Glossary

​ See ATA

A film soundtrack format used for overseas markets. Called a completely filled mix, the four-track stereo M&E mix is ready for the addition of dubbed languages. The M&E tracks should include background sound effects and room tone for every scene, i.e., all sound except dialog.

A digital, discrete six channel mix of Left/Center/Right/Left surrond/Right Surround/ Subwoofer mix. 5.1 is not a specific surround format tied to any particular company or codec. However, all the hardware is the same for any 5.1-based system except for the codec. It is planned that CDs, laserdiscs, and DVDs will have an ID flag to let the decoder know which codec was used, enabling decoders to recognize all incoming bitstreams and automatically switch modes and process the incoming signal appropriately. It is a listening platform and hardware concept for a surround loudspeaker system. See DTS, Dolby Digital, HDTV, CDS, LFE.

The left-hand part of a stereo signal. Also known as L.

See Voice of the Theater.

Dolby laboratories' low-bit-rate codec system used in its Dolby Digital format film, in both broadcast and consumer video formats.

See Voice of the Theater.

See Voice of the Theater.

A comparison between two recordings of the same material; pre- and post-equalization, or pre- and post-effects, or any other comparison between two similar audio devices.

The part of the motion picture reproduction system in a theater that contains the sound transducer (such as an optical analog track reader or digital sound format decoder), preamp, noise reduction, and matrix decoding, where applicable. The A-chain equipment decodes the sound in preparation for the B-chain and loudspeakers.

See serial(2).

Akai Digital Audio Multitrack. A format developed by Akai in 1987 for recording twelve tracks of digital audio data on a standard Video-8 cassette and which allows the synchronization of multiple decks for 24- or 36-track recording. The tape runs at four times the normal Video-8 speed and gives about 15 minutes of recording time at 44.1kHz.

Always In Record. The practice in a recording session to record virtually everything on the off-chance that something which was not formally recorded as a take will be useful.

Film footage used to introduce or provide backup material for a live video broadcast.

 The primary dialog track cut by the picture editor. The B-track and subsequent tracks would be used for overdubs.

See Dolby noise reduction.

An equalization curve first applied to sound level meters in an attempt to make their measurements correspond better to the perceived loudness of sounds, decreasing the sensitivity of the meter to frequencies below 1kHz. An important note is that the bottom octave (32Hz) is attenuated by almost 40dB; the second octave (63Hz) by 26dB, and the third octave (125Hz) by 16dB. See B-weighting, C-weighting, equal loudness curves, SPL.

See concert pitch.

Advanced Audio Coding. A flexible streaming format that supports multichannel audio including subwoofer and embedded data channels, using a variety of sample rates up to 96kHz. AAC is being developed as a successor to MPEG-2.

A binaural microphone developed by Head Acoustics.

Advanced Authoring Format. A cross platform interchange format used in the creation, editing, and distribution of media content in an all-digital environment.

(Avid Audio eXtension) An audio plug-in format created by Avid, used exclusively by Pro Tools and Pro Tools HD. See AAX DSP, AAX Native

(Avid Audio eXtension) A digital signal processing audio plug-in format created by Avid, used exclusively by Pro Tools HD. See AAX, AAX Native

(Avid Audio eXtension) An audio plug-in format created by Avid, used exclusively by Pro Tools and Pro Tools HD See DSP, AAX

In the US, this means recording with a spaced pair. In Europe, this means recording with a coincident pair.

Term for a 23-minute or 2,050’ maximum reel of film specially made for theater screening. The AB-reel may originally have been made from two 1,000’ edit reels; "Projection reel 1AB" would have been originally been reel #1 and reel #2 during editing and mixing. (In the event that the total footage of the first three editing/mixing reels added up to less than 2,050’, there may be a projection reel "1ABC," but this is rare.) It is becoming more commonplace to edit films in AB reel format as the magnetic film units are gradually replaced with DAWs. AB-reels are also known as "big reels" or "2,000-foot reels." AB-reels are not the same as A/B-rolls, in which the camera negative is checkerboarded into two strands, allowing for simple optical effects such as fades and dissolves to be made when making original-negative prints (see EK Negative) called interpositives. This latter process is not limited to two (A, B) rolls, but can involve as many rolls of film as desired, e.g., a camera negative cut in four strands would have a "D-roll."

Absolute time. Timecode which is the actual running/recording time in HH:MM:SS, where 00:00:00 is the head of the tape. For example, DATs use ABS timecode. See also feet/frames.

The ability of a material to absorb, rather than reflect, sound waves. A higher absorption coefficient means better acoustical damping. See bass trap, boundary effect, standing wave, Sabins.

Alternating current. The current flows in both directions.

A form of ADPCM(Adaptive Delta (Differential) Pulse Code Modulation). It was first used in 1985 for digital radio (sound-only) applications and since adopted for other DBS (direct broadcast satellite) services, including soundtrack-with-video, satellite communication networks, and digital cable radio systems. AC-1 has a data rate between 220 kbps and 325 kbps.

A transform encoding/ decoding scheme for audio compression developed by Dolby labs that uses 256-band transform coding at a data rate of 128 kbps or 192 kbps on two channels. Used in the Dolby Fax System and also DP5xx encoding.

A multichannel, digital, split-band, perceptual coding scheme developed by Dolby Labs. It produces a 5.1 channel format, using lossy compression. Designed to be the matrixing format for DVD and surround-sound with HDTV broadcasts. Versatile, in that parameters such as bit-rate and number of channels, it can be tailored to particular applications, unique in that AC-3’s data bits are distributed dynamically among the filter bands as needed by the particular frequency spectrum or dynamic nature of the program. Data rates vary from 32 kbps for a single mono channel to as high as 640 kbps for 5.1 format. See Dolby Digital.

See bias.

Coupling between electronic circuits that passes only time-varying signals (i.e., alternating current), not direct current.

A newly developed codec based on a soft data compression ratio of between 2:1 and 3:1. Used in the Dolby Digital Dubber, it is designed specifically to record eight tracks of 20-bit material on removable media, including Iomega Jaz and MO drives. AC-M is said to be optimized for multiple record/replay generations. Initial tests have reported as many as 14 codec processes being possible with no audio degradation.

Active Combining Amplifier. See combining amplifier.

See optical track.

The name of the standard mono optical track that has existed since the beginning of sound on film. Standards were codified in 1938, although the standard has changed somewhat through the years. The standard specifies a flat response throughout the range of 100Hz-1.6 kHz and is down 7dB at 40 Hz, 10dB at 5 kHz, and 18 dB at 8 kHz. Also called an N-Curve. See also X-Curve.

The visual countdown that precedes the first program frame of a motion picture. Symbols and numbers on the academy leader are used for aligning the various film reels and the optical track for composite printing, for aligning the workprint and edited soundtracks for mixing, and for timing the change-over from one reel of film to another during projection. Academy leader contains one number per foot following the Picture Start, with 11, 10, etc., leader to three. (As projected, these numbers appear upside-down.) Named after the Academy of Motion Picture Arts and Sciences, which sets all film format standards. See also leader, SMPTE Universal leader, plastic leader, fill leader, LFOP.

Specifically, the Samuel Goldwyn Theater at the Academy of Motion Pictures Arts and Sciences on Wilshire Boulevard in Los Angeles, considered the best-sounding theater in the world. Academy members screen films at the Academy Theater prior to voting on them for the Oscar awards.

An indication that the tempo of a piece of music should gradually be increased.

 A recording technique using a midfield mike to add more room presence. Accent pickup should sound natural and not overly present.

The usable working area in front of a microphone is defined by the polar pattern and is called the acceptance angle.

In a musical scale, the accidentals are the extra sharp and flat notes that are not part of the diatonic series. For example, in the key of C on the piano, the accidentals are the black keys.

The British manufacturer of edgecoding machines.

See baffle.

A partition placed between two sources of sound, or between a sound source and a microphone, to prevent sound from passing through. The baffle, or screen, may be made of any material with a high absorbtion coefficient. Most baffles are designed as movable partitions, and are used to isolate individual instruments in recording studios.

A squealing sound when the output of an audio circuit is fed back (in phase) into the circuit’s input. See feedback.

The power carried by sound waves per unit area in a direction perpendicular to that area. See sound pressure level.

(1) A type of design for the housing of highly directional microphones that enhances the rejection of off-axis sources. Two or more concentric tubes in front of (and sometimes around) the capsule create a compact series of folded pathways through which all sounds approach the diaphragm. Those arriving on-axis reach the capsule via these paths in phase coherence. Off-axis sounds, due to the different lengths of the passways, reach the diaphragm and are partially or fully removed due to phase cancellation. (2) A type of speaker enclosure in which sound waves emanating from the rear of the woofer cone travel through a long, folded interior path before coupling with the outside. This extends bass response considerably.

A device placed in front of a high-frequency speaker that disperses or directs the sound in the desired pattern. Normally used to increase the angle of dispersion, either horizontally, vertically, or both.

A loudspeaker designed for, or used in, a sealed enclosure. Typically, a low-frequency loudspeaker baffle where most of the damping of the cone is the result of the elasticity of the air in the sealed cabinet.

The science or study of sound and its interaction with the human hearing mechanism.

(1) An audio device that requires a power source such as from an AC line or battery, as opposed to passive. Sometimes amplifying components such as transistors or ICs are called active circuit elements. (2) See MIDI patchbay.

 See crossover network.

An equalizer that employs active components such as transistors or ICs in its processing circuits. A pre-amplifying circuit generally follows each stage of actual equalization, boosting the signal level to restore unity gain. See also equalizer.

A type of loudspeaker that has amplification circuitry built-in. In addition, a true active monitor system utilizes active equalization and active crossovers to precisely contour the system sound. If there is only one amplifier driving all transducers, and/or there is no active equalization or crossover circuitry, the terms powered speaker or powered monitor are preferred.

A MIDI system message that carries no note data or control instructions, but simply indicates to a receiving device that the MIDI line is in working order.

A slow or leisurely tempo: 66-76 bpm.

Alesis Digital Audio Tape. A second-generation (1992) MDM. An 8-track, S-VHS-based digital audio recorder. Like the Tascam DA-88, ADAT systems record digital audio on consumer videocassette formats and provide for interlocking up to 16 8-track, rack-mount recorders in sample-accurate (48kHz) sync for up to 128-track recording. ADAT is a 16-bit format, currently supported as well by Panasonic, using T-180 S-VHS tape. ADAT-II is a newly proposed 20-bit S-VHS format used by newer Alesis and Studer 8-track recorders. See also DTRS.

 Apple Desktop Bus. The original serial interface for the keyboard, mouse, and other "desktop" peripherals on Apple computers. ADB has been replaced by USB.

Commonly abbreviated A/D converter or just A/D. A device that changes the continuous fluctuations in voltage from an analog device (such as a microphone) into digital information that can be stored or processed in a sampler, DSP, or digital recording device.

The generation of complex musical waveforms in electronic synthesizers by the linear addition of sine wave components whose frequency relationship is a harmonic series. See sample synthesis, sound synthesis, subtractive synthesis.

A control/timing track on the edge of videotape (1", C, and 3/4" formats) that contains control data for quick and accurate location of program material, recorded at the same time as the picture. See control track.

A variable tone control in a preamplifier which allows the adjustment of the boost/cut and the frequency below or above which the gain/attenuation is applied (turnover), but not the rolloff slope of the shelving equalizer. If it were possible to adjust the rolloff slope, the result would be a fully parametric tone control.

Audio Decision List. ADL is also known as AES 31. It is an edit data exchange format for audio mixing and workstation. This ASCII based design provides sample accurate editing. It also includes file locators within a structure to support exchange between disk based audio systems The ADL or AES31 is officially titled: "AES standard for network and file transfer of audio — Audio-file transfer and exchange Part 3: Simple project interchange.

Adaptive Delta (Differential) Pulse Code Modulation. A type of split-band, time-domain audio compression algorithm for 16-bit digital audio based on describing level differences between adjacent samples. Different from conventional linear PCM by coding only level differences between samples, rather than the absolute level of each sample. According to the characteristics of the audio signal, ADPCM adapts the step size represented by each quantizing interval to accommodate rapid changes in level caused by high frequencies or transients, thereby providing an overall reduction in bit rate; the compression ratio is 4:1. There are at least two ADPCM standards: Microsoft and IMA/ADPCM, the latter popular for multimedia applications. See delta modulation, split-band, sub-band, transform coding.

Automatic Dialogue Replacement. Recording of dialog for a scene after it has been shot, usually to replace location sound that is unusable because of street noise, camera noise, etc. The workprint and sync magnetic film transfer for the scene are spliced into continuous loops and projected in a sound studio so that the actors can recreate the phrasing and feeling they had on the set. New takes are recorded on a separate mag film loop and/or other synchronous tape until an acceptable performance is obtained. Also see virgin looping, looping, and lip sync.

Asymmetric Digital Subscriber Line. A high-speed Internet connection that’s called “asymmetric” because it’s capable of much faster downloads speeds than upload speeds. It is faster that ISDN but cheaper than T-1.

Attack/Decay/Sustain/Release, the four segments of a common type of sound synthesizer envelope. The controls for these four parameters determine the duration (or in the case of sustain, the height) of the segments of the envelope. Two additional parameters, D (Delay Time) and H (Hold Time) are available on some synthesizers. See envelope, envelope generator.

Auto Double Tracking. An effect produced by taking a track and copying the material onto another track, delayed by a few ms, then mixing it with the original. Like chorusing, but with a shorter delay. See also double-tracking.

Audio Engineering Society. The professional organization whose members report on new technological developments in audio, and bring together designers, manufacturers, and users of various audio equipment to establish international standards.

 Audio Engineering Society/European Broadcast Union. A standard for encoding multiple channels of digital audio along a serial cable, officially named AES3-1985. The standard specifies usually uses 3-pin XLR jacks and balanced line cables, usually running at +4dBm. Originally designed as a self-clocking system, a subsequent addendum to the specification permitted master clock systems. Two channels of digital audio data are multiplexed on a single conductor within the cable, with a maximum bit depth of 24 bits. Data are transmitted at 64 times the sample rate, allowing the possibility of sending two channels of 24-bit audio (plus ECC) to play the resulting stereo signal in real-time. AES/EBU does not carry the SCMS copy code, and is a self-clocking protocol. It has been adopted by the EIAJ, which calls it the CP-340, Type 1. See also S/PDIF.

Standard for network and file transfer of audio: Audio-file transfer and exchange. A file format that can be converted to AAF lossless-ly. However, this file structure is not as deep as that of AAF and file transfers from AAF to AES31 are lossy

Audio Frequency. This refers to frequencies within the audible range, usually taken to be 20Hz-20kHz. This frequency range is an average; many people hear tones below 20Hz, although most people are virtually deaf above 15kHz or 16kHz. See Hz (Hertz).

Alternative Frequency. See RDS.

(1) Audio Frequency Modulation. A processing scheme used for recording high-quality analog audio in video cassette recorders equipped with "Hi-fi" stereo audio. (2) American Federation of Musicians. The union that represents professional musicians in all their client and employer relations.

(AFL) On many recording and mixing consoles, there is an option labeled AFL, or after-fader listen. This allows the listener to hear the audio after the channel fader has effected the audio signal. The AFL is also known as post-fader listen. The oposite of AFL is pre-fader listen.

A type of MIDI controller data, generated by pressing down on one or more keys after they have reached and are resting on the keybed. Also called pressure. See channel pressure, poly pressure.

American Federation of Television and Radio Artists.

 A third performing rights organization similar to ASCAP and BMI but much smaller. AGAC primarily represents modern classical composers.

 Automatic Gain Control. See AVC.

Advanced Graphics Port. A dedicated high-speed interface between PC memory and a graphics card and monitor. Before AGP existed, graphics cards were fitted into standard PCI slots and could cause audio clicks and pops if they didn’t correctly release the PCI buss for other duties when requested, thereby preventing audio data from reaching a PCI sound card. Nowadays all PCs use AGP graphics technology, which provides faster graphics performance and avoids such audio interference problems. As an aside, music applications don’t use 3D graphics, so musicians don’t need an expensive state-of-the-art graphics card. Far more useful is a good-quality 2D dual-head model that supports two monitor screens, since most music software will let you place the song’s arrange page on one screen, and the software mixing desk on the other, which makes working with complex multi-track songs far more pleasurable.

Containing frequencies that are not whole-number multiples of the fundamental. See inharmonic, clangorous.

Audio Interchange File Format. A common Macintosh audio file format. It can be mono or stereo, at sampling rates up to 48kHz. AIFF files are QuickTime-compatible and support uncompressed mono, stereo, and multichannel audio at many different resolutions and sampling rates, including the CD standard. It was designed to serve as a universal interchange format that allows any program to open a digital recording created by any other program. AIFF is high-quality audio, used in pro-level Mac and PC audio software. As AIFF is a standard for uncompressed audio, Apple introduced AIFF-C which can use MACE and IMA/ADPCM compression with ratios as high as 6:1, but the audio sound quality suffers.

 Intermodulation distortion where one signal will cause amplitude modulation of another signal.

Also known as a Velocity Mic, which converts sound waves traveling in air into an audio signal that travels through the mic cable.

A sealed enclosure. Typically, a low-frequency loudspeaker baffle where most of the damping of the cone is the result of the elasticity of the air in the sealed cabinet.

 A remixed and possibly re-edited version of a film that has any objectionable material removed. The airline film standard is more stringent even than those of the broadcast networks, and is often used as a benchmark for TV viewing.

Abbreviation for auditory integration training.

A type of composition in which the large outlines of the piece, or the procedures to be used in generating it, are determined by the human composer while some details, such as pitches or rhythms, are created by a computer program.

A file on a Mac that serves as a pointer to another file. The most common use for an alias is to sit on a desktop or in a top-level folder, where the real document or application file is nested deep within the file system. This is similar to a shortcut file on a PC-type computer.

 Distortion that is produced when higher harmonic components of the input audio signal sampled by a digital recording device, or generated within a digital sound source, lie above the Nyquist frequency. This happens when the sampling rate is less than twice the frequency of the signal being sampled. The effects of aliasing differ from some other types of distortion in that its pitch changes radically when the pitch of the intended sound changes. Also called foldover. See anti-aliasing filter.

(1) In tape recording, the process of adjusting all parameters of the position and orientation of the tape heads and guides with respect to the tape path. See azimuth. (2) The adjustment or calibration of any parameter of an electronic circuit or device, e.g., program level, bias level, to bring this parameter into conformance with an industry standard. (3) The process of matching mixer and recorder meters so that only one meter needs to be watched during recording. When the mixer and (analog) recorder are both peaking about 0VU, this minimizes the noise and distortion in both units. Ideally, both units would be matched with a steady tone (the C or B two octaves above middle-C, or about 2kHz, for example.) See line-up tone.

A tape loop of audio silence, processed by a recorder with some kind of noise-reduction enabled, such as Dolby-SR (where it is called SR noise) or dbx or dB. Used to check equipment for ground loops or other problems in recordings and/or masters. See Dolby noise. See biased noise.

 A MIDI command, recognized by some but not all synthesizers and sound modules, that causes any notes that are currently sounding to be shut off. The panic button on a synth or sequencer usually transmits All-Notes-Off messages on all 16 MIDI channels.

An electrical circuit with a uniform amplitude response versus frequency response, but with a phase-shift which does not vary in a linear relationship with frequency. (A pure time-delay device such as a digital delay line will have a phase-shift which is directly proportional to frequency, i.e., its phase-shift increases at a constant rate with frequency.) Complex filters often have significant phase distortion because they are not phase linear, and an all-pass network can be designed to correct phase anomalies without affecting the amplitude response.

A term historically related to mediæval note lengths, in which the breve was one of the shortest notes. In modern usage, the term is usually used to denote 2/2 (cut-time). In commercial and popular music, it is frequently used to mean half-time, i.e., play twice as fast. See time signature.

A lively to reasonably fast tempo: 116-150 bpm. Allegretto is a slightly slower tempo than allegro.

(Advanced Linux Sound Architecture) An open source audio interface driver protocol allowing audio interfaces to connect to and be controlled by a computer.

 An electrical current that periodically changes in direction. The rate of alternation is called the frequency and is measured in cycles per second or Hertz. Audio signals are always alternating, the frequency corresponding to the pitches of the sounds the signals represent. See Appendix.

The ability of an FM tuner or receiver to reject amplitude modulation of the received signal and be sensitive only to frequency modulation. Much of the interference and noise in broadcasting appears as amplitude modulation, so a tuner with good AM suppression will have less distortion and noise than a tuner with poorer suppression. Also called AM rejection.

Ambience refers to the acoustical qualities of a listening space, such as reverberation, echoes, background noise, etc. On most music recordings, some of the ambience is recorded along with the music and are, to a certain extent, reproduced in the listening environment, e.g., an organ in a cathedral. See room tone, walla, NC Curve.

An edited roll of magnetic film, or one track of a multitrack tape, assembled by the sound editor in preparation for the final mix of a motion picture or video production, containing the series of room tones or ambient sounds of the various sets and locations in which a scene was shot.

Area of sound where the reverberate sound is louder than it’s source

 Ambient sound which is environmental in nature, such as traffic noise coming through walls, heating or air conditioning, or other extraneous sounds which cannot be turned off or removed.

Sounds such as reverberation, room tone, walla and atmospherics that form a background to the main sound, usually in the context of a film soundtrack of a motion picture, taking place at any given moment. The lack of ambient sound is noticeable because the human hearing system expects it. See also ambient noise.

A system for the reproduction of a three-dimensional sound field, using two or more transmission channels and four or more loudspeakers.

A group that works with MMA on MIDI standards, among other things.

A filter circuit that mimics the amplifier and loudspeaker voicing of an electric guitar and amplifier system.

The unit of measurement for electrical current in coulombs (6.25 x 10^18 electrons) per second. There is one ampere in a circuit that has one ohm resistance when one volt is applied to the circuit. One should not speak of the "flow of current." The current exists; the charge flows. This is analogous to the current in a river, which consists of the flow of water.

 A former manufacturer of videotape recorders, analog audio tape recorders, and associated magnetic tape media. For the historic trivia fan, AMPEX is an acronym based on the founder’s name, Alexander M. Poniatoff Excellence.

An electrical circuit or device designed to increase the current, voltage, or power of an applied signal. An amplifier is an active device and, strictly speaking, should always increase the power of a signal; some amplifiers, such as certain distribution amplifiers, may only reduce the impedance level of the signal for the purpose of driving long lines.

The amount of amplification that an amplifier provides is called its gain. The gain is a ratio of the input signal level to the output signal level and is simply a number. Commonly expressed in dBindB, one should not express the voltage gain of an amplifier in dBindB unless the input and output impedance are matched as the gain of a typical amplifier is not related to its power output capability. For instance, if an amplifier has a voltage gain of 10, it might be said that it has a gain of 20dB because it actually would raise the power level of a signal by 20dB if the input and output impedance were matched. In practice, however, this is very seldom the case, and the true power gain is usually very much different from what would be predicted by the voltage gain. See impedance matching.

 The relative strength (amount) of a signal, without regard to its frequency content. Amplitude is measured by determining the amount of fluctuation in air pressure (of a sound), voltage (of an electrical signal), or numerical data (in a digital application). When the signal is in the audio range, amplitude is perceived as loudness. Amplitude is the measurement of how much energy the sound has, i.e., the total change in air pressure during a single cycle of the sound wave. Amplitude, or sound pressure, is measured in a scale called decibels (dB). An increase of 3dB is equal to a doubling of a sound’s pressure. Amplitude can be expressed as either a negative or positive number, depending on the signals being compared. See also magnitude, SPL.

See frequency response errors, jitter.

The instantaneous amplitude modulation of one signal by another. This results in the formation of sidebands which contain the same information as the original signals, but translated upwards and downwards in frequency. In AM radio transmission, the audio signal is combined with a very high-frequency sine wave, called a carrier, in such a way that the amplitude of the carrier is varied in exact response to the amplitude and frequency of the signal. This is called the amplitude modulation of the carrier. The modulated carrier is transmitted at high power where it is received by radio sets that are tuned to the carrier frequency. The modulated carrier is then demodulated by a process called detection, recovering the original signal. In radio, a circuit that does amplitude modulation is also called a heterodyne.

A special case of offbeat which immediately precedes the first beat of the bar, and hence the bar line. See beat

Capable of exhibiting continuous fluctuations. An audio signal is an electrical replica, or analog, of the waveform of the sound it represents. The voltage of the signal varies up and down (negatively and positively, in electrical terminology) the same way as the sound pressure varies up and down at the microphone. In an analog synthesizer, such parameters as oscillator pitch and LFO speed are typically controlled by analog control voltages rather than by digital data, and the audio signal is also an analog voltage. Compare with digital.

Any method of recording in which the recorded waveform is a continuous representation of the original signal, e.g., conventional magnetic tape recording.

A sequencer into which sounds for storage and playback are fed as analog signals, via analog potentiometers.

The technique of arriving at the desired timbre by filtering waveforms rich in harmonics. Subtractive synthesis is the type generally used on analog synthesizers. This works well on good analog synthesizers, but when used on samples, reducing the number of harmonics usually just makes the sound flat and lifeless.  See subtractive synthesis.

 Commonly abbreviated A/D converter or just A/D. A device that changes the continuous fluctuations in voltage from an analog device (such as a microphone) into digital information that can be stored or processed in a sampler, DSP, or digital recording device.

The camera/projector lens system which squeezes an image, usually originating in a 2:1 aspect ratio) onto film during shooting and "unsqueezes" it during projection. The resulting viewed image has an aspect ratio twice as wide as what was originally recorded on the film, e.g., if the image on the print is 2.2:1, the screen image will be 2.4:1. See also CinemaScope, flat(4), ‘scope.

At a "walking" speed: 76-94 bpm. Andantino can mean either a little faster or a little slower than andante, although it more commonly denotes a little faster.

Without echo. Said of an acoustic which is free-field, and specifically of a room which is designed to produce no reverberation or other echo effects. This is achieved by giving the walls very irregular surfaces of considerable and varying depths so that, in theory, all sound waves which strike them are completely absorbed and not reflected. Anechoic chambers are used to test audio equipment and for other types of acoustic and electromagnetic electro-magnetic research.

See hysteresis.

The anode in any electronic component, such as a silicon diode or a vacuum tube, is the electrode normally connected to the positive voltage.

The first composite print made from the edited picture negative in 35mm film, or the A- and B-rolls of a motion picture in 16mm. Each shot is exposed, color-balanced, and otherwise processed. Further changes and corrections can be made in a second or third answer print, if necessary. In many contracts, the delivery of the answer print is specified because it means that post has ended and release printing can begin, although the release print is usually made from an internegative, not the answer print. The answer print is not the same as a black-track answer print which contains no soundtrack.

Before a signal is subjected to the process of A/D conversion, it must be passed through a lowpass brick-wall filter to remove any components that are higher than the Nyquist frequency. This is because it requires at least two samples per cycle to determine the existence and strength of a frequency component or the A/D process will create aliased signals. See reconstruction filter, decimation, FIR, IIR.

See reconstruction filter.

A place of minimum sound pressure level in a standing wave, as opposed to a node, which is a maximum level.

A condition where two signals have a phase difference of 180˚, or one-half cycle. It should be called out of polarity, phase being a continuous variable rather than discrete.  See phase reversal.

A term used to describe music that is played or sung in alternating sections by two separate groups of performers, widely separated.

Audio over Ethernet is the distribution of uncompressed audio over a low latency network.

Album-Oriented Rock or Adult Oriented Rock. A tendency of some FM radio stations to play longer album tracks.

In an A/D converter, the sample-and-hold circuit would ideally take zero time to determine the level of the signal waveform and to hold this level until the next sample is called for. However, it takes a finite time to charge the holding capacitor in the sample-and-hold circuit, and this is called the aperture time. Because the time required to establish a new value of charge depends on the amount of change in the signal level from one sample to the next, the aperture time will vary with the rate of change in signal level, increasing for high-level, high-frequency signals. The starting time of the sampling aperture is also slightly uncertain, and this is called jitter. In other words, lack of precision in the sampling time leads to amplitude errors in rapidly changing signals. The errors involve aperture time, uncertainty in aperture time, and jitter. The result is distortion of the sampled signal which rises with frequency.

The main menu on a Mac, used to access system utilities (such as Keycaps), applications, files, and control panels. This is the equivalent of the Start menu on a PC-type system.

 A system-wide macro facility on Macs which gives operating system-level control for compatible applications.

Audio Post-Production for Video. The process of preparing the individual soundtracks and the final mix for a videotaped production.

Association of Professional Recording Studios. An industry body set up to ensure a uniform standard of service and practice in the area of sound recording.

The APRS has decided on a standard color-code for tape labelling:

See ISDN.

Artists and Repertoire. The department of a record company that selects the performing groups or artists who will be signed to the label, what songs or compositions each artist will record, and who will work with the artist in the production.

Italian for air (song). Generally indicates a composition for solo voice with accompaniment, also by extension, a lyrical instrumental piece.

The playing of chord patterns by sounding each note in a sequence, rather than simultaneously. An arpeggiator is a device which will automatically produce arpeggiation, given the parameters which control Direction (up or down or random), a Hold button which allows note patterns to be triggered which keep playing when the keys are released, and a Range control which sets the group of notes to be played over.

 (1) A version of a piece of music for resources other than those originally intended. This may be an instrumental version of a vocal number, a piano reduction of an orchestral piece, or may involve altering other parameters of the original, such as its harmony, rhythm, or structure. (2) In sequencers, a term sometimes used for the general layout of tracks, channels, and patches, rather than a complete song. This template can often be saved as a separate file.

The way of characterizing notes (usually in a melody) by the precise control of their individual lengths to produce or eliminate gaps between them. The terms staccato and legato reflect the two extremes of articulation. It is one of the most important ways by which music can be shaped into phrases.

American Society of Composers, Authors, and Publishers. The first performing-rights organization formed in 1908, ASCAP collects fees for broadcast of recorded compositions on radio and television, and for live public performances of music and distributes payments to the copyright holders of these compositions.

American Standard Code for Information Interchange. The most common encoding for transmitting text data digitally. The code employs 8-bit binary words, by which each letter of the English language, numeral, and symbol is uniquely designated.

Audio Stream Input/Output. Steinberg’s protocol for communication between audio hardware and software. Used in Cubase. It bypasses the operating system’s standard multimedia audio drivers to provide multi-channel support, improve audio timing, and reduce monitoring latency. ASIO runs at a lower level (it bypasses much of the Windows OS) and typically manages lower latencies than both MME and DirectSound.

The width-to-height ratio of an image. Specifically in film, the format that the film image is intended to be shown in, most commonly expressed as width relative to height, where the height parameter has been scaled to represent 1 unit. Standard TV screens are 1.33:1, films shown in U.S. theatres are 1.85:1, anamorphic films are 2.4:1. Ratios may also be expressed as integers, e.g., the TV ratio may be expressed as 4x3, and widescreen TVs are 16x9, or an aspect ratio of 1.78:1.

A small irregularity or imperfection in the surface of a magnetic tape. Low-frequency noise in analog tape recordings caused by asperities produce asperity noise in the recording, a type of modulation noise in that the noise is manifested in the band immediately above or below the program signal. See calendering, dropout.

Editing of an audio or video program by making a master copy of the various takes, rather than physically splicing pieces of tape together. Virtually all digital editing is done this way. The opposite of insert editing.

A collection of selected extracts from original sound rearranged to a desired order. See copy editing

Push-buttons on the input modules of a control console that connect, or assign, that particular input to any of the output busses of the console. The signal is routed to the desired tape track of the destination device usually by a matrix of switches in each module of the mixing and/or recording console. This routing process is known as assignment.

Not according to a fixed rate of repetition. An asynchronous signal can occur at intervals which do not necessarily coincide with a fixed-rate system or master clock pulse.

See IDE.

Automatic Track Following. The system used by R-DAT players to ensure that the rotary heads follow the recorded track. This uses a set of signals that is recorded along with the digital data and which are passed to the servo controls to ensure that the tape is correctly positioned with respect to the heads.

Asynchronous Transfer Mode. A high-speed cell-switching transport standard favored by WAN architects. Used as a LAN networking solution. ATM is capable of getting the job done, but inability to guarantee delivery, tiny fixed frame length and sizeable transmission overhead mean that it is best used for enterprise backbones and WANs.

See backgrounds.

 Audio Tape Recorder. This is the analog version. A digital audio tape recorder is called a DTR.

Adaptive TRansform Acoustic Coding. A lossy, 5:1-formatsplit-band perceptual coding and compression scheme for reducing data to be written on a MiniDisc. ATRAC offers a 5:1 data reduction ratio in the case of Minidisc, employing the equivalent of 52 filter bands for spectral analysis and quantization. Later versions of ATRAC vary the size of the sample blocks dynamically between 11.6ms and 1.45ms according to the input signal to allow for temporal masking, providing extremely good resulting sound.

The first part of the sound of a note. In a synthesizer envelope, the attack segment is the segment during which the envelope rises from its initial value (usually zero) to the attack (peak) level (often the maximum level for the envelope) at a rate determined by the attack time parameter. See ADSR.

 (1) The rate of attack of a note. (2) The time it takes for a compressor or limiter to reduce its gain after a strong signal is applied to it. See release time.

The actual attack waveform. See transient.

The reduction, typically by some controlled amount, of an electrical signal.

A potentiometer (pot or pad) that is used to adjust the amplitude of the signal passing through it. The amplitude can usually be set to any value between full (no attenuation) and zero (infinite attenuation). Pots can be either rotary or linear (sliders), may have discrete dentents (more often in older equipment), and can be either hardware or virtual sliders on a computer screen.

An audio plug-in format created by Apple used by many DAWs including Logic Pro, Live, Digital Performer, and Studio One.

An audio file format developed by Sun Microsystems, supported by some PC and Mac audio programs. This format supports stereo and mono files with either 8-bit or 16-bit resolution. It can encode linear files, or use µLaw or ADPCM compression.

 Literally, "I hear" in Latin. The term pertains to any signal, sound, waveform, etc., that can be heard, as opposed to subsonic or ultrasonic sound, radio-frequency signals or video signals.

Usually accompanies track search and allows the user to skip from the end of one track to the beginning of the next missing out any blank or unrecorded passages on the tape.

A parameter in Dolby Digital surround-sound format which refers to the number of channels and their location in for form F/R, where F is the number of front channels and R is the number of rear channels. For example, 5-channel surround is called 3/2 mode, stereo is 2/0, and mono is designated 1/0.

Any dynamic signal processing device that in some sense improves a dull or lifeless sound. It can be a simple as EQ or a complex DSP algorithm. Examples of exciters are the Aphex Aural Exciter, BBE Sonic Maximizer, or SPL Vitalizer. Enhancers combine dynamic equalization with either harmonic synthesis or phase manipulation.

This refers to frequencies within the audible range, usually taken to be 20Hz-20kHz. This frequency range is an average; many people hear tones below 20Hz, although most people are virtually deaf above 15kHz or 16kHz. See Hz (Hertz).

A type of diagnostic recording made with the recording set-up as planned, but with all faders down. Used to make a reference measurement of the noise floor and/or a tape of biased noise.

A type of potentiometer designed for use as a volume control in audio equipment where the resistance varies in a logarithmic, rather than a linear, fashion with rotation of the knob. This gives a better correlation between control rotation and the subjective loudness of the signal.

 Software or hardware that takes a monophonic instrumental or vocal line, analyzes the pitches, amplitude, and timbre, and converts the line to MIDI notes, complete with pitch-bend, MIDI velocity and volume, and possibly additional controller data.

An open MIDI driver specification/standard being promoted by Cakewalk™.

An audio artifact which occurs when several sounds are mixed, all which occur in the same frequency range. This happens because human ears tend to blend simultaneous sounds into a single, composite sound. When several instruments or other sounds emphasize similar frequencies, those frequencies accumulate and can either become too dominant or can cause one sound to mask another. Synonyms:  frequency masking, auditory masking

 (1) The increase of a major or perfect interval by one half-step to make an augmented interval. (2) The appearance of a musical idea in note durations which are longer than those used for its first appearance. This technique was often used in the ployphonic music of the middle ages and renaissance, as well as in contrapuntal music (e.g., fugues) of the baroque and later periods.

Of, relating to, or perceived by the ear.

Audio format supporting high-resolution surround sound (96KHz-24 bit) in all channels that works using an 11.1 speaker system

 In on-line editing, the process by which the edit-programmer produces the edited video master tape according to the instructions on the EDL, without human intervention. This is only possible where footage is consistently lit and exposed.

See quantization.

 One of the electronic operating modes of a multitrack recorder. When auto-input is selected, all channels will remain in sel-sync playback mode until the machine is placed in record mode. Any channels that are in "ready " status will then begin recording and will automatically pass their input signals direct to their outputs. When recording is stopped, these channels return to sel-sync playback mode. Also called stand-by mode.

A device for controlling the transport system of a tape recorder, allowing timecodetime code referencing such as SMPTE. Usually a number of locate points can be stored by the device. Some sequencers have an autolocate facility. Also called zero locate.

A circuit which adjusts the gain of an audio device in inverse proportion to the incoming signal level. An example is a portable tape recorder which is designed for speech recording; when the speaker is close to the microphone, the gain is reduced so as not to overload the tape. Also, a circuit which increases a TV or radio receiver’s gain when it is tuned to weak stations and decreases the gain when it is tuned to strong stations. Called AGC (Automatic Gain Control) in TV.

A system where manual control of a process is replaced or enhanced by computer control, such as mixing desk automation where faders, mutes, and equalization can be controlled in part or in whole by a computer. In write mode, the automation system produces a continuous record of all the actual fader settings and adjustments made by the engineer during a mix. Most systems allow changes on replay, while remembering and recreating previous manipulations of other tracks. The level changes are recorded and recreated by VCAs in each input module of the console. The VCA-produced data can be recorded directly onto a track of the multitrack tape, giving a continuous record of all mixdown fader settings. Or, the VCA outputs can be recorded onto a separate disk. In the latter system, alignment of the fader data with the multitrack master tape is achieve by referring to a common SMPTE timecodetime code recorded on the tape and disk systems. See mute mode, mute-write, null-point, read mode, snapshot automation, update mode, write mode.

 A device for processing a signal so that it can be made to appear at various positions in a stereo image via a remote control or MIDI commands

Benefits the long distance driver by scanning the waveband in the area of travel and automatically storing five or six strongest station in order of signal strength.

 A smaller, independent mixer within the main mixing console which has an output consisting of a mix of everything going into the channels on which the appropriate effects send been turned up.

An assignable, line-level input with no dedicated input source. Generally refers to an input connector in a preamplifier or integrated amplifier, signal processor, mixer, effects device, etc. The aux input has no de-emphasis or other special equalization and accepts line-level signals. Tone controls onf a preamp usually also affect signals sent to the aux input.

A bus allowing a signal to be sent from a mixing desk prior to the main output, usually to provide an input to effects. See effects send.

An extra envelope in a synthesizer that, instead of being hard-wired to a filter or amplitude, is intended as a modulation source that can be applied to various destinations.

A classification of MIDI messages which includes Active Sensing, All Notes Off, Local On/Off, and Reset, and which describes whether the particular MIDI device responds to any of thsese messages.

 Automatic Volume Control. A circuit which adjusts the gain of an audio device in inverse proportion to the incoming signal level. An example is a portable tape recorder which is designed for speech recording; when the speaker is close to the microphone, the gain is reduced so as not to overload the tape. Also, a circuit which increases a TV or radio receiver’s gain when it is tuned to weak stations and decreases the gain when it is tuned to strong stations. Called AGC (Automatic Gain Control) in TV.

Audio Video Interleaved. Microsoft’s answer to Apple’s QuickTime, and not compatible with Macs.

A brand of nonlinear video editing system, which, while not being exactly an industry standard, is the most commonly used digital video editing system.

 In microphones, the direction of maximum sensitivity, generally perpendicular to the surface of the diaphragm or ribbon. In loudspeakers, the line projecting through the center of the voice coil toward the listening area. This is usually the direction in which the speaker exhibits the best overall frequency response. See acceptance angle, off-axis, directional microphone, polar pattern.

In a tape recorder, the azimuth is the angle that the gap in the record or playback head makes with the direction of the tape travel, and it must be precisely 90˚ to ensure proper high-frequency performance.

The film industry’s term for the sound reproduction system, including amplifiers, crossovers and loudspeakers. See A-chain, chain.

See ISDN.

A 1" professional video format developed by Bosch. Although generally considered superior to the standard C-format, B-format equipment is used only in a few production and post-production facilities. B-format video masters must be transferred to C-format for broadcast.

(1) An additional set of inputs to a mixer channel that allows either additional (but not simultaneous) tracks to be assigned a mixer; (2) More commonly these days, a different source of the same information that appears on the A inputs. This latter technique allows a sound editor to work offline on a sequence while the mixer is adjusting the overall EQ and level in automation while playing back from another copy. The material is recorded to tape, after witching inputs, when the editor is finished.

See A-roll.

Frequency correction approximately corresponding to human hearing at 70dB SPL. See A-weighting, C-weighting, equal loudness curves.

The nickname of the Dolby 70mm process that dedicates two of the six tracks on a 70mm print to low-frequency information (signals below 250 Hz). This term is no longer used as the new digital multichannel film sound formats specify a dedicated subwoofer track.

In magnetic recording tape, a thin coating applied to the non-oxide or back surface of the tape to reduce slippage between tape layers, prevent accumulation of static charge, and minimize curling or wrinkling.

In a condenser microphone, the fixed, rigid capacitor element that is charged with an electric polarity opposite to that of the diaphragm.

Subtracting the length (in minutes and seconds) of a recorded segment from the time in a longer program at which the segment is supposed to end. If a three-minute segment is to end a 30-minute program, backtiming will indicate that the end segment needs to roll at 27:00.

 A musical term referring to the second and fourth beats in a four-beat bar, often emphasized by the drummer.

To edit the space between words so that the whole length of a scene, including sections where the take or angle in question is not being used, is contiguous.

Sound effects that sonically define the time and place of a location. Also called ambience, atmos or atmospheres. Backgrounds give a sense of lush sonic effects and placements. More specifically, they usually use pan controls, reverbs, delays, and other positioning tools. BGs are considered sound effects and are not the same as room tone.

Pre-recorded music used by a singer or other musician during a performance and which augments or entirely replaces other performers. This has become increasingly popular as musicians attempt to recreate the sound of their studio recordings live on stage.

In popular music, extra vocal parts which fill in gaps, or harmonize with, the lead vocal line. Usually sung by specialist session singers. Usually abbreviated bvox.

On-stage instrument amplification.

See temporal masking.

 British Academy of Composers and Songwriters. A group being formed among the current Association of Professional Composers, the Composers’ Guild of Great Britain and the British Academy of Songwriters, Composers and Authors, building a larger and more influential "umbrella" organization.

A partition placed between two sources of sound, or between a sound source and a microphone, to prevent sound from passing through. The baffle, or screen, may be made of any material with a high absorption coefficient. Most baffles are designed as movable partitions, and are used to isolate individual instruments in recording studios.

 Hollywood colloquialism for the meeting of the Sound Branch of the Academy of Motion Picture Arts and Sciences in which the members hear ten-minute clips of the seven films that have made the semifinals of the Best Sound Effects Editing award.

BALanced-to-UNBALanced. A transformer device used to convert a singled-ended (unbalanced) signal to a differential (balanced) signal. A bal-unbal is essentially a transformer with one leg of the input and output windings hooked together. More complicated devices may also change impendences at the same time. The most common use for a bal-unbal is a 75 ohms coaxial-300 ohms twin-lead converter used in television.

(1) The amount of relative signal provided to each of two (or more) audio channels. (2) A control on a synthesizer that adjusts the relative volumes of two different sounds which it can voice simultaneously. Not to be confused with pan.

 See mag film.

Audio lines in which the signal current is not carried by the cable shield of a shielded cable. This requires two conductors for the signal, enclosed in a shield, with neither conductor connected to the shield. The circuit utilizes two identical conductors operated so that the voltages on each of them are equal in magnitude, but opposite in polarity with respect to ground. Compare with unbalanced line. See common-mode.

The dynamic behavior of the needle in a meter, such as a VU meter.

(1) An extent along the frequency dimension in which a signal exists is the band. For instance, an octave band is one octave wide. The AF band is 20Hz-20kHz wide. (2) The wider spiraled grooves that separate any two selections on a record. (3) Band is also used to indicate any single selection on a record, cassette, or reel-to-reel tape or CD, i.e., a track.

A signal is said to be band-limited if its frequency content is restricted to a particular frequency range. For instance, the output signal of a CD player is band-limited to 20kHz by the reconstruction filters built into the player.

An audio artifact occurs when several sounds are mixed, all of which occur in the same frequency range. This happens because human ears tend to blend simultaneous sounds into a single, composite sound. When several instruments or other sounds emphasize similar frequencies, those frequencies accumulate and can either become too dominant or can cause one sound to mask another. See auditory masking

A notated form of a piece of music, derived from a full score, usually containing only the music for a single instrument or pair of similar instruments.

A filter that discriminates against signals in a specific frequency band. The most common band-reject filters reject a very narrow frequency band, and they are usually called notch filters. The opposite of a bandpass filter.

A filter which has both a high-frequency and low-frequency rolloff, and only frequencies in between are allowed to pass. When applied to sound synthesis, a bandpass filter makes the waveform sound like it is coming down a phone line as telephone lines cannot reproduce lows or highs. The opposite of a band-reject filter.

 (1) The capacity of the channel through which information can pass. In audio, the rated bandwidth of a device is the portion of the frequency spectrum it can handle without significant degradation. In digital communications, the bandwidth is the amount of data that can be transmitted in a given period of time. (2) The bandwidth of a bandpass filter is the upper rolloff frequency minus the lower rolloff frequency, i.e., the frequency range in Hertz (Hz), or band, passed by the filter.

(1) A set of patches. (2) A related set of items, e.g., a filter bank: a set of filters that work together to process a given signal.

A type of MIDI controller message which specifies which bank of (receiving) sequencer programs to use; a way to get around the 128 program limit specified by MIDI.

 See TT connector.

In written music, a grouping of pulses into a convenient unit falls between two barlines. A barline is a vertical line that crosses the stave at regular intervals. The bar begins with the downbeat and ends immediately before the next downbeat, and will contain a constant number of beats of the type determined by the time signature, e.g., a bar of 4/4 will have four quarter-note beats.

The tendency of the magnetic elements or domains on a magnetic medium to influence one another and to become magnetized in one direction or another as a group rather than individually. This means that a magnetic medium, such as recording tape, has a graininess in its magnetic makeup which is what causes most background noise, or tape hiss. Modulation noise, which is only present in conjunction with a recorded signal, is also caused by the Barkhausen effect, and is sometimes called Barkhausen noise.

See blimp.

 In magnetic recording tape, the thin ribbon of polyester or other plastic material to which the oxide and back coating are applied, measured in mils. For example, the base of most professional recording tape is 1.42 mils thick.

In a MIDI device, the channel on which the device receives fundamental messages governing its operation, e.g., Reception Mode changes. In Mono Mode, the basic channel is the lowest-numbered channel.

The group of instruments or vocalists recorded first during a multitrack session. This group, usually including bass, drums, and standard rhythm section, will be played back through headphones to other instrumentalists who later overdub solos, lead or background vocals, or narration, and other sweetening or sound effects. See also backing track, bed.

The metal frame of a loudspeaker.

The very low end of the audio spectrum, approximately 20 Hz -200Hz or 300Hz.

 An increase in molecular pressure variation, not molecular velocity, which occurs at low frequencies at room boundaries. The pressure nodes for all frequencies build up in the corners of rooms, particularly at the intersections of the walls and floor or walls and ceiling. These regions can roughly be thought of as resonant spaces, and energy can be trapped quite effectively by placing frictional absorbers at the desired quarter wavelength out into the room from the corner. See bass trap, boundary effect.

 A specially designed low-frequency sound absorber to reduce the effects of standing waves in recording studios. It is a tuned absorber and may have a narrow or wide range of frequencies over which it operates. It usually consists of resonant wood panels with absorptive material behind them, or suitably shaped slots in a wall or ceiling. See bass build-up, boundary effect.

When higher frequencies are radiated in a narrower angle than lower ones creating an observable ascending attenuation of SPL above a certain frequency.

A regularly occurring pulse that can be heard or implied. (1) When two periodic signals are less than 20 Hz or so apart in frequency, and if they are mixed together, the amplitude of the combined signals will fluctuate as they alternately reinforce and cancel each other. These amplitude fluctuations cause loudness fluctuations and are called beats. See also difference tone. (2) In music, the sensation of a basic pulse from which all rhythm in the piece is derived. Beats are of three types: a downbeat is a strongly accented pulse, such as the first in the bar; an offbeat is any pulse other than the downbeat; an upbeat, also called the anacrusis, is a special case of offbeat which immediately precedes the first beat of the bar, and hence the bar line.

Adjusting the tempo variations in a recorded piece of music to fit the set tempo of the project. In a DAW, this is done using time stretching tools and cuts to synchronize the transients to the appropriate tempo markers

Background music used underneath a narrator or foreground dialog. Primarily applied to commercial radio or television spots. Also called basic tracks.

The logarithm in base 10 of the ratio of two different levels of power, acoustic or electric. Since large changes in loudness correspond to fractional portions of a bel, the decibel, 1/10 of a bel, is used as the measurement unit of level for sounds and audio signals. See Appendix A.

A type of filter that allows the boost or attenuation of a specified set of frequencies around a center frequency. Bell filters often allow users to adjust the center frequency, Q, and the amount of boost or cut. Bell filters are sometimes also known as haystack filters.

 In film, the editing table which consists of rewind handling reels holding 35mm pictures and mag film, a sprocketed synchronizer that keeps the reels in sync, in addition to providing a count, and a squawk box. See mut.

 To change pitch in a continuous sliding manner, usually using a pitch-bend wheel or lever. An upward bend is created by pushing away from the front of the modulation controller, and creates an increasing pitch, and vice-versa. See bend depth.

The amount of pitch-shift possible if the pitch-bend modulation controller is moved as far as possible. This is usually set to a whole step, but for special effects (such as electric guitar), it could be set to an octave or more.

A professional analog videotape in Beta format, but at an increased tape speed, which gives picture quality comparable with the 1" C format. Betacam also allows separate recording of the red, green, and blue picture information via its RGB mode for computer use. This capability gives much better control of edge-cuts in special effects. Often called Beta for short.

A system used for color videotape recording, developed by Sony for consumer systems. Generally acknowledged to give higher picture quality than VHS.

 Background. The walla in a commercial or other video production, over which other sound effects, music, and dialog are dubbed. bi-directional microphone : A figure-eight microphone.

An electronic reference signal used by mag recorders, editing stations, and film projectors. See bi-phase modulation, pilot tone, neo-pilot, and control track.

In SMPTE timecode generation, the electronic process that produces the signal containing the SMPTE data itself. A 1.2k Hz square wave is momentarily modulated to 2.4kHz with each new bit of location information coming from the master clock.

An electronic pulse used by some film equipment and other motor-driven devices. Similar to a bi-phase signal, but different in the way directional information is provided. See also tach pulse.

Short for bi-amplification. A two-way crossover network.

(1) Bias is the voltage or current that establishes the intrinsic noise floor of an active device. (2) In an analog tape recorder, bias is an ultrasonic signal, usually between 100k Hz-200kHz, which is mixed with the audio signal and applied to the recording head, reducing distortion by reducing the hysteresis inherent in the tape recording process. This process is known as AC bias because the bias current is alternating. The ideal setting of analog bias involves a compromise between the MOL of the tape, noise, and third-harmonic distortion. In general, classical recordings use a bias setting with lower distortion and lower MOL; rock or other recordings prefer a higher distortion in order to get the highest S/N ratio. Digital recorders do not require bias as the signal consists only of a bitstream of 0s and 1s, regardless of the audio frequency being recorded. (3) See electric microphone.

 A tape loop of audio silence, processed by a recorder with some kind of noise-reduction enabled, such as Dolby-SR (where it is called SR noise) or dbx or dB. Used to check equipment for ground loops or other problems in recordings and/or masters. Sometimes called an alignment recording. See Dolby noise.

 (1) A barrel into which strips of film hang, suspended from a row of pins or small nails above. Also called an editing (2) In tape duplication, the container or housing that holds a tape loop to be duplicated.

 A special tape that is used in cassette duplication equipment. It contains both sides of the tape and is either run at a very high speed or, for higher quality dubs, in real-time.

Literally, "having two ears." Because humans have two ears spaced apart by the width of the head, the human hearing mechanism can make use of amplitude, phase (arrival time) and spectral (frequency response) cues to help determine the direction from which a perceived sound is coming. See binaural synthesis.

A type of recording-playback system where two microphone inputs are specially processed to simulate the frequency-dependent time delays that would occur between the ears on a human head. The binaural localization cues are preserved, and the listener is able to achieve localization of sounds as if s/he were actually at the site where the recording was made, despite the fact that binaural recording has no ability to accurately image the sound. Also called dummy head recording.

A liquid or gelatinous medium in which oxide particles are suspended for application to magnetic recording tape. Usually consists of a solvent that evaporates, and an adhesive substance which that, when dry, permanently bonds the oxide to the base.

A type of terminal which allows wires, such as loudspeaker wires, to be connected to the output of an amplifier with alligator clips, banana plugs or bare wire.

Film sound slang for a mixing "top sheet," indicating the layout and content of the premixes. The layout is usually one column per premix.

 "Basic Input-Output System." An operating system which resides on ROM and is used to control disk access, exclusively. Used in some samplers and sequencers to control the internal hard drive.

A type of loudspeaker design where the sound emanates from the sides of the monitor, specifically designed to be surround-sound monitors. These type of speakers work well for ambient material, but less well for dialog, soundtrack, or main effect sound. This is opposed to a direct radiator speaker which distributes the sound in front or a tripole design which is a combination of a direct radiator and a bipole.

Extraneous high-pitched whistles sometimes present in tape-recorded signals where the high-frequency content of the signal causes beats with the bias signal. Also used to refer to high-pitched interference in AM radio reception.

Binary digit. The representation of data using base-2 arithmetic, i.e., a series of ones and zeroes. Digital audio is encoded in words that are usually 8, 12, 16, 20 or 24 bits long (the bit depth). Each added bit represents a theoretical improvement of about 6 dB in the S/N ratio.

The number of data bits used to encode each sample point. Bit depth determines the accuracy of a sampler, converter, or other digital device in capturing momentary changes in a sound’s amplitude. Typical bit depth is 16 bits, which is good for capturing loud sounds, but less good for sounds in a quieter range. Also called bit resolution.

See bit depth.

A technique for lossless compression which, rather than encoding the entire data word, only bit cells with data (ones) are stored, and the null data (zeroes) are removed. For example, if only 19 bits of a 24-bit word contain data, only those bits are transmitted.

A feature on some A/D converters, digital recorders, DAW or other digital devices to choose word lengths to accommodate various output channels, such as a choice between six outputs at 20-bit resolution, or four output channels at 24-bits per sample.

 "Burnt-In Timecode." Video that shows the SMPTE time on-screen in a window along with the picture, eliminating the need to watch a time-code reader. Accurate in still-frame. Sometimes called a "window dub."

A subjective term for the sharp onset or attack of a musical instrument, especially brass instruments. Excessive bite can result from positioning microphones too close to the instrument or from distortion caused by a momentary overload. See attack transient.

A type of clock reference, this is essentially a video signal without any picture and without any positional information. Also known as house sync. As a black-burst signal is typically distributed throughout a recording/editing facility as the facility master clock due to the extremely accurate clock signal provided. See video black.

 A version of the answer print which has no sound, i.e., it is "silent," made from the original camera negative. The first answer prints are usually black-track in order to proceed with the color timing, even though post-production sound has not been finalized.

The recording of a periodic signal on a blank video tape which marks the start of each video frame. See video black, video sync, control track.

The blanking interval occurs at the end of each video frame, during which video information is absent. The interval occurs when the CRT electron gun scanner goes from the bottom-right corner of the screen to the beginning of the next field (4) in the top-left corner.

See crosstalk, channel separation.

A solid cover for a motion picture camera, designed to completely contain camera noise. A barney is a padded cover for a portable camera that attenuates but does not eliminate camera noise.

(1) The recording of a periodic signal on a blank video tape which marks the start of each video frame. This control track (2) can be used to count time for editing purposes, but is prone to slip and lose count during winding. (3) Plotting actor, camera and microphone placement, and movement in a production.

 A CD specification for data, as opposed to sound or video.

A stereo miking technique which uses two figure-eight microphones, crossed at a 90˚ angle, set up as closely as possible to one another. This is also sometimes called coincident figure-eights. See also coincident pair.

A performing rights organization: It collects license fees on behalf of its songwriters, composers and music publishers and distributes them as royalties to those members whose works have been performed. Other performing rights organizations include ASCAP and AGAC.

(1) Bayonet-Nut Coupler. A two-conductor, low voltage, locking connector most commonly used for the connection of video and high-frequency clock signals. (2) Blimped Newsreel Camera. (ex: The 35mm Mitchell model Camera)

A synonym for a recording console or mixer, a mixing console. (2) Short for a film storyboard.

(1) In general recording, any sort of microphone stand with extending sections that allows the microphone to tilt and be pointed at a target, such as above a performer or some section of an orchestra, etc. Also called a fishing rod, although the latter usually refers to a lighter-weight rig more suitable for close miking. (2) The LFE in a mix. See baby boom.

To mix a sample with a backward version of itself.

Boost refers to an increase in amplitude, usually of a specific frequency or within a frequency band range. Equalizers, the most common of which are tone controls, cause boost or cut of selected frequency ranges.

A single control which has "no change" at its center-point. If the knob is rotated counter-clockwise, the input is attenuated; rotated clock-wise, the input is amplified.

 An arrangement where the apparent impedance of a circuit element is reduced by applying an appropriate feedback voltage to it, improving the linearity of a circuit, thus reducing its distortion. It is especially useful in circuits that are required to carry a very wide range of power or voltage levels. Used in power amplifier output stages.

 In multitrack recording, the process of recording several tracks and mixing those sounds down to one or two unused tracks. For example, on an 8-track recorder, you could record six tracks, bounce them down to the two remaining tracks, freeing up the original six tracks for recording use.

A sound reflection effect due to room modes ( standing waves) which accumulates at walls. sound wave reflections appear to make the localized sound level increase as all of the room modes terminate at the boundary (wall). Essentially as the wavefront approaches the wall, the amounts of molecular motion become smaller and smaller while the pressure differences become greater and greater as the wall resists the motion of the air molecules, the wall becoming a pressure node. The rigidity of the wall surface determines how much the pressure rises, i.e., how much of the pressure is reflected versus how much is absorbed. This occurs on a mode-by-mode basis at each resonant frequency. At very low frequencies, nothing large is rigid. However, at higher frequencies, the boundary effect is more pronounced, e.g., frequencies above 100 Hz in a room with typical walls. A related effect is often observed at a control room window, where the window itself will resonate at one or more resonant frequencies so that the window passes the resonant frequencies through to the (recording) space on the other side, somewhat reducing the boundary effect within the control room, but not providing sound isolation from the adjacent space(s). This last effect is worse for lower frequencies as higher frequencies tend to be absorbed by the glass in the window. Also called the pressure zone effect. See absorption coefficient, bass build-up, bass trap.

 A boundary microphone uses a small condenser microphone capsule mounted very near a sound reflecting plate, or boundary, so there is no delay in the reflected sound. Direct and reflected sounds add in-phase over the audible range of frequencies , resulting in a flat response, free of phase cancellations, excellent clarity and reach, and the same tone quality anywhere around the microphone. Boundary microphones have a directional response that is either half- omni, half- cardioid or half- supercardioid. An example of a boundary microphone is a PZM (pressure zone microphone).

Beats Per Minute. The usual measurement of tempo.

Bits per second. A metric used to measure the volume of data transferring.

Bytes per second. The smallest unit of measurement used in data transferring.

In a piece of music, a break is a solo or section of reduced instrumentation, or even complete silence. In modern usage the term usually implies an opportunity for an instrumental solo.

A type of jack socket fitted with switching terminals, so that insertion of a plug breaks an existing connection. Also called a normalled connection

On synthesizers and samplers, the specific value at which the tracking of scalable parameters, such as velocity, starts to take effect, or at which the nature of the scaling changes.

A device which a performer blows into, bites, or presses with the lips, allowing the articulated sound to be digitally recorded by a synthesizer or sampler. Breath controllers can control volume, filter frequency or amount of LFO. They incorporate a device known as a stress bridge.

Audible fluctuations in the noise level of a signal caused by poorly adjusted or unsuitable noise reduction systems which produce a variable changing program level and/or noise floor. Also called pumping, noise pumping or breathing. Pumping is caused by the action of a compressor, occurring when one loud sound source causes severe gain reduction in the compressor. With each loud sound, the level of the other instruments will decrease sharply. Pumping occurs during program material. Breathing, on the other hand, occurs when the program stops long enough for the compressor to cease its gain reduction, suddenly boosting the noise floor of the program. Quantization noise can also exhibit breathing. See also compander.

A very sharp filter which masks any frequency outside the passband, for example, the lowpass filter at the input of an A/D, used to prevent frequencies above the Nyquist frequency from being encoded by the converter. See aliasing, reconstruction filter, anti-aliasing filter, decimation, FIR, IIR.

(1) Meter bridge. A structure mounted at the rear of a mixing desk, or on other equipment such as a tape recorder, which contains a number of VU or PPM meters. (2) Bridge mode: a method of driving a single load, such as a loudspeaker, from two similar (ideally identical) amplifiers in order to double the power presented to the load; a stereo amplifier operating at 200W per channel could provide approximately 400W into a single load in bridge mode. Many stereo amplifiers designed for sound reinforcement offer this option. See bridged mono. (3) See bridging. (4) Originally an eight-bar section of contrasting material in the middle of a song, but later applied to a linking section of any length. Also called a bridge passage or middle-eight. See also break.

A section of music which links two musical ideas. A bridge is usually used to connect movements in different keys and/or tempos. See bridge

Where the two separate outputs of some stereo amplifiers can be combined to give a mono signal.

A method of combining both channels of stereo power amplifiers to create a doubly powerful single-channel (monaural) amplifier. See bridge(2).

The opposite of impedance-matching. When the input of an audio device is connected to the output of another device, it is a bridging connection if the second device does not appreciably load the first device and essentially no power is transferred. The second device is sensitive to the output voltage of the first device, and this is maximized when the loading is minimized. Most audio connections are bridging, and the load impedance is at least ten times greater than the source impedance. A bridging connection is made by connecting everything in parallel (all the plus inputs connect to the plus output, all the minus inputs connect to the minus output.) This not only allows for a number of loads to be connected to the same source before overloading it, but this also gets the maximum voltage swing possible from the source.

The amount of high-frequency signal present in a sound, which tends to make the sound appear closer. The opposite of darkness.

Including a wide range of frequencies, generally the entire audio range. Usually used in terms of referring to the broadband performance of an audio device with respect to some specification such as noise, <distortion, etc.

To discard recorded sound during a mix. Site brooming is when a director rejects a whole group of effects, often the product of several days work.

Broadcast Television Systems Committee. The FCC committee that decided upon the MTS standards for stereo television sound in the U.S.

A brand name of electronic devices that will maintain synchronization between two tape recorders, a tape recorder and a projector or video playback machine, etc. Used primarily to interlock one or more multitrack recorders to a video playback, for purposes of recording, overdubbing, or mixing music in sync with picture. The device uses SMPTE, time code, for electronic control of all machines.

The cancellation of one signal or frequency component of a signal by another signal with equal amplitude but opposite polarity. See also phasing, flanging, comb filter.

An amplifier with a high input impedance and approximate unity gain. Used, for example, in a mixer at the back panel outputs for headphones and control room monitors to prevent the two loads from overloading the fader output and causing A-rolloff in high-frequency response. A sort of internal distribution amplifier.

A small contact microphone, designed for string and wind instruments which work along similar lines to a piezo pick-up.

A System-Exclusive description of an actual sound sent over MIDI.

A tape demagnetizer that can erase an entire cassette reel or multitrack tape without removing the tape from its carrier. Essentially a powerful electromagnet. Some bulk erasers have circuits built in that automatically fade the magnetic field up from and ultimately back down to zero. This eliminates pops and other erasure noise normally left on tape if the eraser is suddenly turned on or off. Also called a degausser.

A System-Exclusive message of the non-real-time type that allows the exchange of tuning data between MIDI devices as well as other devices such as computers, allowing microtuning or different temperaments by defining a specific pitch value. The frequency range is from 8.1758 Hz to 13,289.73Hz, in steps of one half-step cents, for each of the 128 notes in the MIDI range. Two messages are involved: a bulk tuning dump request message which is transmitted by a device in order to signify that it is ready to receive, and a bulk tuning dump message which contains the data for 128 tuning programs, each containing 128 pitch values.

Small segments of music in a television or film score that usually precede a dissolve. In television, usually used before commercial breaks. burnt-in timecode time code: See BITC.

In a mixer, a path via which the user can route a signal from one or more inputs to a specified destination. Typical destinations include: groups, mix, auxiliary send, foldback, etc. For example, "routing inputs 1-8 to the mix bus" means that the eight input signals appear additively at the mix output.

Alignment film used to set the lateral alignment of the optical film recording areas for replay.

See backing vocals.

Abbreviation for Broadcast Wave

A facility on an effects unit which allows the user to switch the incoming signal directly through to the unit’s output, cancelling the effect so that an A/B comparison may be made quickly between the wet and dry signal. See wet/dry balance.

Comité Consultatif International Radio. An international radio standards committee, whose recommended recording pre-emphasis and post-emphasis curves are standard on all recorders in most European and some other countries. The European analog to the NAB.

The international standard format for professional 1" videotape equipment. Developed by Sony, and sometimes called S-format after that company’s name. See B-format, Betacam, VHS.

Con Repeats. As in, "Play from the beginning with repeats" is written, "D.C. (C.R.)."

See Spectral Recording, noise reduction.

 Unlike A-weighting, C-weighting measures frequencies uniformly over the audio spectrum. An SPL meter will allow the choice of either (or neither) weighting function. See B-weighting, equal loudness curves.

Also written as dB(C) or dBC. C-weighting is a weighting for sound level meters used for high level measurements and peak sound pressure levels. Flat or linear between 31.5Hz and 8kHz, the C-weighting correlates better with the human response to high noise levels.

A musical punctuation, indicating the end of an idea, or preparing the ground for transition to a new one; essentially a juxtaposition of two chords.

To reduce the asperities on the surface of a magnetic tape, the tape is squeezed between large steel rollers; a manufacturing process called calendaring.

See Soundfield microphone.

See phase cancellation.

Slang for pre-recorded, as opposed to live music or visuals.

 See XLR.

Headphones.

See impedance.

A device made up of two metallic plates separated by a dielectric (insulating material). Used to store electrical energy in the electrostatic field between the plates. It produces an impedance to an alternating current. Also called a condenser.

The beginning of a piece of music. See D.C.

In a tape machine, the tape is moved by the effect of friction between a rotating motor-driven pillar, the capstan, and a pinch wheel, also called the capstan idler, that holds the recording tape securely against the capstan when a tape transport is in record or play mode. The capstan motor directly or indirectly drives the capstan and moves the tape past the heads. The capstan itself may be the extended shaft of the capstan motor.

 In a microphone, the diaphragm or actual sound receptor, including, in various types of mics, the moving coil, ribbon, permanent magnet, or fixed condenser plate, and the housing in which these are mounted.

 A directional microphone with an acceptance angle that is most sensitive to sounds coming from the front and sides, while rejecting sounds coming from the rear. Called cardioid because the polar pattern of the microphone is roughly heart-shaped. All directional mics have a proximity effect, whereby sound sources close to the mic will have an exaggerated low-frequency response. Supercardioids and hypercardioids are cardioids, but with a trade-off in the rear lobe. When using supercardioids and hypercardioids as sound reinforcement mics, it is important to note that the maximum rejection is not directly behind the mic as it is with a cardioid, but is off to the side between 110¢ª-126¢ª. However, a pair of hypercardioid microphones used as a stereo X-Y pair yields a very clean cardioid response pattern. See pressure gradient.

(1) A signal that is constant in amplitude or frequency and can be modulated by some other signal. The carrier itself does not transmit any information; all of the intelligence is in the modulation sidebands, which are in a band of frequencies on either side of the carrier frequency. Some signals, such as FM stereo, involve more than one carrier to encode the information, and the lower-frequency carrier is called a subcarrier. The subcarrier is mixed with parts of the audio signal and used to modulate the main carrier. In the receiver, the subcarrier is recovered by demodulation of the main carrier and then demodulated to recover its signal. See amplitude modulation, frequency modulation. (2) In FM synthesis, the carrier is the operator at the bottom of a stack in an algorithm, through which the composite effect of other modulating operators connected to it is heard.

Cinema Audio Society. A Los Angeles-based organization of film and television recording personnel, founded in 1966.

A French word meaning "little box." A cassette is a magnetic tape sound recording format. The cassette was originally intended for dictation, but improvements in fidelity led the Compact Cassette to supplant reel-to-reel tape in most non-professional applications. Uses for the cassette include portable audio, home recording, and data storage for early computers. Between the 1960s and early 1980s, the cassette was one of the three most common formats for prerecorded music along with the LP and the Compact Disc. Cassettes consist of two miniature spools, between which a magnetic tape is passed and wound. These spools and their attendant parts are held inside a protective plastic shell. Two stereo pairs of tracks (four total) or two monaural audio tracks are available on the tape; one stereo pair or one monophonic track is played or recorded when the tape is moving in one direction and the second pair when moving in the other direction. This reversal is achieved either by manually flipping the cassette or by having the machine itself change the direction of tape movement ("auto-reverse").

The Dolby Laboratories device that turns a Cat. No. 22 Dolby A-Type noise reduction card into a 4-band "noise fighter." The precise frequencies of the bands are optimized for production sound problems and differ from those used in standard noise reduction applications. In 1991, Dolby formally introduced SR-type noise reduction, called the Cat. No. 430.

 The cathode in any electronic component, such as a silicon diode or a vacuum tube, is the electrode normally connected to the negative voltage.

Constant Angular Velocity: In a mass storage device, such as a disk, CAV means that the disk assembly rotates at a constant speed, i.e., the data rate will increase for the tracks near the edge, and decrease for tracks near the center spindle. As opposed to CLV.

 See LCRS.

The CD sampling rate is 44.1kHz and there are 32 bits per sample, so the data rate of the encoded analog data is 1.41Mbps, but the inclusion of parity, synch, and subcode bits raises the real data rate to 4.3Mbps. A CD will hold about 650 Mb, or about 74 minutes of stereo, 16-bit audio. The digital portion of the CD audio system is not stereophonic, but sequential monaural. See also Control and Display signals. Compare with Direct Stream Digital. The CD file format is defined by ISO 9660. For more information on CD standards, please see Sound on Sound, "Compact Disc Formats" by Mike Collins, January 1998. See also SACD. (Super Audio Compact Disc.)

Computer Enhanced Digital Audio Restoration. A British-developed system for the restoration and preservation of old audio recordings. See also NoNoise.

 A notched position in the range of a variable control, allowing the user to return the control to precisely that position, such as the midpoint between the left and right channels in a balance control. Use to denote the flat position on tone controls, etc.

The frequency that is boosted or attenuated most by the operation of any parametric equalizer or other similar processing device or circuit. See Q.

 In a transformer, the electrical midpoint of the windings made accessible for external connection. Used, for example, in delivering power to balanced line condenser microphones. See phantom power, Appendix B.

Abbreviation for Computer graphics.

 Also called iron. An integrated system composed of separate audio and/or video recording, processing, or playback circuits and/or devices which are used in conjunction with one another to produce one output result. See B-chain, program chain, signal chain, side chain.

See projection.

See projection.

An independently processed or recorded signal. (1) An electrical signal path. In analog audio (such as a mixer), each channel consists of separate wired components. In the digital domain, channels may share wiring, kept separate through logical operations. (2) A system for independently addressing up to sixteen separate MIDI devices over a single MIDI cable. MIDI provides definitions for 16 channels which transmit not audio signals, but digital control signals for triggering synthesizers and other devices. MIDI data are associated with a particular channel by virtue of a Channel ID Number that is interwoven with other MIDI data being recorded. A track holds data that (depending on the sequencer) may or may not be restricted to one MIDI channel. MIDI’s 16-channel limitation has been overcome by employing multiple independent MIDI ports that each route sixteen channels, offering the possibility of hundreds of channels. (3) The left or right signals of a stereo audio system, or the left, right, center, surround and/or subwoofer signals of a multichannel system, such as LCRS or 5.1. (4) In film, A complete, self-sufficient recording setup. A production channel would include a recorder, mixer, microphones, headsets, etc. A transfer channel would include a 1/4" tape deck, a 35mm mag recorder, a resolver, and a monitoring system.

 In a recording console, the group of buttons or switches by which the signal from any input channel can be assigned to one or more busses, and thereby be sent to one or more tracks of the multitrack recorder.

 The actual bits being read from a digital medium are greater than the number strictly required to encode the audio signal. This is because of ECC and synchronization bits, etc. For example, with a CD, the audio bitrate is 1.41Mbps, but the channel bit rate is actually three times as high, 4.32Mbps.

 An insertion point in a mixer channel which opens up the signal path and allows an outboard device to be inserted in-line. The output point (the place where the signal is routed to the outboard device) is called the channel insert send, and the place where the effected comes back into the mixer is called the channel insert return. The actual point at which the channel signal path is broken with the insert connection is not standard among all consoles. Some are between the preamp and equalizer sections, some after the equalizer, but before the fader, and some are post-fader. Some are switchable with an internal jumper or other modification. If, for example, the channel insert send is post-fader, the fader setting will affect the action of a compressor that is inserted into the channel’s signal path. On the other hand, a post-fader insert is good when it is desirable to send a single channel’s signal direct to a tape track, making the fader into a convenient record-level control. See normalled connection.

 A class of MIDI messages which only affect devices on a MIDI network set to a particular channel, i.e., all non-system messages. Channel messages may be of either Channel Mode or Channel Voice type. See MIDI.

See MIDI mode.

The record section of the signal chain in a mixer. See also monitor path.

A type of MIDI channel message that is applied equally to all of the notes on a given channel; the opposite of poly pressure, in which each MIDI note has its own pressure value. Also called aftertouch, channel pressure is generated on keyboard instruments by pressing down on a key or keys while they are resting on the keybed. Also called channel key pressure.

 The amount of crosstalk between the channels of a stereo system. It is the inverse of inter-channel crosstalk, as measured in decibels. A small amount of crosstalk is equivalent to a large channel separation.

Channel separation refers to the amount of crosstalk between the channels of a stereo system. It is the inverse of interchannel crosstalk, as measured in decibels. A small amount of crosstalk is equivalent to a large channel separation.

One of multiple identical sections in a mixering console from the mic preamp and phantom power (if present) to the bus outputs, and typically includes the input pad, EQ, and signal routing, including pan, effect sends and effects returns, and main channel fader, and optionally an automation interface. There is one channel strip per mixer input.

A classification of MIDI channel message relating specifically to a musical performance, where features of the performance (notes, articulation, etc.) are individually described by a unique message. Channel Voice messages include Note On, Note Off, Polyphonic Key Pressure, Channel Pressure, Program Change, Pitch-bend, and Controller Change. These messages all include a specific channel number, allowing similar messages to address different devices on the same MIDI network. The message will only be implemented by a receiving device whose channel number matches that of the message.

 See MIDI mapping.

 See termination.

A measure of the quantity of electricity and its unit is the coulomb. In an electrical circuit, charge consists of negative charges, or electrons. A positive charge can be thought of as simply an absence, or deficiency of electrons. Charge is what is moving in an electric current. See ampere

 (1) A musical score or arrangement. The term is used both to designate the conductor’s full score, or any band part. (2) A list of current hit singles or albums.

 (1) The process whereby a slave device attempts to sync to a master clock. (2) In MIDI parlance, to chase means, upon playback, to look backward to earlier MIDI events to see if there were any program or channel change messages prior to the playback point which would affect playback. See controller chasing.

A type of controller for a video or audio recorder that will listen to the SMPTE timecode signal from the master clock device and will adjust its own speed to find the correct time and then will lock into synchronization with the external timecode. Unlike sync-lock, chase-lock controllers respond to changes in timecode sequence.

 See controller chasing.

The practice of connecting the signal ground of a device to the rack rails or other common grounding location on a multi-component electronic system.

When the input signal level to a noise gate hovers near the threshold level, the gate may be unsure if it should be open or closed. It may rapidly open and close, resulting in the audio cutting in and out; this is known as the gate "chattering." To correct this problem, adjust the threshold setting to be slightly lower or higher.

A number derived from arithmetical actions on data, used to check that data has not been corrupted after transmission or recording and replay.

(1) In vinyl record production, the thin thread of acetate lacquer that is carved out of the master disc by the cutting stylus. Also called swarf. (2) A slang term for integrated circuit.

An effect caused by the overuse of single-ended noise reduction systems whereby the low-level signals take on an electronic, "ringing" character, known as chirping. If the signal is very noisy, the noise floor itself begins to sound chirpy, which can be more annoying than the original, broad-spectrum, noise.

The playing of multiple notes simultaneously. The opposite of an arpeggio. See inversion.

(1) A regularly repeated section of a song or other musical composition. (2) A group of singers, also called a choir.

A type of audio effect in which a delayed (30-40ms) or detuned copy of a signal is mixed with the original signal. The mixing process changes the relative strengths and phase relationships of the overtones to create a more complex sound. See ADT, double-tracking. The mixture becomes extremely complex as the relative phases of the signals cause partial cancellation and reinforcement over a broad frequency spectrum. The simplest way to achieve chorusing is to detune one synthesizer oscillator from another to produce a slow beating between them. See comb filter.

Pertaining to the full twelve-note scale, as opposed to the eight-note diatonic scale.

 A new system of digitally recording motion picture sound format introduced by the Optical Radiation Corporation, a division of Kodak, in 1990, for the film "Dick Tracy" for digital sound on 35mm or 70mm film formats via a laser beam, which reportedly combines the dynamic and frequency ranges and low distortion of the CD on six discrete channels. Five channels encompass the full audio bandwidth and the sixth is designated a subwoofer channel, containing only the lowest frequencies. The CDS-encoded film is capable of being shown with conventional stereo optical sound, but requires a special sound system to reproduce the six channels digitally. First used in 1990, this format lasted only two years and is now obsolete. See AC-3, 5.1.

The trademark of a widescreen camera system developed by Twentieth Century Fox, the first true stereophonic motion picture sound system which had the sound tracks on the same film with the picture. First used in 1953, CinemaScope was responsible for popularizing the anamorphic film format.

A widescreen system comprising three 35mm cameras/projectors running in interlock with 7-track mag film.

A widescreen system comprising three 35mm cameras/projectors running in interlock with 7-track mag film

Cross Interleaving Reed-Solomon Code. The combined error detection and correction scheme used in CDs. See interleaving.

 Also known as the Cycle of Fifths. A way of thinking of the twelve major and minor keys as a circle, arranged in steps of a fifth, which can be read in either direction. Starting from Cmaj and proceeding clockwise, the key signature of each new key gains one sharp until Fmaj is reached. At that point, FA becomes GBmaj and the cycle continues, removing a flat at each step until back to C. If one goes counter-clockwise, the circle is a series of perfect cadences, with each new tonic key becoming the dominant of the next. For this reason, the Circle of Fifths is often used for modulation, especially to or back from a remote key, i.e., a key on the far side of the circle.

A way of thinking of the twelve major and minor keys as a circle, arranged in steps of a fifth, which can be read in either direction. Starting from Cmaj and proceeding clockwise, the key signature of each new key gains one sharp until Fmaj is reached. At that point, FA becomes GBmaj and the cycle continues, removing a flat at each step until back to C. If one goes counter-clockwise, the circle is a series of perfect cadences, with each new tonic key becoming the dominant of the next. For this reason, the Circle of Fifths is often used for modulation, especially to or back from a remote key, i.e., a key on the far side of the circle.

A complete path that allows electrical current from one terminal of a voltage source to the other terminal.

A headset with a large cushion which surrounds the ear to exclude external noise, unlike supra-aural or intra-aural designs.

 See SDMI.

Meaning containing partials that are not part of the natural harmonic series, i.e., partials which are not whole-number multiples of the fundamental frequency. Clangorous tones often sound bell-like.

 In written music, a symbol placed at the beginning of the stave which assigns a pitch to a specific line on the stave, and by inference, to all of the other lines and spaces. Three clef symbols are commonly used, derived from the medieval forms of the letters G (&), F(?), and C(B).

A click track records a series of clicks, like a metronome, on one channel of a multitrack tape recorder or one channel on a MIDI sequencer. The click track is used to synchronize the recording of subsequent tracks by playing it back via headphones to the musicians while they are overdubbing the added tracks.

A distortion caused by cutting off the peaks of audio signals. Clipping usually occurs in an amplifier when its input signal is too high or when the volume control is turned up too high. A clipped waveform contains a great deal of harmonic distortion and sounds very rough and harsh. Hard clipping results in very sharp edges on the waveform, producing the maximum amount of high-harmonic content. Soft clipping produces rounded edges of the clipped waveform and is much less grating on the ears and tweeters than hard clipping as it contains much less very high-frequency energy. Different amplifiers produce different clipping effects; tube amps often produce soft clipping. See full code, digital black.

 (1) Any of several types of timing control devices, or the periodic signals that they generate. Clock pulses are usually derived from crystal-controlled oscillators. See also MIDI Clock, master clock. (2) In recording sessions for jingles or film scores, a stopwatch.

An artifact of digital-to-analog conversion that creates staircase-like changes in voltage produced by the converter. Most clock noise is caused by shifts in the zero-crossing times. See quantization noise, reconstruction filter.

See master clock.

The precision (measured in ppq) with which a sequencer can encode time-based information. A sequencer’s internal clock is always set to some ppq value, and this setting is one of the main factors that determine how precisely the sequencer can record time-dependent information. The actual clock speed is usually determined by the bpm setting. See MIDI clock.

 A closed-loop system is one that modifies its behavior based on the different behavior between an output variable and a set point, i.e., it relies on feedback to determine its output. The opposite of an open-loop, an example of a closed-loop system is a household thermostat. In audio, the main use for closed-loop systems is in power amplifier output stages. See bootstrap.

See Foley.

An internet communications network (either a Wide Area Network or a private network) in which a data center performs a range of services such as data storage or remote apps and programs (cloud computing) See WAN, LAN

 Constant Linear Velocity. As opposed to CAV, a mass storage system that has a disk whose speed varies to keep the data rate at the read/write head constant, regardless of the location of the data on the disk. The CD standard specifies CLV format.

Common Mode Rejection Ratio. In a balanced line connection, the common mode is the noise whose phase is common on both lines. The degree of attenuation of the common mode by a differential amplifier is called the rejection ratio, and like other level-modulating devices, the output change is measured in dB.

Speakers that have two separate drive units (normally a small woofer and tweeter) moving independently. Essentially this gives better high frequency response and hence greater range and clarity.

The phenomenon of human aural discrimination among sounds of equal loudness, e.g., the ability to hear one conversation out of many at a party. Related to auditory masking.

Musical symbol (C) which references a section of music which is to be repeated from or repeated to, as in "D.C. al Coda," meaning "play from the beginning to the Coda." A coda is a musical passage that gives a sense of completion to a movement or work, possibly an extended cadence or a substantial passage. A codetta is a short musical passage that links the subject and answer in a fugue, or at the end of the first passage of a sonata.

A display of the SMPTE timecode numbers, usually corresponding to each frame of picture viewed on a monitor. These numbers appear in a window that replaces a portion of the program image, usually at the bottom right of the screen. Depending on the equipment used, the timecode data in the window can be either generated in real-time from the source of the SMPTE timecode data, or have been recorded as a permanent part of the picture on a particular copy of a pre-recorded program. BITC is generally recorded onto copies of footage that will be used for off-line editing. The source of the SMPTE timecode data in this latter case is the sync track or control track of the original video footage.

COder-DECoder. A device that digitizes an input waveform, eliminating redundant information, reducing the number of bits needed to carry the same data, then decoding the data at the receiving end, hopefully with a high degree of sonic transparency. See PCM, PWM.

The process of altering the form of a signal, such as from analog to digital. Or, the coding may be used to allow a transformation of the signal not possible in its original form such as in noise reduction. Or, coding may be used to take advantage of effects inherent in the coding process itself, such as in PCM which allows recording and playback with low noise. The complementary stage to coding is decoding, a process which attempts to reconstruct the original signal, i.e., from digital back to analog for use with loudspeakers. See codec.

The magnetic field strength required to bring any specific type of recording tape, when fully saturated, to complete erasure. Measured in Oersteds, and abbreviated

The polarity relationship between two complex sounds or signals being combined, measured at any instant. Total phase coherence indicates complete phase alignment or full signal reinforcement. Incoherence, to any degree, designates a partial to complete phase difference, producing partial to total phase cancellation. The opposite of incoherent. See also isochronous

Also known as an X-Y or XY pair, this is a microphone configuration which commonly uses two cardioid or figure-eight microphones mounted at right angles to one another, the latter preferred only for special applications. This is called a coincident pair because the two microphones are mounted as closely as possible to each other so that the sound being captured arrives at both microphones at exactly the same time, regardless of the direction of the source. All coincident configurations have to use directional microphones in order to create the necessary level differences between the two channels of the stereo system; omnidirectional microphones do not produce level differences proportional to the angle of incident sound. This technique is favored by many broadcast applications because of good mono compatibility. Recording with a coincident pair is called XY recording in the US and the UK, AB recording throughout Europe, and also crossed pairs, or normal stereo. In the US, "AB recording" means a spaced pair. See also Blumlein pair, near-coincident pair. Contrast with spaced pair.

An analog video, composite video signal generated by a video-signal generator used to keep the chrominance subcarrier synchronized in a color television signal.  See videoblack.

Subtle distortion which results in a change in the timbre of a sound without that sound being otherwise noticeably distorted, such as a smearing type of distortion produced by intermodulation distortion. More prevalent at high audio frequencies.

A type of notch filter that produces a series of very deep notches, or dips, in its frequency response. The spacing of the notches along the frequency axis is at multiples of the lowest frequency notch. A comb filter is produced when a signal is time-delayed and added to itself. Frequencies where the time delay is one-half the period and multiples of these frequencies are cancelled when the signals are combined because they have opposite polarity, usually used to filter out 60Hz hum and its associated harmonics. If the signals are of equal strength, the cancellation is perfect and the notches are infinitely deep on a decibel scale. See common mode. Also called timbral interference cues.

An amplifier, also called a summing amplifier, combines two or more signals prior to sending them to a single audio bus, signal processor, tape recorder track, or other destination. For example, on a mixer, if an aux send controls on all channels of a console feed a combining amplifier, whose output can be routed to a reverb system, cue or headphone amp, the monitor amplifier, etc. There are also devices which are active combining amplifiers, called an ACA, as well as passive combining networks

 A typically passive network in which two or more signals are combined before being sent to a single bus, signal processor, or other destination.

A combination of loudspeaker(s) and amplifier in one unit, usually portable. Used by guitarists, keyboard players, etc. for stage amplification.

See diatonic comma.

A technical term for composite magnetic print.

 Referring to equal voltages induced in the two wires of a signal-carrying pair. In a balanced line circuit, the signal voltages are of opposite polarity in the two signal wires. Any voltage which appears with the same polarity on each wire is called a common-mode voltage. Usually noise, such as a 60Hz hum, is induced in audio cables equally and in the same direction, and so is a common-mode voltage. If the signal is connected to a differential amplifier input, the common-mode voltages will cancel, while the signal voltages, being of opposite polarity on each input terminal, will add together. This is the reason why balanced lines are less prone to induced noise from external influences. See CMRR.

The measurement of how well a balanced circuit rejects a common mode signal. See also CMRR

A technical term for composite optical print.

 Short for compressor/expander. A compander is a device for noise reduction in audio devices such as tape recorders. The compander will reduce the dynamic range of the signal before sending it to be recorded. The compression makes the softer passages louder so the dynamic range recorded on the tape is less than it would be if it were not compressed. Then, on playback of the tape, the signal is expanded; that is, the softer passages, which are too loud on the tape, are reduced in volume to match the original signal, restoring its dynamics. In the expansion, which is similar to a fast-acting AVC, the noise introduced by the tape recording process is effectively reduced because the music, when loud, masks the noise, and during the soft passages, the volume is turned down, making the noise comparatively softer. Digital companding allows a device to achieve greater apparent dynamic range with a lower bit depth. See dbxSeedBx, Dolby noise reduction.

An A/D-D/A pair which uses a non-linear scale, i.e., one that has larger steps towards peak amplitude and smaller steps towards minimum amplitude. This scale increases the ability of the converter to resolve small changes in low amplitude signals, reducing distortion, but with the penalty of increased noise. The overall effect is that of a compressed analog input signal and a resulting expanded digital output. See compander.

Companding noise reduction works by first sending the source matrial through a compressor, thus compressing the source material's dynamic range (in this case by a factor of 2) before being recorded to a medium known for noise such as magnetic recording tape. The compressed audio becomes contaminated with noise, but is passed through an expander during playback, and the noise from the recording medium is masked by expansion of the dynamics. This results in a reduction of perceived noise. See also dbx™ noise reduction.

 (1) The degree to which different pieces of equipment can be used together or are interchangeable, e.g., whether a tape recorded with one type of NR can be replayed on a tape player equipped with another type of NR. (2) See mono compatibility.

Any pair of audio signal processing procedures which perform two equal and opposite processes on the signal, one before recording, the other after playback. Noise reduction and tape recorder pre- and post-emphasis are examples. See encoding.

See 4-track.

 The overall frequency response modification produced when a signal passes through more than one equalizing circuit in the same device, or through several equalizers in a series.

Film print that contains a sountrack.

See time signature.

(1) The process of reducing the dynamic range of an audio signal by reducing the peaks so as to be able to boost the low levels. For every dB of compression applied, the S/N ratio is worsened by 1dB, assuming that the make-up gain is set so that the maximum levels of the compressed and uncompressed signals are the same, as the quieter parts of the original signal, plus any noise contained in these regions, will be raised in level. (2) A dynamic-range problem in loudspeakers caused by nonlinearity under conditions of high input power levels. At very high levels, the acoustic output increases more slowly or ceases to increase altogether as the input power increases, producing nonlinear distortion, i.e., a frequency response curve very different for very high levels. (3) Data compression used on digital audio files is a process ADPCM, MACE, for example. (4) The opposite of rarefaction whereby a quantity of data is reduced in order to occupy less storage space. See ATRAC.

A specialized mid- or high-frequency speaker consisting of a small diaphragm and voice coil coupled to a large magnet structure. The unit is mounted to a horn which acoustically matches the impedance of the driver to the impedance of the air and shapes the signal. Expensive due to the precise tolerances required, compression drivers are substantially more efficient than traditional direct-radiating cone speakers.

 (1) The ratio of the dB change from input level to output level effected by a compressor, once the threshold has been exceeded. (2) In data compression, the ratio of the number of bytes of uncompressed to compressed data, an indication of the space-saving efficiency of the compression algorithm.

A device for reducing the effective dynamic range of an input signal by preventing it from rapidly exceeding or falling below a selected amplitude threshold. The first part of a compander, it is used to make loud parts of a signal softer and soft parts louder. Beyond the threshold, the ratio of the signal’s input level to its output level (e.g., 2:1, 4:1) is user-selectable. A compressor is commonly used to keep mic levels within an acceptable range, but because it can slow a signal’s rate of decay below the threshold, compressors are also used to add sustain to instruments such as electric guitar and bass. The limiter acts like a compressor, but operates only at the top end of the dynamic range. The limiter has a faster attack time (1µs to 1ms) than the compressor alone (1ms to 10ms). A compressor/limiter is inserted between the outputs of a MIDI soundcard, synthesizer, or mixer and the inputs of the mixdown deck. See hard knee compression, soft knee compression, Limiter

 (1) A Salt Lake City-based company that makes portable wireless transmitters and receivers. (2) The generic name for wireless headphone feeds to directors and for wireless timecode feeds to slates.

Established by ISO in 1955, the agreed reference frequency of 440Hz, for the note A above middle-C, notated A=440

Established by ISO in 1955, the agreed reference frequency of 440Hz, for the note A above middle-C, notated A=440.

A condenser, or capacitor, mic capsule has a conductive diaphragm and a metal backplate placed very close to the diaphragm. They are charged with static electricity to form two plates of a capacitor. When sound waves strike the diaphragm, it vibrates, varying the spacing between the plates. In turn, this varies the capacitance and makes a signal analogous to the incoming sound waves. There are two types of condenser mics: the true condenser and the electret condenser. In the former, the diaphragm and backplate are charged with a voltage from a circuit. In the latter, the diaphragm and backplate are charged by an electret material, which is in the diaphragm or on the backplate. All true condenser mics need a power supply to operate, such as a battery or phantom power. In general, condensers have a smooth, detailed sound with a wide, flat frequency response--usually up to 15kHz-20kHz, useful for cymbals or instruments that need a detailed sound, such as acoustic guitar, strings, piano, or voice. Condenser mics tend to be more expensive and fragile than dynamic microphones. Note that omnidirectional condenser mics have deeper lows than cardioid condensers, making the former a good choice for pipe organs and bass drum. See also boundary microphone.

The reciprocal of resistance, or electric current divided by voltage. The traditional unit of conductance is the mho (ohm spelled backwards). See also impedance.

A material which exhibits efficient thermal or electrical transference through itself is said to have a high conductivity. Conductivity in a material is rated as its resistivity, the inverse of conductivity, in ohms per meter. Metals have a high conductivity owing to the large number of free electrons in metal atoms which efficiently transfer the current or heat from one part of the material to another. An insulator, on the other hand, is a material with few free electrons, and hence does not readily pass heat or current.

 The vibrating diaphragm of a dynamic or moving coil loudspeaker, usually made of paper and shaped roughly like a cone.

(1) To re-edit sound stems to match a new version of the picture edit, which is the final matching of all music, dialog and/or special effects to the video image. This may involve synchronization, editing, and re-recording one or more of the components of the final sound/video mix. (2) To assemble sound elements from their original sources to match their location in a picture edit, often with the assistance of an EDL.

See mixer.

 Literally "sounding together." Musical tones that are consonant sound harmonious or in tune when sounded together rather than discordant or harsh. Musical intervals composed of tones that have relatively simple frequency ratios are more consonant than ones with more complex ratios. The most consonant interval is considered to be the octave, which has a frequency ratio of 2:1.

See wrap.

 A chemical compound which, when applied to plugs, sockets, or other metallic electrical connection, improves the electrical conductivity between the metal surfaces, making a better, less noisy contact.

A mic that is physically attached to the body of an instrument or other sound source. It is primarily the vibration of the contact microphone’s body itself that is the transducer. By comparison, other microphones contain an internal diaphragm or membrane that vibrates in response to sound carried to it through the air, while the capsule of the microphone itself remains motionless. See also bug, piezo pick-up.

Convert sound waves in a dense medium (wood, metal, skin) into an audio signal. Sometimes used on acoustic stringed instruments such as guitar, mandolin, violin, etc. Usually of the crystal type, occasionally capacitive.

Film sound slang for Dolby Laboratories’ peak limiter designed specifically for controlling the dynamics of program material during SVA printmastering.

A MIDI Real-Time system message, correctly written Song Continue.

 A type of MIDI channel message that allows dynamic, real-time control changes to be made in notes that are currently sounding. There are 128 possible continuous controllers on each of 16 MIDI channels, and each of these controller types can have any data value between 0 and 127. Modulation (such as pan or volume) is an example of a true MIDI continuous controller. Continuous controller 1 is always the modulation wheel; controller 7 is the instrument’s main volume. See controller.

 A software feature where the DAW will create a new clock based on incoming SMPTE timecodetime code to enable recording to the DAW from an ATR. The result is that the sample rate of the DAW will vary continuously, effectively speeding up and slowing down to track the timecodetime code variations. Continuous sync requires dedicated hardware, and may not be available on all DAWS. For example, ProTools™ has a feature for continuous sync which is necessary when syncing continuously to an ATR while recording on any digital machine as the SMPTE timecodetime code-based clock is not guaranteed to be at the precise sampling rate. The ProTools Slave Driver™ does the sample rate conversion in the ProTools hardware so that the audio quality of the digital data isn’t compromised.

See envelope generator.

Also called PQ codes. In the CD format, eight additional bits are added to each frame of audio data; this means that a byte of information is available from the disc every 136 µs. Each bit in the added byte is given a one-letter name, P-W. Thus, eight separate subcodes can be recorded on and recovered from the CD. So far only P and Q are used: the P-code is used for the pause signal between musical tracks and at the end of the last track, and the Q-code tells the player if the recording is two- or four-channel (no quadraphonic CD player is yet available). The Q-code also contains timing information about the tracks and identifies the country of origin and date of the recording. No standard has been defined for the use of the other six subcodes.

The part of a synthesizer that tells the sound generators and controllers what to do to make a given note. These modules include envelope generators, LFOs, the keyboard itself, and the modulation and pitch-bend wheels. These allow control of some aspects of a synthesizer’s sound by sending signals to the sound generators and modifiers telling them now to behave. For instance, the keyboard sends a signal to the oscillator telling it what frequency to play. Also called modulation modules.

A file which becomes a part of the Mac’s system software, giving the user either control over or adding functionality to various aspects of the operating system, peripherals, or applications. See also extension.

 (1) One track of a multitrack magnetic tape recorder used for recording special signals that provide control information to the recording console during automated mixdown. (2) A dedicated track prerecorded with a pilot tone, used on video tape which marks the start of each video frame in order to resolve playback speed by controlling and synchronizing the video frames. Can be used to count time for editing, but is prone to slip and lose count during winding. In 1" and " formats, SMPTE timecode information is sent to a separate address track which creates confusion about the names of both of the tracks. See blacking, sync-locksync lock.

 A voltage, usually varying, used in synthesizers to control various parameters of the signal being produced. Control voltages are used for envelope control, pitch control, and filter bandpass and rolloff frequency control, etc. Suitable control voltages can be generated in various ways, one of the most straightforward of which is by a standard keyboard. See VCA, VCF, VCO.

(1) Any device, for example, a keyboard, wind synth controller, or pitch-bend lever, capable of modulating a sound by altering the action of some other device. (2) Any of the defined MIDI data types used for controlling the on-going quality of a sustaining tone via a controller message. In many synthesizers, the controller data category is more loosely defined to include pitch-bend and aftertouch data. See continuous controller.

A Channel Voice message which allows for musical effects such as vibrato or sustain on currently active voices.

A sequencer feature whereby whenever playback is requested, the sequencer looks back for the most recent controller, pitch-bend, aftertouch, and similar parameters and sets everything accordingly so that playback started in the middle of a song replays correctly.

 A sequencer feature whereby whenever playback is requested, the sequencer looks back for the most recent controller, pitch-bend, aftertouch, and similar parameters and sets everything accordingly so that playback started in the middle of a song replays correctly.

 (1) In any linear system or device, the output signal is a function of the input signal and the characteristics of the device. The interaction between the input and the device is described by a mathematical infinite integral called convolution. The output is the input convolved with the impulse response of the device. The spectrum of the output of a device is simply the spectrum of the input multiplied by the frequency response of the device via the FFT. (2) The modulation of one audio file by another. For example, the use of a hand-clap echo sample could be convolved with a guitar chord sample to produce an echo effect which sounds like it was produced by the guitar.

The process of re-recording or copying selected extracts from original sound or video recordings and rearranging their order as they are copied, so that the copy will have all the desired segments in the correct order. This copy is called an assembly, and will generally need fine editing in order to meet timing or other production requirements.

An audio interface driver protocol allowing audio interfaces to connect to and be controlled by a computer, created by Apple.

See rolloff frequency.

See distortion.

See figure-eight microphone.

UK slang for cardioid microphone. See supercardioid or hypercardioid microphone.

The coulomb is the unit of electric charge (C), and is the quantity of electricity transferred in one second by a current of one ampere.

A slang term for footage numbers or cues for specific events in a film or videotape. Also called footage counts. See feet/frames.

The process of or means by which energy is transferred from one system or medium to another. For example, the coupling of acoustic energy from a loudspeaker to the surrounding air.

 (1) centimeters per second. The speed of movement of tape past a tape read/record head, also denominated in ips, inches per second. (2) cycles per second, or Hertz.

 Central Processing Unit. The brain of the PC. Its clock speed determines how many software plug-in effects and soft synth notes can be run simultaneously in real time. The two most popular manufacturers of CPUs are AMD and Intel.

The CPU generates a significant amount of heat when it’s running, so to ensure that it doesn’t overheat its normally fitted with a finned metal heat sink to help dissipate the heat, and some sort of fan that generally blows cool air onto the heat sink. The standard fans can be quite noisy, which is not good in a recording studio. To keep down the noise you can purchase a replacement fan that is not as noisy.

 Cyclic Redundancy Check. A system of recording a checksum number along with data in order to detect, and in some cases, correct any corruption of the data. See ECC.

 A gradual increase in loudness of a musical sound.

The ratio between the average amplitude as shown on a VU meter and the instantaneous amplitude as shown on a peak meter. The human ear is very sensitive to this difference.

The distance from a loudspeaker where the direct sound is equal in intensity to the reverberant sound. See also free-field.

A velocity threshold effect in a synthesizer in which one sound is triggered at low velocities and another at high velocities, with a fade-out/fade-in transition between the two. If the transition is abrupt rather than gradual, the effect is called cross-switching rather than cross-fading. Cross-fading can also be initiated from a footswitch, LFO, or some other controller.

A sample-editing feature found in many samplers and most sample-editing software, in which some portion of the data at the beginning of a loop is mixed with some portion of the data at the end of the same loop, so as to produce a smoother transition between the end of the loop and the beginning of the loop replay.

See interleaving.

Cross-modulation test. A means of determining correct exposure on a track negative to result in minimum distortion on a positive print. Tests are conducted to determine the relationship of specific optical cameras to specific laboratories.

A velocity threshold effect in a synthesizer in which one sound is triggered at low velocities and another at high velocities, with an abrupt transition between the two. If the transition is smooth rather than abrupt, the effect is called cross-fading rather than cross-switching. Cross-switching can also be initiated from a footswitch, LFO, or some other controller. Also called velocity-switching.

A velocity threshold effect in a synthesizer in which one sound is triggered at low velocities and another at high velocities, with an abrupt transition between the two. If the transition is smooth rather than abrupt, the effect is called cross-fading rather than cross-switching. Cross-switching can also be initiated from a footswitch, LFO, or some other controller. Also called velocity-switching.

See coincident pair.

 A type of distortion present in some amplifiers which increases for low-level signals. In many amplifiers, the output devices are connected so that one of them is active during the positive half of the waveform, an the other one is active for the negative half. There is a region near zero current where the signal is transferred from one to the other. If this is not done smoothly, there will result a small discontinuity in the output waveform. This discontinuity causes higher-order harmonic distortion, and being constant in value, is more noticeable with low-level signals than with stronger ones. See crossover frequency.

The frequency above and below which an audio signal is divided into two bands, each of which is directed to a separate destination. Precisely, the frequency at which each of the two bands is attenuated 3dB by the crossover network.

A dividing network that splits a full-spectrum signal into two or more frequency bands and routes them to feed the various components of a speaker system. Passive crossover is a speaker design whereby multiple speakers are in a single enclosure and sound frequencies are separated and sent to be driven by the appropriate speaker. A two-way biamp monitoring system has a high-frequency loudspeaker (tweeter) and a low-frequency speaker (woofer). A three-way triamp system has a third speaker to reproduce midrange frequencies. Active crossovers divide a line-level output signal from a mixer or other sound source and route the resulting signals to individual amplifiers optimized for the different speaker components.

In multichannel audio transmission systems, such as tape recorders, record players, or telephone lines, a signal leaking from one channel to one or more of the others is called crosstalk. Also known as bleeding. See channel separation.

A system for generating a sync signal that will ensure proper synchronization of film footage and its corresponding sync sound without using any sync reference, such as 60Hz AC line frequency. Two piezoelectric crystals, each tuned to the same high frequency, are installed in the camera and recorder. The crystal in the camera precisely controls the motor speed during shooting. The crystal in the recorder produces a pilot tone that is recorded on tape in the same way which the camera motor speed would be recorded through the conventional sync cable. Because the two crystals are tuned identically, the dailies, when synched with the magnetic film copy of the original sound takes, will maintain perfect sync once the slate marks are aligned. Crystal sync generators are also installed in portable video cameras and VTRs. See Nagra, neo-pilot.

Clock time. See RDS.

 (1) (noun) A piece of music for a specific scene or event in a film. A specific part of a film soundtrack which correlates to a visual event in the film is called a cue point or hit point. (2) (verb) To position a sound source to activate at a specific time. See Real-Time MTC Cueing. spot.

A wall-mounted or movable box that receives one or more monitor mixes from the recording console, and that has jacks to plug in several sets of headphones used to send the cue mix to singers or instrumentalists in overdubbing, a narrator, or other studio talent. Also called a headphone box. See cue system.

A line drawn on the workprint, meant to be seen during projection in post-dubbing or scoring, which gives the actor or conductor a visual cue to begin.

 A list of the footages and frames, beginning with 00:00, at which specific shots begin and end. Used by the re-recording mixer who needs to know which sounds or music must be played as the final mix proceeds. See also edit decision list, feet/frames.

The blend of live inputs and/or previously recorded tracks sent by the mixing engineer to the headphones of performers playing or singing in the studio. Also called the headphone mix.

A tape machine operating mode in which the tape lifters are defeated while the playback electronics remain operative. Used most often during editing, thus also called edit mode.

See Real-Time MTC Cueing.

 A track sheet for mixing that gives locations of edited sounds on a track-by-track bases, either in film footages or in timecode numbers. See binky.

The entire electronic circuitry contained within the recording console that allows the engineer to adjustably feed sound from any input module to the cue mix, then out to the musicians’ or singers’ headphones via the cue amp and cue boxes.

The entire electronic circuitry contained within the recording console that allows the engineer to adjustably feed sound from any input module to the cue mix, then out to the musicians’ or singers’ headphones via the cue amp and cue boxes.

The entire electronic circuitry contained within the recording console that allows the engineer to adjustably feed sound from any input module to the cue mix, then out to the musicians’ or singers’ headphones via the cue amp and cue boxes.

A track of recorded music and/or clicks which are sent over headphones to the musicians and/or singers to assist them in overdubbing additional music/vocals. If the track is simply tempo clicks, it is known as a click track.

To locate a desired point, at which a specific sound event happens on a reel of tape, and to position that point just ahead of the playback head on a tape recorder. When playback begins, the desired sound will be heard immediately.

 The flow of electrical charge measured in amperes.

(1) (verb) To attenuate amplitude of a signal or particular frequency band; the opposite of boost. (2) To produce the master for a vinyl LP. (3) In film, the instruction used to terminate filming. (4) (noun) A musical selection on a record, tape, or CD, or a particular edited version of film.

On a hard-disk audio editing system, a term used to denote the ability of an audio editing program to move and/or copy sections of the recorded audio to a new location in the track or to other tracks.

 Sound effects that are taken from a sound library and edited, usually as opposed to recorded Foley effects. See pull, M&E.

A switch or button that mutes an audio signal on a mixer.

An edited track of a film soundtrack which is ready to be used either as a track in a premix (music or effects), or as a (dialog) track in the final mix.

 See rolloff frequency.

 Sound editor.

Da Capo. Italian for "Head," meaning "Play from the beginning."

See D-sub(miniature) connector.

 Dal Signo. "Play from the sign(D).

Also called a D-connector, or a D-type connector. A type of connector commonly found on computers and data transmission devices, including SCSI devices and computer monitors. D-type connectors have a "D-shaped" angled housing, and have 9-pin, 15-pin, and 25-pin configurations, designated DE-9, DA-15, and DB-25, respectively.

 See D-sub(miniature) connector.

(1) Digidesign Audio Engine.™ A Macintosh application that can run behind other applications, such as sequencers or ProTools,™ handling the transfer of audio data to/from the hard disk. DAE is licensed by many sequencer developers to avoid writing their own low-level I/O code. (2) Digital Audio Extraction. The process of capturing CD-Audio tracks digitally from a CD-ROM drive to hard drive, using software such as Astarte’s CD Copy™ or OMI’s Disc-to-Disk.™

 Uncut footage shot each day during production. If picture editing is on film, with picture and synchronized magnetic film, those elements when edited together become the workprint and worktrack. Used to chart the progress of the film and for preliminary music cuts. Also called rushes.

 (1) Damping is the addition of friction to a resonance in order to remove energy from a mechanical system, reducing the magnitude of vibration at resonant frequencies. For example, the reduction of movement of a speaker cone, due either to the electromechanical characteristics of the speaker driver and suspension, or the effects of pressure inside a speaker enclosure. The electrical analog of friction is resistance, and it is used to damp resonating electrical circuits, such as crossover networks and filters. See also Q. (2) Acoustic fiberglass material used inside speaker enclosures.

 (1) A factor defined as the rated load divided by the amplifier output impedance. (2) The ability of an amplifier to control the motion of a loudspeaker cone after a signal disappears, i.e., its ability to defeat the natural ringing tendency of the body (cone) in motion. An amplifier with a high damping factor looks more like a kind of short circuit to the speaker, reducing its vibration when the signal stops. The damping factor of an amplifier will vary with frequency, and sometimes a manufacturer will publish a curve of damping factor vs. frequency. The effect of high damping factors is most audible at low frequencies, where the primary resonance of the woofer cone, called hangover, is reduced in level.

An audio-over-ip format which gives low-latency multichannel audio the ability to be available over ethernet by taking digital audio and turning it into packets containing information for the audio’s source, destination, network address, and timing.

See Digital Audio Recorder. Any type of audio recording system which records upon a digital medium, such as DAT or hard disk. DAT or DCC recorders, digital dubbers, digital multitracks, and hard-disk recording systems are all example of digital audio recorders. These recorders are an alternative to analog recorders, such as traditional cassette or reel-to-reel formats which do not convert the waveform to a digital representation prior to writing it to the recording medium.

The amount of low-frequency, or corresponding lack of high-frequency, components of a sound. Reverberation from distant objects usually has fewer high frequencies and sounds darker than reverb from close objects. The opposite of brightness.

 Digital Audio Stationary Head. A standard format for ensuring compatibility between Sony PCM-model digital multitrack recorders which that use stationary, rather than rotating heads. Originally, the DASH format was designed to support 2-track, 8-track, 16-track, and 24-track recorders using reel-to-reel tape. The 8-track and 16-track machines were never marketed, and a 2-track model is no longer in production. The DASH specification now includes double-density, thin-film heads that allow 48-track recording on the same tape originally used by the 24-track devices. DASH-format machines are backward compatible: 24-track machines can be used with newer models, and a project can be started on a 24-track machine and completed on a 48-track recorder, if needed, as the data from tracks 25-48 are written into the spacing between the original 24 tracks. DASH tapes run 30ips at up to 48kHz, with 44.1kHz and other sampling rates supported. In addition to 24-track or 48-track recording, DASH format provides two analog cue tracks and one track each for control and timecode signals. The format covers a wide range of versions, such as from 2-48 tracks and tape speeds from 12-76cps, and was agreed on by Sony and Studer, among others. DASH format recorders are currently manufactured by Sony and Studer. See also S-DAT, ProDigital.

 Digital Audio Tape. There are two formats: R-DAT which uses a rotating head assembly similar to a VCR, records diagonally across the tape, and includes a four-channel format which would permit recording of ambisonics; and the S-DAT which uses a stationary head and records several linear, parallel tracks of digital signals. There are no known commercial S-DAT products. The DAT standard format specifies a small cassette that provides up to two hours of 16-bit, linear, sequential monaural, PCM digital recording at a sampling rate of 32kHz, 44.1kHz, or 48kHz. Also called a DCAC, Digital Compact Audio Cassette. See also Digital Compact Cassette.

See compression(3).

 A controller change message which is used to set some parameter in the receiving device, for example, the data increment and decrement switches on a synthesizer.

A packet of memory contents being transmitted from a sending device to a receiving device, usually in the form of MIDI System-Exclusive data, or stored in RAM. Also called a bulk dump or block transfer.

A pot fitted to a device such as a synthesizer which allows parameters within the device to be adjusted for programming, etc.

 A sequencer software feature which allows programs and/or devices to reduce the amount of MIDI data produced by continuous controllers such as pitch-bend, aftertouch, etc. This is accomplished by only keeping the continuous controller data when the parameter changes, as opposed to sending all bytes of data all of the time.

A portable device for the replay of previously recorded MIDI data, used in live performances.

Digital Audio Workstation. See workstation.

See decibel.

An industry-standard connector for serial machine control of professional audio and video transports. Developed by Sony, also called the Sony 9-pin.

Direct Broadcast Satellite. See AC-1.

a noise reduction system that uses a companding noise reduction to reduce noise. Dbx is connected into a recording system in the same way that a Dolby system is. It provides up to 30dB of noise reduction, but unlike Dolby noise reduction, the dbx system works over the entire audio frequency range, using a 2:1/1:2 compression/expansion ratio. Dolby-encoded and dBx-encoded tapes are incompatible. Systems using dbx noise reduction are typically more expensive than systems using Dolby.

 Digitally Controlled Amplifier. Sometimes short for Digitally Controlled Attenuator. The DCA of a digital synthesizer modifies the amplitude of the signal generated by the DCO. It is the digital analog of a VCA.

 Digitally Controlled Amplifier. Sometimes short for Digitally Controlled Attenuator. The DCA of a digital synthesizer modifies the amplitude of the signal generated by the DCO. It is the digital analog of a VCA.

See Digital Compact Cassette.

Digitally Controlled Oscillator. The microprocessor-controlled sound generator used in a digital synthesizer. The DCO directly generates the original signal that is used as the fundamental for the sounds created by the synthesizer. The keyboard tells the DCOs what pitch to produce; the audio signal may then be altered by sound modifiers, including a DCW, DCA, differentiators and integrators, and various modulators and limiters. The digital equivalent of the analog VCO.

Digital Delay Line. See delay line.

The complementary equalization which follows pre-emphasis. Sometimes redundantly called post de-emphasis.

A special type of compressor that operates only at high frequencies, usually above 3kHz-4kHz. It is used to reduce the effect of vocal sibilant sounds. De-essers are usually used only for vocal music.

Acoustically absorptive. The opposite of live. See also LEDE.

The interval between two quantities plotted along an axis where the second quantity is ten times the first. A frequency ratio or interval of 10:1, as opposed to an octave, which is a 2:1 ratio. Sometimes the rolloff of a filter or equalizer is expressed in dB/decade, rather than in dB/octave. A rolloff of 20dB/decade is equal to A rolloff of 6dB/octave. The decade interval has no musical significance, but is used in the discussion of logarithmic quantities such as decibels.

 (1) The time it takes for a sound to reach minimum loudness; the end of a sound. (2) The second of the four segments of a typical ADSR envelope. The decay control determines the amount of time it takes for the envelope to fall from the peak reached at the end of the attack segment to the sustain level. If no additional energy is put into the sound source (e.g., a cymbal), then the decay is the time during which the sound falls from the loudest point back to silence. (3) The time taken for reverberation to die away. See decay time, reverberation time,RT-60.

 The number of decibels per second by which echoes or reverberation of a sound diminish once the sound has stopped. Depending on the sound source and environment, the decay rate may be linear, i.e., a constantly decreasing number of dB per second, or it may begin to decay slowly and then fall off rapidly, or the reverse. Also, various frequencies of the sound may decay at different rates.

See reverberation time.

 A triangular array of omnidirectional microphones , a type of true spaced-microphone recording technique, where the central channel is distributed equally to left and right. This yields a very stable central image, avoiding the hole-in-the-middle which is problematic with many space-pair arrangements. A variant on the Decca tree places three cardioid microphones (L,C,R) in a triangle configuration. In all cases, the width of the tree is typically one-half to one-third the sound field width, and the center microphone is slightly closer to the performers. See binaural recording.

 A unit of measurement used to indicate audio power level, literally one-tenth of a bel, where the bel is a power ratio of 10:1. Technically, a decibel is a logarithmic ratio of two power measurements, which means that there is no such thing as a dB measurement in isolation. In order to measure a signal in dB, you need to know what power (watts, volts) it is referenced to and the impedance of the reference system. Number of dBs = 10 log (P1/P2), where P1 and P2 are the two powers being compared, and where the log is base-10. Imprecisely, 1dB is the smallest increment in loudness detectable by a careful listener. An increase of about 3dB is a doubling of electrical (or signal) power; an increase of 10dB is ten times more power, but is only a doubling of perceived loudness. Some commonly used power ra-tios, expressed in dB: However, Number of dBs = 20 log (V1/V2), where V1 and V2 are the two voltages being compared, and where the log is base-10. This means that the answer is twice what it would be for a ratio of powers. In other words, dou-ble the voltage and the level goes up by 6dB; halve the voltage and the level goes down by 6dB. See Appendix A.

A form of digital filtering whereby audio data is oversampled and then decimated to the required 44.1kHz. In practice, the sampling rate is 64 or 128 times 44.1kHz. A digital brick-wall filter is then applied to the data, resulting in a perfectly phase linear transformation. [This type of filter is impossible in the analog domain due to the phase-shift caused by very steep roll-off filters.] See FIR, IIR. After the data have been filtered below the Nyquist frequency, the next step is decimation where the data are re-sampled to produce an output stream of 44.1kHz, with the attractive result that the excess data thereby provides increased bit-resolution. See anti-aliasing filter, reconstruction filter, DSD.

In a tape recorder transport, the heavy metal plate on which the headstack, rollers, and other transport components are located.

(1) In signal processing, restoring a signal to its original state by reprocessing the signal in a complementary manner, e.g., a NR system’s re-expansion of the signal during playback. (2) In digital recording, the entire process converting the encoded data stream back into an analog signal, including the process of error correction, i.e., digital-to-analog conversion.

 A mathematical process for separating two signals that have been convolved. See convolution.

A musical term indicating a gradual reduction in loudness.

A musical term indicating a gradual reduction in loudness.

A control that can be used to mute a signal on a mixer.

 A qualitative term that denotes the clarity of a sound. A sound with poor definition may, like some woodwinds in their middle ranges, be easily mistaken for a similar sound. In recording, the apparent definition of a sound can be increased by boosting the frequency band characteristic to the specific sound of the instrument, and cutting other frequencies it has in common with other sounds in the mix.

 (1) The first stage of a five-stage D(elay)AD(ecay)SR envelope, which delays the beginning of the envelope’s attack segment. See ADSR. (2) An audio effect which temporarily suppresses the beginning of a sound, producing echo, chorusing, phasing, and flanging effects. A modulated digital delay effect which varies the time and/or intensity of the delay effect over time. See double tracking. (3) A signal processor used for flanging, chorusing, and echo, that holds its input for some period of time before passing it to the output, or the algorithm within a signal processor that creates delay. Also used in artificial reverberation systems and to provide delayed sound to certain loudspeakers in time-coherent sound reinforcement systems. (4) See MIDI delay.

Used to simulate an acoustic echo or reverberation. There exist both digital delay lines (DDL) and analog delay lines as well. The original delay lines were made by using tape recorders to record a signal while playing it back on the same machine. The distance between the record and reproduce heads causes a time delay; this technique is called tape delay. See tape delay.

Used to simulate an acoustic echo or reverberation. There exist both digital delay lines (DDL) and analog delay lines as well. The original delay lines were made by using tape recorders to record a signal while playing it back on the same machine. The distance between the record and reproduce heads causes a time delay; this technique is called tape delay. See tape delay.

A type of modulation which creates a series of echoes when the modulation source is boosted. The greater the amount of feedback, the more repetitions of each echoed event.

 In the UK, often, and more properly called delta-sigma modulation. A type of PCM which differs from most other digital encoding schemes in that the signal, after being sampled at a fast rate, is encoded as the difference between successive levels, rather than as the absolute level of each sample. Delta modulation requires a very high sampling rate, usually around 700kHz, but the digital words need for each step contain one bit, whereas conventional PCM samples at only about 45kHz but requires 14-16 bit words. The "delta" phase of delta modulation involves taking the difference of the reconstructed signal and the incoming signal to adjust the output to minimize the quantization error; the "sigma" part involves the summation of the differences to reconstruct the original signal, although there are a number of variant algorithms based on this basic theme. The reason for the popularity of delta modulation-type converters is the inherent linearity of the process. See also ADPCM.

In the UK, often, and more properly called delta-sigma modulation. A type of PCM which differs from most other digital encoding schemes in that the signal, after being sampled at a fast rate, is encoded as the difference between successive levels, rather than as the absolute level of each sample. Delta modulation requires a very high sampling rate, usually around 700kHz, but the digital words need for each step contain one bit, whereas conventional PCM samples at only about 45kHz but requires 14-16 bit words. The "delta" phase of delta modulation involves taking the difference of the reconstructed signal and the incoming signal to adjust the output to minimize the quantization error; the "sigma" part involves the summation of the differences to reconstruct the original signal, although there are a number of variant algorithms based on this basic theme. The reason for the popularity of delta modulation-type converters is the inherent linearity of the process. See also ADPCM.

See delta modulation.

See SMF.

A device that recovers the audio signal from a modulated carrier waveform. Also called a detector. See amplitude modulation and frequency modulation.

(1) In stereophonic reproduction of music, depth refers to the perceived relative distance between the listener and the various instruments in the sonic image. (2) In a digital delay or flanger, a parameter which modulates the length of delay around the specified delay time. Because this happens in real-time, the pitch of the input signal is varied, causing the output signal to have an apparent vibrato effect. The speed of this vibrato is set by a rate control.

See depth (1)

Dolby-Encoded Stereo. A noise reduction system employed in the reproduction of stereo optical tracks in movie theaters.

See DAW, mixer.

(1) See frequency modulation. (2) See level-sensing circuit.

(1) (noun) A control that allows one oscillator to sound a slightly different pitch than another. (2) (verb) To slightly change the pitch of one oscillator relative to another, producing a fuller sound.

Also called a direct box. (1) The use of some form of mechanical or electrical pick-up mechanism on an instrument for the purpose recording or amplification. DI also refers to the connection of an electronic keyboard or power amplifier feeds to a mixer. A DI consists of (usually) a small electronic box into which an instrument is plugged and the electroacoustic pick-up attached to the instrument itself. Pick-ups can be electro-magnetic, as on electric guitars, piezo devices, and also contact microphone, also called bugs. All types of pick-up have unbalanced outputs at mic-level (~-50dBu), so the DI box has to balance the signal and drive it to the mixing desk. DIs can be passive or active in the typical sense. Active DIs have some form of electronic amplification built-in; this is only a buffering amplifier, separating the instrument pick-up from the rest of the DI, yielding no significant gain. Active DIs offer better sound and playability over passive devices, but require batteries, phantom power, or some other means of powering the internal amplifier. (2) Any device used to convert unbalanced lines to balanced lines.

There is a wide difference in the apparent loudness between different TV programs’ audio content. In DTV, with by standard is AC-3 encoded, a program producer chooses one of 31 different dialog normalization (abbreviated DN or "Dialnorm") values and this parameter is carried within the AC-3 datastream, where each step represents a 1dB change in level. The DN value is the difference in dB between the maximum level possible (0 dBFS) and the average loudness level of the program material. The smaller the difference between the maximum and program average levels, the lower the DN value is assigned. The lower the DN value, the lower the output volume of the AC-3 decoder is set in direct proportion, meaning that subjectively louder programs will be played back at lower volumes than those in which the average program level is less loud. This supposedly will obviate the user having to adjust the volume control between programs, once the audio listening level is set by the user.

The edited track on magnetic film containing the dialog portion of a film’s sound. Sometimes there may be a separate track for each actor in a scene, requiring the tracks to be mixed down to a single track. The "D "part of DME.

 The membrane part of a microphone’s capsule or cone of a loudspeaker that moves in response to sound waves or an incoming signal, respectively.

A musical scale of eight notes spanning one octave, consisting of an ascending pattern of two whole-steps, a half-step, three whole-steps and another half-step. There are two types of diatonic scale in common use in western music: the diatonic major scale and the diatonic minor scale. Music which includes notes outside of the diatonic in which the piece is written is said to be chromatic.

After playing the Circle of Fifths, i.e., twelve ascending perfect fifths, followed by seven descending octaves, the pitch discrepancy between the ending note and the starting note is called the diatonic comma, or the Comma of Pythagoras. This discrepancy amounts to a little over 1%, or about one-sixth of a half-step and gives rise to various temperaments in an attempt to distribute the error as harmoniously as possible. See scale construction, syntonic comma.

After playing the Circle of Fifths, i.e., twelve ascending perfect fifths, followed by seven descending octaves, the pitch discrepancy between the ending note and the starting note is called the diatonic comma, or the Comma of Pythagoras. This discrepancy amounts to a little over 1%, or about one-sixth of a half-step and gives rise to various temperaments in an attempt to distribute the error as harmoniously as possible. See scale construction, syntonic comma.

Dichotic generally refers to headphone listening where each ear hears a different signal, as opposed to diotic, where both ears hear the same signal. See also monotic.

A tone produced by combining two tones which are not part of a harmonic series, having a frequency difference of 20Hz or greater. Any slower than 20Hz, and the difference of the two notes will be perceived as a pulse, called beating. Also called a resultant tone. Also called a Tartini tone.

Usually one of the signal input terminals of an amplifier is connected to the chassis of the amplifier, i.e., it is grounded. The amplifier is then sensitive to the voltage difference between the input terminal and ground. However, in a differential amplifier, neither input terminal is grounded. Instead, the amplifier is sensitive to the voltage difference between the two inputs. Used in professional microphone preamplifier where a low-level signal has to go some distance, a differential amplifier cancels the hum induced by the proximity of the two input wires to a source of interference. In the UK, a differential amplifier is called an inverting amplifier. See differential input, common mode.

Signal input response to amplitude differences between two out-of-phase signals. Used in a balanced wiring system where the two wires carry signals that are identical, but 180° out-of-phase. The phase difference means that as a signal increases in voltage along one line, its mirror image on the other line decreases. This is useful because signals, such as hum and noise which have accumulated along a cable acting as an antenna, that are in phase are cancelled. See common mode, differential amplifier.

The output of an amplifier designed to provide two signals that are completely identical, but of opposite phase.

A highpass filter which can accentuate the higher-frequency harmonics and transients of a sound envelope. Compare with an integrator module.

The bending of a sound wave around an obstacle and the reflection of a sound wave from an obstacle in its path are called diffraction. It is frequency dependent. Where the wavelength is short compared to the obstacle, reflection will occur as well as bending of the wave front. When the wavelength is long with respect to the obstacle, little reflection will occur and the bending will be more pronounced. See also refraction.

In audio, the opposite of analog. The representation of audio or video as a series of encoded binary amplitude values, rather than as a continuous waveform.

A consumer audio device that attempts to simulate the acoustics of an auditorium or other room by adding suitable time delays and synthetic reverberation to recorded signals.

The application of digital technology to the recording, processing, and reproduction of music is somewhat loosely called digital audio, as opposed to analog.

An alternative to AM and FM broadcasting with audio quality comparable to that of the CD, it does not suffer from fringe area fading or multipath distortion, and requires less radiated power than conventional broadcasting (1kW versus 50kW for AM and up to 100kW for FM.)

See grabbing

See Digital multitrack.

 Any type of audio recording system which records upon a digital medium, such as DAT or hard disk. DAT or DCC recorders, digital dubbers, digital multitracks, and hard-disk recording systems are all example of digital audio recorders. These recorders are an alternative to analog recorders, such as traditional cassette or reel-to-reel formats which do not convert the waveform to a digital representation prior to writing it to the recording medium.

Any type of audio recording system which records upon a digital medium, such as DAT or hard disk. DAT or DCC recorders, digital dubbers, digital multitracks, and hard-disk recording systems are all example of digital audio recorders. These recorders are an alternative to analog recorders, such as traditional cassette or reel-to-reel formats which do not convert the waveform to a digital representation prior to writing it to the recording medium.

 In digital audio, a term which means complete silence. Digital black is calculated by taking the sample word length (e.g., 16, 20, or 24 bits) and multiplying this bit depth by 6dB, a number which represents the dynamic range represented by one bit. In a 16-bit system, for example, full code represents 96dB, the maximum amplitude that the system is capable of encoding without clipping. Digital black is at the opposite end of that dynamic range, or 96dB down from full code amplitude.

A type of recording format announced by Philips in 1990, designed to compete with the R-DAT format. The system allows for the recording and playback of analog cassettes as well as DCCs on the same machine. Uses PASC (Precision Adaptive Subband Coding), derived from the MPEG-1, Layer 1 data reduction system to provide data compression (lossy) for the recording of digital audio on wide magnetic tape at ips. This format has not been widely adopted. Sometimes called DCAC for Digital Audio Compact Cassette. See also MiniDisc, DAT, and CD.

A type of recording format announced by Philips in 1990, designed to compete with the R-DAT format. The system allows for the recording and playback of analog cassettes as well as DCCs on the same machine. Uses PASC (Precision Adaptive Subband Coding), derived from the MPEG-1, Layer 1 data reduction system to provide data compression (lossy) for the recording of digital audio on wide magnetic tape at ips. This format has not been widely adopted. Sometimes called DCAC for Digital Audio Compact Cassette. See also MiniDisc, DAT, and CD.

Film industry term for a multitrack digital recorder, usually having eight tracks per unit, that use removable hard drives or magneto-optical drives as the recording medium. The term is partly a misnomer because previous film sound terminology had used dubber to describe a copying device as opposed to a recording device.

A device for recording multiple channels of digital audio data at various sampling rates. Two formats have survived: the Sony/Studer DASH format and MDM machines of either ADAT or DTRS type. The first digital multitrack recorder was introduced in the late 1970s by 3M, a 32-track recorder called the Digital Audio Mastering System.

A device for recording multiple channels of digital audio data at various sampling rates. Two formats have survived: the Sony/Studer DASH format and MDM machines of either ADAT or DTRS type. The first digital multitrack recorder was introduced in the late 1970s by 3M, a 32-track recorder called the Digital Audio Mastering System.

The manipulation and modification of signals in the digital domain, possibly after having undergone analog-to-digital conversion.

See DTV

DTV’s audio specification provides up to six discrete channels of 5.1-format audio, where the LFE channel is band-limited to 25Hz-120Hz. DTV has been developed specifically for the home theater market, as an improvement to the ProLogic system.

See delay.

Commonly abbreviated D/A, D/A converter or DAC. A device that changes the sample words put out by a digital audio device into analog fluctuations in voltage that can be sent to a mixer, amplifier, or speaker. All digital synthesizers, samplers, and effects devices have DACs at their outputs to create audio signals, as the transducers in loudspeakers are inherently analog devices.

The solution for a piracy and duplication protection scheme developed jointly by Sony and Philips which writes copyright data encrypted within the CD/DVD etc. disc itself. This scheme would, for example, encode discs with a country code so that these discs would only play on players from the same country. This is presumably better than older forms of digital copy protection which tried various pilot tones or random number generators, failing ultimately because the results were either too audible or too easy to circumvent. In a digital watermark, the copyright data are stored as a modulation of the width of the injection-molded pits. Duplicating the watermark would require the same equipment as that which produced the disc stamper, the distribution of which is presumably tightly controlled. It is also possible to synchronize the modulation of the pit widths so that there is a visible pattern formed on the disc pit substrate itself, making an "analog" watermark (without the need for water, of course.) In addition to the watermark and country codes, identifiers for the mastering house and pressing plant, glass master number, ISRC catalog numbers, etc. can be stored. The digital watermarking technology has been called Pit Signal Processing (PSP) which works by modulating the strength of the laser used to record the digital data onto the glass master. One by-product of the watermarking process is that the EFM used to encode audio data onto the CD master allows the pits to vary in length between 3-11 units. These slight errors in length, or "jitter" result in slight timing errors which can cause a smearing of the stereo image as well as an increase in HF noise. The more rigid requirements of pit length control in watermarking should result in a significant reduction of pressing-induced jitter, just generally improving the CD production process.

A 5.1-format theater surround-sound system which uses six A six-channel discrete analog channels and for surround-sound on a CD-ROM interlocked to either a 35mm or a 70mm print with timecode. Lossless. The DTS codec provides for data rates from 256kbps to 1536kbps, focusing on 1141kbps as the optimum for transparent sound quality. DTS was originally developed for the film industry, however, there are a number of CD titles currently released in DTS format. A DTS CD carries six channels of digital audio in 5.1 format in 20-bit words at a 44.1kHz sample rate, with a compression ratio of about 3:1. An additional decoder is needed to play a DTS CD on a standard CD player. First used in 1993 for the film, "Jurassic Park." See also Dolby Digital Surround Sound, Dolby Stereo.

See DCA.

The microprocessor-controlled sound generator used in a digital synthesizer. The DCO directly generates the original signal that is used as the fundamental for the sounds created by the synthesizer. The keyboard tells the DCOs what pitch to produce; the audio signal may then be altered by sound modifiers, including a DCW, DCA, differentiators and integrators, and various modulators and limiters. The digital equivalent of the analog VCO.

A DCW varies the timbre of synthesized sound by modifying the harmonic content of the tone produced by a DCO. See waveform.

Synonym for decrescendo.

 (1) The reduction of a major or perfect interval by one half-step to make a diminished interval. (2) The appearance of a musical idea in note durations that are shorter than those used for its first appearance. The opposite of augmentation. (3) A method ornamentation where notes of long duration are broken into a number of shorter notes, often at different pitches, e.g., a trill.

 Deutsche Institut für Normung. A German standards organization that proposed a set of connector configurations in the early 1960s. The standard MIDI connector is the 5-pin DIN where:

See pilot tone.

A circuit element that will pass current in one direction only, from the anode (positive) to the cathode (negative). Used to make DC from AC.

Literally, "with two ears." Diotic generally refers to headphone listening whereby the two ears hear the same signal, as opposed to monotic, where only one ear hears the signal. See also dichotic.

A parametric equalizer with an extremely narrow Q, designed to remove noise in a small band such as that from a camera or light.

In loudspeaker design, a dipole radiator is a system which radiates forwards and rearwards with equal energy, but with opposite polarity. Examples of dipole radiators are electrostatic loudspeakers and planar speakers. Some cone-type speakers have dipole radiators. For a dipole radiator to have adequate low-frequency response, it must be very large to prevent the rear wave from canceling the front wave. Also, the dipole radiator must not be placed close to and parallel to a wall, working best when not near reflective surfaces.

See DI.

 A connection between two devices that allows both DC and AC between them.

Current in only one direction. DC always has the same direction, from the positive to the negative terminal. Compare with AC.

See reverberant field.

A system for cutting a metal mother on a record mastering lathe, eliminating the lacquer master and metal master steps. Release pressings made from a stamper are thus only two stems from the DMM and thus have less noise and distortion than those made by the older, five-step process. The DMM process is also used in CD mastering.

 A recording console output taken directly after the input module and main channel fader, but before the panpot and output bus assignment switches. This output is sometimes used to avoid crosstalk that may be introduced if the signal is allowed to flow through the complete circuit.

 A optical (photographic) sound recording that, when processed, results in a track that can be played and edited; now obsolete.

A loudspeaker which does not have a horn between the moving element and the air is called a direct radiator. Most direct radiator-type speakers are for home use, while horn-type speakers are preferred for sound reinforcement applications. Direct radiators generally provide smoother, more uniform response, while horns are much more efficient, providing a greater output level for a given power input. Also, horns have greater directivity, which is desirable in sound reinforcement systems. See compression driver.

(1) See reverberant field. See also free-field, critical distance. (2) The sound received at the recording console from an electronic instrument when using a direct box.

: A proprietary CD/DVD data format proposed by Sony and Philips for use in the SACD . DSD bandwidth is normally 2.844Mb per channel (64 times 44.1kHz), with optional sampling rates of 32 or 128 times 44.1kHz,yielding a slightly higher data rate than that required by 24-bit/96 kHz resolution conventional A/D-D/A systems. DSD uses a delta modulated ADC to generate a 2.8224MHz, 1-bit signal, a rate chosen as a simple multiple of the lowest common high-fidelity PCM sampling rate, 44.1kHz. The 1-bit datastream is recorded directly to disk, avoiding the decimation and oversampling stages, inherently improving the resultant audio quality, simplified error protection, and there is no need to frame the data into words. Sony claims that the sampling rate is so high that it more nearly approximates the original analog signal, allowing equalizers and other effects processors to better simulate analog effects. It is claimed that DSD can have frequency response up to 1MHz, or up to a dynamic range (within the audio bandwidth) of 120dB, equivalent to about a 20-bit resolution. A number of DSP algorithms are available which allow the optimization of either bandwidth or dynamic range. [Note that these benefits apply only to those players which support the DSD standard, as a replay on any conventional CD, DVD or another digital format would require the decimation and framing steps.]

A proprietary CD/DVD data format proposed by Sony and Philips for use in the SACD . DSD bandwidth is normally 2.844Mb per channel (64 times 44.1kHz), with optional sampling rates of 32 or 128 times 44.1kHz,yielding a slightly higher data rate than that required by 24-bit/96 kHz resolution conventional A/D-D/A systems. DSD uses a delta modulated ADC to generate a 2.8224MHz, 1-bit signal, a rate chosen as a simple multiple of the lowest common high-fidelity PCM sampling rate, 44.1kHz. The 1-bit datastream is recorded directly to disk, avoiding the decimation and oversampling stages, inherently improving the resultant audio quality, simplified error protection, and there is no need to frame the data into words. Sony claims that the sampling rate is so high that it more nearly approximates the original analog signal, allowing equalizers and other effects processors to better simulate analog effects. It is claimed that DSD can have frequency response up to 1MHz, or up to a dynamic range (within the audio bandwidth) of 120dB, equivalent to about a 20-bit resolution. A number of DSP algorithms are available which allow the optimization of either bandwidth or dynamic range. [Note that these benefits apply only to those players which support the DSD standard, as replay on any conventional CD, DVD or other digital format would require the decimation and framing steps.

Philips’ proprietary technology for lossless 2:1 data reduction in digital recordings on SACD. DST is optimized for audio-type signals, allowing sufficient storage capacity for double the old CD standard of 74 stereo minutes. By incorporating DST into the SACD standard, it is possible to store two complete 74-minute versions of audio material so as to combine a stereo DSD track, and a 6-channel surround mix, plus other data, text, graphics, and video, all on the single, high-density DASD layer.

 Philips’ proprietary technology for lossless 2:1 data reduction in digital recordings on SACD. DST is optimized for audio-type signals, allowing sufficient storage capacity for double the old CD standard of 74 stereo minutes. By incorporating DST into the SACD standard, it is possible to store two complete 74-minute versions of audio material so as to combine a stereo DSD track, and a 6-channel surround mix, plus other data, text, graphics, and video, all on the single, high-density DASD layer.

 A type of analog LP mastering in which a master tape is not used. The signal directly from the control console is used to cut the original acetate disc. This means a direct-to-disc recording cannot be edited, and is made live. (2) Recording digital audio data onto a hard disk for replay or editing.

A method of recording in which the instruments and vocals are mixed and recorded directly onto a stereo half-track or DAT. If analog, no further changes or remixing is possible. The fidelity, edibility, and relatively low-cost of direct-to-two-track digital recording has revived the popularity of this medium for making master tapes, especially those intended for release on CDs. Also called live-to-two-track.

A microphone which does not have a spherical polar pattern, i.e., a microphone which is not omnidirectional, such as a cardioid, figure-eight, etc. having an acceptance angle of less than 360¢ª. See directivity.

Describes the angle of coverage of a loudspeaker system or microphone acceptance angle, both in the vertical and horizontal planes. High directivity equates to a narrow angle of coverage. The directivity factor is a measure of the directionality of the sound output of a loudspeaker. See also Q.

A collection of application programming interfaces for handling tasks related to multimedia, especially game programming and video, on Microsoft platforms. Originally, the names of these APIs all began with "Direct", such as Direct3D, DirectDraw, DirectMusic, DirectPlay, DirectSound, and so forth. The most common audio effects plug-in format used by Windows™ software.

 Refers to a 1:1 relationship of recorded tracks on an audio medium or film print and the resulting number of speaker channels. Contrast with matrixed sound.

 Traditionally means the five-speakers-behind-the-screen system made popular by the Todd-AO 70mm process (although first used for Cinerama). Today the term sometimes means six nonmatrixed tracks, assigned to L,C,R,LS,RS,Subwoofer. See also 5.1.

Traditionally means the five-speakers-behind-the-screen system made popular by the Todd-AO 70mm process (although first used for Cinerama). Today the term sometimes means six nonmatrixed tracks, assigned to L,C,R,LS,RS,Subwoofer. See also 5.1.

A direct output from a mixer channel, which services only that one channel.

 A direct output from a mixer channel, which services only that one channel.

A CD production process where the entire disc is written in one burn; the laser is never turned off. Ideal for audio, disk-at-once mode allows gaps between tracks of any length (except the first track, which must have a 2-3 second gap.) Compare with track-at-once.

A CD production process where the entire disc is written in one burn; the laser is never turned off. Ideal for audio, disk-at-once mode allows gaps between tracks of any length (except the first track, which must have a 2-3 second gap.) Compare with track-at-once.

(1) The spreading of sound waves as they leave a loudspeaker. (2) Another term for refraction.

 (1) The spreading of sound waves as they leave a loudspeaker. (2) Another term for refraction.

The distance between some measured position of a moving object, e.g., a speaker cone, and its static position. Also applies to the position of air molecules in a sound wave. See rarefaction.

 The opposite of close miking. In recording, the placement of one or more microphones relatively far away from the sound source. This technique picks up a substantial portion of reverberant sound, and is therefore used to make most classical or orchestral recordings to capture the sound as closely as possible to that experienced by an audience. In the studio, close and distant mics on any instrument or group may be blended at the mixer to achieve the desired sonic image. See depth.

The opposite of close miking. In the recording, the placement of one or more microphones relatively far away from the sound source. This technique picks up a substantial portion of the reverberant sound and is therefore used to make most classical or orchestral recordings to capture the sound as closely as possible to that experienced by an audience. In the studio, close and distant mics on any instrument or group may be blended at the mixer to achieve the desired sonic image. See depth.

Also called correlated noise. Any (usually) unwanted sound which varies with the input signal. (1) Any undesirable change in the characteristics of an audio signal of six types: The two types of (i) nonlinear distortion are intermodulation distortion and harmonic distortion. Other types of distortion are (ii) frequency distortion (pitch-shift), (iii) phase distortion (time shift), (iv) transient distortion,(v) scale (volume) distortion, and (vi) frequency modulation distortion. There are other factors which cause music reproduction to be untrue to the original but which are not considered distortion, such as background noise, and a lack of directional realism and proper ambience due to the use of too few channels of reproduction. See noise. (2) A sound modulation technique whereby the original waveform is distorted intentionally.

 Also called correlated noise. Any (usually) unwanted sound which varies with the input signal. (1) Any undesirable change in the characteristics of an audio signal of six types: The two types of (i) nonlinear distortion are intermodulation distortion and harmonic distortion. Other types of distortion are (ii) frequency distortion (pitch-shift), (iii) phase distortion (time shift), (iv) transient distortion,(v) scale (volume) distortion, and (vi) frequency modulation distortion. There are other factors which cause music reproduction to be untrue to the original but which are not considered distortion, such as background noise, and a lack of directional realism and proper ambience due to the use of too few channels of reproduction. (2) A sound modulation technique whereby the original waveform is distorted intentionally. See noise.

A noise-based rounding method used to add a tiny amount of controlled noise to a digital audio file to make other, more objectionable errors less obvious and/or to convert from one word size down to a smaller word, e.g., from 24-bit resolution to 16-bits. Redithering is a dithering process used in digital-to-digital signal processing to distinguish it from the dithering process used during the original A/D conversion. There are several ways to dither: Add white noise at about half or one-third the value of the LSB, or half the level that a system can transmit. Thus, for a 16-bit converter encoding a range of 2V, the LSB is equivalent to a difference of 30.5µV (2÷65,536V), yielding dithered noise of about 10-15µV;Second-Order dither in which the dither signal (white noise) is processed by a highpass filter to remove low-frequency components. This makes the noise less apparent to the ear, since humans are less sensitive to high-frequency noise than other levels;Noise-shaping in which the dither signal is run through a set of filters to provide the most energy in regions where the ear is the least sensitive;(Triangular Probability Density Function) where before the noise is shaped, it has a different spectral content than ordinary white noise. It has better noise modulation performance, which is how the noise affects the signal itself;Sony’s SBM (Super Bit-Mapping) where the audio is run through a processor with an algorithm that maps a series of higher-resolution samples to a series of lower-resolution samples;The Apogee UV-22, which is not really dither. Instead, it uses a periodic signal centered around 22kHz that has good performance in terms of audibility and noise modulation. Placing the signal this high in the audio spectrum makes it very difficult to hear and produces fewer effects on the character of the signal.

Dynamic Link Library. Files used by PC-type computer application programs to provide additional functionality to the computer’s operating system. The equivalent of an extension file on a Mac.

Dynamic Link Library. Files used by PC-type computer application programs to provide additional functionality to the computer’s operating system. The equivalent of an extension file on a Mac.

 Downloadable Sounds Level 2. An improvement over DLS which moves closer to enabling multitrack audio and MIDI programs to be used by audio and MIDI hardware and software interchangeably. This is accomplished by transferring entire multisampled instruments, along with the MIDI data, to the end-user’s platform, resulting in playback as the author intended. In addition to the specification of DLS-1, DLS-2 specifies: resonant filter control, adds delay and hold to envelope segments, effects routing for chorus and reverb, no limit to the number of regions in any soundbank, each reason can have independent envelope and filter data, and is extensible to include other forms of synthesis. Ratified by the MMA in early 1999. See also MPEG-4.

 Downloadable Sounds Level 2. An improvement over DLS which moves closer to enabling multitrack audio and MIDI programs to be used by audio and MIDI hardware and software interchangeably. This is accomplished by transferring entire multisampled instruments, along with the MIDI data, to the end-user’s platform, resulting in playback as the author intended. In addition to the specification of DLS-1, DLS-2 specifies: resonant filter control, adds delay and hold to envelope segments, effects routing for chorus and reverb, no limit to the number of regions in any soundbank, each reason can have independent envelope and filter data, and is extensible to include other forms of synthesis. Ratified by the MMA in early 1999. See also MPEG-4.

DownLoadable Samples. A standard which allows multimedia games or other programs to contain samples which would be downloaded to the playback hardware. This is a hybrid between streaming digital audio (direct audio from CD-ROM, for example) and MIDI, which alters predefined sounds. DLS is an attempt to solve the problems inherent in the ambiguity of the GM specification and the somewhat random sounds produced by the playback hardware. DLS-1 specifies: 16-zone multisampled soundbanks with 128 zones for drum banks, pitch-shifting, ADSR for amplitude and pitch, and LFO for amplitude or pitch. DLS-1 was ratified by the MMA in 1997. See DLS-2.

Digital Linear Tape. A tape-based computer backup format developed by Quantum Laboratories.

Digital Linear Tape. A tape-based computer backup format developed by Quantum Laboratories.

Direct Memory Access. A digital logic design which allows peripheral devices to communicate directly with the system memory, rather than requiring the central processor to stop whatever processing it was doing to control communications between an attached device (usually having some kind of I/O function) and the computer memory.

Direct Memory Access. A digital logic design which allows peripheral devices to communicate directly with the system memory, rather than requiring the central processor to stop whatever processing it was doing to control communications between an attached device (usually having some kind of I/O function) and the computer memory.

Dialog, Music, and Effects. The three basic stems of film soundtracks, originally meant to denote the 35mm 3-track master mix of academy mono films.

 Dialog, Music, and Effects. The three basic stems of film soundtracks, originally meant to denote the 35mm 3-track master mix of academy mono films.

Dialog, Music, and Effects. The three basic stems of film soundtracks, originally meant to denote the 35mm 3-track master mix of academy mono films.

There is a wide difference in the apparent loudness between different TV programs’ audio content. In DTV, with by standard is AC-3 encoded, a program producer chooses one of 31 different dialog normalization (abbreviated DN or "Dialnorm") values and this parameter is carried within the AC-3 datastream, where each step represents a 1dB change in level. The DN value is the difference in dB between the maximum level possible (0 dBFS) and the average loudness level of the program material. The smaller the difference between the maximum and program average levels, the lower the DN value is assigned. The lower the DN value, the lower the output volume of the AC-3 decoder is set in direct proportion, meaning that subjectively louder programs will be played back at lower volumes than those in which the average program level is less loud. This supposedly will obviate the user having to adjust the volume control between programs, once the audio listening level is set by the user.

Dynamic Noise Reduction. Similar to Dolby, but was not effective.

A 7.1 (surround sound technology/audio format for creating and playing back multichannel audio (codec) developed by Dolby Laboratories. It expands on existing surround sound systems by adding height channels, allowing sounds to be interpreted as three-dimensional objects.

The 5.1-channel digital format created by Dolby Laboratories, first used in 1992 for "Batman Returns." In current usage, the term applies to both the Dolby 35mm theatrical format, which contains the data printed optically between the sprocket holes, and for video formats, such as DVD, laserdisc, and DTV. AC-3, as Dolby Digital was first called, used RF modulation of the digital signal onto one of the analog tracks, making it possible to fit an entire movie, along with the already existing digital tracks, onto a conventional laserdisc; a demodulator was needed to recover the audio back into a digital bitstream. The Dolby Digital format is a surround-sound, split-band, perceptual coding scheme. AC-3 was designed as a 5.1 multichannel format , using approximately 13:1 lossy compression, and is specified as the matrixing format for DVD and DTV. Also used in HDTV broadcasts, SR-D, and DSD cinema productions. Versatile, in that parameters such as bit-rate and number of channels can be tailored to particular applications, unique in that the data bits are distributed dynamically among the filter bands as needed by the particular frequency spectrum or dynamic nature of the program. Data rates vary from 32kbps for a single mono channel to as high as 640 kbps for 5.1 format. The data rate is 320kbps for film, 384kbps for laserdisc, and 384kbps or 448kbps for DVD, although the maximum throughput for the specification is 640kbps. Dolby’s current decoder can accept incoming data at 32kHz, 44.1kHz, or 48kHz sample rates, with bit depths of 16, 18, or 20 bits. The commercial competitor to the Dolby Digital format is DTS. See metadata, audio coding mode.

See ISDN.

A matrixed surround-sound system which combines multichannel LCRS audio in such a way that the encoded signal forms a stereo-compatible, two-channel format for recording and broadcasting. Originally developed in 1977 for "Star Wars," and now in wide use. As with any matrix system, it is impossible to completely recover the original multichannel signals with perfect isolation. The decoder disguises this problem through a steering process which emphasizes the signal emanating from its appropriate loudspeaker by canceling out a portion of the crosstalk in adjacent channels. See also free encoding, Pro Logic.

The Dolby SR analog allows the comparison of the recorded Dolby noise on a tape to that generated by the decoder, with four continuous seconds of noise to identify the generator, and two 2-second sections of noise indicating that the monitoring is off-tape. This allows for confirmation of correct EQ settings as well as playback verification. The broadband reference signal used to correctly calibrate the different Dolby codecs is called Dolby noise. See Dolby tone.

 A type of two-ended, dynamic noise reduction for magnetic tape recording and playback. The essential difference between a compander and the Dolby system is that the Dolby system is frequency-dependent. The compander was developed to reduce distortion. Dolby applies companding to frequency variations in addition to signal amplitude variations, adjusting gain as frequency changes. The Dolby-A and Dolby-SR systems are used for professional recording in studios. Consumer tape decks use either a Dolby-B or Dolby-C system. Dolby-B operates only at high frequencies and reduces tape hiss by about 10dB. Dolby-C works over a slightly wider frequency range, providing a noise reduction of up to 20dB. All of the Dolby systems operate on quiet passages, below levels of about -10VU. Very strong signals, such as over 60dB or at frequencies below 500Hz are not affected by the Dolby system because these signals are not degraded by tape noise. When the recorded signal is played back, the Dolby circuit reduces the accentuated high-frequency signals so that the frequency response of the record/playback system is flat, hence reducing also the high-frequency tape hiss, improving the S/N ratio of the taped music. See asperity, Barkhausen effect, compander, dbxcompander,dbx, spectral recording.

A type of two-ended, dynamic noise reduction for magnetic tape recording and playback. The essential difference between a compander and the Dolby system is that the Dolby system is frequency-dependent. The compander was developed to reduce distortion. Dolby applies companding to frequency variations in addition to signal amplitude variations, adjusting gain as frequency changes. The Dolby-A and Dolby-SR systems are used for professional recording in studios. Consumer tape decks use either a Dolby-B or Dolby-C system. Dolby-B operates only at high frequencies and reduces tape hiss by about 10dB. Dolby-C works over a slightly wider frequency range, providing a noise reduction of up to 20dB. All of the Dolby systems operate on quiet passages, below levels of about -10VU. Very strong signals, such as over 60dB or at frequencies below 500Hz are not affected by the Dolby system because these signals are not degraded by tape noise. When the recorded signal is played back, the Dolby circuit reduces the accentuated high-frequency signals so that the frequency response of the record/playback system is flat, hence reducing also the high-frequency tape hiss, improving the S/N ratio of the taped music. See asperity, Barkhausen effect, compander, spectral recording.

 A four-channel perceptual coding scheme developed by Dolby Labs where an LCRS audio signal is converted into two channels of analog audio, then recovered to yield discrete left, center, right, and mono surround channels. This is a hardware version of their surround decoder originally developed for the Dolby Motion Picture 4:2:4 matrix surround-sound system, developed for the Star Wars picture in 1977. If a subwoofer is used, it is generally fed by lowpass-filtering a mix of the three front channels at the receiver.

A complex type of two-ended, dynamic split-band, noise reduction compander system that outperforms Dolby-A, -B, or -C systems and also results in reduced distortion in most cases. This was developed as an upgrade to the professional Dolby-A, featuring an improvement in tracking accuracy and sliding bands, and closest to C-type noise reduction. S-Type noise reduction is the consumer analog. See SR.D.

 A system developed by the Dolby company for placing a digital audio soundtrack onto 35mm film, first used in 1992. The soundtrack includes a Dolby Digital mix, as well as an SR analog stereo optical track. The data are compressed and printed onto the film between the sprocket holes. See surround-sound, perceptual coding, Dolby surround-sound, spectral recording.

 The original Dolby Surround system which used four audio channels carried on a stereo optical track on 35mm film, using Dolby perceptual encoding. On 70mm film, six audio channels are recorded on discrete magnetic tracks laid onto the film. In the broadest and most common usage, the trademark that appears on movie prints, advertisements, and posters which means that a given film has been released in prints that employ Dolby A-Type noise reduction encoding. Beginning in 1987, Dolby-SR has been available on 35mm stereo optical prints. Dolby Stereo on 70mm usually means four discrete primary channels (LCRS) with the left-center and right-center tracks dedicated to low-frequency information (below 250 Hz). The four tracks are normally use A-Type encoding, although selected 70mm films, since 1987, have utilized Dolby-SR encoding. See film soundtrack.

The digital release format developed by Dolby Laboratories and THX for use in "Star Wars: Episode One--The Phantom Menace." Three surround tracks are derived by matrix-encoding them in the two previously existing surround tracks. This should not be referred to as a 6.1-channel format because the additional surround channel is not a discrete channel.

A reference tone, usually recorded at the head of a Dolby-A recorded tape, by which the threshold levels of the Dolby noise reduction system are adjusted for proper encoding and decoding of the companded signal.

 In magnetic recording tape, the smallest ferric oxide particle that can be considered as a separate magnet. Defined as molecules of ferric oxide, or, less than one billionth of a gram of material. See Barkhausen effect.

See Circle of Fifths, key.

A hardware device that plugs into a parallel or a USB port, acting as copy protection for a particular software application.

 The apparent change in the pitch of a sound when the source of the sound is moving with respect to the listener. Also called Doppler distortion.

Disk Operating System. The original operating system for PC-type computers. Much of the Windows™ operating system is written in DOS; NT is not.

(1) A method of producing sound motion pictures where the soundtracksound track is recorded on a magnetic tape recorder which is separate from the video recorder and which is synchronized with the movement of the film in the camera, projecting a film with the picture on 35mm film, in interlock with the soundtrack, most commonly on mag film. The synchronization was originally done by recording a special tachometer signal on one track of the tape, but is now done with timecode. Examples of double-system sound are film and Nagra, film and mag dubber, videotape and audio tape. (2) A film or video production that utilizes sound recorded on a separate tape recorder, such as a DAT or Nagra. This term is still used even if the video recorder is also simultaneously recording the sound. Compare with single sound system. See mut. Also called sep mag.

Originally, double-tracking meant the recording of a vocal track on one tape recorder track, then listening to this while recording another similar track. The two tracks are combined and re-recorded into a single track, which will sound more diffuse due to slight differences in the two original tracks as double-tracking produces a slight chorus effect to voices. In this case, it is also called re-tracking. Double-tracking can be done with DSP which introduce a small randomly varying time delays to one signal and then combine it with the original signal. See delay(3), stereoizing.

 If a loudspeaker is driven too hard in its low-frequency range, it will produce second-harmonic distortion, sometimes with greater amplitude than the fundamental. The doubled frequency sounds one octave higher than the fundamental, and is often not musically annoying. This is called frequency doubling, or simply doubling.

See beat.

A mix derived from a multichannel (usually 5.1 format) source to create a compatible stereo, mono, or other version of fewer channels. The common use of downmixing today occurs in consumer Dolby Digital products to play back a 5.1-channel DVD mix either via Dolby Pro-Logic decoding or in standard two-channel stereo for headphones. In those instances, an Lt-Rt or an Lo-Ro respectively, are the result.

Opposite of uptime. This refers to the amount or percentage of time that a product, system, or service is operational.

See noise reduction.

A family (DP521/DP522/DP523/DP524) of 2-channel codecs used for point-to-point and point-to-multipoint signal distribution: ISDN, Switched-56, T1 or DS-3 networks; recording/post-production studio interconnection with or without video; and voice-over and other applications. AC-2 and AC-3 perceptual coding algorithms are supported to provide audio transfers at a total data rate between 56 kBps and 384 kBps. With AC-3, single-channel, two-channel, and composite stereo algorithms are supported.

Digital Read After Write, an erasable CD that can be re-recorded.

On a Hammond organ with tonewheels, a slider that shortens the distance between the axle bearing the wheels and the transducer which converts their spinning patterns into an audio signal. This has the effect of introducing a particular harmonic into the sound to alter its timbre. While similar in purpose to a stop on a pipe organ, it has the advantage of being variable in intensity as opposed to a stop’s simple on/off action. Drawbars have been retained on more recent electronic organs of the Hammond type, but their function is now to act as simple faders that adjust the gain of different oscillators.

 In magnetic tape recording, any extended deviation from the nominal tape speed. Drift can be due to excessive take-up tension, improper capstan motor control, etc.

 (1) A power amplifier which increases the amplitude of a voltage,current, or power signal, (2) any direct radiator speaker, or (3) the term used to describe the chassis loudspeaker, mid-range unit, or tweeter elements of a loudspeaker system (as opposed to speaker system which covers both cabinet and drivers.) (4) A software program which enables communication between a particular make and model of a hardware device and the computer’s operating system, usually necessary for some kind of I/O device such as a soundcard, printer, or scanner. The problem of outdated hard-disks and soundcard drivers is particularly problematic.

A version of the SMPTE timecode used for color video recording where two frames are dropped at the beginning of each minute, except at the beginning of every tenth minute, devised to compensate for the difference between the NTSC (US) standard of 29.97fps and a real-time counter. The difference equates to 108 frames per hour. To avoid this confusion, most audio-only synchronization applications specify a non-drop time code.

See punch-in.

(1) In analog magnetic tape recording, the quality of the recorded signal depends on the uniformity of the magnetic coating of the tape. If its sensitivity varies on the tape, the signal level will be reduced periodically, and these reductions in level are called dropouts, their combined effects resulting in an increased noise level in the reproduced signal. See asperity, calendering. In a digital recording, a dropout is caused by an irrecoverable data error. (2) In timecode, a loss of sequence in the linear timecodetime code count.

An acoustic isolation booth or small room primarily intended as an enclosure for the recording of drums, traps or other percussion instruments and their players. Acoustically sealed off from the main recording space, drum booths have bass traps to prevent loud percussive transients from being heard. Some drum booths are not fully enclosed. This type of booth does not provide complete isolation, but does avoid the small-room problem of standing waves and lower midrange resonances that can give enclosed booths an unnatural, closety sound.

 A set of pads which have a similar response to the heads of acoustic drums when struck with sticks. They are made for two purposes: to quiet drum practice, and, when fitted with suitable transducers, to play electronic (usually sampled) drum sounds. If equipped with MIDI, drum pads can also act as a controller, allowing drummers to trigger any type of synthesized sound across a MIDI network.

Consisting entirely of the original, unprocessed sound. The output of an effects divide is 100% dry when only the input signal is being heard, with none of the effects created by the processor itself. Lacking in reverberation. Compare with wet, flat.

This refers to the amount of dry signal relative to the amount of reverb or other effect-processed wet sound.

 The name of the original Dolby Laboratories recording/monitoring unit used by re-recording stages during a Dolby Stereo mix. Prior to the 2-track print master, the unit is used for 4:2:4 monitoring purposes, encoding a 4-channel composite mix into two tracks and then decoding it back into four channels. Later versions of these units include theSEU4 and SDU4 units which offer, respectively, the ability to encode and decode print masters, although without either the container of the optical track simulation featured in the DS4. The DS10 contains a magneto-optical recorder for theatrical Dolby Digital mixes and also records the Lt-Rt SR-encoded print master. None of the above units can be purchased; their use is free for films that have paid for the appropriate license fee and/or trademark agreement.

See DS4

See Direct Stream Digital

Digital Signal Processor. Broadly speaking, all changes in sound that are produced within a digital audio device, other than changes caused by simple cutting and pasting of sections of a waveform, are created through DSP. A digital reverb is a typical DSP device.

Direct Time Lock. An early MIDI synchronization system developed by MOTU. See MTC.

Digital Tape Recorder. This is the analog version. An analog audio tape recorder is called an ATR.

 A 16-bit format used on Tascam and Sony MDMs, providing up to 108 minutes of 16-bit, 8-track record time on an NTSC-120 Hi-8mm videocasette. See ADAT.

 See stereo optical print.

(DIGITAL THEATER SOUND) An audio codec capable of supporting 32 speaker locations and works with any configuration in a hemispherical layout. This "object-based" encoding supports height speakers and can control vocal levels.

Digital TeleVision. DTV’s audio specification provides up to six discrete channels of 5.1-format audio, where the LFE channel is band-limited to 25Hz-120Hz. DTV has been developed specifically for the home theater market, as an improvement to the ProLogic system.

 Speakers which have two cones; the second cone is usually much smaller and glued directly to the centre of the main cone although it has just one voice coil.

(1) (verb) In the most general sense, to dub is "to copy," although in film sound it has many similar meanings. Dub can refer to the act of replacing dialog, usually via ADR, either in the original language or in a foreign language. (2) Dubbing is also the common name for re-recording.

See ProDigital.

 See final mix.

 Film sound term for a playback-only mag machine. These were previously known as dummies. See digital dubber.

(1) The act of re-recording sound effects, location sound, music, dialog, and/or Foley. Usually used to refer to the substitution of a foreign language or other replacement for the original dialog track in a film or TV production. (2) The process of making a copy or copies of a recorded analog or digital original. (3) To mix together onto a single track all of the separate edited soundtracksound tracks of a film or television production. See re-recording, dubbing theater and transfer.

 Also called a dub stage. A special studio where music is blended with dialog and sound effects for the final soundtracksound track. A dubbing theater is actually a small movie theater, with a large screen and full theater surround system. A row of seats is removed from the middle of the theater and a large mixing console specially designed for film sound is put in place. There is also a machine room which houses dubbers and projectors, isolating the noise from the recording studio in which the DME stems can be recorded in sync with film projected on a screen visible through a window. The screen hangs in a theater equipped with the mixing console which controls the sounds played back by all the dubbers, other prerecorded sources, and the sounds being recorded in the studio. The theater itself is designed to approximate the acoustics of a public cinema. Also called mixing studio, re-recording studio, re-recording stage, or theater.

(verb) To lower the level of music to accommodate dialogue or other sound effects.

A high-power resistor that is connected to the output of a power amplifier to make the amplifier function as though it had a loudspeaker connected to it. A dummy load circuit can be used to test amplifier performance as it would perform when connected to loudspeakers, or in a device such as a speaker simulator so that the amplifier always sees a high-impedance load at the output stage, even if no loudspeaker is connected.

See edit switch.

The technical term for the variable area optical soundtrack format used on all 35mm mono and stereo soundtracks. See SVA, Dolby Stereo, DTS Stereo.