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A domestic and semi-professional form of quarter-inch (6.35mm) jack plug. Available with either two or three terminals and referred to as TS (tip-sleeve 2-terminal) or TRS (tip-ring-sleeve 3-terminal). Widely used for electric instruments in an unbalanced mono format (TS), for stereo headphones in an unbalanced stereo format (TRS), and for balanced mono line-level connections on semi-pro equipment (also TRS).
Although similar in overall length and diamter to the professional B‑Type Plug, the two formats are not compatible. The large angular tip of the A-type plug will damage B-type sockets, while B-type plugs will not mate correctly with A-type sockets.
A miniature version of the A-type plug is also available called the Mini-jack, with a 3.5mm diameter and in both TS and TRS formats. Widely used for domestic headphone and input connections on consumer portable equipment, as well as for CV and gate connections on modular synthesizers.
An electrical filter which used when taking audio measurements of equipment, and designed to mimic the relative insensitivity of the human ear to high and low 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 form of weighting filter 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.) (An explanatory video is available when subject link is clicked.)
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 due to the passing of sound waves.
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 (<1Hz).
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.
An open-celled expanded polyurethane or melamine foam that allows sound waves to enter and flow through the foam to dissipate their energy and thus prevent the sound waves from being reflected. The density and depth of the foam affects the frequency range over which it is effective as an acoustic absorber.
A general 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 acceptable reverberation times at all frequencies, and a neutral overall sound character.
The term typically describes an electronic circuit containing transistors, integrated circuits (ICs), tubes or other devices that require power to operate, and which are capable of amplification. Also used to describe loudspeaker monitoring systems which employ separate amplifiers to power each drive unit individually.
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. Now more commonly used as a shorthand term for the ADAT Lightpipe digital connection format.
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' and its conenctors are physically identical to that used for the TOSlink optical S/PDIF stereo interface found on many digital consumer hi-fi devices. However, while the light-fibre itself can be used interchangeably for either format, the S/PDIF and ADAT interfaces are not compatible in any other way.
A system for generating audio waveforms or sounds by combining basic waveforms or sampled sounds at different pitches, prior to further processing with filters and envelope shapers. The Hammond tonewheel organ was one of the first additive synthesizers, allowing harmonically complex waveforms to be created by combining tones of different pitches using 'harmonic drawbars'.
When creating artificial waveforms in a synthesizer, changes in the signal amplitude (or frequency) over time are controlled by an ‘envelope generator’ which typically has controls to adjust the Attack, Decay, Sustain and Release times, triggered by the pressing and subsequent release of a key on the keyboard.
Acronym for the Audio Engineering Society (aes.org), a professional body which develops and defines a wide range of Standards in audio engineering and technology, as well as running trade shows and conferences around the world, and publishing a peer-reviewed technical journal.
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 at base sample rates. A double-rate format with half the channel count is also available.
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, control and status data, with up to 24 bits per audio sample and supporting sample rates up to 384kHz.
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 10V phantom power supply conveys remote control and clocking information through an ingenious modulation technique.
An AES standard which defines the use and pin-outs of 25-pin D-sub connectors for eight-channel balanced analogue audio travelling in a single direction in or out of a device, and an alternative bi-directional eight-channel digital interfacing (four AES3 threes in each direction). It conforms fully with the established Tascam interface standard. (Extended information available when topic link clicked.)
A system used within mixing consoles to allow specific signals to be monitored at the level set by their fader, typically after the pan-pot for channel AFL. Aux sends, matrix outputs and so forth are generally monitored AFL rather than Pre Fade Listen (see PFL and Solo).
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. This resulting in unwanted anharmonic distortion component frequencies (not musically related) being added to the wanted signal.
Also known as a Phase-Rotator. An electronic filter circuit that doesn't change the amplitude of any signal passing through it, but which alters the phase of the signal at different frequencies in a non-linear way. Used as a core element of phaser effects pedals, and also in broadcast processors to help make asymmetrical audio signals symmetrical (which allows a slightly higher output level before clipping).
A means of convenying information on a carrier wave by varying its instantaneous amplitude. In an audio application, like AM radio, the wanted audio signal's amplitude is used to create corresponding deviations in the carrier wave's amplitude. A demodulator can detect these deviations and thus recover the base-band audio signal. This form of modulation contrasts with frequency modulation (FM) where the carrier wave's amplitude remains constant but its frequency is varied to convey the wanted signal.
A description of sound reflections in a confined space which add to the original direct 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 and the room a sonic character.
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.
This term describes a type of oscillating arrangement involving a non-linear relationship between the restorative force and the oscillation amplitude. As a result, the oscillating period is dependent on the amplitude of oscillation. This is not the case in a true harmonic oscillation.
Anharmonic distortion involves the introduction of frequencies not musically related to the source signal, usually through some form of modulation process, such as aliasing in digital equipment which is overloaded or has poor anti-alias filtering, or scrape flutter in analogue tape machines.