Audio Terminologies & File Formats That You Should Know

1 . Acoustic Terminologies

2 . Audio editing terminologies

3 . Hardware Terminologies

4 . General audio terminologies

5 . Audio file formats

Acoustic Terminologies:

Acoustic Foam — 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.

Acoustic Treatment — 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.

Bass Trap — A special type of acoustic absorber which is optimized to absorb low-frequency sound waves.

Boundary — 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.

Decoupler (also isolator) — A device intended to prevent the transmission of physical vibration over a specific frequency range, such as a rubber or foam block.

Early Reflections — The initial sound reflections from walls, floors, and ceilings following a sound created in an acoustically reflective environment.

Fidelity — The accuracy or precision of a reproduced acoustic sound wave when compared to the electrical input signal.

Flutter Echoes — 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 which can be experienced in a larger hall when sound from a stage is reflected strongly from the rear wall.

Isolation Room — 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.

Isolator (also decoupler) — 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.

Audio editing terminologies:

Backup — 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!

Chord — Three or more different musical notes played at the same time.

Chromatic — A scale of pitches rising or falling in semitone steps.

Clipboard: The clipboard is where sample data is saved when you cut or copy it from a data window. You can then paste, mix, or cross fade the sample data stored on the clipboard with another data window. This sample data can also be used by other Windows applications that support Sound data on the clipboard, such as Sound Recorder.

Clipping — When an audio signal is allowed to overload the system conveying it, clipping is said to have occurred and severe distortion results. The ‘clipping point’ is reached when the audio system can no longer accommodate the signal amplitude –either because an analogue signal voltage nears or exceeds the circuitry’s power supply voltage, or because a digital sample amplitude exceeds the 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 sine wave source signal becomes more like a square wave. In an analogue system clipping produces strong harmonic distortion artifacts at frequencies above the fundamental. In a digital system those high frequency harmonics cause aliasing which results in an harmonic distortion where the distortion artifacts 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.

Click Track — An audible metronome pulse which assists musicians in playing in time.

Clone — An exact duplicate. Often refers to digital copies of digital tapes.

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

Cross fade: Mixing two pieces of audio by fading one out as the other fades in:

Cross fade Loop: Sometimes a sample loop cannot be easily created from the given source material. In these instances, a cross fade can be applied to the beginning and end of the loop to aid in the smooth transition between the two. The Cross fade Loop function provides a method of creating sampling loops in material that is otherwise difficult to loop.

Cut and Paste Editing — The ability to copy or move sections of a recording to new locations.

DC Offset: DC offset occurs when hardware, such as a sound card, adds DC current to a recorded audio signal. This current results in a recorded waveform that is not centered around the baseline (-infinity). Glitches and other unexpected results can occur when sound effects are applied to files that contain DC offsets. Sound Forge software can compensate for this DC offset by adding a constant value to the samples in the sound file.

In the following example, the red line represents the baseline.

Destructive Editing: Destructive editing is the type of editing whereby all cuts, deletes, mixes and other processes are actually processed to the sound file. Any time you delete a section of a sound file in Sound Forge software, the sound file on disk is actually rewritten without the deleted section.

Drag and Drop: A quick way to perform certain operations using the mouse. To drag and drop, you click and hold a highlighted selection, drag it (hold the left mouse button down and move the mouse) and drop it (let go of the mouse button) at another position on the screen.

Dubbing — The practice of transferring material from one medium to another, or of adding further material to an existing recording (cf. Over-Dub).

Erase — To remove recorded material from an analogue tape, or to remove digital data from any form of storage media.

File — 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.

Frame Rate: Audio uses frame rates only for the purposes of synchronizing to video or other audio. To synchronize with audio, a rate of 30 non-drop is typically used. To synchronize with video, 30 drop is usually used.

Frequency Modulation (FM): Frequency Modulation (FM) is a process by which the frequency (pitch) of a sound is varied over time. Sub audio frequency modulation results in pitch-bending effects (vibrato). Frequency modulation within audio band frequencies (20 Hz – 20,000 Hz) creates many different side-band frequencies that drastically alter the timbre of the sound.

Frequency Spectrum: The frequency spectrum of a signal refers to its range of frequencies. In audio, the audible frequency range is between 20 Hz and 20,000 Hz. The frequency spectrum sometimes refers to the distribution of these frequencies. For example, bass-heavy sounds have a large frequency content in the low end (20 Hz – 200 Hz) of the spectrum.

Invert Data: Inverting sound data reverses the polarity of a waveform around its baseline. Inverting a waveform does not change the sound of a file; however, when you mix different sound files, phase cancellation can occur, producing a “hollow” sound. Inverting one of the files can prevent phase cancellation.

Loop — The process of defining a portion of audio within a DAW, and configuring the system to replay that portion repeatedly. Also, a circuit condition where the output is connected back to the input.

Marker: A marker is an anchored, accessible reference point in a file. Markers can be used for quick navigation.

Nondestructive Editing: This type of editing involves a pointer-based system of keeping track of edits. When you delete a section of audio in a nondestructive system, the audio on disk is not actually deleted. Instead, a set of pointers is established to tell the program to skip the deleted section during playback.

Normalize: Refers to raising the volume so that the highest level sample in the file reaches a user-defined level. Use normalization to make sure you are using all of the dynamic range available to you.

One-Shot: One-shots are RAM-based audio clips that are not designed to loop. Things such as cymbal crashes and sound bites could be considered one-shots. Longer files can be treated as one-shots if your computer has sufficient memory.

Zero-Crossing: A zero-crossing is the point where a fluctuating signal crosses the baseline. By making edits at zero-crossings with the same slope, the chance of creating glitches is minimized.

Zipper Noise: Zipper noise occurs when you apply a changing gain to a signal, such as when fading out. If the gain does not change in small enough increments, zipper noise can become very noticeable. Fades are accomplished using 64-bit arithmetic, thereby creating no audible zipper noise.

Hardware Terminologies:

A-Type Plug — 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-Weighting — 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)

AC — 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 pressure.

Active — Describes a circuit containing transistors, ICs, tubes and other devices that require power to operate, and which are capable of amplification.

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

A/D [A-D] Converter — A device which converts an analogue audio signal into a digital representation.

ADAT Lightpipe — 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.

AES3 — 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.

AES10 — 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.

AES11 — 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).

AES17 – And AES standard that defines a method of evaluating the dynamic range performance of A-D and D-A converters.

AES42 — 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.

AES59 — 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.

AFL — 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).

Amp/Amplifier — 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).

Audio Interface — 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.

Autolocator — 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.

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

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

Azimuth — the alignment of a tape head which references the head gap to the true vertical relative to the tape path. (cf. Wrap and Zenith).

B-Type Plug — 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)

Back Electret — 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.

Balanced Wiring — 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 — the signal is conveyed ‘differentially’ and the receiver detects the voltage difference between the two signal wires. Any interference instils the same voltage on each wire (common mode) because the impedance to ground is identical for each, and as there is therefore no voltage difference between the signal wires, the interference is ignored completely by the receiver.

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.

Bass Response — 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.

Bantam Plug — 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)

Blumlein Array — 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.

BNC — A type of bayonet-locking, two-terminal connector used for professional video and digital audio connections.

Boom — 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.

Boundary Layer Microphone — A specialized 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.

Buffer — An electronic circuit designed to isolate the output of a source device from loading effects due to the input impedance of destination devices.

Bus — (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.

C-Weighting — A form of electrical filter which is designed to mimic the relative sensitivity of the human ear to different frequencies at high sound pressure levels (notionally 100 Phons or about 87dBA SPL). Essentially, the filter rolls-off the low frequencies below about 20Hz and the highs above about 10kHz. This filtering is often used when making measurements of high-level sounds, such as when calibrating loudspeaker reference levels. (See also A-Weighting and K-Weighting)

Cabinet — 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.

Cabinet Resonance — 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.

Capacitor — 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 ‘condenser’.

Capacitor Microphone — 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.)

Capsule — An alternative term for a transducer which converts acoustic sound waves into an electrical signal.

Carbon Microphone — (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.

Cardioid — 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).

CD-R — 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’

CD-R Burner — A device capable of recording data onto blank CD-R discs.

Channel — 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.

Chip — A slang term for an Integrated Circuit or IC.

Clocking — The process of controlling the sample rate of one digital device with an external clock signal derived from another device. In a conventional digital system there must be only one master clock device, with everything else ‘clocked’ or ‘slaved’ from that master.

Close-Miking — 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’.

Coincident — 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.

Common Mode Rejection — A measure of how well a balanced circuit rejects an interference signal that is common to both sides of the balanced connection.

Compact Cassette — 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.

Compander — 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.

Conductor — A material that provides a low resistance path for electrical current.

Cone — 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.

Console — An alternative term for mixer.

Contact Cleaner — A compound designed to increase the conductivity of electrical contacts such as plugs, sockets and edge connectors. (cf. De-Oxidising Compound)

Control Voltage — 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.

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

Daisy Chain — 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.

DAT — 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.

DC — Direct Current. The form of electrical current supplied by batteries and the power supplies inside electrical equipment. The current flows in one direction only.

DCC — 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.

DBX — 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.

DCO — Digitally Controlled Oscillator. Used in digitally-controlled synthesizers.

DDL — An abbreviation of Digital Delay Line, used to create simple delay-based audio effects.

DDP — 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.

De-emphasis — A system which restores the spectral balance to correct for pre-e

De-Oxidising Compound — A substance formulated to remove oxides from electrical contacts. (cf. Contact Cleaner)

Decca Tree — 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.

Detent — 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.

DI — 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.

Diaphragm — 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.

DI Box — 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.

Digital (cf. Analogue) — 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.

DIN Connector — A consumer multi-pin connection format used for vintage microphones, some consumer audio equipment, and MIDI cabling. Various pin configurations are available.

Diode-Bridge Compressor — 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.

Disc — Used to describe vinyl discs, CDs and MiniDiscs.

Disk — An abbreviation of Diskette, but now used to describe computer floppy, hard and removable data storage disks.

Dome — 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.

Double-lapped Screen — 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

DSP — Digital Signal Processor. A powerful microchip used to process digital signals.

Drive unit — 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’.

Drum Pad — A synthetic playing surface which produces electronic trigger signals in response to being hit with drum sticks.

Dynamic Microphone — 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.

Envelope generator — 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.

E-PROM — Erasable Programmable Read Only Memory. Similar to ROM, but the information on the chip can be erased and replaced using special equipment.

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

EuCon — 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.

Expander Module — A synthesizer with no keyboard, often rack mountable or in some other compact format.

Fader — A sliding potentiometer control used in mixers and other processors.

Ferric — A type of magnetic tape coating that uses iron oxide.

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

Figure of Eight — 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.

FireWire — 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.

Flash Drive — 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.

Floppy Disk — An obsolete computer disk format using a flexible magnetic medium encased in a protective plastic sleeve.

Fukada Tree — 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.

Gain Staging — 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.

Galvanic Isolation — 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.

Gate (CV) — 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.

Gooseneck — A flexible tube often used to support microphones or small lights. Sometimes also known as a ‘Swan Neck’.

Ground — 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.

Ground Loop / Ground Loop Hum — 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.

Hard Disk Drive (cf. Solid-state Drive) — 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.

Head — The part of a tape machine or disk drive that reads and/or writes information magnetically to and from the storage media.

Hub — 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.

IC — An abbreviation of Integrated Circuit, a collection of miniaturised transistors and other components on a single silicon wafer, designed to perform a specific function.

IEM — In-Ear Monitor. A wirelessly-connected foldback monitoring system, often used by musicians on stage with in-ear earpieces.

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

Microphones have a very low impedance (150 Ohms or so) while microphone preamps provide an input impedance of 1,500 Ohms or more. Line inputs typically have an impedance of 10,000 Ohms and DI boxes may provide an input impedance of as much as 1,000,000 Ohms to suit the relatively high output impedance of typical guitar pickups.

Inductor — A reactive component that presents an increasing impedance with frequency.

Initialise — Resetting a device to its ‘start-up’ state. Sometimes used to mean restoring a piece of equipment to its factory default settings.

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

Input Impedance — The input impedance of an electrical network is the ‘load’ into which a power source delivers energy. In modern audio systems the input impedance is normally about ten times higher than the source impedance — so a typical microphone preamp has an input impedance of between 1500 and 2500 Ohms, and a line input is usually between 10 and 50k Ohms.

Insulator — A material that does not conduct electricity.

Interface — A device that acts as an intermediary to two or more other pieces of equipment. For example, a MIDI interface enables a computer to communicate with MIDI instruments and keyboards.

Intermittent — Something that happens occasionally and unpredictably, typically a fault condition.

Intermodulation Distortion — A form of non-linear distortion that introduces frequencies not present in and musically unrelated to the original signal. These are invariably based on the sum and difference products of the original frequencies.

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

IPS — Inches Per Second. Used to describe tape speed. Also the Institute of Professional Sound.

Isopropyl Alcohol — A type of alcohol commonly used for cleaning and de-greasing tape machine heads and guides.

Jackfield — 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.

Jack Plug — 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.

K-Weighting — A form of electrical filter which is designed to mimic the relative sensitivity of the human ear to different frequencies in terms of 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.

Lay Length — 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.

LED — Light Emitting Diode. A form of solid state lamp.

LCD — Liquid Crystal Display.

LFO — Low Frequency Oscillator, often found in synths or effects using modulation.

Logic — A type of electronic circuitry used for processing binary signals comprising two discrete voltage levels.

Loom — A number of separate cables bound together for neatness and convenience.

Low Frequency Oscillator (LFO) — An oscillator used as a modulation source, usually operating with frequencies below 20Hz. The most common LFO waveshape is the sine wave, though there is often a choice of sine, square, triangular and sawtooth waveforms.

Loudspeaker (also Monitor and Speaker) — A device used to convert an electrical audio signal into an acoustic sound wave. An accurate loudspeaker intended for critical sound auditioning purposes.

General audio terminologies:

Acoustic Signature: The acoustic signature of a system is data containing all of the sound characteristics of a system. This includes such things as reverb time, frequency response and other timbral qualities. Impulse files used by Acoustic Mirror can be thought of as acoustic signatures.

Additive Synthesis — 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.

ADSR — 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.

AES — Acronym for Audio Engineering Society, one of the industry’s professional audio associations. (www.aes.org)

A-Law: A-Law is a compounded compression algorithm for voice signals defined by the Geneva Recommendations (G.711). The G.711 recommendation defines A-Law as a method of encoding 16-bit PCM signals into a nonlinear 8-bit format. The algorithm is commonly used in United States telecommunications. A-Law is very similar to µ-Law; however, each uses a slightly different coder and decoder.

Aliasing: A type of distortion that occurs when digitally recording high frequencies with a low sample rate. For example, in a motion picture, when a car’s wheels appear to slowly spin backward while the car is quickly moving forward, you are seeing the effects of aliasing. Similarly, when you try to record a frequency greater than one half of the sampling rate (the Nyquist Frequency), instead of hearing a high pitch, you may hear a low-frequency rumble.

Ambience — 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.

Amp (Ampere) — Unit of electrical current (A).

Amplitude — The waveform signal level. It can refer to acoustic sound levels or electrical signal levels.

Amplitude Modulation: Amplitude Modulation (AM) is a process whereby the amplitude (loudness) of a sound is varied over time. When varied slowly, a tremolo effect occurs. If the frequency of modulation is high, many side frequencies are created that can strongly alter the timbre of a sound.

Analog: When discussing audio, this term refers to a method of reproducing a sound wave with voltage fluctuations that are analogous to the pressure fluctuations of the sound wave. This is different from digital recording in that these fluctuations are infinitely varying rather than discrete changes at sample time. See Quantization.

Analogue (cf. Digital) — 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.

Analogue Synthesis — A system for synthesizing sounds by means of analogue circuitry, usually by filtering simple repeating waveforms.

Arming — 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.

Arpeggiator — 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.

ASCII — American Standard Code for Information Interchange. An internationally

Audio Data Reduction — 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).

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

Balance — This word has several meanings in recording. It may refer to the relative levels of the left and right channels of a stereo recording (eg. Balance Control), or it may be used to describe the relative levels of the various instruments and voices within a mix (ie. Mix balance).

Bandwidth — The range 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.

Baseline: The baseline of a waveform is also referred to as the zero-amplitude axis or negative infinity.

Beats Per Minute (BPM): The tempo of a piece of music can be written as a number of beats in one minute. If the tempo is 60 BPM, a single beat will occur once every second.

Bias — 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.

Bit: The most elementary unit in digital systems. Its value can only be 1 or 0, corresponding to a voltage in an electronic circuit. Bits are used to represent values in the binary numbering system. As an example, the 8-bit binary number 10011010 represents the unsigned value of 154 in the decimal system. In digital sampling, a binary number is used to store individual sound levels, called samples.

Bit Depth: The number of bits used to represent a single sample. For example, 8- or 16-bit are common sample sizes. While 8-bit samples take up less memory (and hard disk space), they are inherently noisier than 16- or 24-bit samples.

Bit Rate — 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.

Bi-Timbral — A synthesizer than can generate two different sounds simultaneously

Bouncing — The process of mixing two or more recorded tracks together and re-recording these onto another track.

BPM — Beats Per Minute.

Buffer: Memory used as an intermediate repository in which data is temporarily held while waiting to be transferred between two locations. A buffer ensures that there is an uninterrupted flow of data between computers. Media players may need to rebuffer when there is network congestion.

Bus: A virtual pathway where signals from tracks and effects are mixed. A bus’s output is a physical audio device in the computer from which the signal will be heard.

Byte: Refers to a set of 8 bits. An 8-bit sample requires one byte of memory to store, while a 16-bit sample takes two bytes of memory to store.

Cut-off Frequency — The frequency above or below which attenuation begins in a filter circuit.

Cycle — 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).

Damping — 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.

DANTE — 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.

DAW — (Digital Audio Workstation): A term first used in the 1980s to describe early ‘tapeless’ recording/sampling machines like the Fairlight and Synclavier. Nowadays, DAW is more commonly used to describe Audio+MIDI ‘virtual studio’ software programs such as Cubase, Logic Pro, Digital Performer, Sonar and such-like. Essentially elaborate software running on a bespoke or generic computer platform which is designed to replicate the processes involved in recording, replaying, mixing and processing real or virtual audio signals. Many modern DAWs incorporate MIDI sequencing facilities as well as audio manipulation, a range of effects and sound generation.

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

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.

dB/Octave — A means of measuring the slope or steepness of a filter. The gentlest audio filter is typically 6dB/Octave (also called a first-order slope). Higher values indicate sharper filter slopes. 24dB/octave (fourth order) is the steepest normally found in analogue audio applications.

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

Dithering: Dithering is the practice of adding noise to a signal to mask quantization noise.

Dolby — 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 multiband encode/decode processing that raised low-level signals during recording, and reversed the process during playback. Dolby’s surround sound systems started with an analogue 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.

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

Dynamic Range: The difference between the maximum and minimum signal levels. It can refer to a musical performance (high-volume vs. low-volume signals) or to electrical equipment (peak level before distortion vs. noise floor).

Effect — A treatment applied to an audio signal in order to change or enhance it in some creative way. Effects often involve the use of delays, and include such treatments as reverb and echo.

Envelope — The way in which the amplitude of a sound signal varies over time.

Equivalent Input Noise — 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.

Fast Fourier Transform (FFT) Analysis: A Fourier Transform is the mathematical method used to convert a waveform from the Time Domain to the Frequency Domain.

Since the Fourier Transform is computationally intensive, it is common to use a technique called a Fast Fourier Transform (FFT) to perform spectral analysis. The FFT uses mathematical shortcuts to lower the processing time at the expense of putting limitations on the analysis size.

The analysis size, also referred to as the FFT size, indicates the number of samples from the sound signal used in the analysis and also determines the number of discrete frequency bands. When a high number of frequency bands are used, the bands have a smaller bandwidth, which allows for more accurate frequency readings.

Foldback — 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.

Formant — The frequency components or resonances of an instrument or voice sound that doesn’t change with the pitch of the note being played or sung. For example, the body resonance of an acoustic guitar remains constant, regardless of the note being played.

Frequency — The number of complete cycles of a repetitive waveform that occur in 1 second. A waveform which repeats once per second has a frequency of 1Hz (Hertz).

Frequency Response — The variation in amplitude relative to the signal frequency. A measurement of the frequency range that can be handled by a specific piece of electrical equipment or loudspeaker. (Also see Bandwidth)

FSK — Frequency Shift Keying. An obsolete method of recording a synchronisation control signal onto tape by representing it as two alternating tones.

Fundamental — The lowest frequency component in a harmonically complex sound. (Also see harmonic and partial.)

Gain — The amount by which a circuit amplifies a signal, normally denoted in decibels.

Glitch — Describes an unwanted short term corruption of a signal, or the unexplained, short term malfunction of a piece of equipment.

Group — A 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.

Harmonic — High frequency components of a complex waveform, where the harmonic frequency is an integer multiple of the fundamental.

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

Hertz (Hz) — The standard measurement of frequency. 10Hz means ten complete cycles of a repeating waveform per second.

Head-Related Transfer Function (HRTF): Sounds are perceived differently depending on the direction the sound comes from. This occurs because of the echoes bouncing from your shoulders and nose and the shape of your ears. A head-related transfer function contains the frequency and phase response information required to make a sound appear to originate from a certain direction in 3-dimensional space.

Hertz (Hz): The unit of measurement for frequency or cycles per second (CPS).

High Resolution — 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).

Hiss — Random noise caused by random electrical fluctuations.

Hum — Audio Signal contamination caused by the addition of low frequencies, usually related to the mains power frequency.

Hysteresis — 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.

Hz — The SI symbol for Hertz, the unit of frequency.

Inverse Telecine (IVTC): Telecine is the process of converting 24 fps (cinema) source to 30 fps video (television) by adding pulldown fields. Inverse telecine, then, is the process of converting 30 fps (television) video to 24 fps (cinema) by removing pulldown.

k — (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.

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

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

Load — 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.

Loudness — The perceived volume of an audio signal.

Low-range (low, lows) — The lower portion of the audible frequency spectrum, typically denoting frequencies below about 1kHz

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

Mid-Side recording: Mid-side (MS) recording is a microphone technique in which one mic is pointed directly towards the source to record the center (mid) channel, and the other mic is pointed 90 degrees away from the source to record the stereo image. For proper playback on most systems, MS recordings must be converted to your standard left/right (also called AB) track.

Mix: Mixing allows multiple sound files to be blended into one file at user-defined relative levels.

Multiple-Bit-Rate Encoding: Multiple-bit-rate encoding allows you to create a single file that contains streams for several bit rates. A multiple-bit-rate file can accommodate users with different Internet connection speeds, or these files can automatically change to a different bit rate to compensate for network congestion without interrupting playback.To take advantage of multiple-bit-rate encoding, you must publish your media files to a Windows Media server or a RealServerG2.

Nyquist Frequency: The Nyquist Frequency (or Nyquist Rate) is one half of the sample rate and represents the highest frequency that can be recorded using the sample rate without aliasing. For example, the Nyquist Frequency of 44,100 Hz is 22,050 Hz. Any frequencies higher than 22,050 Hz will produce aliasing distortion in the sample if no anti-aliasing filter is used while recording.

Punch-In: Punching-in during recording means automatically starting and stopping recording at user-specified times.

Root Mean Square (RMS): The Root Mean Square (RMS) of a sound is a measurement of the intensity of the sound over a period of time. The RMS level of a sound corresponds to the loudness perceived by a listener when measured over small intervals of time.

Sample: The word sample is used in many different (and often confusing) ways when talking about digital sound. Here are some of the different meanings:

A discrete point in time which a sound signal is divided into when digitizing. For example, an audio CD-ROM contains 44,100 samples per second. Each sample is really only a number that contains the amplitude value of a waveform measured over time.
A sound that has been recorded in a digital format; used by musicians who make short recordings of musical instruments to be used for composition and performance of music or sound effects. These recordings are called samples. In this Help system, we try to use sound file instead of sample whenever referring to a digital recording.
The act of recording sound digitally, i.e. to sample an instrument means to digitize and store it.

Sample Rate: The Sample Rate (also referred to as the Sampling Rate or Sampling Frequency) is the number of samples per second used to store a sound. High sample rates, such as 44,100 Hz provide higher fidelity than lower sample rates, such as 11,025 Hz. However, more storage space is required when using higher sample rates.

Sample Value: The Sample Value (also referred to as sample amplitude) is the number stored by a single sample.

In 32-bit audio, these values range from -2147483648 to 2147483647.
In 24-bit audio, they range from -8388608 to 8388607.
In 16-bit audio, they range from -32768 to 32767.
In 8-bit audio, they range from -128 to 127.

The maximum allowed sample value is often referred to as 100% or 0 dB.

Sampler: A sampler is a device that records sounds digitally. Although, in theory, your sound card is a sampler, the term usually refers to a device used to trigger and play back samples while changing the sample pitch.

Secure Digital Music Initiative (SDMI): The Secure Digital Music Initiative (SDMI) is a consortium of recording industry and technology companies organized to develop standards for the secure distribution of digital music. The SDMI specification will answer consumer demand for convenient accessibility to quality digital music, enable copyright protection for artists’ work, and enable technology and music companies to build successful businesses.

SCSI MIDI Device Interface (SMDI): SMDI is a standardized protocol for music equipment communication. Instead of using the slower standard MIDI serial protocol, it uses a SCSI bus for transferring information. Because of its speed, SMDI is often used for sample dumps.

Sign-Bit: Data that has positive and negative values and uses zero to represent silence. Unlike the signed format, twos complement is not used. Instead, negative values are represented by setting the highest bit of the binary number to one without complementing all other bits. This is a format option when opening and saving RAW sound files.

Signed: Data that has positive and negative twos complement values and uses zero to represent silence. This is a format option when opening and saving raw sound files.

Signal-to-Noise Ratio: The signal-to-noise ratio (SNR) is a measurement of the difference between a recorded signal and noise levels. A high SNR is always the goal.

The maximum signal-to-noise ratio of digital audio is determined by the number of bits per sample. In 16-bit audio, the signal to noise ratio is 96 dB, while in 8-bit audio its 48 dB. However, in practice this SNR is never achieved, especially when using low-end electronics.

Small Computer Systems Interface (SCSI): SCSI is a standard interface protocol for connecting devices to your computer. The SCSI bus can accept up to seven devices at a time including CD ROM drives, hard drives and samplers.

Society of Motion Picture and Television Engineers (SMPTE): SMPTE time code is used to synchronize time between devices. The time code is calculated in hours:minutes:second:frames, where frames are fractions of a second based on the frame rate. Frame rates for SMPTE time code are 24, 25, 29.97 and 30 frames per second.

Sound Card: The sound card is the audio interface between your computer and the outside world. It is responsible for converting analog signals to digital and vice-versa. There are many sound cards available on the market today, covering the spectrum of quality and price. Sound Forge software will work with any Windows-compatible sound card.

Streaming: A method of data transfer in which a file is played while it is downloading. Streaming technologies allow Internet users to receive data as a steady, continuous stream after a brief buffering period. Without streaming, users would have to download files completely before playback.

Telecine: The process of creating 30 fps video (television) from 24 fps film (cinema).

Tempo: Tempo is the rhythmic rate of a musical composition, usually specified in beats per minute (BPM).

µ-Law: µ-Law (mu-Law) is a companded compression algorithm for voice signals defined by the Geneva Recommendations (G.711). The G.711 recommendation defines µ-Law as a method of encoding 16-bit PCM signals into a nonlinear 8-bit format. The algorithm is commonly used in European and Asian telecommunications. µ-Law is very similar to A-Law, however, each uses a slightly different coder and decoder.