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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.
Delay
The time between a sound or control signal being generated and it auditioned or taking effect, measured in seconds. Often referred to as latency in the context of computer audio interfaces.
Digital
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. (cf. Analogue)
Digital Reverberator
A digital processor which simulates acoustic reverberation.
DMA
Direct Memory Access. Part of a computer operating system that allows peripheral devices to communicate directly with the computer memory without going via the central processor (CPU).
Dolby Noise-Reduction
A manufacturer of analogue and digital audio equipment in the fields of tape noise-reduction systems and cinema and domestic surround-sound equipment. Dolby’s noise-reduction systems included types B, C and S for domestic and semi-professional machines, and types A and SR for professional machines. Recordings made using one of these systems must also be replayed via the same system. These systems varied in complexity and effectiveness, but essentially they all employed the principlals of spectral noise-masking in ever-more complex ways using multiband encode/decode processing to raise low-level signals during recording, while reversing the process during playback.
Dynamic Microphone
A type of microphone that works on the electric generator principle, such as Moving Coil and Ribbon mics. An acoustical sound wave impacts 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.
Dynamic Range
The amplitude range, usually expressed in decibels (dB), between the loudest signal that can be handled by a piece of equipment and the level at which small signals disappear into the noise floor. (See AES17)
Equaliser
A device which allows the user to adjust the tonality of a sound source by boosting or attenuating specific ranges of frequencies. Equalisers are available in the form of shelf equalisers, parametric equalisers and graphic equalisers — or as a combination of these basic forms. (cf Filter)
Expander
A hardware or software plug-in device designed to increase the dynamic range, typically by reducing the volume of low level signals (below a set threshold), or to increase the volume of high level signals (above a threshold). (See Compressor.)
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 (wedge-shaped loudspeakers placed on the floor).
Hertz (Hz)
The standard measurement of frequency. 10Hz means ten complete cycles of a repeating waveform per second. (Previously defined as 'cycles per second or CPS).
High-Pass Filter (HPF)
A filter which passes frequencies above its cut-off frequency, but attenuates lower frequencies. Also known as a 'low-cut' filter.
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.
Impulse Response (IR)
An impulse respsonse (IR) is the time-domain equivalent of the much more familiar frequency (and phase) responses in the frequency-domain. A very brief click (technically, a Dirac delta function) which theoretically contains all frequencies at equal amplitude, is passed through the device under test. The resulting output is the 'impulse response' of that device and uniquely describes its signal processing behaviour. Impulse responses are very convenient for digital signal processing applications as the source impulse is very similar to a single digital sample value.
Loudness-Normalisation
The practice of matching the perceived loudness of different material to a given target loudness value. To accommodate varying peak levels, the medium requires an approporiate headroom margin. Loudness-normalisation is now the default form for HDTV broadcasts, as well as most audio streaming services, although the target loudness level currently varies between different platforms. Loudness-normalisation is measured using the LUFS or LKFS scale. (See LUFS, Peak-Normalisation, Mastering, Loudness Wars).
MADI
Multichannel Audio Digital Interface. Originally specified by the Audio Engineering Society (AES) as AES10 in 1991. This unidirectional digital audio interface shares the same core 24-bit audio and status data format as AES3, but with different 'wrapping' to contain 56 or 64 synchronous channels at base sample rates, or 28/32 channels at 96kHz. It can be conveyed over unbalanced coaxial cables, or via optical fibres.