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A domestic and semi-professional 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.)
A device which converts an analogue audio signal into a digital representation. (Cf. D-A Converter.)
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.
AC Coupling (sometimes also known as DC-Blocking) is an electronic engineering arrangement that allows an audio (or any other alternating) signal to be passed through a connection while simultaneously preventing any DC bias or offset voltage on the source signal from getting through. In other words, AC coupling rejects any DC components within a signal, passing only the AC elements. The simplest form of AC coupling is a series capacitor in the signal line and, in effect, it forms a high-pass filter with a very low turnover frequency.
AC coupling is employed widely in audio circuitry to isolate the DC operating condition of one stage of circuitry from affecting the next, and to protect parts of an audio chain from the potentially damaging effects of DC voltages.
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, integrated circuits (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 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, then the receiving device will recognise a fault condition and switch off all notes. Not all MIDI devices support active sensing.
Acronym for Alesis Digital Audio Tape, referring to the company's popular range of digital eight-track tape machines released in the early 1990s.
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 48 kHz) 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 96 kHz) but carries only four channels, while S/MUX4 operates at quad rates (176.4 and 192 kHz) 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, Decay, Sustain 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.
Acronym for Audio Engineering Society, one of the industry's professional audio associations. (www.aes.org)
An AES standard which defines the serial MADI interface (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 converters.
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 (AES) and the European Broadcasting Union (EBU), 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 peak-peak. 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 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.
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 Pre Fade Listen (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 ('polyphonic aftertouch'), but rather detect the overall pressure by means of a sensing strip running beneath the keys. Aftertouch is typically 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 (eg. 13A).
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. (cf. Digital)
A system for synthesizing sounds by means of analogue circuitry, usually by filtering simple repeating waveforms.
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 a 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.
A device (or software) that allows a MIDI instrument to sequence around any notes currently being played. Most arpeggiators also allow the sound to be sequenced over several octaves, so that holding down a simple chord can result in an impressive repeating sequence of notes.