A

ABS

An abbreviation often used for Absolute Time, or Absolute Time Code. The name is pretty indicative of what it is. See WFTD archive Absolute Time Code for a detailed definition.

Absolute Phase

A positive pressure to a microphone diaphragm will (in most mics) produce a positive voltage at its output. If the correct polarity (see WFTD archive polarity) of the signal is maintained throughout the signal path this should ultimately produce a positive voltage at the speaker terminal, which will (on most speakers) cause the speaker to move forward creating a positive pressure wave in the listening position. This is known as absolute phase (see also WFTD archive phase): The original polarity of the source sound is thus reproduced in phase by the loudspeaker for listening.

Absolute Time Code

Absolute Time Code (ATC) is generally recorded in the subcode (see WFTD archive "subcode") or control track region of any digital tape. This is the code that digital tape machines use to locate specific points on a tape for autolocation or other functions. In some machines it is even used to synchronizes the tape to other equipment. ATC is precisely accurate and usually conforms to the IEC standard which is easily converted to the more commercially used SMPTE time code. Unlike SMPTE, ATC always begins at zero at the beginning of a digital tape and increments one frame at a time until recording stops. Some DAT machines have the ability to function without ATC on a tape while others simply will not play a tape without it. These days most all machines record it automatically so it will always be on every tape.

Absorption

In acoustics (as opposed to paper towels), the opposite of reflection. Sound waves are "absorbed" or soaked up by soft materials they encounter. Studio designers put this fact to work to control the problem of reflections coming back to the engineer's ear and interfering with the primary audio coming from the monitors. The absorptive capabilities of various materials are rated with an "Absorption Coefficient," which is a measure of the relative amount of sound energy absorbed by that material when a sound strikes its surface. (See also WFTD "Anechoic")

A-B Stereo

Sometimes known as Time Difference Stereo, A-B Stereo is a stereo miking technique that employs two spaced omnidirectional microphones to capture a stereo image. The microphone spacing introduces small differences in the time or phase information contained in the audio signals (according to the relative directions of the sound sources). As the human ear can sense these time and phase differences in audio signals and use them for localization, they will act as stereo cues to enable the listener to "capture the space" in the recording, and experience a stereo image of the sound-field. Omnidirectional microphones and A-B Stereo are often the preferred choice when the distance between microphone and the sound source is large. One reason is that true omnidirectional microphones are able to capture the true low frequencies of a sound-source regardless of the distance, while directional microphones are influenced by the proximity effect. Directional microphones will therefore exhibit loss of low frequencies at larger distances. (See Spaced Omni)

AC-1

AC-1 was Dolby's first digital audio coding scheme. First adopted by systems providers in 1984 when bit rate reduction was in its infancy, AC-1 is a refined form of adaptive delta modulation (ADM), whereby changes in the signal amplitude from moment to moment are transmitted, rather than the absolute values. In addition to coding amplitude changes, a system of dynamic pre- and de-emphasis is used to minimize the audibility of coding noise. It was intended for mass broadcasting applications at a time when the digital signal processing "horsepower" that we are used today just wasn't available. The solution was to use a simple and therefore cheap decoder driven by a fairly complex encoder. The technology is in use in satellite and cable delivery systems. Encoders and decoders are both sold and licensed.

AC-2

Another of Dolby Labs' sound encoding schemes. Dolby AC-2 is a perceptually based adaptive transform coding algorithm that combines very high audio quality with a low bit rate, thus substantially reducing the data capacity required in such applications as satellite and terrestrial links and digital audio storage media. The digital algorithm developed by Dolby uses a multi-band approach to take advantage of psychoacoustic masking. A bit allocation scheme based on 80% fixed allocation and 20% adaptive allocation keeps the complexity of the codec relatively low. Dolby both manufactures professional codecs incorporating AC-2 (such as Dolby FAX) and licenses the technology to other manufacturers for inclusion in their products.

AC-3

Also known as Dolby Digital, AC-3 is an advanced perceptual coding technology for transmission and storage of up to five full-range channels (Left, Center, Right, Left Rear, Right Rear), plus a supplemental bass-only effects channel (referred to as a .1 channel due to the smaller number of bits needed for the information). It accomplishes this in less space than is required for one linear PCM coded channel on a compact disc. Dolby Digital is a more powerful and flexible coding system than AC-2 and provides a feature set including: 1) down mixing for optimal reproduction in mono, stereo, and Dolby Pro Logic compatible configurations as well as full 5.1 channel sound; 2) carriage of dynamic range and dialog level control information to decoders; and 3) operation over a wide range of bit rates. Dolby Digital can be heard on the soundtracks of a thousand plus films, and on the current generation of laser discs. Dolby Digital is being used on the audio tracks on DVD, and will be the standard audio on the new high definition television (HDTV) system going into operation in the United States.

Access Time

This is the time it takes from when a disk (or disc) command is sent, until the disk (or disc) reaches the data sector requested. Access time is a combination of latency (WFTD 7/2/97), seek time, and the time it takes for the command to be issued. Access time is important in data intensive situations like hard disk recording, multimedia playback, and digital video applications. Lower access times are better. Keeping your drives in good shape with periodic de-fragging, etc. will ensure that your drive is providing the fastest access times it can.

Acoustic Suspension

A type of speaker design using a sealed cabinet. Primarily used for low frequency enclosures, acoustic suspension designs use the air mass within the cabinet as a "spring" to help return the relatively massive speaker to the rest position. This allows heavier, longer throw drivers to be used, but results in a less efficient design requiring more amplifier power.

Active Sensing

A MIDI message sent by some MIDI devices all the time to confirm that they are still there and hasn't gone off line. It carries no other information and hasn't been widely implemented in the industry. It comes from the early days of MIDI when some companies thought that if MIDI was interrupted it would be a good idea for instruments to change all notes to an off state. When implemented, a MIDI device will usually wait about 300 ms for the active sensing signal to confirm MIDI information is still being sent. If it is not received within this time limit, all notes are shut off and the operation is returned to normal.

ADAT

Today's word may seem a bit basic, but we did have a request for it (and far be it for us to ignore a request). ADAT Ü An acronym for Alesis Digital Audio Tape. Taken from the acronym "DAT" (see also WFTD R-DAT), ADAT is the name Alesis chose in the early 1990's for their ground breaking product, which records eight tracks digitally on a standard 1/2" SVHS video cassette. The ADAT has been arguably the most significant technology/price breakthrough for recording studios in the last 20 years and has undoubtedly changed the face of modern recording forever. The ADAT has gone through several generations and is currently a 20-bit digital format. The ADAT optical connections for transferring digital data 8-tracks at a time have become a standard of the industry and are used in a wide range of products from many manufacturers.

A/D Converter

An A/D (Analog to Digital) converter is an electronic device who's function is to convert analog voltages into a digital representation of electrical one's and zero's which can be stored, manipulated, and later retrieved or converted back to analog. In the domain of audio recording these converters are found built in to virtually all digital audio products such as DAT machines and digital signal processors. There are also a variety of high quality stand alone converter boxes which will accept line or mic level analog signals and output digital equivalents which can then be input directly into a digital device.

Additive Synthesis

The process of constructing a complex sound using a series of fundamental frequencies (pure tones or sine waves). Each of the fundamental frequencies usually has its own amplitude envelope which allows independent control of each partial (harmonic). Pipe organs or Hammond organs are both instruments which are based on additive synthesis. Some modern synthesizers have employed additive synthesis techniques, but other techniques such as FM (see WFTD archive FM Synthesis) and physical modeling (see WFTD archive Physical Modeling Synthesis) have proven to be easier to develop and still very effective at producing a wide variety of sounds.

ADSR

Abbreviation for Attack, Decay, Sustain, and Release. These are the four parameters found on a basic synthesizer envelope generator. An envelope generator is sometimes called a transient generator and is traditionally used to control the loudness envelope of sounds, though some modern designs allow for far greater flexibility. The Attack, Decay, and Release parameters are rate or time controls. Sustain is a level control. When a key is pressed, the envelope generator will begin to rise to its full level at the rate set by the attack parameter, upon reaching peak level it will begin to fall at the rate set by the decay parameter to the level set by the sustain control. The envelope will remain at the sustain level as long as the key is held down. Whenever a key is released, it will return to zero at the rate set by the release parameter.

AES

The Audio Engineering Society (AES) is a professional society of audio people who work to set standards for the audio community. The AES serves the audio industry by stimulating and facilitating advances in the constantly changing field of audio. It encourages and disseminates new developments through annual technical meetings and exhibitions of professional equipment, and through the Journal of the Audio Engineering Society (JAES), the professional archival publication in the audio industry.

AFL

AFL (After Fade Listen) is used in mixing boards to override the normal monitoring path in order to monitor a specific signal at a predefined point in the mixer. Unlike PFL (see WFTD archive "Pre-Fade Listen"), the AFL signal by definition is taken after the fader of a channel or group buss such that the level of the fader will affect the level heard in the AFL monitor circuit. AFL is sometimes also taken after the pan pot which also allows the engineer to monitor the signal with the pan position as it is in the mix. AFL is a handy way to monitor a small group of related instruments by themselves with all of their eq, level, and pan information reproduced as it is in the overall mix. An AFL circuit that includes pan information is often called "solo" (see WFTD archive "solo") or "solo in place" depending upon who builds the mixer.

Aftertouch

Aftertouch is MIDI data sent when pressure is applied to a keyboard after the key has been struck, and while it is being held down or sustained. Aftertouch is often routed to control vibrato, volume, and other parameters. There are two types: The most common is Channel Aftertouch (also known as Channel Pressure, Mono Aftertouch, and Mono Pressure) which looks at the keys being held, and transmits only the highest aftertouch value among them. Less common is Polyphonic Aftertouch, which allows each key being held to transmit a separate, independent aftertouch value. While polyphonic aftertouch can be extremely expressive, it can also be difficult for the unskilled to control, and can result in the transmission a great deal of unnecessary MIDI data, eating bandwidth and slowing MIDI response time.

AGC

An abbreviation for Automatic Gain Control. AGC circuits do just what it sounds like; they automatically adjust the level of incoming audio so it can be recorded and/or mixed properly in whatever the device is. The way they work is almost exactly like a typical compressor and the end result often sounds the same. They have most often been employed in low cost audio and video recorders to both avoid the expense of costly knobs and pots, and so equipment is less complicated to use. We pro's always want access to our levels, but consumers often find it an unnecessary complication (this is sort of the same as the automatic versus manual transmission debate).

AIFF (or AIF)

AIF is known as AIFF on the Apple platform and stands for Audio Interchange File Format. Standardized by Apple, AIFF has been the most common file type for sound files on the Macintosh for years (with Sound Designer files a close second). In order to more easily work with PC systems (and the PC.xxx mentality) the second "F" is often dropped from the name.

AIT

AIT stands for Advanced Intelligent Tape, and was developed by Sony for the purpose of efficiently and reliably storing large amounts of data. AIT drives use a special 8mm tape, some of which have a built in memory chip called the MIC. The MIC chip is capable of storing information specific to that piece of media and in certain situations, it will improve data access and reliability. AIT drives provide large capacities and very fast data transfers by using a 68-pin Fast/Wide SCSI-2 interface.

Algorithm

A step-by-step problem-solving procedure, especially an established, recursive computational procedure for solving a problem in a finite number of steps. Algorithm's can be thought of as similar to computer programs. They are often run as subroutines to normal operations of computing devices. Algorithms are used in all sorts of DSP devices to carry out specific aspects of their functionality.

Aliasing

In digital sampling and recording, aliasing is digital distortion that occurs when the frequency being sampled is higher than one-half the sample rate (called the Nyquist Frequency). Essentially, when a frequency exceeds the Nyquist Frequency, it is "folded over" and becomes an audible component of the signal. Most digital recorders have filters, etc., to prevent aliasing from occurring. In samplers, aliasing also becomes apparent when a sample has been "stretched" too far in pitch...

All-Notes-Off

A special MIDI message used to turn of any notes that might be playing on a MIDI instrument. It is often used to recover from erroneous stuck notes that may be playing due to a fault elsewhere in the system. Some MIDI equipment has an associated button called a Panic Button that generates the all-notes-off message and broadcasts it over all MIDI channels. Sometimes an all-notes-off command can be used to stop an instrument from sounding before some specific commands are issues, such as patch change commands.

Alternating Current (AC)

Electrical current flow that reverses direction on a periodic basis. This is the way our normal household electricity works. Alternating the current flow makes it easier for electrical power to be efficiently distributed over vast distances. In the United States our current changes direction at a rate of 60 times per second (60 Hz). Audio signals are also alternating, with the frequencies corresponding to the frequencies of the sounds present.

Amperage

A measure of electrical current flow, also called amps for short. It literally equates to the number of electrons in a conductor flowing past a certain point in a given amount of time. Without going into an electronics course here, current is "drawn" from a supply of power due to the presence of a voltage being placed across a load. Uh, in English please? A typical electrical outlet is a good example. There is a 120 volt potential there (think of the voltage as "pressure"), but no current flow. Put some type of load across the outlet (like a light bulb) and you now have a complete circuit in which electrical current can flow. Current will flow according to how much resistance there is to the flow. High resistance (impedance) will yield less current flow than low resistance. Amps are a measure of this current flow. A direct short (no resistance) will cause a very large amount of current to flow and, if the supply is capable of delivering enough amperage, the wire will eventually heat up and melt (possibly causing a fire in the process). That's why we have fuses and circuit breakers. They limit current flow by opening (the opposite of shorting) a circuit before this happens. A 15 amp breaker will trip when the load(s) in that circuit try to pull in excess of 15 amps of current.

Amplitude

In physics and electronics amplitude is literally the maximum absolute value of a periodically varying quantity. In layman's terms it is the strength of a signal or sound without regard to its content. Amplitude measurements of audio signals generally refer to the signal voltage, which is only one component of what determines power (watts), or the ability to do work. Thus it is important to understand that amplitude alone does not singly determine power (or loudness in audio), but does affect it. In the physical world the amplitude of a sound is measured in dB of SPL (Sound Pressure Level), which again does not define the true sound power or intensity (many people are confused about this), only the sound level at a point in time. We'll cover this in more detail later when we define those words.

Analog

Literally, an analog is a replica or representation of something. Examples: In audio signals, changes in voltage are used to represent changes in sound pressure. On vinyl records, groove depth is an analog for sound pressure levels. On magnetic tape recorders, changes in magnetism are an analog for changes in sound pressure. Note that in all these examples, the signal analog is a continuous representation, as opposed to the quantized, or discrete "stepped" representation created by digital devices (see also WFTD "Quantization Error"). Since analogs rely on physical measurements, the accuracy of the representation will be limited only by the precision of available measuring techniques (not taking in account the characteristics of various storage media, transducers, etc.).

Analog Synthesizer

An analog (see WFTD archive Analog) synthesizer uses voltage controlled analog modules to synthesize sound. The concept of a variety of analog modules, all of which can interconnect via a standardized voltage control system, was invented by Dr. Robert Moog. The three main voltage controlled modules in an analog synthesizer are: Voltage Controlled Oscillator (VCO), Voltage Controlled Filter (VCF), and Voltage Controlled Amplifier (VCA). The oscillator (see WFTD archive Oscillator) generates a periodic waveform, the filter (see WFTD archive Filter) is usually employed to remove certain frequencies from the waveform, and the amp is used to vary the attack and decay characteristics (see WFTD archive ADSR).

Anechoic

Literally, without echoes. Anechoic refers to the absence of audio reflections. The closest thing to this situation in nature is the great outdoors, but even here there are reflections from the ground, various objects, etc. It is almost impossible to create a truly anechoic environment, as there is no such thing as a perfect sound absorber. At high frequencies, it is possible to create near-anechoic conditions, but the lower the frequency, the harder this is (Absorption is wavelength dependent. As an example, a 100 Hz wave is about 10 feet long; the absorber must be at least 1/2 a wavelength deep to function properly. It quickly becomes impractical to create a large enough space with enough material in it to absorb low frequencies).

It is not desirable to create anechoic or near-anechoic conditions in a recording studio. The total absence of reflections skews perception, and will not result in good recording or mixing decisions. Anechoic chambers are used for testing and spec'ing microphones and loudspeakers, as well as for a variety of other audio measurements.

ANSI

An acronym for the American National Standards Institute. Several engineering societies and government agencies founded ANSI in 1918. It is a private nonprofit membership organization supported by a diverse constituency of private and public sector organizations. ANSI does not itself develop American National Standards (ANSs); rather it facilitates development by establishing consensus among qualified groups. ANSI also promotes the use of U.S. standards internationally, advocates U.S. policy and technical positions in international and regional standards organizations, and encourages the adoption of international standards as national standards where these meet the needs of the user community.

ASIO

An abbreviation for Audio Stream Input/Output architecture. Developed at Steinberg, it is the software engine that is the fundamental access method to the audio hardware for Cubase VST and is being employed in a growing number of hardware and software systems for doing audio on computers.

The computer manufactures and operating system vendors target the "Multimedia" market and have implemented audio playback and recording capabilities specifically for it. This market however is based on stereo playback and recording, it did not require synchronization between other Media in the beginning, and multi channel operation wasn't necessary. So far the only professional solutions have been proprietary expensive hardware based systems.

ASIO addresses all areas for pro-audio recording including flexibility with sample rates and bit depths as well synchronization between different media like audio, MIDI and video. As a result the user gets a low latency, high performance, easy to set-up and control recording solution. The audio hardware can be either one or more sound cards with multiple audio input and output ports that conform to the ASIO specifications. ASIO exists for PC (Windows) and Macintosh systems currently.

Asperity

A small imperfection or irregularity in the surface of magnetic tape. Some gross imperfections can be visible on inspection of the tape, but most are not. Asperities are numerous in all tapes and produce asperity noise, heard as a sort of low frequency rumble, in tape recorded signals.

ASPI

A rare acronym within an acronym, it stands for Advanced SCSI Programmers Interface. ASPI is a layer of software for the PC that manages communication between all kinds of software and your peripheral SCSI devices. It comes in to play when dealing with SCSI on the PC because SCSI is not part of the normal PC operating architecture.

Asynchronous

The opposite of synchronous. A mode of SCSI operation where each byte is requested, sent, and acknowledged before the next is requested, sent etc. In synchronous SCSI transfers the byte does not have to be acknowledged before the next byte is sent, it only needs to be eventually acknowledged. Most modern SCSI drives can be set to allow synchronous transfers or not. Most SCSI devices can work synchronously or asynchronously, but some systems or SCSI controllers require one method or the other in order to perform correctly (Digidesign equipment, for example, prefers to have synchronous transfers enabled).

Asynchronous also refers to some system commands in the Macintosh, such as copy and format, which take place in the background without tying up the CPU (on new versions of the OS).

Attack

In audio terms, the beginning of a sound. What type of attack a sound has is determined by the sound's attack time, or how long it takes for the volume of the sound to go from silence to maximum level. It is critical to consider the attack time of sounds when applying processing. Compression, gating, and other types of processors can destroy a sound's attack, changing the character and quality of the audio. Reverbs can also be affected by attack time; careful use of a 'verb's predelay parameter will allow you to optimize the reverb for different types of attacks.

Attenuation

A decrease in the level of a signal is referred to as attenuation. In some cases this is unintentional, as in the attenuation caused by using wire for signal transmission. Attenuators (circuits which attenuate a signal) may also be used to lower the level of a signal in an audio system to prevent overload and distortion.

Audio Interchange File Format (AIFF)

A common digital audio file specification, AIFF allows a variety of applications running on different platforms to easily share audio files. Electronic Arts published the AIFF spec in 1985. Since then, it has been widely used on Mac, PC, and Atari computers, as well as in a variety of digitally based music instruments. Most digital audio editing software will import and export AIFF files, making the format well suited for situations where more than one program or platform must access audio data. Kurzweil's K2000 and K2500 will also recognize AIFF files, making them ideal for exporting samples to and from computer-based sample editing software.

Aux Send

Slang for Auxiliary Send, a circuit pathway (or bus) in a mixing console that supplies an independent mix, which can be routed to an external (auxiliary) device such as an effects processor or monitor system. Most modern consoles have several aux sends on each channel so several devices can process the input to any channel or groups of channels.