the signal
on playback or reception.
An audio processor designed to brighten audio material using techniques such as dynamic equalisation,
phase shifting and harmonic generation.
The way in which the amplitude of a sound signal varies over time.
An electronic circuit capable of generating a control signal which represents the envelope of the sound
you want to recreate. This may then be used to control the amplitude of an oscillator or other sound
source, though envelopes may also be used to control filter or modulation settings. The most common
example is the ADSR generator.
Erasable Programmable Read Only Memory. Similar to ROM, but the information on the chip can be
erased and replaced using special equipment. (See ROM)

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A device which allows the user to adjust the tonality of a sound source by boosting or attenuating a
specific range of frequencies. Equalisers are available in the form of shelf equalisers, parametric
equalisers and graphic equalisers or as a combination of these basic forms.
A means of describing the intrinsic electronic noise at the output of an amplifier in terms of an equivalent
input noise, taking into account the amplifier’s gain.
To remove recorded material from an analogue tape, or to remove digital data from any form of storage
A control protocol developed by Euphonix which operates at high-speed over an Ethernet connection. It is
used between control surfaces and DAW computers to convey information about the positions of faders,
knobs, and buttons and to carry display information.
A modular synthesizer format developed by Doepfer in the mid-1990s for its A ‑100 system, but which has
since become a standard format embraced by most modular synth manufacturers. It uses 3U high panels
to accommodate standard Eurocard-sizes electronics, with 3.5mm plugs for patching and a +/-12V dual-
rail power supply (sometimes with a separate 5V rail) connected via a ribbon cable.
In MIDI terms, an event is a single unit of MIDI data, such as a note being turned on or off, a piece of
controller information, a program change, and so on.
An audio processor that works by synthesizing new high frequency harmonics.
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.)
A synthesizer with no keyboard, often rack mountable or in some other compact format.
A sliding potentiometer control used in mixers and other processors.
A type of magnetic tape coating that uses iron oxide.
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.
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.
The accuracy or precision of a reproduced acoustic sound wave when compared to the electrical input
Describes the polar response of a microphone or loudspeaker that is equally sensitive both front and rear,
yet rejects sounds coming from the sides. Also called Bipolar.
A container for stored digital data that usually has a meaningful name. For example, a Standard MIDI File
is a specific type of file designed to allow sequence information to be interchanged between different
types of sequencer.
An electronic circuit designed to attenuate a specific range of frequencies. (See low-pass, high-pass and
band-pass.)
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.
A computer interface format based upon the IEEE 1394 standard and named FireWire by Apple
computers (Sony’s i.Link format is also the same interface). FireWire is a serial interface used for high
speed isochronous data transfer, including audio and video. FireWire 400 (IEEE 1394-1995 and IEEE
1394a-2000) or S400 interface transfers data at up to 400Mb/s and can operate over cables up to
4.5metres in length. The standard ‘alpha’ connector is available in four and six-connector versions, the
latter able to provide power (up to 25V and 8 watts). The FireWire 800 format (IEEE 1394b-2002) or S800
interface uses a 9-wire ‘beta’ connector and can convey data at up to 800Mb/s.

An effect which combines a modulated delay with the original signal, using feedback to create a dramatic,
sweeping sound.
A large capacity solid-state memory configured to work like a conventional hard drive. Used in digital
cameras and audio recorders in formats such as SD and CF2 cards, as well as in ‘pen drives’ or ‘USB
memory sticks’. Some computers are now available with solid state flash drives instead of normal internal
hard drives.
An obsolete computer disk format using a flexible magnetic medium encased in a protective plastic
A high-speed variation in replay speed causing rapid 'fluttering' pitch variations. See Wow and Flutter or
Scrape Flutter

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Short time-span sound echoes which can be created when sound waves bounce between opposite walls
in a small or moderately sized room. A shorter version of the ‘slapback’ echo whch can be experienced in
a larger hall when sound from a stage is reflected strongly from the rear wall.
A system for making one or more separate mixes audible to musicians while performing, recording and
overdubbing. Also known as a Cue mix. May be auditioned via headphones, IEMs or wedge monitors.
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.
A procedure required to ready a computer disk or digital tape for use. Formatting organises the medium
into a series of ‘electronic pigeon holes’ into which data can be stored. Different computers often use
different formatting systems.
The process by which the available space on a disk drive gets split up into small, sometimes unusable,
sections due to the storing and erasing of files.
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).
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)
Frequency Shift Keying. An obsolete method of recording a synchronisation control signal onto tape by
representing it as two alternating tones. (Also see timecode)
A 7-microphone array surround-sound, broadly equivalent to the stereo Decca Tree. Conceived by Akira
Fukada when he worked for the Japanese state broadcaster NHK. The front Left, Centre and Right
outputs are generated from a trio of mics arranged in a very similar way to a Decca Tree, with the left and
right outriggers spaced 2m apart, and the centre mic 1m forward. The Rear Left and Rear Right channels
come from mics spaced 2m apart placed and 2m behind the front outriggers. Instead of using omni mics
like a Decca Tree, all five mics are usually cardioids, aimed 60 degrees outwards to maximise channel
separation. These five mics are usually supplemented with an extra pair of omni outriggers placed midway
between the front and rear mics.
The lowest frequency component in a harmonically complex sound. (Also see harmonic and partial.)
Shorthand term for Effects.
The amount by which a circuit amplifies a signal, normally denoted in decibels.
The act of optimising 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.
Electrical isolation between two circuits. A transformer provides galvanic isolation because there is no
direct electrical connection between the primary and secondary windings; the audio signal is passed via
magnetic coupling. An opto-coupler also provides galvanic isolation, as the signal is passed via light
modulation.
A synthesiser control signal generated whenever a key is depressed on an electronic keyboard and used
to trigger envelope generators and other events that need to be synchronised to key action.
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.)
A universally agreed subset of the MIDI standard, created to enable manufacturers to build synthesizers,
synth modules and plug-in instruments that exhibit an agreed minimum degree of compatibility.
Describes an unwanted short term corruption of a signal, or the unexplained, short term malfunction of a
piece of equipment.
A universal SysEx command which activates the General MIDI mode on a GM instrument. The same
command also sets all controllers to their default values and switches off any notes still playing by means
of an All Notes Off message.
A flexible tube often used to support microphones or small lights. Sometimes also known as a 'Swan
An form of equaliser 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.
An alternative term for the electrical Earth or 0 Volts reference. In mains wiring, the ground cable is often
physically connected to the planet’s earth via a long conductive metal spike.

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A condition created when two or more devices are interconnected in such a way that a loop is created in
the ground circuit. This can result in audible hums or buzzes in analogue equipment, or unreliability and
audio glitches in digital equipment. Typically, a ground loop is created when two devices are connected
together using one or more screened audio cables, and both units are also plugged into the mains supply
with safety ground connections via the mains plug earth pins. The loop exists between one mains plug, to
the first device, through the audio cable screen to the second device, back to the mains supply via the
second mains plug, and round to the first device via the building’s power wiring. If the two mains socket
ground terminals happen to be at slightly different voltages (which is not unusual), and small current will
flow around the ground loop. Although not dangerous, this can result in audible hums or buzzes in poorly
designed equipment. Ground loops can often be prevented by ensuring that the connected audio
equipment is powered from a single mains socket or distribution board, thus minimising the loop. In
extreme cases it may be necessary to disconnect the screen connection at one end of some of the audio
cables, or to use audio isolating transformers in the signal paths. The mains plug earth connection must
NEVER be disconnected to try to resolve a ground loop problem as this will render the equipment
potentially LETHAL.
A mixed collection of signals within a mixer that are combined and routed through a separate fader to
provide overall control. In a multitrack mixer several groups are provided to feed the various recorder track
inputs.
Roland's own extension to the General MIDI protocol.
Graphical User Interface (pronounced ‘Gooey’). A software program designer’s way of creating an
intuitive visual operating environment controlled by a mouse-driven pointer or similar.
The conventional means of computer data storage. One or more metal disks (hard disks) hermetically
sealed in an enclosure with integral drive electronics and interfacing. The disks coated in a magnetic
material and spun at high speed (typically 7200rpm for audio applications). A series of movable arms
carrying miniature magnetic heads are arranged to move closely over the surface of the discs to record
(write) and replay (read) data.
High frequency components of a complex waveform, where the harmonic frequency is an integer multiple
of the fundamental.
The addition of harmonics that were not present in the original signal caused by non-linearities in an
electronic circuit or audio transducer.
1. The part of a tape machine or disk drive that reads and/or writes information magnetically to and from
the storage media.
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.
The standard measurement of frequency. 10Hz means ten complete cycles of a repeating waveform per
A filter which passes frequencies above its cut-off frequency, but attenuates lower frequencies.
The upper portion of the audible frequency spectrum, typically denoting frequencies above about 1kHz.
A misnomer, but used to refer to digital formats with long word-lengths and high sample rates, eg. 24/96
or 24/192. Audio resolution is infinite and identical to analogue systems in properly configured digital
systems. Word-length defines only the system’s signal-to-noise ratio (equivalent to tape width in analogue
systems) , while sample rate defines only the audio bandwidth (equivalent to tape speed in analogue
systems).
Random noise caused by random electrical fluctuations.
Normally used in the context of the USB computer data interface. A hub is a device used to expand a
single USB port into several, enabling the connection of multiple devices. Particularly useful where
multiple software program authorisation dongles must be connected to the computer.
Audio Signal contamination caused by the addition of low frequencies, usually related to the mains power
frequency.

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A condition whereby the state of a system is dependent on previous events or, in other words, the
system's output can lag behind the input. Most commonly found in audio in the behaviour of ferro-
magnetic materials such as in transformers and analogue tape heads, or in electronic circuits such a
'switch de-bouncing'. Another example is the way a drop-down box on a computer menu remains visible
for a short while after the mouse is moved.
The SI symbol for Hertz, the unit of frequency.
An abbreviation of Integrated Circuit, a collection of miniaturised transistors and other components on a
single silicon wafer, designed to perform a specific function.
In-Ear Monitor. A wirelessly-connected foldback monitoring system, often used by musicians on stage
with in-ear earpieces.
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.

An impulse respsonse is the time-domain equivalent of the much more familiar frequency (and phase)
responses in the frequency-domain. A very brief click (technically, a Dirac delta function) which
theoretically contains all frequencies at equal amplitude, is passed through the device under test. The
resulting output is the 'impulse response' of that device and uniquely describes its signal processing
behaviour. Impulse responses are very convenient for digital signal processing applications as the source
impulse is very similar to a single digital sample value.

A reactive component that presents an increasing impedance with frequency. (Also see Capacitor.)
Resetting a device to its 'start-up' state. Sometimes used to mean restoring a piece of equipment to its
factory default settings.
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)
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 input impedance of between 1500 and 2500 Ohms, and a line
input is usually between 10 and 50k Ohms.
A material that does not conduct electricity. (Also see conductor)
The nominal signal level generated by an electric instrument like a guitar, bass guitar or keyboard.
Typically around -25dBu. Instrument signals must be amplified to raise them to line-level.
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.
Something that happens occasionally and unpredictably, typically a fault condition.
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.
The input/output connections of a system.
Inches Per Second. Used to describe tape speed. Also the Institute of Professional Sound
Interrupt Request. Part of the operating system of a computer that allows a connected device to request
attention from the processor in order to transfer data to it or from it.

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A separate room or enclosure designed to provide acoustic isolation from external noise. Often used
alongside a studio's main live room to record vocals or drums, for example, without spill from other
instruments.
A device intended to prevent the transmission of physical vibrations over a specific frequency range, such
as a rubber or foam block. The term can also be applied to audio isolation transformers, used to provide
galvanic isolation between the source and destination, thus avoiding ground loops.
A type of alcohol commonly used for cleaning and de-greasing tape machine heads and guides.
A system of panel-mounted connectors used to bring inputs and outputs to a central point from where
they can be routed using plug-in patch cords. Also called a patchbay.
A commonly used audio connector, usually ¼ inch in diameter and with either two terminals (tip and
sleeve known as TS) or three (tip, ring, sleeve called TRS). The TS version can only carry unbalanced
mono signals, and is often used for electric instruments (guitars, keyboards, etc). The TRS version is used
for unbalanced stereo signals (eg for headphones) or balanced mono signals.
Specialised words associated with a specialist subject.
A hardware controller in the form of a rotary encoder which is often used to enable audio scrubbing in a
DAW or audio editing platform.
(lower-case k) The standard abbreviation for kilo, meaning a multiplier of 1000 (one thousand). Used as a
prefix to other values to indicate magnitude, eg. 1kHz = 1000Hz, 1kOhm = 1000 Ohms.
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.

A form of electrical filter which is designed to mimic the relative sensitivity of the human ear to different
frequencies in terms of pereceived 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)
The time delay experienced between a sound or control signal being generated and it being auditioned or
taking effect, measured in seconds.
The distance along the length of a cable over which the twisted core wires complete one complete turn.
Shorter lay lengths provide better rejection of electromagnetic interference, but make the cable less
flexible and more expensive.
Light Emitting Diode. A form of solid state lamp.
Liquid Crystal Display.
Low Frequency Oscillator, often found in synths or effects using modulation.
Least Significant Byte. If a piece of data has to be conveyed as two bytes, one byte represents high value
numbers and the other low value numbers, much in the same way as tens and units function in the
decimal system. The high value, or most significant part of the message is called the Most Significant Byte
or MSB.
see ADAT Lightpipe.
An automatic gain-control device used to restrict the dynamic range of an audio signal. A Limiter is a form
of compressor optimised to control brief, high level transients with a ratio greater than 10:1.
A device where the output is a direct multiple of the input with no unwanted distortions.
A nominal signal level which is around -10dBV for semi-pro equipment and +4dBu for professional
see LUFS
An electrical load is a circuit that draws power from another circuit or power supply. The term also
describes reading data into a computer system.
A function to allow the keyboard and sound generating section of a keyboard synthesizer to be used
independently of each other.
A type of electronic circuitry used for processing binary signals comprising two discrete voltage levels.
A number of separate cables bound together for neatness and convenience.

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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.
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.
A filter which passes frequencies below its cut-off frequency, but attenuates higher frequencies.
A device used to convert an electrical audio signal into an acoustic sound wave. An accurate loudspeaker
intended for critical sound auditioning purposes.
The perceived volume of an audio signal.
The practice of matching the perceived loudness of different material to a given target loudness value. To
accommodate varying peak levels, the medium requires an approporiate headroom margin. Loudness-
normalisation is now the default form for HDTV broadcasts, as well as most audio streaming services,
although the target loudness level currently varies between different platforms. Loudness-normalisation is
measured using the LUFS or LKFS scale. (See LUFS, Peak-Normalisation, Mastering, Loudness Wars).

The practice of trying to make each new commercial music release sound subjectively louder than any
previous release, on the misguided notion that louder is more exciting and results in more sales. A
relationship between the average loudness of 45rpm singles and sales was noticed in America from
jukebox plays, and that led to the first loudness war. However, the advent of the CD really ramped up the
situation, with music becoming ever-more dynamically compressed to squeeze the average level higher
and higher towards the 0dBFS peak level. This desctructive trend is, thankfully, now being slowly reversed
with the ubiquity of loudness normalisation adopted by most online audio streaming services and
broadcasters.
The lower portion of the audible frequency spectrum, typically denoting frequencies below about 1kHz
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. (See this article on the loudness metering
concept.)
Abbreviation for milli, meaning a multiplier of 1/1000 (one thousandth). Used as a prefix to other values to
indicate magnitude, eg. 1mA = 0.001A.
Abbreviation for mega, meaning a multiplier of 1,000,000 (one million). Used as a prefix to other values to
indicate magnitude, eg. 1MOhm = 1,000,000 Ohms or 1000k Ohms.
Multichannel Audio Digital Interface. Originally specified by the Audio engineering Society (AES) as
AES10 in 1991. This unidirectional digital audio interface shares the same core 24-bit audio and status
data format as AES3, but with different 'wrapping' to contain 56 or 64 synchronous channels at base
sample rates, or 28 channels at 96kHz. It can be conveyed over unbalanced coaxial cables, or via optical
fibres
Also called magnetic compensation (which is usually a more accurate description). A means of restricting
the radiation range of the stray magnetic field from a drive unit’s permanent magnet which might otherwise
interfere with the correct operation of moving-coil meters or CRT television monitors. While it is possible to
enclose a magnet in a soft-metal case to prevent a stray magnetic field this becomes very expensive for
large magnets, and so a more common approach is to affix additional small external magnets with
opposite polarities to cancel out the unwanted stray field.

A device which controls slave devices. Often used to refer to synchronised recorders, or digital clocking
Traditionally the sequencing of individual recordings to form a cohesive album of material, and to apply
corrective equalisation and dynamics processing to ensure a consistent sound character and to optimise
playback on the widest possible range of sound systems. Appropriate signal processing may also be
applied to make the mastered material suitable for its intended medium (such as controlling transient
peaks and dynamics and mono-ing the bass for vinyl records, etc).

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A facility found mostly in live-sound mixing consoles used to create separate mixes which can be sent to
many different output destinations. Essentially, a Matrix creates 'a mix of mixes' since derived from pre-
mixed output signals such as subgroups, auxes or main outputs. This is in contrast to the normal console
mixing facilities which are derived from input channel signals. A matrix can be any size, and is usually
described in terms of numbers of inputs and outputs, such as 2x2 or 8x8, or 8x6 etc with a level control at
each junction of the matrix so that the desired amount of any source can be sent to any output and it is
this feature that makes a Matrix more versatile than a conventional subgrouping system. Matrix sends
might be used to create appropriate feeds for disparate destinations like PA front fills, green-room
foldback, camera recorders, and so on.
The loudest sound pressure level that a device can generate or tolerate.
Megabyte. Nominally 1,000,000 (one million) bytes of data, but in fact, because computer memory works
in with binary, the actual value is 1,048,576 bytes (220).
A term describing the tuning mechanism of a guitar.
Modular Digital Multitrack. An obsolete term for hardware digital recorders that can be used in multiples to
provide a greater number of synchronized tracks than a single machine.
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.
A list of choices presented by a computer program or a device with a display window.
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).
The nominal signal level generated by a microphone. Typically around -50dBu. Mic level signals must be
amplified to raise them to line-level.
A device used to convert an acoustic sound wave into an electrical signal.
A specialised IC at the heart of a computer which performs calculations and other data manipulations,
following software instructions.
The middle portion of the audible frequency spectrum, typically denoting frequencies between about
300Hz and 3kHz.
Musical Instrument Digital Interface. A defined interface format that enables electronic musical
instruments and computers to communicate instructional data and synchronise 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 synchronisation
purposes.
A device that gives a visual readout of MIDI activity when connected between two pieces of MIDI
A type of controller message used to select alternate banks of MIDI Programs where access to more than
128 programs is required.
A term used to describe the physical interface by means of which the musician plays the MIDI synthesizer
or other sound generator. Examples of controllers are keyboards, drum pads, wind synths and so on.
Also known as MIDI Controllers or Controller Data, these messages convey positional information relating
to performance controls such as wheels, pedals, switches and other devices. This information can be
used to control functions such as vibrato depth, brightness, portamento, effects levels, and many other
parameters.
A standard file format for storing song data recorded on a MIDI sequencer in such a way as to allow it to
be read by other makes or model of MIDI sequencer.
A chart, usually found in MIDI product manuals, which provides information as to which MIDI features are
supported. Supported features are marked with a 0 while unsupported feature are marked with a X.
Additional information may be provided, such as the exact form of the Bank Change message.
The socket used to receive information from a master controller or from the MIDI Thru socket of a slave
A device or sequencer function that enables two or more streams of MIDI data to be combined.
A sound generating device with no integral keyboard.
A MIDI Sound Source capable of producing several different sounds at the same time and controlled on
different MIDI channels.
MIDI information can be interpreted by the receiving MIDI instrument in a number of ways, the most
common being polyphonically on a single MIDI channel (Poly-Omni Off mode). Omni mode enables a
MIDI Instrument to play all incoming data regardless of channel.

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Every key on a MIDI keyboard has its own note number ranging from 0 to 127, where 60 represents
middle C. Some systems use C3 as middle C while others use C4.
The MIDI message sent when note is played (key pressed).
The Message sent when key is released.
The MIDI connector used to send data from a master device to the MIDI In of a connected slave device.
The MIDI connections of a MIDI-compatible device. A Multiport, in the context of a MIDI Interface, is a
device with multiple MIDI output sockets, each capable of carrying data relating to a different set of 16
MIDI channels. Multiports are the only means of exceeding the limitations imposed by 16 MIDI channels.
A type of MIDI message used to change sound patches on a remote module or the effects patch on a
MIDI effects unit.
An alternative term for MIDI Thru box.
A description of the synchronisation systems available to MIDI users - MIDI Clock and MIDI Time Code
The socket on a slave unit used to feed the MIDI In socket of the next unit in line.
A device which splits the MIDI Out signal of a master instrument or sequencer to avoid daisy chaining.
Powered circuitry is used to 'buffer' the outputs so as to prevent problems when many pieces of
equipment are driven from a single MIDI output.
Made from natural or synthetic minerals in the form of threads or fibres tangled together to form a
moderately dense ‘blanket’ which permits but impedes air flow and is useful in the creation of sound
absorbers, often employed as a cheaper and more efficient alternative to polyurethane form.
The positions on the walls or ceiling where, if the surface was covered with an optical mirror, one or both
loudspeakers could be seen in the reflection. The mirror point is essentially any position on a boundary
where sound waves from a sound source usually a monitor loudspeaker will be reflected directly to the
listening position. This is therefore the ideal location to place an acoustic absorber to prevent audible
reflections.
A device used to combine multiple audio signals together, usually under the control of an operator using
faders to balance levels. Most mixers also incorporate facilities for equalisation, signal routing to multiple
outputs, and monitoring facilities. Large mixers are also known as ‘desks’ or ‘consoles’.
The characteristic distribution of resonant low frequency sound waves within a confined space such as a
A process of analysing 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 equaliser can be
analysed and its properties modelled by digital algorithms to allow its emulation within the digital domain.
See Room Modes
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.
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.
A single channel of audio.
One note at a time.
a synthesizer that can play only one note at a time (see also poly-synth and paraphonic)
The main circuit board within a computer into which all the other components plug or connect.
A dynamic microphone where the diaphragm supports a coil of wire which moves within a magnetic field.
When sound causes the diaphragm to vibrate a small electrical current is generated within the coil. The
same technology is used in reverse for a moving coil loudspeaker, in which a powerful current is passed
through the coil, causing the diaphragm (cone) to move in response.
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
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A format used for transmitting synchronisation instructions between electronic devices within the MIDI
An abbreviation of 'multiple output' (also known as a 'parallel-strip' in BBC parlance). Refers to a line-level
signal splitting or distribution facility typically found on patchbays in which three or more sockets are wired
together to allow an input signal to be shared with multiple destinations. As an entirely passive facility the
operation relies on a very low source impedance and high destination (bridging) impedances to minimise
the loss of signal level. Microphone 'mults' tend either to use transformers with multiple secondary
windings or active buffer or distribution circuitry.

The creation of several samples, each covering a limited musical range, the idea being to produce a more
natural range of sounds across the range of the instrument being sampled. For example, a piano may
need to be sampled every two or three semitones in order to sound convincing.
A synthesizer, sampler or module that can play several parts or different sounds at the same time, each
under the control of a different MIDI channel.
A recording device capable of recording several 'parallel' parts or tracks which may then be mixed or re-
recorded independently.
the physical angle between two microphones, used to specify various microphone array configurations
(eg. 90 degrees for a Blumlein pair, or 110 degrees for an ORTF array).
A means of arranging two or more directional microphone capsules such that they receive sound waves
from the directions or interest at slightly different times due to their physical spacing. Information about the
directions of sound sources is captured in the form of both level differences between the capsule outputs,
generated by aiming directional polar patterns in different directions, and the timing differences caused by
their physical spacing. Specific forms of near-coincident microphones include the ORTF and NOS
arrangements.
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.
A system for reducing analogue tape noise or for reducing the level of hiss present in a recording. (See
DBX and Dolby).
A system using spectrally-shaped dither to improve the perceived signal-to-noise performance of a digital
audio system.
An addition to the basic MIDI spec that allows Controllers 98 and 99 to be used to control non-standard
parameters relating to particular models of synthesizer. This is an alternative to using System Exclusive
data to achieve the same ends, though NRPNs tend to be used mainly by Yamaha and Roland
instruments.
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.
A socket is said to be normalised when it is wired such that the original signal path is maintained unless a
plug is inserted into the socket. The most common examples of normalised connectors are the insert
points on a mixing console.
A specific form of near-coincident microphone array devised by the Nederlandse Omroep Stichting
(NOS), the Dutch national broadcaster. The technique employs a pair of small-diaphragm cardioid
microphones mounted with a mutual angle of 90 degrees and spaced apart by 30cm. The theoretical
stereo recording angle is 81°.
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.

A slotted plastic or bone component at the headstock end of a guitar neck used to guide the strings over
the fingerboard, and to space the strings above the frets.
When a frequency or pitch is transposed up by one octave, its frequency is doubled.
A process carried out while a recording is not playing. For example, some computer-based processes
have to be carried out off-line as the computer isn't fast enough to carry out the process in real time. Also
used to refer to a remote-controlled machine which is not currently active.

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Directional microphones are inherently more sensitive to sound from one direction, and the direction of
greatest sensitivity is referred to as the principle axis. Sound sources placed on this axis are said to be
‘on-axis’, while sound sources elsewhere are said to be ‘off-axis’
The unit of electrical resistance.
A microphone or loudspeaker polar pattern with equal sensitivity in all directions (often abbreviated to
Omni). Also the MIDI mode where data on all channels is recognised.
A break in an electrical circuit that prevents current from flowing. (see Short Circuit)
A tape machine where the tape is wound on spools rather than housed within a cassette.
A high-resolution networked communication protocol for computers, synthesizers and other audio
The basic software that enables a computer to load and run other programs.
The concept of configuring a computer in such as way as to maximise its performance for certain tasks. In
the context of a machine being used as a DAW, optimisation might involve disabling sub-programs that
access the internet regularly or intermittently, such as email hosts, automatic program update checkers
and so on. It might also include the structure of the hard drive, or the separation of program data to a
system drive and audio data to a separate drive to minimise access times and maximise data throughputs.

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.
A specific form of near-coincident microphone array devised by the Office de Radiodiffusion Télévision
Française (ORTF) at Radio France, the French national broadcaster. The technique employs a pair of
small-diaphragm cardioid microphones mounted with a mutual angle of 110 degrees and spaced apart by
17cm. The theoretical stereo recording angle is 96 degrees.
An abbreviation for 'oscillator' or 'Open Sound Control'.
A circuit designed to generate a periodic electrical waveform.
see Polarity
The effective internal impedance (resistance which many change with signal frequency) of an electronic
device. In modern audio equipment the output impedance is normally very low. Microphones are normally
specified with an output impedance of 150 or 200 ohms, although some vintage designs might be as low
as 30 Ohms.
The nominal output voltage generated by a microphone for a known reference acoustic sound pressure
level. Output sensitivity is normally specified for a sound pressure level of one Pascal (94dB SPL), and
may range from about 0.5mV/Pa for a ribbon microphone, to 1.5mV/Pa for a moving coil, and up to 20 or
30mV/Pa for a capacitor microphone.
Recording new material to separate tracks while auditioning and playing in synchronism with previously
recorded material.
The intentional use of overloaded analogue circuitry as a musical effect.
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 Nyquest Theorum, 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 obvous and unpleasant in comparison to analogue
overloads.
a component of a complex sound which has a higher frequency than the fundamental frequency, but
which is not necessarily related by a simple integer multiple (cf. harmonics)
A resistive circuit for reducing signal level.
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.
A means of connecting two or more circuits together so that their inputs are connected together, and their
outputs are all connected together.

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A variable value that affects some aspect of a device's performance.


An equaliser with separate controls for frequency, bandwidth and cut/boost.
Often used to descibe a multi-oscillator mono-synth which can be configured to allow the oscillators to be
controlled independently from the keyboard, allowing two or more notes to be played simultaneously.
The combination of fundamental and overtones together are called particals. (cf. harmonic)
A circuit with no active elements.
A loudspeaker which requires an external power amplifier, the signal from which is passed to a passive
cross-over filter. This splits and filters the signal to feed the two (or more) drive units.
An alternative term for a Program, referring to a single programmed sound within a synthesizer that can
be called up using Program Change commands. MIDI effects units and samplers also have patches. (see
also Bank)of panel-mounted connectors used to bring inputs and outputs to a central point from where
A system
they can be routed using plug-in patch cords. Also called a Jackfield.
A short cable used with patch bays.
Peripheral Component Interconnect: an internal computer bus format used to integrating 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.
Pulse Code Modulation the technique used by most digital audio systems to encode audio as binary
The maximum instantaneous level of a signal.
The practice of removing all headroom to maximise the peak level of an audio signal on a particular
medium. So all peak-normalised material will have the same maximum peak value, but the perceived
loudness is likely to vary between different tracks. The technique became popular and commonplace with
the advent of digital formats such as CD, where the signal level was typically engineered to reach 0dBFS.
(See also Loudness-Normalisation, Mastering and Loudness Wars)

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

An audio connector developed by RCA and used extensively on hi-fi and semi-pro, unbalanced audio
equipment. Also used for the electrical form of S/PDIF digital signals, and occasionally for video signals.
The part of a guitar that converts the string vibrations to electrical signals. Also the stylus/cartridge
assembly used to replay vinyl records.
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)
The musical interpretation of an audio frequency.
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.
A device for changing the pitch of an audio signal without changing its duration.

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A self-contained software signal processor, such as an Equaliser or Compressor, which can be ‘inserted’
into the notional signal path of a DAW. Plug-ins are available in a myriad of different 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.

Consumer recorders, such as MP3 recorders, are often equipped with a microphone powering system
called ‘Plug-In Power’. This operates with a much lower voltage (typically 1.5V) and is not compatible with
phantom powered mics at all.
The directional characteristic of a microphone (omni, cardioid, figure-eight, etc).
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'.
The ability of an instrument to play two or more notes simultaneously. An instrument which can only play
one note at a time is described as monophonic.
The most common MIDI mode that allows and instrument to respond to multiple simultaneous notes
transmitted on a single MIDI channel.
A synthesizer that can play more than one note at a time (eg. eight or sixteen notes), each with an
independent signal chain of oscillators, filters, and envelope generators.
A device placed between a sound source and a microphone to trap wind blasts such as those created by
a vocalist’s plosives (Bs, Ps and so on) which would otherwise cause loud popping noises as the
microphone diaphragm is over- driven. Most are constructed from multiple layers of a fine wire or nylon
mesh, although more modern designs tend to use open-cell foam.
A connection for the input or output of data.
A gliding effect that allows a sound to change pitch at a gradual rate, rather than abruptly, when a new
key is pressed or MIDI note sent.
Work done to a recording.
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).
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.
A unit designed to convert mains electricity to the DC voltages necessary to power an electronic circuit or
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.
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.
Peak Programme 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)
Pulsed Per Quarter Note. Used in the context of MIDI Clock derived sync signals.
Process for adding Pause, Cue and other subcode information to a digital master tape in preparation for
CD manufacture.
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.
A system for applying high frequency boost to a sound before processing. When the corresponding de-
emphasis is applied any noise contribution from the processing is reduced.
A signal derived from the channel path of a mixer before the channel fader. A pret-fade aux send level is
unaffected by channel fader changes. Normally used for creating Foldback or Cue mixes.
An effects unit or synth patch that cannot be altered by the user.
An alternative term for Aftertouch.
The undesirable process that causes some magnetic information from a recorded analogue tape to
become imprinted onto an adjacent layer. This can produce low level pre or post echoes.

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A device designed to treat an audio signal by changing its dynamics or frequency content. Examples of
processors include compressors, gates and equalisers.
A MIDI message designed to change instrument or effects unit patches.
A relatively small recording studio facility, often with a combined recording space and control room.
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.
Similar to a square wave but non-symmetrical. Pulse waves sound brighter and thinner than square
waves, making them useful in the synthesis of reed instruments. The timbre changes according to the
mark/space ratio of the waveform.
A means of modulating the duty cycle (mark/space ratio) of a pulse wave. This changes the timbre of the
basic tone; LFO modulation of pulse width can be used to produce a pseudo-chorus effect.
The action of placing an already recorded track into record at the correct time during playback, so that the
existing material may be extended or replaced.
The action of switching a tape machine (or other recording device), out of record after executing a punch-
in. With most multitrack machines, both punching in and punching out can be accomplished without
stopping
A form ofthe tape.
audio compressor which uses Pulse Width Modulation to determine the energy in the audio
signal over time. In essence, the audio signal is chopped up at a very high rate and the width of the
resulting pulses is adjusted to control the average energy over time, and thus provide signal attenuation.
When done well, this is the fastest form of compressor with the lowest distortion artefacts.
A type of boundary layer microphone.
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.
Part of the process of digitising an analogue signal. Quantisation is the process of describing or
measuring the amplitude of the analogue signal captured in each sample, and is defined by the
wordlength used to describe the audio signal eg. 16 bits.
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.
A standard equipment sizing format allowing products to be mounted between vertical rails in
standardised equipment bays.
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.
A digital tape machine using a rotating head system and a tape cassette.
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.
A term used to imply a standard audio CD. The name comes from the fact that the original specifications
documents for the audio CD created by Sony and Philips had a red cover! Recordable CD-Rs are
described as 'orange book' for similar reasons.
The way in which sound waves bounce off surfaces.
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 .‘
Opposition to the flow of electrical current. Measured in Ohms.
The characteristic of a filter that allows it to selectively pass a narrow range of frequencies. See Q.
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.

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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.
An abbreviation for Radio Frequency.
Unwanted interference into an audio system from external RF signals.
An alternative form of capacitor microphone which uses the capacitive capsule as the tuning element of a
radio-frequency oscillator. Sound waves arriving at the capsule change its capacitance, which varies the
frequency of the RF oscillator to produce an FM signal. This is immediately demodulated by the
microphone's internal circuitry to produce the audio output. The advantage of this approach is that the
capsule works in a very low-impedance environment (as opposed to the very high-impedance
environment of a traditional DC-biased capacitor mic), making it immune to the effects of humidity which
can cause unwanted noise in conventional capacitor mics. This technology was invented by Sennheiser
and is used in their MKH range of microphones.
A dynamic microphone where the sound capturing element is a thin metal ribbon diaphragm suspended
within a magnetic field. When sound causes the ribbon to vibrate, a small electrical current is generated
within
A set the ribbon.
of requests/demands that an artist or band (or their management) ask of the hosting venue as
criteria for performing. A Technical Rider would typically specify the size and layout of staging, required
equipment for lighting (truss weight limits, power requirements, numer of follow-spots, lighting plots or
designs, etc) and sound (input channel counts, PA power, number of monitors, effects, DI boxes, backline
amps and instruments, quality of equipment etc) and possibly also the operating and rigging staff
provided. A Hospitality Rider is a list of requests to ensure the comfort of the artist(s), such as the number
of dressing rooms, private bathroom/shower, food and beverage requirements, a number of
complimentary (comp) tickets for guests, security arrangements, and so on. There may also be additional
riders covering other aspects such as a Merchandise Rider detailing the space and stands provided for
selling merchandise, the rates, exclusivity rights, and so forth.
A device that accepts and processes two input signals in a particular way. The output signal does not
contain any element of the original input signals but instead comprises new frequencies based on the sum
and difference of the input signals' frequency components. The best known application of Ring Modulation
is the creation of Dr Who’s Dalek voices, but it may also be used to create dramatic instrumental textures.
Depending on the relationships between the input signals, the results may either be musical or extremely
dissonant - for example, ring modulation can be used to create bell-like tones. (The term 'Ring' is used
because the original circuit which produced the effect used a ring of diodes.)

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.
The rate at which a filter or equaliser attenuates a signal once it has passed the turnover frequency.
An abbreviation for Read Only Memory. This is a permanent or non-volatile type of memory containing
data that can't be changed once programmed. Operating systems are often stored on ROM as the
memory remains intact when the power is removed.
Acoustic resonances within an enclosed space or room. These occur at specific frequencies where the
source sound is reflected from the room's boundaries to reinforce and/or cancel with itself to create
standing waves. This results in some areas in the room with very boomy or exaggerated pitches, and
others where the pitch may be almost completely absent. The resonant frequencies involved relate directly
to the sound wavelength and room dimensions, and is particularly prevalent at low frequencies.

A hardware controller comprising a knob or dial which can be rotated in either direction without end-stops.
A digital encoder of some kind attached to the shaft translates the movement into a digital code that can
indicate both direction and speed of rotation to the controlling software of a device.
A copy or clone of an original tape for use in case of loss or damage to the original.
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.
The number of times an A/D converter samples the incoming waveform each second.
Usually refers to a feature whereby random amplitude values are generated at regular intervals and then
used to control another function such as pitch or filter frequency. Sample and hold circuits were also used
in old analogue synthesizers to 'remember' the note being played after a key had been released.

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The acronym stands for 'Serial Advanced Technology Attachment' and is a computer interface employed
for connecting standard ATA hard drives to a computer motherboard. The SATA interface supersedes the
PATA (parallel ATA) interface which has been used since the 1980s. A variant of the SATA interface,
called eSATA (with the 'e' standing for 'external'), permits the connection of external hard drives. it uses a
slightly different connector but is otherwise a very similar interface.

So called because it resembles the teeth of a saw, this waveform contains both odd and even harmonics.
As analogue recording tape moves across the heads or other non-moving parts in the tape path it can
vibrate at a high frquency (typically above 100Hz) due to a rapid stick-slip action, and this causes a form
of intermodulation distortion. Often mechanical dampers and rollers are placed in the tape path to prevent
scrape flutter.
A term taken from the practice of editing analogue tape where the tape was manually dragged back and
forth across the replay head to locate the required edit point using an action similar to the cleaning action
of 'scrubbing'. The term is now routinely used in DAWS and audio editing software platforms where the
audio is played forwards or backwards at variable speeds, usually to locate an edit or cue point. A Jog
Wheel is often used as the hardware controller for scrubbing.

Pronounced SKUZZY, an abbreviation for Small Computer Systems Interface. An obsolete interfacing
system for using hard drives, scanners, CD-ROM drives and similar peripherals with a computer. Each
SCSI device has its own ID number and no two SCSI devices in the same chain must be set to the same
number. The last SCSI device in the chain should be terminated, either via an internal terminator or via a
plug-in terminator fitted to a free SCSI socket.

The original tape recording made during a recording session.


A device for recording and replaying MIDI data, usually in a multitrack format, allowing complex
compositions to be built up a part at a time.
a mechanical isolator intended to prevent the transfer of vibrations which may be transmitted through a
microphone stand from reaching a microphone where they would otherwise produce unwanted low
frequency sound.
A very low resistance path that allows electrical current to flow. The term is usually used to describe a
current path that exists through a fault condition. (See Open Circuit)
A high-frequency whistling or lisping sound that affects vocal recordings, due either to poor mic technique
or excessive HF equalisation.
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.
An electrical representation of an audio event.
The route taken by a signal from the input of a system to the output.
The ratio of nominal or maximum signal level to the residual noise floor, expressed in decibels.
The waveform of a pure sinusoidal tone with no harmonics.
A device for removing or attenuating the noise component of a recording or transmission system without
pre-conditioning the signal. Most digital noise-reduction systems are of the Single-ended type.
The term 'slate' comes from the silent film practice of writing the scene, take and shot numbers with chalk
on a slate and holding it up in front of the camera before the action starts, so that the film editor can
identify the material. A role now replaced by the 'clapper-board' which adds an audio synchronisation
marking facility as well. In an audio context, a slate is a verbal identification recorded just before each take
to help identify it subsequently. This is normally achieved by using a talkback microphone routed to the
main, group and/or direct outputs of a mixer. The console slate function often mixed a low frequency tone
in with the microphone signal to help make locating the start of each take much easier when fast-winding
the tape against the playback head. Each slate ident would be heard as a short, steady mid-frequency
tone.
A device under the control of a master device. Often used to refer to synchronised recorders, or digital
clocking devices.
see Switching Power Supply
The Society of Motion Picture and Television Engineers (www.smpte.org) an American standards body.
The term is also used to refer to a form of time code developed for the film industry but now extensively
used in music and recording. SMPTE is a real-time digital code describing hours, minutes, seconds and
film or video frames. Usually conveyed as an audible warble.

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Sample Multiplexing (see ADAT ).


A term used to describe a cable used to carry multiple individual audio signals, typically between a stage
and mixing console in live sound applications.
A large capacity solid-state memory configured to work like a conventional hard disk drive. Some
computers are now available with solid state flash drives instead of normal internal hard disk drives. Also
used in digital cameras and audio recorders in formats such as SD and CF2 cards, as well as in ‘pen
drives’ or ‘USB memory sticks’.
A dedicated interface to transfer audio signals in and out of a computer. A Sound Card can be installed
internally, or connected externally via USB2 or FireWire, and they are available in a wide range of formats,
accommodating multiple analogue or digital audio signals (or both) in and out, as well as MIDI data in and
out.
An early recording technique pioneered by Les Paul and others which was a simple form of overdubbing
to build up a mix of sources. Also, the world's best recording technology magazine (see
www.soundonsound.com).
The use of materials and construction techniques with the aim of preventing unwanted sound from
entering or leaving a room.
A means of arranging two or more microphone capsules such that they receive sound waves from
different directions at different times these timing differences being used to convey information about the
relatice directions of those sound sources. The technique is usually used with omnidirectional
microphones, although directional mics can also be employed. The best known form of spaced array is the
Decca Tree. Mono-compatibility is often reduced because the timing differences between channels often
results in comb-filtering colouration when the channels are summed to mono.

Sony/Philips Digital Interface. Pronounced either ‘S-peedif’ or ‘Spudif’. A stereo or dual-channel self-
clocking digital interfacing standard employed by Sony and Philips in consumer digital hi-fi products. The
S/PDIF signal is essentially identical in data format to the professional AES3 interface, and is available as
either an unbalanced electrical interface (using phono connectors and 75ohm coaxial cable), or as an
optical interface called TOSlink.
An accurate loudspeaker intended for critical sound auditioning purposes.
Unwanted sound picked up by microphones on one instrument from other nearby instruments.
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!
Song Position Pointer (MIDI).
A symmetrical rectangular waveform. Square waves contain a series of odd harmonics.
see Stereo Recording Angle
see Solid-state Drive
A standard file format that allows MIDI files to be transferred between different sequencers and MIDI file
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)
A connection box terminating a multicore cable (see Snake) which is usually placed on a stage for the
easy connection of individual microphone cables.
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.

A system for programming a sequencer in non-real time.


By convention, two channels of related audio which can create the impression of separate sound source
positions when auditioned on a pair of loudspeakers or headphones.
The angle over which sound sources can be captured by a microphone array. For a stereo array with a
stereo recording angle of 90 degrees, sound sources can be placed ±45 degree relative to the array’s
centre front axis, with a source at the extreme angle appearing fully left or right in the stereo image.

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A problem affecting some brands of analogue tape after a long time in storage. A breakdown of the
binder causes the oxide to shed, and the tape tends to adhere to the tape heads and guides when played.
A short term cure can be affected by baking the affected tape.
The practice of recording a time code signal onto one track of a multitrack tape machine to facilitate
subsequent synchronisation.
Frequencies below the range of typical monitor loudspeakers. Some define sub-bass as frequencies that
can be felt rather than heard.
Hidden data within the CD and DAT format that includes such information as the absolute time location,
number of tracks, total running time and so on.
See Group
The process of creating a new sound by filtering and shaping a raw, harmonically complex waveform.
A specific type of efficient loudspeaker system intended to reproduce only the lowest frequencies
(typically below 120Hz).
A sudden increase in mains voltage.
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 ±30 degrees and
straight ahead), with two behind (at ±120 degrees or thereabouts), supplemented with a separate
subwoofer.
Part of the ADSR envelope which determines the level to which the sound will settle if a key is held down.
Once the key is released, the sound decays at a rate set by the Release parameter. Also refers to a
guitar's ability to hold notes which decay very slowly.
See Gooseneck
The optimum position for a microphone, or for a listener relative to monitor loudspeakers.
(Also SMPS) A type of power supply that uses mains power to drive directly a high frequency oscillator so
that a smaller, lighter transformer may be used. These power supplies are commonly used because they
can be made to accept a wide range of mains supply voltages, and are thus universal.
A system for making two or more pieces of equipment run in synchronism with each other.
The creation of artificial sound.
An electronic musical instrument designed to create a wide range of sounds, both imitative and abstract.
(System Exclusive) A part of the MIDI standard that allows manufacturers to define their own specific
message formats, commonly used to dump and load a specific product’s patch data.
A system designed to enable voice communication between rooms.
The part of a tape machine that transfers magnetic energy to the tape during recording, or reads it during
playback.
The rate of the 'beat' of a piece of music measured in beats per minute.
A steady, fixed level tone recorded onto a multitrack recording, or passed over a signal connection to test
the signal path and act as a reference when matching levels.
Total Harmonic Distortion. A measure of the linearity of a device. The THD+N measurement includes the
noise contribution as well and is an indication of the quality of an audio product.
A MIDI connector which passes on the signal received at the MIDI in socket.
A bi-directional computer interface based on the PCI Express protocol, used for both data transfers and
to connect display monitors (it supports DVI, HDMI, and VGA monitors via adapters). Introduced by Apple
in 2011, Thunderbolt 1.0 supports bi-directional data transfers at 10Gbit/s, while Thunderbolt 2.0
(launched in 2013) operates at twice that speed. This means a Thunderbolt 2.0 interface (20Gbit/s) is five
times faster than USB3.0 and 25 times faster than FireWire 800. The physical Thunderbolt port is the
same as Apple's mini-DisplayPort connector, and can be used to integrate FireWire, USB and Ethernet
connections via appropriate adapters or hubs. Thunderbolt 3.0 uses the USB-C connector to carry up to
40Gbit/s, and has a 100W power transfer capability with appropriate cables.

The tonal 'colour' of a sound.


Referring to the tones that can be created by a synthesizer (see multi-timbral and bi-timbral)
see S/PDIF.

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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.
The process of recording individual tracks to a multichannel recorder. Tracking is also often discussed in
the context of MIDI guitar synthesizers or controllers where the MIDI output attempts to track the pitch of
the guitar strings.
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.

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 recognisable character of the sound source.
When the length of an electrical cable is shorter than about 10% of the wavelngth of the signal it conveys,
the voltage and current are effectively the same at all points along the cable. However, if the cable is
longer than 10% of the can be considered to propogate as electromagnetic waves along the cable, and
this condition is referred to as a 'transmission-line'. At 20Hz the electrical wavelength is well in excess of
2000 miles, and even at 20kHz it is over six miles, so there is no need to consider transmission line theory
in normal audio interconnections. However, it is very important in radio-frequency installations as the
relevant cable length is about 20cm for 100MHz signals, and just 20mm at 1GHz. A transmission-line can
be constructed in many different physical forms, such as spaced parallel wires or coaxial cables, but all
are generally of uniform cross-sectional area and have a defined 'characteristic impedance' per unit
length. To prevent the signal being reflected from the end of the cable it musty be terminated at both ends
in source and destination impedances which match the characeteristic impedance. The term is often also
used (usually erroneously) to describe a form of loudspeaker cabient design in which the lowest
frequencies are guided down an open-ended tube of considerable length lined with materials which allow
the lowest frquencies to pass but absorb the higher frequencies.

A subjective term used to describe audio quality where the high frequency detail is clear and individual
sounds are easy to identify and separate.
A form of modulation of the amplitude of a sound using an LFO. (cf. vibrato)
A device for converting one form of energy to another. Microphones and Loudspeakers are good
examples of transducer converts between mechanical and electrical energy.
To shift a musical signal by a fixed number of semitones.
A symmetrical triangular shaped wave containing odd harmonics only, but with a lower harmonic content
than the square wave.
A type of quarter-inch jack plug with three contacts (Tip, Ring and Sleeve), used either for stereo
unbalanced connections (such as on headphones) or mono balanced connections (such as for line-level
signals). Physically compatible in size with the TS (Tip, Sleeve) quarter-inch jack plug used for electric
guitars and other instruments.
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.
A metal bar within a guitar neck which is tensioned so as to counteract the tendency for the neck to bend
under the tension of the strings.
see Bantam Plug
see Valve
The colloquial term to describe a loudspeaker drive unit optimised for the reproduction of high
frequencies. (See Woofer).
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.
To play the same melody using two or more different instruments or voices.

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A condition where the output signal is the same amplitude as the input signal; the overall system gain is
then x1 or unity.
Universal Serial Bus. A computer interface standard introduced in 1996 to replace the previous standard
serial and parallel ports more commonly used. The USB1.1 interface operated at up to 12Mb/s, but this
was superseded in 2000 by USB2.0 which operates at up to 480Mb/s. Most USB interfaces can also
provide a 5V power supply to connected devices. USB3.0 was launched in 2008 and is claimed to operate
at rates up to 5Gb/s, but it is only now (2011) starting to appear on hardware. USB connectors can be
indentified by having a blue insert in the Type-A sockets and the letters SS (SuperSpeed). The latest
edition, USB3.1 (launched in 2013) offers transfer rates of 10Gb/s, approaching that of the first generation
of Thunderbolt interfaces. Like earlier USB interfaces, the USB3 format provides a 5V power supply, but
the current rating has been increased from USB2.0's 200mA (1.0 Watts) to 900mA (4.5 Watts), and
sockets designated as charging sockets can manage 1500mA (7.5 Watts).
is the latest (2014) incarnation of a Universal Serial Bus interface, carrying bidirectional data at speeds of
10Gb/s. USB-C is not compatible with previous generations (USB 1.1, 2.0, 3.0, 3.1) as it has a very
different 24-pin connector which is symmetrical and employed at both ends of a free cable, and all devices
have the same socket. Power is optionally available over USB-C conenctions at 5V with a current capacity
of either 1.5A (7.5W) or 3A (15W)

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.

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.
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.
See VCA. VCA compressors tend to be fast-acting (at least in comparison to opto-compressors), a wide
dynamic range, and low distortion.
Found in large mixing consoles. The fader levels of a number of separate channels assigned to the VCA
group can be adjusted together by the VCA Group fader but without mixing their signals together. Usually
referred to as a DCA Group in a digital console.
Computer display screen (See also Monitor).
The rate at which a key is depressed. This may be used to control loudness (to simulate the response of
instruments such as pianos) or other parameters on later synthesizers.
A signal processor that imposes a changing spectral filter on a sound based on the frequency
characteristics of a second sound. By taking the spectral content of a human voice and imposing it on a
musical instrument, talking instrument effects can be created.
See Isolation Room
The capacity of a synthesizer to play a single musical note. An instrument capable of playing 16
simultaneous notes is said to be a 16-voice instrument.
Pitch modulation using an LFO to modulate a VCO. (cf. Tremolo)
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)
A guitar effects device where a bandpass filter is varied in frequency by means of a pedal control.
Unit of electrical power.
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.
A graphic representation of the way in which a sound wave or electrical wave varies with time.
A colloquial way of denoting how many separate frequency bands are reproduced by a loudspeaker.
Most are two-way systems with a woofer and tweeter, but some are three way with a woofer, midrange
and tweeter.

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


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).
The precise timing of digital audio samples is critical to the correct operation of interconnected digital
audio equipment. The 'metronome' that governs sample timing is called the Word Clock (sometimes
conjoined to 'Wordclock', or abbreviated to 'Wclk'). However, word clock does more than merely beat time;
it also identifies the start and end of each digital word or sample, and which samples belong to the left or
right channels. Digital interfaces such as the AES3 and S/PDIF embody clock signals within the data
stream, but it is often necessary to convey a discrete word clock between equipment as a square wave
signal running at the sampling rate. Dedicated word clock inputs and outputs on digital equipment
generally use BNC connectors.
A cyclical variation in replay speed which affects the pitch of the recorded material. Wow is a low-speed
variation (nominally below 4Hz) which typically occurs once per revolution of the device and may be
caused by an off-centre hole on a vinyl record, or a sticking tape on a reel-reel machine. Flutter is a higher
speed variation (nomninally above 4Hz) and can often be perceived as a form of intermodulation
distortion.
The parameter of tape head alignment which relates to the rotation angle of the head in a vertical axis
which determines how the tape arrives and leaves the head gap.
To save data to a digital storage medium, such as a hard drive.
Yamaha's alternative to Roland's GS system for enhancing the General MIDI protocol so as to provide
additional banks of patches and further editing facilities.
A very robust and latching connector commonly used to carry balanced audio signals such as the outputs
from microphones or line level devices. An XLR is a type of connector developed by US manufacturer,
Cannon, and used widely in professional audio systems. The company’s original X-series connector was
improved with the addition of a latch (Cannon XL) and a more flexible rubber compound surrounding the
contacts to improve reliability (Cannon XLR). The connector format is now is available in numerous
configurations, from many different manufacturers, and with several different pin configurations. Standard
balanced audio interfaces - analogue and digital - use three-pin XLRs with the screen on pin 1, the ‘hot’
signal on pin 2 and the ‘cold’ signal on pin 3.
A specific way of mounting two directional microphone capsules such that they both receive sound waves
from any direction at exactly the same time. Information about the direction of a sound sources is captured
in the form of level differences between the two capsule outputs. Commonly, the two microphones in an
XY array are mounted with a mutual angle of 90 degrees, although other angles are sometimes used. The
two capsules will have the same polar pattern, the choice of which determines the stereo recording angle
(SRA). The XY configuration is entirely mono-compatible because there are no timing differences between
the two channels.
A form of adapter cable that passively splits the output of a source to feed two or more destinations. Y-
leads may also be used in unbalanced console insert points in which case a stereo jack plug at one end of
the lead is split into two monos at the other for separate send and return connections. A Y-lead must
never be used to combine signals.
The parameter of tape head alignment relating to vertical alignment head and whether it is leaning
forward or back relative to the tape path. (cf. Azimuth and Wrap)
The point at which a signal waveform crosses from being positive to negative or vice versa.
Audible steps that can occur when a parameter is being varied in a digital audio processor.

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