The Tech-T Glossary

Part 2

A - F

G - H - I - J - K - L

M - S

T - Z

G

Gain: How much an electronic circuit amplifies a signal is called its "gain". In most specs or references you will see gain expressed as a decibel value (i.e. 6 dB of gain). Occasionally, you may sometimes see gain expressed as a straight numeric ratio (a voltage gain of 4 or a power gain of 2).
Graphic Equalizer: A type of EQ that is configured to provide a graphic display of the EQ settings. Years ago equalizers were all rotary knob based. When units began to arrive on the scenes that had 15, 30, or even 45 bands (frequencies) they could EQ at once, it became difficult to see what was going on at a glance. Looking at a row of 30 knobs to get an overall idea of the EQ curve is pretty difficult. So equalizers that used sliders instead of knobs were developed and quickly won the favor of engineers due to their improved ergonomics. People liked how easy it as to see the overall EQ curve at a glance, but they also just liked using sliders more than knobs (something that we'd already figured out about mixers). The ONLY thing that makes an EQ "graphic" is this configuration of being able to see the curve at a glance. Contrary to popular belief there are graphic EQ's that have the same features as parametric EQ's, including Q controls and sweepable frequencies. Most graphic EQ's, however, only give you control of cutting or boosting a pre-selected set of frequencies at a pre-selected Q.
Green Book: Also known as CD-I or CD Interactive, this book is a subset of the White Book. It specifies an operating system, an API (Application Programming Interface) and playback hardware. Philips is the only manufacturer supporting this technology.
Ground: In electricity, a large conducting body, such as the earth or an electric circuit connected to the earth, used as a reference zero of electrical potential. A conducting object, such as a wire, that is connected to such a position of zero potential for the sake of "grounding" an electronic device.; In electricity an electrical path to ground is provided by code at all electrical outlets for safety. In the event of a failure or circumstance that might normally cause the operator (or anyone coming in to electrical contact with connected equipment) to be the return path for the electrical current the ground can shunt this current away safely and return it to earth, which is where it is trying to go anyway. The alternative of the operator being the return path can cause electrocution.; In audio, ground usually refers to either the electrical ground mentioned above, or to an audio shield. An audio shield is not always a ground and should never be used as a safety ground. That they are often at ground potential is a function of how they may be connected to other equipment. Many audio devices have the ability to disconnect their signal paths entirely from electrical ground as a way to prevent hum or ground loop problems.
Ground Lift: In some situations, it may be useful or become necessary to defeat a signal's path to ground. For example, if your studio is plagued with ground loops, or noise issues, breaking a path to ground may solve the problem. Ground lift is a switch found on many pieces of audio equipment which disconnects audio signal ground from earth or chassis ground.; Using ground lift switches is considered to be far safer than the old "3-to-2 prong AC adapter" solution (which is heartily NOT recommended). However, occasionally ground lift switching will not be as effective in resolving problems as the brute force AC adapter might be.
Group Delay: A characteristic of electronic components that causes different frequency ranges to be delayed by varying amounts. Typically, highs and mids will be delayed less than low frequencies. By using careful designs, engineers can take advantage of this to reduce true harmonic distortion, and improve sound quality.
Guard Band: A thin, unrecorded area between the recorded tracks on magnetic tape. The physical separation provided by a guard band helps to cut down on crosstalk between adjacent signals on the tape.

H

Hangover: In the domain of sound reproduction, hangover is the tendency of a loudspeaker cone to continue moving after reproducing a sound, or especially, a transient. This is both a low frequency and high frequency phenomenon, and can only be reduced by adding damping to the system. One way of doing this is to increase the damping factor of the amplifier.
Harmonic: In audio a harmonic is sort of the opposite of a fundamental, though technically the fundamental is also considered a harmonic. Pretty confused? Harmonics of a particular waveform are multiples of its fundamental frequency. The first multiple is obtained by multiplying the fundamental frequency by one. Therefore in a strict sense the first harmonic is the same value (frequency) as the fundamental. The rest of the "harmonic series" (2x, 3x, 4x, etc.) of a sound make up the basic character, or timbre, of the sound based upon all of their relative amplitudes.
Harmonic Distortion: Since no electronic device is perfectly linear (meaning the output exactly equals the input) harmonic distortion is a fact of life in all audio components. Most audio signals have harmonics associated with them (a perfect sine wave is one notable exception), and that is what gives them their characteristic sound. An oboe sounds different from a violin mostly because of the harmonic series produced as part of their distinct sounds. The corresponding difference in the shape of their respective waveforms is easily distinguished when viewed on an oscilloscope or a computer audio editing program. Harmonic distortion is the result of a device subtly, or not so subtly, changing the shape of the waveform which alters the relative levels of various harmonics associated with that sound. The more harmonic distortion there is the more the sound will begin to take on the quality we all know and love that we call "distorted".
Headroom: The difference between the normal operating level of a device, and the maximum level that device can pass without distortion. Music generally has wide variations in dynamic range; without enough headroom, you'll find your gear clipping (distorting) far too frequently! There are a variety of other places where it is desirable to have large amounts of headroom (i.e. when mixing signals together); in general the more the better!
Hertz: Abbreviated Hz, and named for Heinrich Hertz, a 19th-century German physicist who first investigated radio waves. Hertz is technically defined as the inverse of the time required for one complete cycle of a wave. Thus, a 10 Hz sine wave takes 1/10 of a second to complete a full cycle. In practice, it is the frequency or number of wave cycles occurring per second. In the audio range this equates to what we perceive as pitch.
Hypercardioid: A polar pattern name typically used to describe microphone pick up characteristics. Hypercardioid patterns are similar to cardioid patterns in that the primary sensitivity is in the front of the microphone. They differ, however, in that the point of least sensitivity is at the 150 - 160 and 200 - 210 degree positions (as opposed to directly behind the microphone in a cardioid pattern). Hypercardioid microphones are thus considered even more directional than cardioid microphones because they have less sensitivity at their sides and only slightly more directly behind. Hypercardioid microphones are frequently used in situations where a lot of isolation is desired between sound sources.
Hysteresis: In almost any mechanical, magnetic or electronic system, the system's response to rising input is different from its response to falling input. This is obviously an important factor in the response of a recorder or other audio component - you want both the positive and negative parts of the waveform represented accurately! This is part of the reason for things like bias on tape recorders.

I

IEEE 1394: A subset of the SCSI-3 standard, also known as Firewire, IEEE 1394 is a new high speed data exchange protocol developed at Apple. Occasionally it is referred to as "serial SCSI" because it is a serial protocol and conforms to SCSI standards as well. It is now a common interface on new digital video equipment and is beginning to be used in audio as well (MOTU's 2408 system for example). IEEE 1394 is fast: it starts at 100 Megabits per second and goes on up past 400 Mbs, easily handling the bandwidth required for a 30 frame-per-second 640x480 pixel datastream from a prosumer video camera. IEEE 1394 supports asynchronous transfers, as well as isochronous transfers so that a stream of video from a video camera can co-exist on the same bus with another sending device, yet the bus will still carry the video images continuously without discontinuities. Another benefit is that it is a hot swappable technology and allows 63 devices on a buss with auto termination and identification.
Imaging: The ability to localize a sound in a stereo field or mix is called imaging. Several things will affect the ability of a speaker system to image accurately: How matched the speakers are in construction and level (volume), exactly matched phase, and the interaction of the speakers with the listening environment will all be critical in determining imaging. Assuming that the first items are determined by the speaker manufacturer and your system set up, careful acoustic treatment in your room can often make the largest difference in the clarity, stereo spread, and imaging of your studio monitoring system.
Impedance: Measured in ohms, impedance refers to the resistance of a circuit or device to AC (alternating current). Such an AC circuit could be any two audio devices connected together, like a speaker and an amp, passing audio signals. All other things being equal, more power (watts) will flow through a speaker with a low impedance than one with a high impedance. This will also put a greater strain on the amplifier to try to produce this power. If the impedance is too low your amp will not be able to handle it and bad things will happen. Most modern electronic audio devices have extremely high input impedances so they can be driven by very low power outputs. This is one of many reasons why high quality audio equipment can be built so much less expensively these days.
Infrasonic: Refers to sounds or signals whose frequencies are below the normal human hearing range, generally considered to be 20 Hertz. The lowest audible frequency is not easy to absolutely define as it depends strongly on level. Some experiments have found that hearing can extend down to 10 Hz at very high levels. Sometimes the term "subsonic" is wrongly used to mean infrasonic. Subsonic actually refers to the speed of sound propagation through a medium and has noting to do with frequency or pitch.
Interleaving: A method of reducing errors in digital data, interleaving distributes and intermingles the consecutive bits or words of data, spreading them over a wider area on the storage media, and scattering potential errors. This helps protect against consecutive errors when the data is read back. A variety of different schemes for interleaving are used for CD, DAT and other medias, but as an example, on DAT one method uses one record head to write the right channel's even samples and the left's odd samples, while the other head writes the left channel's even samples, and the right's odd samples. In the case of a burst error (such as those caused by a dirty head), only half of either channel's samples will be affected, allowing interpolation to conceal the lost data.
Inverse Square Law: Useful when setting up a microphone or speaker, the inverse square law states that, in a free field the intensity of sound drops by 6 dB for each doubling of distance from the source. Now, none of us ever work in a truly free field (no reflective surfaces), but for most applications these numbers are accepted as workable. In real world terms, this means that for each time you double the distance between your sound source and a listener or microphone, the power of the audio drops by 75% - a fairly significant amount! How much is this in terms of volume? Well, it depends on the source you consult, both 6 dB and 10 dB have been convincingly listed as doubling or halving the volume (let's just say it's subjective and leave it at that...) - regardless, 6 dB is a very noticeable drop in level! Consider this the next time you place a microphone or speaker: Rather than just cranking up or attenuating the mic preamp or amplifier level for gain control, look at the distance to your source.

J

Jam Sync: Jam Sync is a mode available on many time code readers/generators/synchronizers. When jam synching, a TC reader/generator looks at external, incoming time code. It then "jams" or creates new time code based on that signal, identical to the original. There are two reasons for doing this: First, to replace degraded or defective time code. Second, when copying a time code track from one tape to another (mainly for analog tape, where each copy generation degrades the code signal).
Jitter: In a digital recorder or sampler, errors in the timing of sample acquisition due to rapid amplitude changes is called jitter (also known as Sample Offset Uncertainty). Jitter introduces some distortion and phase discrepancy into an audio signal. Higher frequencies are more susceptible to jitter than lower. In most modern gear, use of clocked buffers for digital data streams has minimized the effects of jitter.

K

Keyboard Scaling: Keyboard scaling is a parameter found in many modern keyboards. Its function is to provide a way for a sound to be altered smoothly across the range of the keyboard. This is accomplished by using the key number as a modulation source and routing it to some parameter the user wishes to alter. Level scaling changes the loudness of the sound, while filter scaling changes its brightness. In some synthesizers key scaling can be routed to many parameters simultaneously with different strengths and polarities. The practical application comes out of programmers trying to make acoustic type sounds more realistic across the entire range of the keyboard.
kilo (lower-case) versus Kilo (upper-case): kilo - A standard prefix (abbreviated "k") representing 1000. For example, a 4 kiloHertz (kHz) sine wave has a frequency of 4,000 Hertz.; Kilo - A standard prefix generally used in reference to computer equipment. Abbreviated "K", it was developed to represent the binary value of 2 to the 10th power (1024). Thus, 8 Kilobytes (Kb) of memory is 8 times 1024, or 8,192 bytes.

L

Land: The area between the pits on a compact disc.

Latency: You'll see this term used in a variety of contexts, but it means essentially the same thing: Latency is the amount of time it takes for a device to respond to a command. In a MIDI instrument, it might be the time between a "note on" message being sent, to when that note actually sounds (might be different figures for single notes, chords, or multitimbral operation). In a hard drive, there is rotational latency; in a hard disk recording system, there is latency in getting all the tracks to sound, and so on.
Lavalier: A lavalier (a.k.a. Lapel Mic) is a small microphone designed to be worn on clothing or to hang around one's neck. They are used in applications where a large hand held microphone would either be too cumbersome or unsightly, or both. They are typically made with an extreme low frequency rolloff to reduce rumble and noise from moving against clothing.
LEDE - Live End, Dead End: LEDE is a trademarked term for a particular acoustic design. In an LEDE studio, the area around the monitors is deadened, or made absorbent acoustically. The remainder of the room (behind the listener) is made "live" or reflective. The main principle is that the arrival of reflections at the console is in a specific order: 1. direct sound from the monitors; 2. First studio reflection (from the recording room, through the mics and monitors); 3. First control room reflection (off the back wall, assuming it is 10 feet or so behind the engineer). The idea is that by staggering these arrivals, the control room reflections don't interfere with monitoring recorded studio acoustics.
Leveler: A leveler is a device which uses an audio gain circuit to increase or decrease the level of audio passing through it based on parameters set by the user. The purpose is not specifically to reduce the dynamic range of a signal like in a compressor or limiter (though that is often what happens), but to simply have an audio signal stay at roughly the same volume for an extended period of time. Most levelers are not sensitive to peaks of a short duration, but affect very gradual changes to audio level as input levels gradually change over time. Sometimes levelers are referred to as automatic gain controls (AGC). A high quality leveler will work far better and more transparently than the typical AGC control you might have used that was built into an audio or video tape recorder.
LFE: Low Frequency Effects, a term used in surround sound mixing. Low-frequency effects are mixed to a separate so-called LFE channel in modern movie sound production. The LFE channel carries non-essential effects enhancement - such as the low-frequency component of an explosion - often at higher levels than the other channels. The idea behind a separate LFE channel is to provide the extra low-frequency headroom needed to put low-frequency signal components on equal psychoacoustic footing with midrange signal components which require less energy for the same perceived loudness.
Limiter: A limiter is a dynamics processor very similar to a compressor. In fact, many compressors are capable of acting as limiters when set up properly. The primary difference is the ratio used in reducing gain. In a limiter, this ratio is set up to be as close to infinity:1 as possible (no matter how much the input signal changes, the output level should remain pretty much constant). The idea is that a limiter establishes a maximum gain setting, and prevents signals from getting any louder than that setting.; Like compressors, limiters are used for a variety of applications. A few: Maximizing signal levels while preventing distortion when using digital recorders, preventing overload in a signal chain, setting a maximum volume level to protect users of in-ear monitors, protecting speakers and amplifiers from clipping, and so on. Any time you want to establish a maximum gain setting and prevent signals from passing it, a limiter is your tool of choice!
Linear Interpolation: In digital recording, linear interpolation (also known as averaging) is a method of error concealment. This method looks at the values before and after a bad digital word and replaces the error with an average of those values. This method works acceptably well provided there is only a single error; multiple consecutive errors become problematic when using linear interpolation.
Linearity: In an analog to digital, or digital to analog converter, linearity refers to the ability of the converter to record or reproduce various signals, particularly low level signals, at the correct amplitude. For example, when a bit changes from 0 to 1 in a D/A converter, there must be a corresponding change in the analog signal output. Any non-linearity results in audio signal distortion. For a variety of reasons, it is impossible for a converter to be exactly linear, but high-quality units can come very close. In addition to poor design or quality, non-linearity can result from bit weighting errors, thermal or physical stress, aging, temperature variations, and other factors.
Line Level: Literally, the average voltage of an electronic audio signal. While technically any voltage over 25 millivolts RMS is considered line level, in the modern audio world we narrow the scope a bit to the two line level references in use today: Balanced "pro" gear runs at around +4 dBm (1.23 volts), while unbalanced "semi-pro" gear operates at approximately .316 volts (-10 dBV). "Pro" and "semi-pro" may be almost meaningless terms anymore, but the two operating levels must still be dealt with. The important thing is to match the levels of the gear you are using so that -10 equipment isn't directly feeding +4 equipment, and vice versa. If you use gear of both levels, there are various level matching devices on the market to properly interface the items.
Load: In electrical terms a load is something that dissipates power and does some work. The work done may take many forms, including generating heat as almost always happens as a side effect of work being done. Without a load no power can be transferred. A speaker is the load for a power amp. In order for current flow to occur a complete circuit must exist. In order for the circuit not to be a short-circuit (a decidedly bad thing) a load must be present to the power the amp. The power amp drives power through the circuit by way of increasing the voltage at its outputs and as a result the load (speaker) draws current and does work. In this case two major forms of work occur: The speaker moves and generates sound, and heat is produced. Any device you plug into an electrical outlet can be considered a load (toaster, light bulb, etc). Plug in too many devices drawing too much current and you will "load down" the power delivery system (another bad thing). In order to protect against this power delivery systems have fuses and circuit breakers to break the circuit when current flow gets too high. Many power amps employ current limiting devices in their output stages to limit current flow without interrupting the audio. It's sort of a self regulating protection system (back in the old days the amp just blew up). An important thing to understand is that a load will DRAW from an available pool of power all of the current it needs to operate at the given voltage. This is somewhat simplified, but in principle remains fundamentally true for all electrical systems. A speaker's impedance rating is an indication of what kind of load it presents to an amplifier. An appliance's current or amperage rating is exactly the load it will place on the electrical system. The reason a speaker cannot be rated in exact terms of current usage is because the voltage and frequencies presented to it constantly change. Impedance is a way of approximating a speaker's resistance to a varying voltage and frequency signal.; Also related to us is acoustical loading. The efficiency of a loudspeaker depends to some extent on the acoustic load placed on it by the way it couples to a cabinet and the surrounding structures. A speaker placed in the throat of a horn, for example, will see a higher acoustic impedance than a speaker placed in a free space.
Low Frequency Oscillator (LFO): An oscillator is an electronic circuit which produces periodic or regularly repeating waveforms; i.e. sine, square, sawtooth, or triangle waves. An LFO is an oscillator producing these waveforms at a very low frequency or pitch. These slowly vibrating, generally subsonic waves (0 - 20 Hz or so) are often used to modulate or change a parameter in a synth, sampler, or effects processor. One common application is modulating the pitch of an audio oscillator with an LFO; this results in vibrato. If the volume of an audio oscillator is modulated with an LFO, the result is tremolo. Just about any time you see a "modulation" control on a device, it is controlling an LFO, and being used to periodically change some parameter.
Lowpass Filter: A filter specifically designed to remove frequencies above the cutoff frequency, and allow those below to pass unprocessed is called a lowpass filter. The effect of a lowpass filter is to turn down high frequencies. Common examples include the "treble" controls on many lower end radios and stereos, the passive "tone" controls often found on electric guitars and basses, hi-cut filters on consoles, and of course, this type of filter is found on many synthesizers.

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