The Tech-T Glossary

Part 1

A - B - C - D - E - F

G - L

M - S

T - Z

2:1 Rule of Ambience
To capture an equal amount of room ambience, a cardioid microphone must be placed twice as far from a source as an omnidirectional pattern microphone. Keep this in mind the next time you are trying to capture some of a room's natural sound when recording!
3/2 Pull Down
One of two types of pull down pertinent to synchronization. 3/2 pull down is a specific type of pull down employed when film is transferred to video tape. The problem is that film is generally shot at 24 FPS (Frames Per Second) and video in the United States is generally around 30 FPS. It is desirable to have each film frame correspond to a video frame, but because they operate at different frame rates this is impossible. It is also not acceptable to speed up the film to video's 30 FPS during transfer to video (unless it's one of those old Charlie Chaplain movies). The best compromise has been to employ 3/2 pull down during the telecine transfer, which reconciles the 24 FPS of film with the 30 FPS of NTSC video by scanning in the second field of video twice on every other frame of film. This effectively creates an extra 25% of visual filler to occupy the six extra frames of video that occur each second. Amazingly this process is pretty much undetectable to those who aren't looking for it and is the way film has been transferred to video for many years now.
A subset of the Dolby AC-3 sound playback standard (a.k.a. Dolby Digital), and the specific format sound data is in that corresponds to that standard. This is the current state of the art home theater surround sound technology. It means that there are five channels of information (left, center, right, left rear, right rear) and one active sub channel (the .1 channel). It is also a major standard that is becoming part of the DVD standard, which means there will be numerous releases in this format for the next few years.
70-Volt System
A type of speaker distribution system where transformers are used at the output of an amplifier and at each speaker in order to provide a constant voltage of, in this case, 70.7 volts that can be tapped by multiple speakers. These lines can be run great distances with less loss and can have many speakers on them as compared to typical high current speaker lines. These types of systems are generally employed in situations where an amplified signal must be distributed over vast areas without a need for very high SPL in any one area. This is typically the type of P.A. system you will see in schools, churches, business offices, and commercial facilities like malls and shopping centers.
Absolute Time Code
Absolute Time Code (ATC) is generally recorded in the 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 synchronize 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.
In acoustics, 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.
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 use 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.
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.
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 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 command is sent, until the disk reaches the data sector requested. Access time is a combination of latency, 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.
An acronym for Alesis Digital Audio Tape. Taken from the acronym "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.
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.
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.
AES/EBU protocol
AES (Audio Engineering Society) EBU (European Broadcast Union). A professional transmission that conveys two channels of interleaved digital audio data through a single, two-conductor XLR cable (for example, a standard microphone cable).
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, 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" or "solo in place" depending upon who builds the mixer.
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.
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.
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.
The adjustment of an analog tape machine's tape head and electronic circuitry to standardize playback and record frequency and signal levels within industry accepted standards for reasons of compatibility.
Alignment tape
A reference reproduction tape used for aligning analog tape machines.
The process by which a signal level is increased by a device according to a specific input/output ratio.
The distance above or below the centerline of a signal's waveform. The greater the distance or displacement from the centerline, the more intense the pressure variation, electrical signal, or physical displacement within a medium.
Analog-to-digital (A/D) converter
A device that converts analog signals into digital form.
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 decision. Anechoic chambers are used for testing and spec'ing microphones and loudspeakers, as well as for a variety of other audio measurements.
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. 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.).
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.
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).
Acronym for Adaptive Transform Acoustic Encoding. An algorithm used for audio encoding that reduces the size of audio files. It basically works on the concept that low amplitude frequency components in a program signal are temporarily masked by adjacent frequency high amplitude components, and thus can be removed from the audio signal without a significant reduction in quality. The resulting decrease in data allows for smaller sound files. Opinions differ on how much of a sacrifice in quality there is for different types of music, but it is a commonly used algorithm. More recently ATRAC Pro has been developed. It attempts to address specific concerns of the pro audio community, most notably sound quality. There are other forms of ATRAC that are optimized for specific data storage mediums.
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.
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.

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Literally, bandwidth is a frequency span. Beyond that definition, its meaning will depend somewhat on context. For example, the bandwidth of a bandpass filter is the upper cutoff frequency minus the lower cutoff frequency (cutoff frequency being the filter's -3 dB point). The audio bandwidth is generally given as 20 Hz to 20,000 Hz, although there are harmonic components of audio that extend far above the 20k point. In most situations where bandwidth is given as an audio spec, the wider the frequency range the better. Be sure that when comparing bandwidth on different devices, that the same spec is being expressed. For example, some effects devices cite their bandwidth spec based on the dry, or unprocessed signal, while others give the bandwidth of the actual processed sound. The difference between these two specs (both listed as "bandwidth") can be substantial!
A word used to describe patch bays or cables. For all practical purposes it has the same meaning as TT (Tiny Telephone). Western Electric/AT&T invented the Long Frame plug for switchboards, which is why it's called a 'phone' plug. Switchcraft invented the Tini Telephone and registered Tiny, Mini, TT, etc. In the 1960's, ADC was formed by a group of x-Soundcraft employees and they needed a name that would not infringe on Switchcraft's trademark. The Bantam boxing class was the source for the name. Switchcraft now uses the terms "Bantam-Type™", "TT™" and "TT-Jax™" in the same sentence.
Bass Reflex
A type of speaker cabinet design. Bass Reflex cabinets use an opening, or port, in the speaker cabinet to enhance bass frequencies. The idea is that the sound pressure generated by the back of the woofer (inside the cabinet) is routed out the port, where it is mixed with the sound coming from the front of the woofer. By careful design of port size and position, the amount of low frequencies and how low they extend can be greatly modified.
Baud Rate
The transmission speed (or rate) of a modem. Named for French telegrapher Emile Baudot, who developed a five-level telegraph code. Baud rate is roughly one half dot cycle per second in Morse code, one bit per second in binary signals, or other values depending on the coding system used. Note that baud rate is not the same thing as bit rate, although the two are often used interchangeably - with a modem, its baud rate can be quite different from its bit rate.
You'll often see MIDI specified as 31.25 KBaud. Technically, this is not correct. The spec should actually be a bit rate of 31,250 bits per second.
Beats (or "Beating")
The result of combining two sounds less than 30 Hz or so apart in frequency together, beating is the alternate reinforcement and cancellation of amplitude in the combination sound (over around 30 Hz in difference results in an rougher "out of tune" sound, rather than distinct beats). Most easily heard when the original sounds are of equal volume, the frequency of the beats will be the difference in frequency between the two signals. Beats are common in most musical instruments, and are often used for tuning; when the instruments are in tune, beating disappears. When complex sounds are combined, beating occurs between various partials in the signals - listen to a piano for a good example of this. Beating is not restricted to musical instruments, it can occur between any two signals or sounds.
          1. On tubes, bias is a small direct voltage applied to the grid to move the operating point of the device into a more linear range so as to reduce distortion.
          2. On FETs, bias is a small direct voltage applied to the gate to move the operating point of the device into a more linear range so as to reduce distortion.
          3. One bipolar transistors, bias is a small direct current applied to the base to move the operating point of the device into a more linear range so as to reduce distortion.
          4. In magnetic tape recorders, bias is a very high frequency signal (often in the 100 kHz or higher range) mixed with the audio signal during recording. The purpose of bias is to reduce distortion by rapidly saturating the tape in both directions, minimizing the time spent in its non-linear magnetic hysteresis curve. In more basic terms, bias makes it easier for the tape to respond to the audio signal in a linear fashion, and reduces distortion.

In all cases, the level of the bias must be carefully adjusted to achieve the best results. Too little bias increases distortion; too much reduces signal level and diminishes high frequencies.

Binding Post
A type of electrical terminal, a binding post is most commonly found as the output connector on a power amplifier, or as the connectors on a speaker cabinet. A binding post is a very versatile connector, accepting banana plugs, alligator clips, bare wire, and others. Generally, binding posts are color coded, with the black connection going to ground, and the red connecting to hot. Binding posts offer fast, easy connections, and provide reasonably good surface area contact for good conductivity.
Bi-phase Sync
Bi-phase is an older synchronization technology used in the film industry. Typically, the clock was derived from a box that hung off of large film mag recorders. This box emitted a pulse that allowed sync. Working with pulses alone, bi-phase sync did not provide location information, making it a rather limited solution. (Anyone remember working with MIDI clocks before Song Position Pointer and MIDI Time Code was introduced? The situation was similar...)
A contraction of "binary" and "digit," a bit is a number used in a digital information system. A bit is easy to deal with in electronic terms, having only two values, 0 and 1, alternatively expressed as "off" and "on"; "low" and "high", or "absence of voltage" and "presence of voltage". Bits are commonly grouped into binary "words" or bytes.
Bit Mapping
The action of rearranging digital data in such a way that much of the information that would require a larger digital word can be encoded into a word of lower bit depth, thereby producing a higher quality signal than what would normally be possible for the given bit depth. This is distinct from dithering which is more of a randomization of the signal at low levels. Two popular bit mapping schemes right now are the Apogee UV22 process and Sony's Super Bit Mapping. Both claim at least 20-bit performance on 16-bit recordings.
Blumlein Microphone (or Blumlein Pair)
Named for Alan Blumlein (chief engineer at EMI in London during the 1930's, and a pioneer in stereo audio), a Blumlein pair uses two coincident bi-directional (or figure 8) pattern microphones set up at 90 degrees to each other. This stereo miking technique provides a strong center image, and good room ambience. When using this technique, absolute polarity in the entire audio system is essential, mic distance from the source is critical in balancing ambience with direct sound, and since so much ambience is captured, a good sounding room is critical.
A type of coaxial connector often found on video and digital audio equipment, as well as on test devices like oscilloscopes. In audio gear, BNC connectors are normally used to carry synchronizing clock signals between devices. BNCs are bayonet-type connectors, rather than screw on or straight plugs. They are named for their type (Bayonet), and their inventor, Neill Concelman.
The process of combining several tracks together and re-recording them onto another track is called bouncing. This is normally done to free up tracks for more recording. For example, you might have three background vocals recorded on tracks 1 to 3. By combining these tracks with a mixer, and routing them to track 5, tracks 1 to 3 can be erased and used to record other materials. Keep in mind that bouncing does require that you pre-decide on levels, EQ, etc. for the tracks being bounced - once they are combined, it is impossible to adjust their relative levels (overall level and EQ of the bounced track can, of course, still be adjusted).
In audio terms, breathing is the change in audible level of background noise due to use of noise reduction or other processing. If the processing is not set up correctly, background noise will tend to "jump up" in level during breaks in program material.
Brickwall Filter
Some low pass filters are designed with such a steep cutoff slope that they resemble a "brick wall." These types of filters have often been used as anti-aliasing filters for A/D converters. Designers prefer not to use them because their severe nature often has negative side effects on the unfiltered audio such as phase shift and non-linearity near the cutoff point.
Refers to the cancellation of one signal or frequency component of a signal by another signal of equal amplitude but opposite polarity. Sometimes this is called phase cancellation. It also is a phenomenon that is part of the sound of a phaser or flanger. As they sweep through their range various frequencies are accentuated or (nearly) cancelled producing their characteristic "whooshing" effect. Hum bucking, as in hum bucking guitar pickups, is the bucking of frequencies we associate with hum (60 Hz in the United States). In this case the cancellation is of EMI that is being picked up by the guitar's pickup, which is acting as a transformer picking up various fields nearby.
In audio terms, a Bus is a point in a circuit where many signals are brought together. For example, most electronic items have a Ground Bus where all of a device's individual ground paths are tied together. In mixers, we have Mix Busses, where multiple channels' signals are brought (or blended) together; Aux Busses, where feeds from channels are brought together to be routed to an external processor or monitor send, etc. In general, the more busses a mixer has, the more flexible the routing capabilities of that mixer will be.

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Cache RAM
Most of the RAM (Random Access Memory) that computers use is inexpensive dynamic RAM. In modern computers, dynamic RAM is actually too slow to keep up with the bus speeds. To compensate for this, computers incorporate a small amount of expensive Cache (or Static) RAM, which is fast enough to keep up with system speeds (this is also called a Level 2 or L2 cache). All modern CPUs have a small cache built into the chip itself (8-16k). External cache memory can also be added to improve performance. PowerPC's require at least 256k per CPU (or more). Pentium Pro chips have a built-in L2 (256 or 512k) cache that runs at a full 200 MHz, greatly speeding performance. How much cache RAM you should have depends on the amount of total RAM your computer has. Unless you are running serious graphics systems, 256 or 512k should be fine. Increasing to 1 MB will only provide a minor performance improvement.
An electronic component, also sometimes called a condenser. Capacitors, which come in many shapes and sizes, basically do nothing more than store electrical voltage, somewhat like a battery. They are comprised of two (or more) electrical conductors (plates) separated by an insulator. Current cannot flow "through" a capacitor without breaking it (by breaking down the insulator). Thus they do not pass any DC, but do pass AC in varying degrees by the charging and discharging of the two plates. In electrical circuits their behavior in terms of the time it takes them to store and release their voltage charge can be used in multiple ways. Their use can be as simple as removing a DC component from an AC signal or short-term power for memory in electronic instruments, but they are also used in very sophisticated audio and digital circuits (as control elements). Capacitors are important ingredients in most amplifiers, EQ circuits, power supplies, oscillators, clocks, filters, and the list goes on and on. They are everywhere. A typical mixing board could have hundreds or thousands of capacitors.

There has been a great deal of eyebrow raising about the relative merits of different types of capacitors used in audio circuits. Many audiophiles hold that the quality of these components can have a significant impact on the audio quality (especially long term) of a device. This type of thing is one of many that may separate more expensive equipment from the cheap stuff.
In a tape recorder the capstan is the rotating shaft that drives the tape past the heads. Usually the tape is squeezed between the capstan and a rubber wheel known as the pinch roller. The capstan is what controls the speed of the tape and is usually attached to a "flywheel" so that a smooth and consistent tape speed is maintained.

The tape speed is not always equal to the surface speed of the capstan as one would expect. A phenomenon known as "belting" can cause tape to be moving faster than the capstan turns, especially with small diameter capstans. Also, thick tape will move slightly faster than thin tape. This means that a very slight, but audible, rise in pitch occurs whenever a spliced (therefore thicker) piece of tape moves over the capstan.
A microphone polar (pickup) pattern. Characterized by strong sensitivity to audio from the front of the mic, good sensitivity on the sides (at 90 degrees, 6 dB less than the front), and good rejection of sound from the rear, the cardioid pattern can almost be visualized as a "heart-shaped" pattern (hence its name).
The ability to reject sound from the rear makes cardioid patterns very useful in multi-miking situations, and where it is not desirable to capture a large amount of room ambience. Popular in both studio and live use (where rear rejection cuts down on feedback and ambient noise), cardioid mics are used for a very high percentage of microphone applications.
Keep in mind that like all non-omnidirectional mics, cardioid mics will exhibit pronounced proximity effect.
Cart Machine
Broadcasters' slang for "Tape Cartridge Machine," which is a playback machine that uses endless loops of tape in plastic cartridges. Commercials and announcements have historically been recorded on "carts," and some of the playback machines can handle a dozen or so of them, something like a record changer. Nowadays there are various forms of digital technology taking the place of these old machines, but many are still in service.
In musical terms a cent is 1/100 of a semitone. Cents are a common and convenient way of describing very small increments of pitch in musical terms. Keep in mind that the relationship between frequency and pitch is not linear so describing small pitch differences or changes in terms of frequency can be very confusing.
Channel Separation
The crosstalk, or bleed of audio signals from one channel to another. The amount of channel separation is inversely related to the item's crosstalk spec; i.e. a low crosstalk spec indicates high channel separation.
Used in reference to headphones. "Around the ear". Circumaural headphones encircle the ear, and provide a good seal. Typically, circumaural phones use a "closed" design, and provide good audio isolation.
Class D
A class of amplifier design. Class D pertains to digital switching amplifiers, which are very popular these days. They are characterized by output transistors that turn on and off at full rail (supply) voltage at varying frequency depending on the signal intensity (volume) and particular area on the waveform being amplified. At the highest part of the waveform on either the + or - side, the output devices are on for a large part of the time, and when the waveform is near zero, they are on less of the time. For best performance, the devices must be able to switch on or off 10 times the highest frequency to be amplified, e.g., 200 kHz for the 20 kHz limit of human hearing in a full range amplifier. For subwoofer amplifiers, they do not have to be able to switch at such high frequency. The drawback is that the + and - sides of the amplifier cannot be on at the same time or there will be amplifier failure, so there can be a (relatively) large "deadtime" right before the waveform moves from the + to the - side (or - to +), that results in distortion. The shorter the deadtime, the less the distortion. Because they do not have to store vast amounts of reserve power in their power supplies these amps can often be built without the large capacitors and transformers used in traditional amplifier designs. This saves on cost and makes them potentially very light weight.
Class H
A class of amplifier output design. If an amplifier has more than one voltage rail (DC voltage delivered by the power supply), then it is designated Class H. This is a very efficient type of amplification. The output transistors of an amplifier have to dissipate, in heat (watts), the difference between the rail voltage and the voltage across the speaker terminals, multiplied by the current (Ohm's law). So, when there is a low rail voltage for use during periods of low volume, and a high rail voltage for use during loud volume, the output transistors don't have to dissipate very much power when the volume is low. This causes less drain on the power supply and makes it possible to build a very lightweight design. The drawback is distortion at mid-volume when the amplifier has to go back and forth between the two (or more) rail voltages.
A specific type of distortion. If a signal is passed through an electronic device which cannot accommodate its maximum voltage or current requirements, the waveform of the signal is sometimes said to be clipped, because it looks on a scope like its peaks have been clipped off by a pair of scissors. A clipped waveform contains a great deal of harmonic distortion and often sounds very rough and harsh. Clipping is what typically happens when an audio amplifier output is overloaded or its input over driven.

Interestingly, light to moderate clipping does not usually reduce the intelligibility of some signals, especially speech. In fact, it has been shown that clipped speech is easier to understand than normal speech in noisy environments. A probable reason for this is the increased high frequency content that accompanies this type of distortion, which can make a signal stand out more among other sounds and noises. Aphex and some other companies have been using this principle for years in their "exciter" type products. By adding the right amount of distortion at the right frequencies a signal will sound almost clearer and more distinct amidst other sounds, thus standing out more in a mix.

In the digital realm however, clipping is undesirable as it is not pleasing to the ear or musical with just about any input signal. Clipping will occur with digital circuits with a signal that exceeds 0dB full scale.
In audio terms, coincident is normally used in the context of stereo microphone pairs. The idea is to get the capsules of the two mics as close together as possible to minimize phase problems in the final recording. Often the mics are directional (i.e. cardioid) and are "stacked" one atop the other, commonly at an angle of 90 degrees. Another coincident miking approach is called "MS" or "Mid-Side". Here a bi-directional (figure 8) and cardioid mic are placed close together. By combining the outputs of the two mics in varying amounts, the apparent width of the stereo field can be changed.
A type of noise reduction used in audio equipment, a compander circuit is a combination of a COMPressor and an exPANDer. The signal is compressed before recording it to tape (which maximizes the signal to noise ratio), then expanded as the tape is played back. As the signal is expanded, tape noise tends to be "pushed down," resulting in a quieter signal.
A compressor is a device that reduces the dynamic range of an audio signal. First a threshold is established. When the audio signal is louder than this threshold, its gain is reduced. The amount of gain reduction applied depends on the compression ratio setting. For example, with a 2:1 ratio, for every 2 decibels the input signal increases, the output is allowed to increase only 1 decibel. A variety of other parameters in the compressor will also affect its performance processing specific signals; attack time, release time and others are very important.
There are a variety of uses and applications for compressors, the most obvious one being to control the dynamic range of a live performance so that it will fit into the fairly narrow dynamic range of recorders, etc. Other applications include making a signal's average level higher, increasing the apparent sustain on a guitar, evening out a vocal or bass guitar performance, fattening up sounds, and on and on. The list of possibilities is extensive!
Condenser Microphone
The condenser microphone is a very simple mechanical system, with almost no moving parts compared to other microphone designs. It is also one of the oldest microphone types, dating back to the early 1900's. It is simply a thin stretched conductive diaphragm held close to a metal disk called a backplate. This arrangement basically produces a capacitor, and is given its electric charge by an external voltage source. This source is often phantom power, but in many cases condenser mics have dedicated power supply units. When sound pressure acts on the diaphragm it vibrates slightly in response to the waveform. This causes the capacitance to vary in a like manner, which causes a variance in its output voltage. This voltage variation is the signal output of the microphone. There are many different types of condenser microphones, but they are all based on these basic principles.
Constant Q
On most graphic equalizers, changing the gain of a frequency band also changes the Q or bandwidth of that band. (As the slider is pushed up, the width of the band affected becomes wider. This also increases the overlap between adjacent frequency bands). On an EQ with Constant Q, the bandwidth remains constant no matter how far the gain is boosted or cut. This allows the EQ's effect to remain more predictable and controllable as there is less interaction between adjacent bands.
Continuous Controller
In MIDI terms, a continuous controller (CC) is a MIDI message capable of transmitting a range of values, usually 0-127. The MIDI Spec makes 128 different continuous controllers available for each MIDI channel, although some of these have been pre-assigned to other functions. CC's are commonly used for things like MIDI controlling volume (#7), pan (#10), data slider position (#6), mod wheel (#1) and other variable parameters.
Use of continuous controllers in performance and sequencing can be a major factor in adding life to MIDI music - but beware, over-use of CC messages can result in MIDI log-jam, where the amount of data being sent is more than the bandwidth of MIDI can support. (Most sequencers support commands for "thinning" CC data if this becomes an issue)
Interestingly, pitchbend is technically NOT a continuous controller. Because of the greater resolution wide bends require (to prevent "stair-stepping"), pitchbend has been assigned its own dedicated MIDI message type.
Abbreviation for Central Processing Unit. The chip on a computer's motherboard which ultimately controls all the activity of the computer. Standard Macs have a 680x0 chip (x = 0, 2, 3, or 4) manufactured by Motorola. PowerPC Macintoshes use a new RISC (Reduced Instruction Set Computing) chip designed by a conglomerate of computer hardware manufactures, including Apple, IBM, and Motorola. Most IBM compatible computes use a chip based on Intel's X86 architecture. These days most electronic instruments (keyboards, drum machines, etc.) and digital tape machines have a CPU which controls all of the functions of the machine.
Critical Distance
When dealing with acoustics, critical distance is the point at which the volume of a sound source is equal to the volume of reflections from that source off of other surfaces. Control of the volume and timing of these reflections is an important part of creating an accurate listening environment.
A technique commonly used in editing audio. One sound is faded out as another fades in, allowing for a smooth transition between the two. Crossfading is also common in samplers, where it is used to smooth loop transitions (crossfade looping), and sound design to create hybrid sounds (one sound morphing or turning into another). While we often think of this as a digital process, audio engineers have been using two channel faders on a mixing console to crossfade between two signals or tracks for many years.
In multi-channel audio systems, crosstalk is signal bleeding or leaking from one channel to another. Mixers, tape recorders, and many other pieces of gear are all susceptible to this problem. In most modern gear, crosstalk is not a major concern, but be aware that older gear can have significant amounts of bleed between channels!
Cutoff Frequency
In a filter, the cutoff frequency is the point where the response is 3 dB down in amplitude from the level of the passband. Beyond the cutoff frequency, the filter will attenuate all other frequencies, depending on the design of the filter. On a sweepable shelving EQ or filter, what you are "sweeping" (or changing) is the cutoff frequency. To our ears, this changes the point at which the filter is operating.

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D-Sub and DB are prefixes used to describe a type of multi-pin connectors that happen to be commonly used in audio equipment. The original manufacturer, ITT Cannon, adopted the "D" designation as the lead character in their part numbers signifying the connector type. The shell size, or capacity, is next in the part number: A=15 pin, B=25 pin, C=37 pin, D=50 pin and E=9 pin (not originally produced). This type connector can also be specified with many different styles and quantities (up to its capacity) of pin: high power, coax and combinations. The most common connector, early on, was the 25-pin size, which was used on RS232 ports (a common computer port). Hence DB25M means "D" type, "B" shell, 25 pins, Male pin. Note that a 15-pin female would be DA15F. D-Sub is short for the current industrial tag, D-Subminiature, used by almost all of the manufacturers.
Damping Factor
Technically, the damping factor of a system refers to the ratio of nominal loudspeaker impedance to the total impedance driving it (amplifier and speaker cable). In practice, damping is the ability of the amplifier to control speaker motion once signal has stopped. A high damping factor means that the amplifier's impedance can absorb the electricity generated by speaker coil motion, stopping the speaker's vibration.
Other points:
          1. Damping varies with frequency. Some manufacturers publish a damping curve for their amps.
          2. The effects of damping are most apparent at low frequencies, in the range of the woofer's resonance. Well damped speakers sound "tighter" in the low end. Low damping factors result in mushy or indistinct bass.
          3. Speakers connected in series or parallel will experience the same damping factor from the amp. Impedance determines damping factor, not speaker wiring.
          4. Higher impedance speakers increase system damping factor.
          5. The damping factors you see published as amp specs are for the amp only, not referenced to an entire system. Higher is better, and you'll often see quite high numbers, 200, 300, even 3000 or higher.
          6. System damping factors over 10 are generally acceptable. The higher the better.
          7. How to calculate a system's damping factor: First, calculate the output impedance of the amp into, say, an 8-ohm speaker (use the nominal impedance of whatever speaker you are using for your own calculations), and a 100 foot 12 gauge speaker cable. Let's also say we have an amp with a published damping factor of 3000. Since damping factor is the ratio of speaker impedance to amp output impedance, you can work backwards, dividing 8 by 3000, giving us .0027 ohms amp output impedance. You must also consider the impedance of the speaker cable; 12 gauge wire is in the range of .0016 ohms/foot (cable catalogs sometimes publish this spec). For a 50 foot speaker cable, you've got 100 "feet" of impedance (50' out, 50' back) giving a total cable impedance of around .16 ohms (note this is much higher than the amp's impedance - one reason larger speaker wire is better for long runs!). This makes the total output impedance .1627 - pretty low. The system damping factor will then be 8 ohms divided by .1627, resulting in a very good score of 49.
DASH stands for Digital Audio Stationary Head and pertains to a format of digital recorders. Back in the days when digital recording was in its infancy it was not yet clear whether most recorders would use rotating heads (like DAT machines) or stationary heads. Early on DAT was called R-DAT for just this reason. There was also S-DAT, but it was far less used in favor of the DASH acronym that was already in use (and because stationary head DAT machines never got off the ground). Most of the real high-end digital audio multitrack machines (those made by the likes of Sony and Mitsubishi) are DASH machines. These big machines use a reel of special digital tape that runs past a stationary head at (relatively) high speeds. They look almost like analog reel to reel machines to the uninitiated, but generally cost three or four times as much money.
Decay Time
The time it takes for the sound pressure level of reverberations to drop in level by 60 dB (one millionth) from their original strength. This is sometimes also called "reverb time" or RT-60. Carefully setting the decay time allows you to have the mix be as "wet" as you desire, without making things muddy or unclear...
Decca Tree
A stereo miking technique. A Decca Tree configuration is characterized by having three omnidirectional microphones in a "T" shaped setup. Two of the microphones are positioned about two meters apart. The third microphone is positioned between the first two, but about 1.5 meters forward (closer to the source) of them. This configuration is sometimes used for orchestral recordings and film scoring due to its natural sound with good separation. It is useful in film because the image doesn't usually cause problems with Dolby or other surround processes. In many cases the Neumann M50 (or now, the newer TLM50) is used as the center microphone because of its unique directional characteristics and smooth sound.
A decibel (named for Alexander Graham Bell) is a tenth of a bel, and is used as an expression of power. Here's where the confusion arises: A decibel isn't a measure of ANYTHING; it is a ratio of two power levels. Because of the way our ears perceive volume, these ratios follow a logarithmic curve, expressing them as a decibel keeps things easier to deal with. Here are a few convenient decibel figures worth remembering: One decibel is commonly taken as the smallest volume change the human ear can reasonably detect. Doubling the POWER of an amplifier results in a 3-dB increase, which is a "noticeable" volume increase. Doubling the VOLUME of a sound is a 6-dB increase (you may occasionally see 10 dB listed as the "double-volume" figure, 6 dB is the more mathematically correct number). By doing the math, you can see that truly doubling your volume actually requires 4 times the amplifier power! Keep these figures in mind the next time you are comparing the specs of two pieces of equipment.
elay is an electronic device designed to store a signal for a specified period of time and then release it, thereby delaying the signal relative to other parts of an audio program. Delays are often used to create echo effects, where a particular signal may repeat several times, with each repeat being lower in level than the prior one. In a digital delay the signal is simply stored in memory chips until it is needed. The longer a signal needs to be stored, and the higher the sample rate and bit depth, the more memory is required, so early digital delays tended to suffer from some of the same problems as their analog counterparts. Delay technology is at the core of most time based effects such as flangers, chorus units etc. They have also been widely used in broadcast applications over the years to provide a few seconds of delay to "live" broadcasts. These few seconds can be used by an attentive engineer to "bleep" out things like curse words, etc.
A device for removing magnetism from the heads and metal tape path components of tape machines. When magnetic (magnetized) tape passes across these metal parts it tends to magnetize them as well. In general, this is not a good thing! When these parts become magnetized they tend to degrade the signals recorded and played on subsequent tape passes through the transport. Notably, high end may be compromised, signal levels can be reduced, and so on. Opinions vary as to how often a machine should be demagged. Check your tape machine's documentation for its manufacturer's recommended maintenance schedule.
In the audio world diaphragm refers to the component in a microphone that vibrates sympathetically with air disturbances such as sound waves. It is typically a circular shaped very thin piece of mylar or other delicate low mass material that will range from .2 to 2 inches in diameter. When the diaphragm in a microphone vibrates it generates an electrical signal often by either moving an attached coil of wire in and out of a magnetic gap (in the case of moving coil microphones) or by changing the distance between it and another electrically charged plate (as in condenser microphones). These electrical impulses are then present at the output of the mic and ready for amplification as an audio signal.
An acronym for Deutsche Industrie Normung (also seen as Deutsche Institute fur Normung and Deutsche Industrie Norm), DIN is a German organization that establishes standards for industry. One common place you'll encounter DIN standards in America is with circular multi-pin plugs, like those found on the ends of MIDI cables. Other DIN standards exist, including noise specs, rack measurements, signal EQ standards and more. It is important to note that they do not necessarily match up with the corresponding American standards.
The angle of effective coverage for sound radiated from a speaker. When looking at speaker specifications, you'll see this listed with two components, horizontal and vertical (i.e. 90 degrees x 60 degrees).
Distribution Amplifier
A Distribution Amp, or DA, is a low power amplifier designed to transparently split an input signal to several outputs. For example, the stereo outputs of a mixing console might be run to a DA, where they would be split to simultaneously feed a cassette deck, a DAT recorder, a CD recorder, and so on. Using a distribution amp prevents various problems caused by passively splitting ("Y"-ing) an output to feed multiple sources.
Literally, dither is noise added intentionally to a digital recording. Low level signals are difficult for digital gear to record; the sampling machine simply has difficulty deciding whether the necessary bits should be turned on or off, creating "quantization noise." By adding a small amount of very controlled noise to the original signal, the bits can be made to positively switch on or off, improving low level sound resolution. The noise used is often "shaped" to be in-offensive to human ears. Good dithering algorithms, whether hardware or software based, can make an incredible difference in the sound quality of a digital recording!
A term used in surround sound mixing. When left, right, and center channels are available, a sound can be placed in front of the listener by mixing it entirely to the center channel, or by splitting it equally between the left and right channels. Compared to sending the track directly to the center channel, mixing the track to the left and right channels creates the impression of an extended sound source. Whether a narrow or wide source is desired depends on the situation, and many surround panners provide a so-called divergence control, which adjusts the left/center/right panning parameters to control the portion of front-placed sounds mixed to the center channel.
Diversity Receiver
In wireless microphone applications, diversity receivers are often used to improve reception of RF signals. A diversity receiver utilizes two separate, independent antenna systems. The receiver looks at the signal coming in from the each antenna, and determines which one is the stronger. It then switches to that stronger signal. The receiver is constantly comparing to see which antenna is providing the better signal, and can quickly switch from one to the other as signal strength changes.
Dolby Digital
Refer toAC-3
The Doppler effect, named after a German physicist, is the apparent change in pitch of the sound that occurs when the source of the sound is moving relative to the listener. For example: A car horn will sound higher in pitch as it approaches, and lower in pitch after it passes us. This is one principle that is employed in a rotating speaker system like a Leslie. The rapid movement of the horn to and away from the listener creates a sort of vibrato effect. There are many modern effects units that simulate the Leslie sound, and also offer other types of Doppler effects.

If a loudspeaker is producing both low and high frequencies, the low frequencies will cause the cone to move alternatingly toward and away from the listener (obviously high frequencies do this too, but the lows are much more pronounced). As this is happening the perceived pitch of the higher frequency sounds rise and fall at a rate (or rates) equal to the low frequencies moving the cone. This is actually Frequency Modulation of the high frequency by the low frequency, and is called "Doppler Distortion." It manifests itself as a sort of "muddiness" of the sound.

Double Tracking
The process of recording a track, then recording a second track while listening to the first and duplicating it. When the two tracks are played back together, the result is a slight "chorusing" and fattening of the signal due to minor pitch and timing differences between the two performances. Double tracking is an effective tool, and has been used extensively in most pop music styles. Vocals, melodic parts, rhythm guitars, and solos are common candidates for doubling, tripling or even more.
Digital Signal Processing.
Making a copy of a recorded tape is called dubbing, and the resulting copy is known as a "dub".
Dynamic (Microphone)
A dynamic mic is one in which audio signal is generated by the motion of a conductor within a magnetic field. In most dynamic mics, a very thin, light, diaphragm moves in response to sound pressure. The diaphragm's motion causes a voice coil which is suspended in a magnetic field to move, generating a small electric current. Generally less expensive than condenser mics (although very high quality dynamics can be quite expensive), dynamics feature quite robust construction, can often handle very high SPLs (Sound Pressure Levels), and do not require an external power source to operate. Because of the mechanical nature of their operation, dynamic mics are commonly less sensitive to transients, and may not reproduce quite the high frequency "detail" other types of mics can produce. Dynamic mics are very common in live applications. In the studio, dynamics are often used to record electric guitar and drums.
Dynamic Range
The dynamic range of a sound is the ratio of the strongest, or loudest part to the weakest, or softest, part; it is measured in dB. An orchestra may have a dynamic range of 90 dB, meaning the softest passages are 90 dB less powerful than the loudest ones.

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Edit Decision List
An Edit Decision List (EDL) is simply a list of desired takes and edits that will be used from the master recording, along with notes on where the cuts and edits will be performed. EDLs are commonly used in post production and video work.
A measurement of how much of the input electrical energy to a speaker is converted into sound. The remaining energy is converted to heat. Most direct radiator speakers are 1 or 2 percent efficient; a horn-loaded speaker might approach 20 percent, some reach as high as 30 percent. High efficiency means that a lower powered amplifier can be used to produce the same level, but there is also a case to be made for less efficient speakers actually being more accurate due to better damping and less susceptibility to resonance.
A type of microphone design, similar to condenser. Basically, there is a permanently charged plate in the mic element. As the diaphragm moves in response to sound pressure, it creates a changing capacitance with the plate. The big advantage to using electret (also called back-electret, or occasionally prepolarized condenser) technology is that it does not require an external polarizing voltage (battery or phantom power). In some cases, the microphone includes an impedance changing preamp that requires battery or phantom power, but the electret element itself does not require voltage. Electret mics can lose their charge in high humidity and high temperature environments, so some care should be used in storing and using them. If the electret loses charge, the mic's sensitivity will suffer, resulting in a reduced signal to noise ratio.
Envelope Generator (EG)
The envelope of a sound can be explained as a variation that occurs to it over time. How a sound starts, continues, and disappears in terms of pitch, harmonic content, and loudness is a function of its envelope. An envelope generator is a circuit or algorithm found in most synthesizers that provide a means to apply these kinds of changes to a sound over time.
Based on the root word, equal, an equalizer is an audio device whose function is to equal out the tonal characteristics of a sound. At least that was the idea back in the days when they were first conceived as a tool used to get flat response in telephone lines and to make up for the deficiencies in audio equipment and acoustic spaces. Nowadays it could more aptly be named an "unequalizer" since they are more often used creatively to alter the relative balance of frequencies to produce desired tonal characteristics in sounds. An equalizer has the ability to boost and/or cut the energy (amplitude) in specified frequency ranges by employing one or more filter circuits. There are many different types of EQ's in use today in many widely varying applications, but they fundamentally all do the same thing.
Equivalent Input Noise (EIN)
A rating of the overall noise performance of an amplifier (typically a microphone preamplifier). Basically, this is a measure of how much noise a mic preamp will add to a microphone's signal. Measurements are normally made with a 150-Ohm resistor on the preamp to simulate the load a mic would present. The theoretical limit on EIN is -130.0 to -131.8 dBm (the thermal noise generated by the resistor). When comparing this spec, keep in mind that larger negative values are better (i.e. -124 is better than -118). But don't place TOO much weight on this spec, most current EIN specs are infinitesimally small (can you REALLY hear the difference between -120 dBm and -122 dBm??)
The opposite of a compressor. Where a compressor takes a given dynamic change and reduces it, an expander increases it, making changes larger. Expanders are used to "un-do" compression in some circuits (companding). More commonly, expanders are used for noise reduction. In this application (downward expansion), a threshold is set at a level below desired audio signals, but above the noise floor. When signal drops below the threshold, expansion is applied, pushing signal even further down, reducing the level of noise. For example, an expander might be set up with a 1:6 ratio. This means that for every 1 dB of input level change the expander sees, it will output a 6-dB change. When a signal drops below the threshold by 2-dB, the output of the expander will drop by 12-dB, similarly dropping the level of any background noise floor.

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In video a field is one of two interlaced images making up a single video frame. The scan lines of each field combine to make one complete image. Fields are thus shown on a television screen at twice the frame rate. In the case of NTSC video used in the United States the field rate is 59.94 Hertz, while the frame rate is 29.97 Frames Per Second (FPS).
Field Effect Transistor (FET)
A particular type of transistor, a FET behaves in a similar fashion to a triode (tube). There are actually several types of FETs, a common one in the pro audio world being the MOSFET (Metal Oxide Field Effect Transistor). FETs have a high input impedance, and respond in a linear fashion. This makes them ideal for condenser microphone preamps, as well as for certain power amplifier designs.
A filter is an electronic device designed to reduce a signal's energy at a specific frequency. A true filter always acts as a subtractive device, not adding anything to the signal. In many filters, an amplifier is often incorporated into the circuit, allowing the frequency to be boosted or cut (active filter). Filters of different frequencies are often combined to create equalizers.
Fire Wire
A flange is the metal rim or the reel part of a reel to reel tape (also called open reel tape), as opposed to the hub. Years ago when tape machines were used to create delays in audio production a process called flanging was invented. It consisted of recording the same signal on two tapes each playing together and then, using pressure to one of the reel flanges, briefly slowing down one of the machines. The short timing discrepancies that result produce a very pronounced comb filter effect. The effect was often modulated by alternating pressure to each machine's reels. One machine would slow down relative to the other, and then the second machine would be slowed beyond the first. It was also possible to route some of the signal being played back into the recording circuit to provide regeneration and resonance effects. Later electronic flangers were invented that used a modulated analog or digital delay line, which was mixed back with the dry signal. While much more convenient than the old open reel approach many engineers agree that the electronic units have never sounded as good as the real thing.
Flat Response
A piece of gear (or a system) is said to have flat response when it outputs all frequencies at equal levels, assuming that a flat signal was used as input. That is to say, no frequency is boosted or cut in level by the "natural" frequency response of the gear. This is desirable for most studio situations; if a mix sounds good on a "flat" system, it should translate well to an end listener's system.
The resonant characteristics of an acoustic sound generator. For example, the distinguishing characteristics of the vowel sounds of a human voice, as determined by that person's physical characteristics; what makes each voice sound unique. These characteristics are actually emphasized frequency bands, and are relatively fixed in frequency despite the pitch of the voice changing.
Like the human voice, a musical instrument also has a fixed set of formants, which give it a unique, recognizable tonal color or timbre. It is this set of formants that allow us to recognize an instrument regardless of the pitch it is playing; the tonal color remains relatively static.
When a computer writes or re-writes a file to a hard disk, it doesn't necessarily write the file as one contiguous block of information. For a variety of reasons, it may put different pieces of the file in different places on the drive. More and more files become fragmented as time passes. This results in more wear and tear on the drive mechanism as it jumps around to read the files, and also in a significant slowdown in access times. The solution to this problem is to defragment your drive. Defragmenting (also known as "defragging" or "optimizing") means to re-order the files so that they are each stored as one contiguous chunk of data. A variety of disk utilities will perform this function for you. One of the things that fragments a drive fastest is hard disk recording. It is wise to be aware of how fragmented your drive is when recording, as this can seriously affect system performance. (Some manufacturers recommend optimizing if your drive has as little as 5% fragmentation.)
Free Field
A speaker or sound source is operating in a free field if there are no reflecting surfaces around the source. Technically, there is no such thing as a true free field - there's always SOMETHING for sound to bounce off of (although an anechoic chamber comes pretty close) and anytime there is a reflective surface, the response of the speaker is being changed.
FreeMIDI is a complete MIDI operating system for the Macintosh and handles all MIDI communication between various pieces of hardware, including the Mac CPU and MIDI interfaces, and any Free MIDI compatible MIDI software. It ships free of charge and is automatically installed with all Mark of the Unicorn music software products. It comes in the form of a FreeMIDI system extension, an optional OMS emulator extension (to emulate the Opcode MIDI System), and a FreeMIDI Folder, which resides in the top level of the System Folder. FreeMIDI automatically detects what type of MIDI interface is connected to the Macintosh modem and/or printer port, automatically detects what MIDI devices are connected to interface (it "knows" over 200 types of devices), and provides the user with a graphical representation of their MIDI studio. It also provides pop-up sound lists for over 100 popular MIDI synthesizers-as, generic support for any General MIDI device, and advanced features such as inter-application communication, and multiple application real-time synchronization.
Literally the number of times something occurs per unit of time. In the audio world the frequency of sound vibrations are directly related to what we hear as pitch, though the relationship is NOT linear. It is also inversely related to wavelength. We use the word frequency, and the values associated with it, as an objective way to speak about sound characteristics. Saying a unit has a frequency response of 20 Hz to 20kHz is much more accepted than specifying the response in terms of pitch.
Frequency Doubling
Generally caused by overloading a low-frequency speaker, frequency doubling makes bass instruments sound an octave higher than they really are. This is because the overdriven speaker is making the second harmonic louder than the fundamental pitch.
Frequency Response/Frequency Range
Frequency Range is the actual span of frequencies that a monitor can reproduce, say from 5 Hz to 22 kHz.
Frequency Response is the Frequency Range versus Amplitude. In other words, at 20 Hz, a certain input signal level may produce 100 dB of output. At 1 kHz, that same input level may produce 102 dB of output. At 10 kHz, 95 dB, and so on. A graph of all the frequencies plotted versus level is the Frequency Response Curve (FRC) of the monitor.
When you see a Frequency Response specification for a monitor, the manufacturer is telling you that for a given input signal, the listed range of frequencies will produce output within a certain range of levels. For example: 20 Hz to 20 kHz 3 dB. For these frequencies, the monitor will output signals that are within a 6 dB (3 dB) range. This does not mean that the speaker won't reproduce frequencies outside this range, it will! But frequencies outside the range will be more than 3 dB off from the reference level.
Abbreviation for Frequency Shift Keying. An audio tone (frequency) modulated by a square wave, which is used both for data transfer and also for sequencer and drum machine synchronization. FSK is the sound that you hear your fax or modem making as it establishes communication. In the early days of electronic music, before MIDI, drum machines or sequencers were synchronized to each other or to a tape machine via this method. Back then the only information transmitted was a rate which was interpreted as tempo by the machines. There was no location information included so the song always had to be started from the very beginning in order to achieve proper sync. If there was any drop out or glitch along the way one had to go back and start at the very beginning of the song to reestablish sync. It was cumbersome and unreliable to say the least, and that is why formats such as SFSK, DTL, SMPTE, and MTC were later adopted.
Full Code
A phrase used in digital audio applications that means a full digital signal. Digital devices (in theory) have a very finite and exact amount of dynamic range depending upon how many bits are used in recording (8-bit, 16-bit, 24-bit, etc). A Full Code signal is the maximum theoretical output of a given digital device. This is when all of the one's and zero's of the digital signal become one's for a given sample. There is no room for any more amplitude once Full Code is reached. If any more level is applied to A/D converters once Full Code has been reached they will produce numeric values that will result in massive distortion of the signal. Of course, you must remember that there are over 44 thousand such samples per second (depending upon your sample rate) so any one sample going over Full Code is not going to be audible. A few hundred or thousand clipped samples, however, is quite audible.
Full Duplex
Full Duplex is a term that comes to us from the telecommunication industry. It is the ability of a line or channel to simultaneously transmit in both directions. In the music industry, we most commonly see this term applied to computer sound cards. A "Full Duplex" audio card is able to both record and playback at the same time - a handy feature if you are performing overdubs!
Fully Normal
A patchbay term that more specifically defines a normal. The terms normal and full normal are often used interchangeably, but because normal is a generic term and could also mean half normal, it is appropriate to use the term full normal. In a full normal patch bay point, plugging a plug into either the top or bottom row of the front patch connectors will break the normal connection between them.
In audio the fundamental is the frequency of the root, or core pitch making up a pitched sound. Except for a few special cases the fundamental is always the lowest frequency making up any pitched sound and is generally the strongest pitch we hear (in a strict Physics sense the fundamental is, by definition, the lowest pitch of a sound). Most sounds are comprised of a combination of a fundamental pitch and various multiples of it known as harmonics. When harmonics are added to the fundamental, which occurs in almost all naturally occurring sounds, the character of the sound is changed. The relationship between the fundamental and the harmonics are what gives each sound its basic timbre.

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