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1/8 Space
When a speaker or sound source is placed in a corner so it is near three surfaces (like the junction of two walls and the floor) it is said to be in 1/8th space. This is similar in concept to half space (up against one wall) and quarter space (at a junction between two walls). When sound sources are placed near surfaces in this way more of the energy gets forward into the listening space (see WFTD Half Space for more info). Putting a source into 1/8th space yields and increase of approximately 3 dB more sound power level than quarter space, and 6 dB more than half space.
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 (see WFTD archive "Sound Pressure Level") 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 Phase
A positive pressure to a microphone diaphragm will (in most mics) produce a positive voltage at its output. If the correct polarity (see WFTD archive polarity) of the signal is maintained throughout the signal path this should ultimately produce a positive voltage at the speaker terminal, which will (on most speakers) cause the speaker to move forward creating a positive pressure wave in the listening position. This is known as absolute phase (see also WFTD archive phase): The original polarity of the source sound is thus reproduced in phase by the loudspeaker for listening.
Accompaniment
In music this refers to additional instrumentation that surrounds or is played along with some feature such as a solo singer, solo instrument or a speaker. For example a singer could have a piano playing along as accompaniment. Similarly, a featured pianist could have a chorus of singers singing along as accompaniment.
Acoustic Lens
In loudspeakers, a mechanical device used to improve the dispersion of high frequencies, so that dispersion is much more uniform across the audible spectrum. The lens is a product of post-World War II Bell Labs research, first described in 1949. The intent is to focus sound in much the same way that an optical lens focuses light. An axiom called Snell's law describes the refraction of sound as it passes through an interface between two materials of differing sound speed.
A high-frequency loudspeaker mechanical acoustic lens spreads a single-point sound source into a parallel wave front. Originally introduced commercially by JBL in the 1950s, they appeared in two primary designs. First was the slant-plate lens, which utilizes a series of plates with carefully calculated hyperbolic shapes, which results in a horizontal response pattern. This is the most commonly seen acoustic lens type. Second is the perforated-plate lens assembly, which consists of a collection of perforated barriers at the horn mouth. These perforated screens are actually ring shaped with varying sizes of center cutouts. Although acoustic lenses gradually fell out of favor through the 1970s and 1980s (partially due to their fragility, which made them risky to use in portable sound reinforcement systems), the technology has re-emerged in some high-end home audio systems (notably Bang & Olufsen) in recent years.
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.
Alnico
A compound word drawn from Aluminum, Nickel, and Cobalt. Alnico (AlNiCo) is a powerful permanent magnet alloy containing iron, aluminum, nickel and one or more of the elements cobalt, copper, and titanium. Alnico magnets have been used in loudspeaker construction since the 1940s, when a particularly high-energy formula (Alnico V) was developed; it had a much greater energy-to-weight ratio than common ferrite (iron) magnets.
Electric guitar manufacturers also were (ahem) attracted to Alnico magnets for pickups due to their consistency and even distribution characteristics. Two different formulas are commonly used - Alnico II and Alnico V.
Ambisonics
A British-developed surround sound system designed to reproduce a true three-dimensional sound field. Based on the late Michael Gerzon's (1945-1996; Oxford University) famous theoretical foundations, Ambisonics delivers what the ill-fated quadraphonics of the '70s promised but couldn't accomplish. Requiring two or more transmission channels (encoded inputs) and four or more decoded output loudspeakers, it's not a simple system; nor is the problem of reproducing 3-dimensional sound. Yet with only an encoded stereo input pair and just four decoded reproducing channels, Ambisonics accurately reproduces a complete 360-degree horizontal sound field around the listener. With the addition of more input channels and more reproducing loudspeakers, it can develop a true spherical listening shell. As good as some think it is, a mass market for Ambisonics has never developed due to several factors. First, the actual recording requires a special tetrahedron array of four microphones: three to measure left-right, front-back and up-down sound pressure levels, while the fourth measures the overall pressure level. All these microphones must occupy the same point in space as much as possible. So far, only one manufacturer (first Calrec, bought by AMS, bought by Siemens, sold, now Soundfield Research) is known to make such an array. Next, a professional Ambisonics encoding unit is required to matrix these four mic signals together to form two or more channels before mastering or broadcast begins. And finally, the consumer must have an Ambisonics decoder, in addition to at least four channels of playback equipment.
Amplifier, Instrument
An instrument amplifier is an electronic amplifier designed for use with an electric or electronic musical instrument, such as an electric guitar or electric piano/keyboard. Instrument amplifiers come in two main forms. The combo amplifier contains both the amplifier and suitable loudspeakers in a single unit. In the other form, the amplifier is separate from the loudspeakers, and joined to them by cables. The separate amplifier is called an amplifier head and is commonly placed on top of one or more loudspeaker enclosures, the amplifier head and loudspeaker enclosures together forming an amplifier stack. An amplifier stack consisting of a head and two loudspeaker cabinets is sometimes called a double stack.
The first instrument amplifiers were probably guitar amplifiers designed for use with electric guitars. Traditional guitar amplifiers provided a great deal of treble boost, and no high treble or low bass response at all. Some better models also provided a spring reverb and/or an electronic tremolo unit, which electric guitarists (following the lead of Fender) have confusingly always called vibrato, and similarly they call a device designed to produce real vibrato a tremolo arm. Nowadays called a whammy bar)
Guitar amplifiers were at first used with limited success with bass guitars and electronic keyboards, but it was quickly recognized that other instruments had different requirements to the electric guitar. A wide range of instrument amplifiers are now available, some general purpose and some designed for specific instruments, and even for particular sounds.
These include:
- Traditional guitar amplifiers, with a clean undistorted sound, a sharp treble roll off at 5KHz or less and bass roll off at 60-100Hz, and often built-in reverb and "vibrato" units.
- Rock-style guitar amplifiers, intended for distortion.
- Bass amplifiers, with extended bass response and tone controls optimized for bass guitars.
- Keyboard amplifiers, with very low distortion and extended, flat frequency response in both directions.
- Acoustic amplifiers, similar in many ways to keyboard amplifiers but designed specifically to produce an "acoustic" sound when used with acoustic instruments with built-in pickups.
Back-Emf
Literally,, back-voltage, is a phenomena found in all moving-coil electromagnetic systems, but for audio is most often used with respect to loudspeaker operation. This term describes the action where, after the signal stops, the speaker cone continues moving (due to inertia), causing the voice coil to move through the magnetic field (now acting as a microphone), creating a new voltage that tries to drive the cable back to the power amplifier's output. If the loudspeaker does too much of this, the cone flops around unpleasantly. It is not pleasant-sounding. To stop back-emf, the loudspeaker must "see" zero ohms looking backward (a dead short), or as close to it as possible from the output of the amplifier.
Baffle
In music, a baffle is a partition that prevents sound waves from interfering with each other. Baffles are used in speaker cabinets. It is the surface that the speaker is mounted to, and its original purpose was merely to prevent sound waves from the rear of the speaker from interfering with the waves coming out of the front of the speaker. Without a baffle they would tend to cancel each other out, especially at low frequencies. Just hold a raw speaker up in the air with a signal running through it to see for yourself. In order for a baffle to work at low frequencies it would have to be very, very large to prevent the long wavelengths from wrapping around and canceling each other. The workaround for this is the speaker cabinet, which encloses the speaker and prevents a lot of interference. Modern speaker cabinet designs have greatly expanded on the basic baffle with all kinds of little tricks and designs to improve the sound. Some basic designs include bass reflex, acoustic suspension, and horn.
Banana Plug
An electrical connector designed to join audio wires such as speaker wires to the binding posts on the back of many power amplifiers or to special jacks known as (surprise) banana jacks. A common configuration of banana plugs is to have two of them molded together and spaced 3/4 of an inch apart, which also happens to be the spacing of the binding post receptacles on the back of power amps. This assembly is commonly called a banana plug, but the more technically correct term is "double-banana plug," or it is sometimes called a "GR" plug, after the General Radio Corporation, which introduced it many years ago.
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.
Beaming
A phenomenon of loudspeakers (including horns and tweeters) where the normal dispersion characteristics of the device breakdown and higher frequencies begin to be projected straight out from the device rather than dispersing into the soundfield. To a listener it will sound like the device is only producing high frequencies when standing directly in front of it. Unless specific steps are taken to reduce or prevent beaming it will generally occur when the wavelength of a sound becomes smaller than the diameter of the device (or the throat of a horn). This means that an 18" speaker begins to get "beamy" at a lower frequency than a 10" speaker and is one reason why speakers in general aren't used to try to reproduce high frequency sounds. A horn, to a certain extent, solves this problem, but they still get beamy at very high frequencies. In the 1970's Constant Directivity horns were developed that vastly improved this performance, though there are some compromises.
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.
Bluetooth
A short-range wireless technology that communicates via a frequency-hopping transceiver over the 2.4-gigahertz radio frequency, a space known as the Industrial, Scientific and Medical (ISM) band. Bluetooth was originally conceived as a low cost, low power, short-range technology that would replace cables on such devices as mobile phone headsets, handsets and portable computers. However, its promoters soon envisioned the creation of "personal area networks" in which computers could be wirelessly connected to printers, audio could be transmitted over short distances (for example, to the rear speakers in surround setups), and remote control of PDAs or other appliances could be easily implemented. Some people have referred to it as a sort of wireless USB, which is a pretty apt description in many respects.
First conceived in 1994 by Ericsson Mobile Communications (now a part of Sony), by 1998 the Bluetooth Special Interest Group included industry giants Intel, IBM, Toshiba and Nokia. Today more than 2000 companies produce or are developing Bluetooth enabled products. Apple Computers incorporate Bluetooth compatibility that allows keyboards, mice and other peripherals to wirelessly connect to the main unit. While Bluetooth originally had a transmission range of only 10 meters, today, three power classes exist for Bluetooth devices, the most powerful allowing transmissions up to 100 meters.
Bluetooth is a different protocol from Wi-Fi, but both occupy a section of the 2.4 GHz ISM band that is 83 MHz wide. Bluetooth uses a technology called Frequency Hopping Spread Spectrum (FHSS) that allows it to hop between 79 different 1 MHz-wide channels in this band whenever it encounters interference from other transmissions.
Class A
A type of amplifier design. When an amplifier's stage devices are passing current at all times, including when the amplifier is at idle (no music playing), whether the amplifier is single ended or push-pull, the amplifier is said to be biased in Class A. Because the current is flowing at all times, an input signal causes the current to be immediately diverted to the speakers, and therefore, the sound is very "fast". In the case of a push-pull amplifier, there is also less crossover distortion when the signal passes from the positive to the negative or negative to positive, since each side of the push-pull section is already "on". If all stages of the amplifier are biased in Class A, and the amplifier operates in Class A to full output (enough current flowing at idle that could be required for full output), it is said to be a "Pure Class A" amplifier. Pure Class A designs are understandably expensive to build and are usually only found in high-end audiophile equipment.
Class AB
A class of amplifier output design. As its name implies, it is sort of a combination of Class A and Class B operation. If an amplifier operates in Class A for only a portion of its output, and has to turn on additional current in the devices for the remainder of its output, it is said to operate in Class AB. Most amplifiers are in this category, and are said to be Class A/B amplifiers, since they operate in two classes. In class AB and B, the amplifier is slower than in Class A because there is a finite time between the application of the input signal and when the devices are turned on to produce a flow of current to the speakers. However, Class AB and Class B are more efficient than Class A and do not require such large power supplies.
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.
Combo Amp
In addition to types of amplification such as solid state and tube, guitar amps come in different configurations. Combo Amps (short for combinations) are self-contained units containing the amplifier and speaker in one cabinet, as opposed to a separate amp “head” and cabinet.
Compression
Aside from the function accomplished with an audio compressor, compression is an area of increased air pressure caused by a sound wave. Sound waves, which are caused by a vibrating source in the atmosphere (such as a speaker), propagate as waves of compressed and uncompressed air pressures. The changes in pressure are very, very minute in comparison to meteorological pressure differences, but our ears are quite sensitive to the vibrations, which we pick up as sound. In a graphical depiction of a cyclical waveform, compression occurs when the wave is in the top segment (approaching what is known as the node).
Compression Driver
Developed by Bell Laboratories in the early 1930's the compression driver is a special type of dynamic loudspeaker (meaning it works just like a dynamic microphone, but in the opposite direction) designed to fit onto the small end of a horn. The horn acts like an acoustic transformer, with the driver providing a high sound pressure level at throat of the horn, with the mouth of the horn providing a large area of low pressure to radiate the sound efficiently into the air. They work by attaching a voice coil to a diaphragm (much like any tweeter) whose surface radiates sound into the horn through a small opening known as the throat, which is where the compression occurs. There are many sophisticated design variables involved in producing a high quality compression driver.
COSM
Abbreviation for Composite Object Sound Modeling. COSM is a powerful modeling technology that Roland premiered in 1995 with the VG-8 V Guitar System, and continues in the newer VG-88 system. It enables guitarists to emulate a range of classic and modern guitars, amps, cabinets, and microphones, plus it can produce "futuristic" synth-like tones. Today COSM can be found in keyboards, digital recorders, mixers, etc. It can model rotary effects, different speaker colorations, and can even approximate expensive microphones using just an ordinary dynamic mic. Its name comes from "composite object" because its core function revolves around breaking audio producing or reproducing devices down to their component parts and creating a set of instructions to emulate how these various parts interact with each other to produce a new composite that can be dynamically controlled. Of course, that's what all modeling is, but Roland coined this name to call attention to it.
Crossover
A crossover is a device designed to divide audio information into smaller frequency ranges to comply with the requirements of different transducers in an audio reproduction system. This is accomplished by running the audio through a set of filters. For example, a two-way crossover may be comprised of a low pass and high pass filter where the low pass filter passes a signal with frequencies more suitable for a woofer and the high pass filter passes frequencies the tweeter can deal with. Crossovers can be passive or active designs. Passive crossovers are usually found inside speaker cabinets along with the speaker components. These often connect to the outside world via a single jack, but sometimes each speaker component also has its own jack in case one wants to bypass the built in passive crossover. Active crossovers are placed before the power amp. In that application each frequency range is given its own power amp and its own drivers. This is where the phrase bi-amping and tri-amping come from. There are a number of different types of filter configurations used in crossovers and they each produce subtly different results. One of the big variables is how steep the roll off is at the cutoff frequency. Common configurations are 12 dB per octave, 18 dB per octave, and 24 dB per octave. Each design has its own strengths and weaknesses, but in general steeper roll offs are considered better in modern applications.
D'Appolito
A loudspeaker configuration developed by and named for Joe D'Appolito, in which a high frequency driver, or tweeter, is positioned between two midrange or low frequency drivers that each cover the same frequency range. Depending on the exact implementation the speakers can be positioned with a vertical and/or horizontal orientation. In either case the two midrange drivers serve a couple of purposes: they combine to create a larger effective woofer or midrange driver size, and they also serve to control the dispersion of the tweeter. The tweeter's output is somewhat corralled or contained by the sound coming from the midrange drivers in a similar way to how two parallel surfaces control dispersion. There are some variations on the design where two same sized woofer/midrange drivers may cover slightly different frequency ranges, however those aren't considered true D'Appolito designs.
(Input from an inSync Reader:)
The D'Appolito design specifies a third order crossover network. The tweeter is coordinated with the woofer so that at the selected crossover frequency, the drivers all have similar horizontal dispersion. (This is not easily accomplished because many drivers behave badly at the extremes of their range.)
The advantage of doing it all correctly is one of the most seamless blending of drivers possible. The result is an absence of any sudden change in directivity with frequency. This may not mean much for monitors where there is a limited listening area, but in a typical room where a large percentage of the sound is reflected by the room, the effect is dramatic.
Damping
In physics this relates to decreasing the amplitude of a wave, whether represented electrically or mechanically. In acoustic instruments we refer to the mechanical context, where we may dampen or reduce the vibration of strings on a piano, guitar, bass, etc. Applying muffling to drums and other instruments would also qualify. In acoustics this could refer to reducing sympathetic vibrations or the acoustic reflectivity of something. For example, applying acoustic absorbers to a wall surface or the inside of a speaker cabinet effectively dampens or reduces reflections.
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:- Damping varies with frequency. Some manufacturers publish a damping curve for their amps.
- 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.
- Speakers connected in series or parallel will experience the same damping factor from the amp. Impedance determines damping factor, not speaker wiring.
- Higher impedance speakers increase system damping factor.
- 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.
- System damping factors over 10 are generally acceptable. The higher the better.
- For the tweaky among you, here's 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.
Decoupling
The process of isolating one stage of an amplifier from another. Decoupling prevents unwanted oscillations (see WFTD Oscillator) and other noises that may occur due to unwanted feedback through common power supply connections (see WFTD Coupling). It also provides further filtering of the power supply to reduce any lingering AC ripple, producing a cleaner DC supply for the low-level preamp stages. This decoupling is often accomplished by adding a resistor in series with the power supply to a gain stage and a large-value electrolytic capacitor from the supply to ground after the resistor, however, there are a number of other designs employed as well.
In acoustics decoupling refers to mechanically isolating masses from one another, particularly masses that are vibrating, such as speaker cabinets. This prevents the undesired transmission sound through additional materials that can result in a compromise in sound quality to he listener or at the microphone.
Dipole
In physics, a pair of equal and opposite electric charges or magnetic poles that are separated by a small distance. This term has been adapted to cover audio and video concepts in two different ways.
In audio a dipole loudspeaker contains two drivers, usually directed 180 degrees in opposition to each other and wired in opposite phase to each other. Dipole loudspeakers are often found in home theater surround systems where they serve as rear (and sometimes side) satellites. Their donut shaped dispersion pattern can be effective for enhancing the sensation of envelopment that is an important part of the surround experience.
In radio and television, a dipole antenna is an aerial half a wavelength long consisting of two rods connected to a transmission line at the center. The most common example of this is the "rabbit ears" antenna that is often used to pick up local television broadcasts. Many wireless monitor and assistive listening system transmitters use dipole antennas.
Direct Current (DC)
Basically direct current is the operational antonym for Alternating Current (AC). The main distinction being that DC flow does not change directions, and in the case of "pure" DC does not vary at all. Direct current is almost always what is used inside of electronic devices to power the various chips and components, but is considered "bad" or harmful in audio signals, especially those going to speakers. This is because DC produces no sound yet uses a lot of power to be reproduced (remember current flow is constant in DC).
Dispersion
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).
Dolby Pro Logic
Dolby's second generation licensed home surround system. A major advantage of Dolby Pro Logic over the preceding system (Dolby Surround) is the use of an active center channel with its own speaker. Conventional stereo systems create a phantom center channel, which is effective for viewers seated directly in front of the television screen. However, for viewers seated off center, the dialog can appear to come from off center. But with Dolby Pro Logic and the use of an appropriately placed center channel loudspeaker, the dialog always appears to come right from the screen, allowing the main left and right stereo speakers to be widely spaced for a good spread on music and effects. Dolby Pro Logic decoders also decode surround information which is typically fed to a pair of surround speakers slightly behind and to the left and right of the listener.
Dolby Virtual Speaker
An algorithm created by Dolby that attempts to reproduce the dynamics and surround-sound effects of a precisely placed 5.1-channel speaker system from a consumer electronics device or personal computer equipped with as few as two speakers.
The algorithm at the heart of Dolby Virtual Speaker technology is based on psychoacoustic parameters that include an understanding of sound from both a technical and an experiential perspective. Dolby Virtual Speaker technology uses biological, psychological, and physical understanding to create the "impression" of additional speakers positioned exactly at the recommended locations for a Dolby Digital sound system with five actual speakers. In other words, audio channels are processed through filters that simulate the sonic signature of a speaker located within an acoustic space.
Dolby Virtual Speaker technology was launched in fall 2002 to the PC industry, and is currently available on select software DVD players from CyberLink, InterVideo, and Nvidia, as well as models from leading PC OEMs (including Sharp, NEC, Sony, Fujitsu, and Hitachi).
Doppler
The Doppler effect, named after a German physicist (how come things are always 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" (subjective audio term #108) of the sound.
Dual Concentric
A term used to characterize certain loudspeakers. The word concentric indicates a common center. Loudspeakers where the woofer and tweeter share a common center point are known as dual concentric (sometimes called coaxial, though this is not as specifically precise). Dual concentric speakers have the advantage of all sound emanating from one point (they are also called "
Efficiency
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 resonances.
Electrostatic
Literally electricity that is not in action, otherwise known as static electricity, which is technically an electrostatic charge. Anytime one surface or point has an electric charge relative to another you have electrostatic energy, which is a form of potential energy. Electrostatic is also a type of transducer design. Most widely employed in loudspeakers, electrostatics (as they are often called) are built somewhat like a large capacitor. There are two plates, one of which can move. A DC bias voltage is applied to them to create an electrostatic charge. Then the AC audio signal is applied, and the interaction between the resulting (alternating) magnetic field and the electrostatic field forces the movable plate to move back and forth. Some audiophiles consider this a more pure and better sounding method of reproducing audio than our typical moving coil dynamic loudspeaker designs. However most tend to be quite expensive and it is debated whether there is an overall improvement in sound - many electrostatic speakers are characterized as having little or no bass punch. In fact separate subs of standard moving coil design are often employed as a workaround for this.
Envelopment
A term used to describe the degree to which an audio signal is perceived as being all around the listener. The term "envelop" literally means to enclose or cover completely. In audio production envelopment has been adopted to characterize a property of surround sound mixes. For example, a 5.1 encoded DVD video or DVD-Audio of a live concert is likely to incorporate more than the sound of the artist in front of a listener. It would also include the sound of the audience and additional room ambience beside and behind the listener, and in some cases the listener is placed on stage with the artist(s) with instruments coming from all sides.
Envelopment is a result of panning and routing signals to multiple speakers in a surround system. In a sense, the "opposite" of envelopment is localization.
Excursion
In audio, excursion relates to a speaker's movement. The excursion is the distance it travels back and forth (in and out) from its nominal resting position. Different types of speakers are designed to accommodate different amounts of excursion. Usually speakers designed to move massive amounts of air such as low frequency drivers or subs have more excursion than high frequency drivers. Those with more excursion may also exhibit poor damping and sound very loose or sloppy, so a designer must find a happy medium and keep in mind the enclosure and intended application when choosing speakers. If a speaker is pushed beyond its limits you may hear a "cracking" sound as the voice coil slams into the bottom of the magnetic gap (during inward movement) or slips out of the gap (during outward movement). This is called overexcursion or "bottoming out" and usually just precedes a failure.
Extended Surround
Star Wars: Episode I was the first of a number of films using an additional rear channel routed to the array of speakers along the back wall of a cinema. In the cinemas, this back channel is not a discrete channel, but is matrixed into the left and right surround channels, much as the center front channel was matrixed into the left and right front channels in earlier matrix optical surround formats. This matrixed back channel is embedded in the soundtrack printmaster, so finds its way into all cinema digital sound formats. DTS uses the name "ES" on its cinema decoder while Dolby calls the same process "Surround EX". Either set of letters stands for Extended Surround.
Feedback
Literally the return of a portion of the output of a process or system to the input. In our discourse (of audio and video production) we mostly encounter feedback when an open microphone is picking up sound from a nearby loudspeaker that is also being used to amplify sound from the same microphone. This forms what is known as a feedback loop. The sound of the room enters the microphone and is then amplified by the speaker. This amplified sound then becomes part of the sound of the room entering the microphone, which causes it to get amplified by the speaker again. If too much of this "feedback" occurs the signal will "run away" and quickly degrade into an oscillation at some frequency. This sound is the "squeal" we've all come to know and hate and is what we typically call feedback (though technically feedback occurred well before the squeal happened). It is also possible to produce electronic feedback. Routing the output of a mixer or effect unit back to its input is a sure way to do this. In fact, many effects are based on using this phenomenon creatively, the most obvious one being an echo with multiple repeats. Feedback and "feedback loops" are also used in all kinds of electronic circuits to achieve specific results. Old analog oscillators are based on electronic feedback.
Ferro Fluid
A liquid that is ferro magnetic, meaning it is attracted to magnetic fields. Outside of the influence of a magnetic field, ferro fluid has a consistency similar to oil, but in the presence of a strong field it becomes relatively stiff. The fluid is often used in loudspeakers (especially tweeters) to conduct heat from the voice coil to the magnet assembly. It is placed in the magnetic gap along with the voice coil of the speaker. The magnetic field keeps it in place, where it serves as a much better conductor of heat away from the coil than air would. The result is that much more power can be applied to the voice coil without burning it out.
Field Coil
A wire coil that, when charged with electrical current, produces a magnetic field. Field coils were utilized in early 20th-century loudspeakers, prior to their replacement by permanent magnets. Remarkably, many audiophiles consider field coil loudspeakers to be sonically superior to models using Alnico or other magnets.
Speakers are the primary source of distortion in the playback chain. Field coil-based drivers, designed properly, drastically reduce these distortion levels. With this technology, the driver is controlled much more accurately. Drivers vibrate at hundreds and even thousands of times per second. Permanent magnets actually lose strength slightly with each vibration. This causes a loss of low-level information and a blurring of the signal. The more complex the music becomes, the more of a problem this loss of control becomes. The permanent magnet essentially modulates the signal.
Field coil drivers, with their own power supplies, do not exhibit these irregularities in strength and so have much less distortion than their permanent magnet counterparts. However, they are significantly more expensive to build into loudspeakers, and that economic fact spelled their doom. A few contemporary manufacturers now build field coil speakers for audiophile sound systems.
Field Coil
A wire coil that, when charged with electrical current, produces a magnetic field. Field coils were utilized in early 20th-century loudspeakers, prior to their replacement by permanent magnets. Remarkably, many audiophiles consider field coil loudspeakers to be sonically superior to models using Alnico or other magnets.
Speakers are the primary source of distortion in the playback chain. Field coil-based drivers, designed properly, drastically reduce these distortion levels. With this technology, the driver is controlled much more accurately. Drivers vibrate at hundreds and even thousands of times per second. Permanent magnets actually lose strength slightly with each vibration. This causes a loss of low-level information and a blurring of the signal. The more complex the music becomes, the more of a problem this loss of control becomes. The permanent magnet essentially modulates the signal.
Field coil drivers, with their own power supplies, do not exhibit these irregularities in strength and so have much less distortion than their permanent magnet counterparts. However, they are significantly more expensive to build into loudspeakers, and that economic fact spelled their doom. A few contemporary manufacturers now build field coil speakers for audiophile sound systems.
Fly
Suspending a speaker in air by means of a cable or rigging system. Some loudspeaker enclosures have "fly points" built into them and are structurally reinforced for flying. Other loudspeakers must be mounted to sophisticated trussing systems that support them from the bottom as if they are sitting on a hard surface.
Foldback
The original term for monitors, or monitor loudspeakers, used by stage musicians to hear themselves and/or the rest of the band. The term "monitors" has replaced "foldback" in common practice.
Free Field
A speaker or sound source is operating in a free field (or space) 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.
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.
Full Range
In reference to loudspeakers, full range means that a device is capable of producing the entire range of human hearing, which is generally known to be from about 20 Hertz to 20 kHz. This term isn't very rigorously used, however, as there is no implied or commonly agreed upon understanding for how evenly the range of frequencies is represented. For example, a device could put out 30 dB more SPL at 500 Hz than at 20 Hz and still be considered "full range." Instead the term is normally used to denote devices that can be, or are being, used in applications where they produce the full range of sound to the best of whatever their abilities are, even if they can't truly reproduce the full range by any reasonable standard. This term is also applied to speaker enclosures in a similar way, even though they may be made up of several different types of drivers that each take care of a specific portion of the frequency range.
Gap
In dynamic transducers such as most loudspeakers and dynamic microphones, the gap is a narrow circular trough in a magnet assembly in which the voice coil resides. The voice coil is attached to the cone of the speaker or mic diaphragm. In the case of a loudspeaker the voice coil becomes energized with electricity from an amplifier, which creates a magnetic field of varying polarity, which causes it to move in and out of the gap, thereby moving the speaker. In the case of a dynamic microphone the action is the opposite: acoustic energy moves the diaphragm, which causes the voice coil to move in and out of the magnetic gap, which generates an electrical signal that can be amplified.
Grid
An electrode component of many vacuum tubes (not present in diodes). The grid acts as a sort of control gate in tubes. An input signal is applied to the grid and as the voltage of the grid is varied by that signal it will attract more or less of the electrons emitted from the cathode, which enables them to pass through to the plate. You can think of it like a water faucet where the input signal is tied to how 'open' the faucet is to the flow of water. This is basic amplifier theory: apply a large voltage from a power supply and use a signal to regulate how much of it gets passed on to the next device (like a speaker). Triode tubes, which get their name from having three electrodes, have one grid that operates as described above. Tetrodes have two grids - one that performs as the grid in a triode does (called a control grid or grid no. 1), and another (called a screen grid or grid no. 2) that is used to reduce the capacitance between the control grid and the plate. Too much capacitance of this sort can cause coupling between the input and output circuits in the tube and make an amplifier unstable - adding the screen grid with a positive voltage applied to it creates an electrostatic shield between the control grid and the plate. Pentodes add yet another electrode called the suppressor grid or grid no. 3. The suppressor grid prevents electrons that may have been dislodged from the plate (called secondary emission), due to the bombardment of the plate by other electrons, from returning to the similarly positively charged screen grid. The electrons are diverted back to the plate, increasing the overall efficiency of the tube.
Half Space
When a speaker or other sound source is placed in a free field, the sound it produces is able to radiate in all directions (depending, of course, on the design of the speaker enclosure). When a sound source is placed against a solid barrier, such as a wall, that same amount of energy is radiated into the space on one side of the barrier only, or into "half space." This has the effect of doubling the amount of sound energy into that half space environment, yielding a 3 dB increase in sound power level. The phenomenon can be particularly noticeable at lower Frequency. Place a stereo speaker up against a wall and you will usually find it puts more bass energy into the listening space. The highs aren't effected as much because they are already pretty directional, and since the tweeter is mounted to the front surface of the cabinet it is already operating in a half space environment. The low frequencies, on the other hand, may be able to pass right through the thin cabinet behind the speaker, but when they encounter the wall (even a standard household wall) more of the energy is reflected back into the room. Many speakers are pre-tuned at the factory to account for this phenomenon.
Hangover
In the domain of sound reproduction (as opposed to inebriation), hangover is the tendency of a loudspeaker cone to continue moving after reproducing a sound, or especially, a transient (see WFTD archive 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 (see WFTD archive damping factor) of the amplifier.
Hangover
When a signal into a loudspeaker suddenly stops, as happens with musical transients, sometimes the speaker cone will continue to move and produce sound due to inertia. This is called hangover. It effects both low frequency and high frequency devices, and is reduced by adding damping to the system. One way of doing this is to increase the damping factor of the amplifier so it has better control of the speaker. Hangover causes poorly damped woofers to sound "boomy" and poorly damped tweeters to sound "hissy."
Helmholz Resonator
A device comprised of a volume of air and an opening to the "outside." The internal volume of a speaker cabinet and its port is an example of a Helmholz Resonator. A bottle is another example. Blowing air across the opening will produce a tone because of the air resonating, and the pitch of the tone will be related to the resonant frequency of the volume. In a vented (ported) speaker enclosure the back wave of air from the driver is used to reinforce the front wave at the resonant frequency. This phenomenon is commonly employed to extend the low frequency range of the speaker/enclosure system.Helmholz Resonators are also employed in acoustics. Enclosing a volume of air (in a box, for example) while allowing limited access to the outside through a series of holes or slits in the surface can create a resonant system that will absorb (or, more accurately, cancel) standing waves and problem frequencies that may be too prominent in a room. If you have one or two frequencies that are too strong in your room a Helmholz Resonator is a very effective way of correcting it.
Holophonics
An acoustical recording and broadcast technology claimed to be the aural equivalent to holography, hence the name. Holophonics is an encode process that occurs during the recording session using a special listening device named "Ringo." It is claimed that "playback or broadcast is possible over headphones or any existing mono or stereo speaker system, with various levels of spatial effect."
Horn
A number of definitions come to mind. In music there is the horn as a brass or wind instrument. The name comes from the early days when they were actually made from animal horns. In sound reproduction a horn is a device (again often shaped somewhat like an animal horn) for focusing and projecting the sound emanating from some audio transducer. Usually a horn is attached to a compression driver, and is used to couple the output of the driver efficiently to a larger area of coverage. But a horn may be used on a loudspeaker as well. And, in fact, some speaker cabinets are known as "horn loaded" because they set the speaker back in a cavity that is used to help control the dispersion and project the audio a great distance (long throw). Horns come in all shapes and sizes depending upon the exact duty they are to perform. Fundamental design philosophies have come and gone over the years. Radial horns have all but been replaced by constant directivity horns in many PA applications due to their greater control over dispersion across wide ranging frequencies. The science of horn design is ongoing and we'll probably continue to see improvements.
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.
Intensity stereo
A term that refers to a stereo sound image that is produced only by the difference in volume of something in the loudspeakers, as opposed to time arrival differences (see Haas Effect).
Intermodulation Distortion (IMD)
The interaction of two or more frequencies in a signal that results in the generation of new frequency components not present in the original signal. These new components have frequencies equal to the sum and difference of the frequencies of the original signals, and integral multiples thereof. IMD is often a major issue in loudspeaker design due to the varying permutations of issues that arise as a speaker cone moves back and forth.
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, we've seen both 6 dB and 10 dB 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...
ISA
Abbreviation for Industry Standard Architecture. A PC computer expansion bus used for modems, video displays, speakers, and other peripherals. PCs with ISA architecture may have some 8-bit and some16-bit expansion slots, but the bus itself is capable of 16-bit data.
ITU 775 Surround
ITU 775 stands for International Telecommunications Union, Operational Bulletin No. 775 in which recommendations are given for a multi-channel surround standard for "5.1" speaker positions.
The ITU-775 setup is sometimes referred to as "3/2 format," indicating a division between a 3-speaker frontal sound stage and a 2-speaker rear "surround". To arrive at this standard, 20 speakers were placed in an anechoic room to find the critical angles for the best speaker placement. For the surround (rear) speakers, a compromise had to be found between 90ş and 135ş. Whereas 90ş (directly on either side of the sweet spot) was the best placing for ambience or "envelopment", 135ş turned out to be best for "surround-placement" or localization in the rear, hence a compromise at 110ş.
In the reference loudspeaker arrangement for mode 3/2, it is recommended that the loudspeakers be placed on the arc of a circle. In those cases where the front speakers must be placed on a straight line, for example, when the center speaker cannot be placed behind the screen, it was recommended that the sound signals be appropriately delayed so that all signals reach the listener's ears simultaneously. It was further recommended that all the front loudspeakers be driven by discrete audio signals. In those cases where the center loudspeaker cannot be placed behind the screen, it must be placed above or below the screen.
Leslie
A (generally) bi-amplified speaker system for use with an electronic organ where there is a rotating baffle in front of a woofer and where tweeter horns also rotate. The Leslie was introduced as an adjunct to the old Hammond tone wheel organ. The rotating parts provide an amplitude and frequency modulation to the sound in a manner that cannot be duplicated with stationary loudspeakers. The effect is a strong vibrato, where the frequency modulation is caused by doppler shift, combined with tremolo caused by the woofer baffle and dispersion pattern of the horns, which are not always facing the listener. Add to that the sonic characteristics of the old tube amps that were used and the somewhat inefficient drivers and the Leslie stands out as having a very distinct sonic signature, especially when combined with a good organ (which may have many sonic idiosyncrasies of its own).
Limiter
A limiter is a dynamics processor very similar to a compressor (see inSync WFTD 10/13). 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!
Line Array
(Simple Definition) - A group of speakers arrayed in a straight line, spaced close together and running with equal amplitude and in phase.How it works - Multiple speakers are carefully spaced apart and stacked on top of each other and fed the same signal. Since the sound source is increased, an increase in acoustic output is obtained on axis of the array, while at some points off axis of the array it creates a cancellation at varying wavelengths (frequencies) which makes the SPL lower. At some points the cancellation may be nearly complete. This phenomenon is known as combing (see Comb Filter), which leads to another phenomenon of loudspeaker arrays called lobing. Combing is a destructive interference that is usually considered a very bad thing in most traditional sound systems. Line arrays, however, use carefully designed and placed speakers to control the combing and lobing thereby creating a concentrated sound on axis, and moving the combing to the side of the cabinet or speaker array. The result is an ability to control where the sound goes and where it does not, which can be very beneficial in auditoriums and many other applications. For example, a PA can be set up so that sound is focused more on the audience and away from hard surfaces such as concrete walls that will produce excess reverberation. An added benefit is that more acoustic energy gets directed toward the desired spots, which means it takes less overall power to achieve a given SPL. In some of the more advanced systems these directional characteristics can even be controlled by remotely adjusting the relative levels of individual speakers within the array.
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.
Lobe
In acoustics and wireless communications, a lobe pertains to a pattern of transmission (in wireless systems and speakers) or pickup (microphones) that is not spherical, or omnidirectional. Essentially the lobe is the portion of a directional pattern bounded by one or two cones of nulls where there is little or no pickup or transmission. For example, a microphone with a figure 8 pickup pattern has two lobes in its pattern, one on each side of the mic. A hypercardioid mic also has two lobes, it's just that the front (desired) one is much more pronounced than the rear. A cardioid mic generally has one big lobe. As soon as you concentrate the energy of any transmission in a particular direction you create one or more lobes by definition. Wireless systems that use directional antennas also have this type of lobing, and so do loudspeaker systems. The characteristics of most lobes will vary by the wavelength of the sound or electromagnetic energy being radiated.
Long Throw
A down field pass (usually of more than 20 yards) in football. In music performance, anything thrown from the audience that makes it on to the stage at a concert (i.e. Beer bottles, shoes, or women's underwear). In music equipment, long throw has references for loudspeakers and loudspeaker enclosures.In a loudspeaker long throw refers to the ability of a speaker cone to travel long distances in and out without encountering nonlinearities in its response. In speaker enclosures (high frequency horns especially) long throw refers to a shape that "focuses" the sound energy in a tighter pattern so that it will travel farther in a coherent fashion. Some horns are designed to spread the energy into a wide pattern for coverage while others are designed to be "long throw." Usually a long throw horn is recognizable by the long "throat" between the driver and the horn opening. Long throw speaker cabinets usually have the speakers recessed into some kind of horn like shape as well.
Loudspeaker
A transducer that converts electrical energy into sound energy, providing the audible sound in equipment such as public address systems, studio monitors, guitar or bass amplifiers, radios, televisions, and home stereos.
A standard dynamic loudspeaker consists of a voice coil, a magnet, a diaphragm and a cone. The electrical energy output of a power amplifier is transmitted as voltage over a wire to the voice coil. The current flowing through the voice coil produces an electromagnetic field that reacts with the stationary magnet in the speaker assembly. The voice coil is attached to a diaphragm, which in turn is attached to the cone. The magnetic fluctuations cause the diaphragm and thus the cone to move, moving air and radiating sound.
There are other types of loudspeaker technology, the best known being electrostatic speakers. These differ from dynamic loudspeakers in that they consist of a thin sheet of electrically conductive film suspended between two wire screens. A high-voltage charge is applied to the film and it is alternately attracted to one screen and then the other. This creates motion, which again radiates sound. Another type of loudspeaker are servo drive loudspeakers. These employ servo driven motors attached to the speaker cone in place of the magnet/wire assembly. This type of speaker is generally only used in subwoofer applications, and even then only rarely.
MIDI Delay
This is one of those terms that has been bantered around in the industry over the years and has come to have several subtly different meanings. The original meaning of MIDI delay refers to the time it takes for any active MIDI circuit to handle the signal. Just passing MIDI into, and then directly out of any device (even without doing anything to it) takes some finite amount of time because of the electronics involved in managing and buffering the signal. This is MIDI delay and in most cases it is usually well under 5 ms. The delay is cumulative though. So if you pass your signal through several devices it may be significantly delayed by the time it gets to the last device. Some people also refer to the time it takes an instrument to respond to MIDI commands as MIDI delay. While true MIDI delay is one component of this, there are other factors, such as the speed of the processor in the device. Some instruments react more slowly as they are asked to do more (for example, play more notes at once), but this is technically not MIDI delay. Some musicians claim to be able to hear/feel MIDI delay and do not like performing in situations where MIDI is used. While it's pointless to dispute what a person says they can perceive, it is important to note that given the speed of sound in air the sound leaving a speaker cabinet on the one side of a 20 foot wide stage would take about 20 ms to reach the ear of a player on the other side.
Mix-Minus
A specialized matrix-mixer where there is one output associated with each input that includes all other inputs except the one it is associated with. (The output is the complete mix, minus the one input.) In this manner, the simplest mix-minus designs have an equal number of inputs and outputs (a square matrix). For example, if there were 8-inputs, there would be 8-outputs. Each output would consists of a mix of the seven other inputs, but not its own. Therefore Output 1, for instance, would consist of a mix of Inputs 2-8, while Output 2 would consist of a mix of Inputs 1 & 3-7, Output 3 would consist of a mix of Inputs 1,2 & 4-7, and so on. Primary useage is large conference rooms, where it is desireable to have the loudspeaker closest to each microphone exclude that particular microphone, so as to reduce the chance of feedback.
Mod Wheel
A mod wheel (diminutive for Modulation Wheel) is a controller found on keyboards such as MIDI controllers and synthesizers, which takes its shape in the form of a wheel mounted perpendicular to the surface of the keyboard. The wheel itself is imbedded in the surface such that only the top half protrudes. The mod wheel is used to add expression or to modulate (change) various elements of a synthesized sound or sample. One typical use is to modulate an LFO in order to produce vibrato. Another would be to control the speed of rotary speaker emulation. There are many other applications as well depending upon the architecture of the instrument being controlled.
In order to create such effects, mod wheels send continuous controller messages (CC), which send the movements of the wheel as well as knobs, sliders, pedals etc. (See WFTD MIDI Control Change) For example, your synth's modulation wheel or lever will almost always send CC1 messages. Each CC has a possible range of 0-127, so when you move the mod wheel down to its rest position, it should send a CC1 with a value of 0, and when you push it up to its highest point it should send a CC1 with a value of 127. CC values are not smooth, they're stepped, that is, a standard mod wheel can send a value of 56 or a value of 57, but it can't send 56.329 or 57.1. Depending on what sound parameter CC1 is controlling, you may hear a slightly grainy, stair-stepped effect (See Zipper Noise) when you move the mod wheel while holding a note.
Monitor
This term has several meanings as applied to audio and video technology.
As a verb, to "monitor' means to listen to a sound source such as a recorded track or a mix.
In a recording environment, monitors are the loudspeakers used to play back the live signals and recorded tracks of a project. Monitor also refers to a special mix (monitor mix) that is provided to the talent, usually through headphones, to give them a reference to the music they are performing. This is sometimes called a cue mix.
In sound reinforcement, monitors refer to the system of loudspeakers and/or in-ear systems that transmit an often-custom mix of the audio program back to the performers.
In computer usage, a monitor is the CRT or flat-panel LCD display screen that provides visual images of your programs and activities.
Mono Bridge
A method of configuring a two channel amplifier so that the two channels can be "ganged" or bridged to be used together on one load. The purpose of this is to take a two-channel amp and create a larger single channel amp that can deliver more power. It works by reversing the polarity of the signal going to one of the amp channels. The same signal is presented to both sides, but with opposite polarity. Then the load, which is presumably a speaker or set of speakers, is connected across the positive lead of both channels. The ground or negative leads are not used. So while the signal drives one of the hot leads in one direction (positive or negative) it will drive the other hot lead in the opposite direction in an otherwise identical way. This creates a difference of voltage between the two channels that is twice as great as either channel by itself referenced to ground, or the negative terminal. The result is more power to the speaker than would be possible from either channel alone. Most modern amps have a special switch to enable mono bridge operation. It basically takes care of the polarity reversal of the signal going to one side of the amp.
Moving Coil
A specific type of dynamic (as opposed to condenser) microphone design. Moving coil microphones are among the most commonly used in music and sound production. The ubiquitous SM-58 and SM-57 mics are examples of moving coil design. These mics work on very simple principles. In fact they work just like a speaker in reverse. The diaphragm has a coil of wire attached to its base. This coil is inserted into a magnetic gap. When changes in air pressure cause the diaphragm to vibrate in and out of the magnetic gap it generates an alternating current in the wire that represents the signal.Moving Coil is also one method used in making phonograph cartridges. Moving coil designs were all but replaced by moving magnet designs (same principle, but the magnet moves instead) in the 1970's. Moving coil phonograph cartridges have very low output (requiring a different preamp) and are very expensive compared to their moving magnet counterparts, but there are some sonic advantages to them including lower distortion and better frequency response.
Neodymium
Pronounced NE - O - Dim - E - Um, and holding atomic number 60 on the periodic table of elements (Symbol = Nd), neodymium is a silvery rare-earth metal element most commonly used for coloring glass. However it is also sometimes used to make magnets. Neodymium magnets are often stronger than magnets made of other materials, and as such come in handy for the audio industry because they enable manufacturers to produce microphones, and/or speaker drivers that are more powerful for a given size. Neodymium based microphones, for example, may have 6 dB (or more) greater output than their non neodymium counterparts.
Noise
Often defined as any unwanted sound that is not related to the wanted sound (if it is related we call it distortion). In electronics it can be further defined as a wide band addition to a signal by any electronic or mechanical component. "Random noise" is the most common type. It is unpredictable and contains a continuous distribution of energy over all frequencies - or at least all frequencies relevant to the system at hand. There are other types of noise, most of which are artificially created and have specific uses.Noise is present in everything. Virtually any device carrying a signal adds noise. This includes, wire, speakers, amplifiers, and so on. Some devices add much more than others. Consequently an important measure of the quality of a signal is its signal to noise ratio, which describes how powerful the signal is in relation to the noise accompanying it.
Omnidirectional
Literally, from all directions. In audio, microphones are said to be omnidirectional if they can detect sound equally from all directions. Speakers are omnidirectional if they radiate sound in all directions equally; this tends to be the case with subwoofers and low frequency drivers. Low frequencies, in general, tend to be omnidirectional, versus high frequencies which tend to "beam" or be very directional.
On-Axis
In our business this generally refers to an audio source that is directly in front of a listener or a transducer such as a microphone. This is at the 0 degree axis in a polar pattern. A microphone will generally produce the "truest" results if the desired source is on-axis (oriented directly in front of the sound source), although some creative engineers have been known to get desirable sounds by using a microphone's off-axis response. For loudspeakers the meaning is similar - when the listener is directly on axis with a speaker he/she will be exactly in front of it. How a speaker's characteristics change as the listener moves more off axis is an important part of the overall response.
Out of Phase
A phrase used to characterize two or more signals whose phase relationship with each other is such that when one is at its positive peak the other is at (or near) its negative peak. This is also commonly referred to as being 180 degrees out of phase.
Phase is a relative value that is measured in degrees (like angles). 90 degrees out of phase is more out of phase than 80 degrees, but less than 100 degrees. 180 degrees out of phase is completely backwards, which is characterized by one signal's highest peak correlating with another's most negative peak. Most signals are not entirely in phase with each other, and it's just as rare for them to be perfectly (180 degrees) out of phase. But people generally say "out of phase" to mean approximately 180 degrees out of phase. People also frequently say "out of phase" when the more technically correct term to use would be "polarity reversed." Phase implies a time delay, where one signal lags behind another one to some degree. Polarity refers to one signal being "backwards" from another. An example of this would be the "phase" switch on many mic preamps and mixing boards. Generally all this switch does is reverse pins two and three on the XLR connector entering the preamp, thereby reversing the "polarity" of the signal. There is no time delay of the signal. Nevertheless this is often referred to as "out of phase." A similar thing happens when you reverse the polarity of the speaker leads to one speaker in a two-speaker setup. That speaker is now operating with the opposite polarity of the other. No time delay was introduced, yet we often refer to this as "out of phase." This confusion occurs because when viewed on a display like an oscilloscope waveforms that are 180 degrees out of phase with each other will not look any different than two that are polarity reversed. Sonically the difference is generally pretty minute as well. So for all practical purposes the two terms can be used interchangeably.
While it is technically true that any two signals not 100% in phase with each other could be referred to by the somewhat generic phrase, "out of phase," we generally don't use that terminology until the signals approach a 180 degree phase relationship with each other.
Pan (Panning)
Comes from the term panoramic, which pertains to large visual scenes that can completely surround a subject. In film work panoramic shots require a camera to be "panned" across the landscape (or whatever the subject is). This terminology was adopted when two-channel (stereo) audio first arrived on the scene. In audio a pan control is used to position an audio track somewhere between the left and right loudspeaker in the stereo soundfield. A pan control generally works by simply reducing the level of a track in one channel, which makes it appear louder in the opposite channel. Modern designs are more sophisticated in their approach, but the basic concept has stayed the same: turn the pan pot to the left and that track comes out of the left speaker.
Passive Radiator
In speaker design a passive radiator is an element that is designed to move sympathetically with the energy in the cabinet. They generally resemble a low frequency driver or woofer, but have no voice coil or any element to actively generate sound. Often they are employed in speakers instead of a port to create a bass reflex type of design. The extra mass of the passive radiator actually lowers the resonant frequency thereby allowing smaller cabinets to reproduce lower frequencies than they would otherwise be capable of. However this extra mass also causes the radiator's movement to be hard to dampen, which in turn can lead to bass hangover and a more "boomy" sound.
Phantom Channel
A special mode in many surround sound systems that reproduces the effect of a center channel through a left and right stereo speaker setup. The mode is designed for users who wish to experience surround listening, such as with Dolby Surround, but who do not yet have a center speaker to reproduce the discrete center channel information. Basically the audio that would normally be sent to the center channel is added to the audio in the left and right speaker channels. This produces a mono image centered between the two speakers, almost as if a real center channel speaker were there.
Phantom Image
In a multichannel audio playback system a phantom image can sometimes be created between any two (or more) of the loudspeakers, creating the illusion of an additional speaker or (more importantly) adding to the overall realism of the soundstage. For example, in a simple left/right speaker setup it is possible to create a convincing phantom center image if the system is well designed and the audio produced well.
Piezo
Short for piezoelectricity or piezoelectric effect. Piezoelectricity is an electric charge that occurs in some substances when they are squeezed or otherwise subjected to mechanical stress. It is also possible to cause these materials to vibrate when a voltage is applied to them. Quartz is one of the better known piezoelectric materials, and is commonly fabricated into small pieces, called "crystals" that are used for frequency standards. A crystal of specific size and shape will vibrate at a predictable and very stable rate when a voltage is applied. This makes them ideal for use in things like watches or clocks for digital audio equipment. Piezoelectric elements have also been used various types of transducers such as phonograph cartridges, microphones and loudspeakers. Piezo microphones can be quite small and still have relatively high output at a low cost; however, their less than ideal frequency response prohibits use in critical applications. Piezo loudspeakers usually come in the form of tweeters, or very high frequency elements. They generally have very low distortion in the 5 kHz and above range, but haven't widely been used in sound reinforcement due in part to their relatively low output levels. It takes dozens of the average piezo tweeter to equal the output of one medium-sized compression driver.
Planar Loudspeaker
A type of dipole loudspeaker design that combines aspects of both dynamic and electrostatic designs. The planar speaker consists of a large plastic sheet with conducting wires imbedded in it. These wires function as the voice coil. Many small magnets in front of and behind the sheet set up a magnetic field so current in the wires causes a force that moves the unit, similar to an electrostatic speaker. Planar speakers suffer from the same directional problems as other dipole loudspeakers, but their impedance is more similar to dynamic designs.
PMPO
Today's word was a special request from an inSync reader who has been seeing it used on line a lot lately. PMPO: Abbreviation for Peak Music Power Output. A very subjective specification designed to help provide "real world" wattage ratings for power amps and speakers. Lots of number games are played with specifications and very few are held to specific and rigid testing standards. Consequently their value in comparing equipment is often less than ideal. Further, many specifications are obtained in lab conditions that don't translate easily to how the device is likely to perform in the real world, which adds more confusion. For example, power amps are often tested with sine waves, pink noise, and other non-musical signals. These signals can be very demanding (in different ways) for the equipment to reproduce, whereas a "typical" music signal is much easier. The purpose of PMPO is to show a product's performance in real world circumstances that are allegedly more meaningful to the user. An amp that may not compare very well in lab tests to a much more expensive model will often show up as much closer in terms of PMPO. Proponents say this puts the specs on paper much more in line with the actual audible difference in the products. Unfortunately the PMPO measurement is far more subjective than most specs because, in addition to the ambiguities many specs suffer from, there is no standard for what "music" means. Ten different manufacturers can still test their product under ten different sets of circumstances. PMPO is mostly used in the consumer hi-fi industry where deep understanding of specifications is often not a concern.
Polarity
In electronics, two points that have opposite electric potentials (one is positive, the other negative). This is not the same as being 180 degrees out of phase (although the results can be similar). Phase implies a relationship with time, polarity does not. What most engineers, consoles and preamps refer to as a "phase" switch is actually a switch reversing signal polarity.Polarity is important when interfacing equipment, particularly speakers - you don't want one cone moving in while the other moves out. Some designers feel that maintaining "absolute polarity" (no polarity reversal in a signal chain) throughout a signal path is important.While tests don't indicate that the ear can hear which polarity is correct, they do show that it may be possible to detect a difference between normal and inverted polarity signals. (Try it for yourself in a critical listening environment: Play a signal though a single speaker, then reverse the speaker wires and play the same signal again - remember to switch the wires back when you are finished!)
Pole Piece
A shaped piece of high permeability metal, usually soft iron, which serves to concentrate and direct the magnetic field of a permanent magnet to maximize efficiency of devices like loudspeakers, magnetic cartridges, and cutterheads. Pole pieces are needed because magnets are hard (expensive) to make in the complex shapes that can be needed. In layman's terms, the Pole Piece is the part of the speaker magnet assembly that the voice coil (see WFTD archive voice coil) slips over. It is the center round piece. Guitar and bass pickups work on similar concepts, though the function is the opposite of a loud speaker. Instead a guitar pickup's job is to turn mechanical vibration into electrical output. Magnetic guitar pickups often have individual pole pieces positioned under each string to help maximize or otherwise tailor their output.
Port
In audio acoustics a port refers to an opening in a bass reflex-type loudspeaker enclosure. Ports are usually tuned very carefully to create certain kinds of resonances and coupling with the air outside the cabinet. They come in many shapes, sizes, and are even found in different locations on speaker cabinets. The purpose is to improve the bass response characteristics of the enclosure, which is often accomplished specifically by creating a controlled resonance of the air at a frequency just below the normal cutoff frequency of the speaker in the given enclosure. Port design principles and how they integrate with overall cabinet (and speaker) designs are a very rigorous science where refinements are still being made.Port also refers to an input or output of some device, usually an electronic device. A SCSI connection is considered a port (often called a SCSI port), serial connections are ports, and so on.Finally port also refers to the concept of taking software written for one type of platform and converting to work with another platform. This is known as 'porting.' For example, when a software company decides they want their Mac based program to also be available for Windows users they can 'port' it over, which is usually much easier than writing entirely new code from the ground up, though there are also advantages to all new code.
Potential Acoustic Gain
A measure of the amount of gain before feedback that can be obtained with a sound reinforcement system that's based on the number of open microphones and distances from source(s) to microphones and listener(s), as well as speaker distances from listener(s) and microphones. These parameters are basically plugged into an equation that involves the application of the inverse square law. A typical equation might look like:PAG = 20 log (D1) - 20 log (D2) + 20 log (D3) - 20 log (D4) - 10 log (NOM)where,PAG = Potential Acoustic GainD1 = Distance between microphone and loudspeakerD2 = Distance between the loudspeaker and the furthest listenerD3 = Distance between the source and the furthest listenerD4 = Distance between the source and the microphoneNOM = Number of open microphonesThere are a number of subtleties to the application of this formula (what you see here is somewhat simplified) that are beyond the scope of this writing, but when applied correctly it can yield a pretty accurate estimation of the performance of a system.
Power Compression
Speaker voice coils are made of copper or aluminum. As these voice coils increase in temperature during normal operation, their resistance increases. Greater voice coil resistance means less power transfer from the amplifier. As a result, the speaker will not play as loud when it's "warmed up" as it did when it was "cold". Some speakers may exhibit 3 to 6 dB of power compression. A mere 3 dB of power compression is equivalent to cutting the available wattage of your power amps in half. Speaker manufacturers who develop systems for use in demanding applications such as concerts or nightclubs spend a great deal of their research and development energy working on ways to keep speaker voice coils cool while in operation.
PrecedenceEffect
Also known as Haas effect. Refers to how we locate sounds based upon time arrival differences between our two ears. Not only does it effect our perception of where the sound is coming from, but it also effects our perception of the volume. The same sound can be presented to each ear at the same volume, but we will hear the one arriving first as louder. This effect can be so drastic that you can be fooled into thinking one of your monitor speakers isn't outputting any sound simply because you are sitting a few milliseconds closer to the other one.
Push-Pull
A type of amplifier design. Push-Pull is a term that originated in the days of tube amplifiers (which are now having a resurgence). In this design two output tubes are connected in such a way that while the current in one is increasing, it is decreasing in the other. The two signals are then combined in an output transformer and passed on to the load (speakers). The modern solid-state version of this is known as the Complementary Symmetry Circuit (though most lay people still call it push-pull), where two transistors operate in a similar fashion. There are several classes of operation of push-pull designs that we will discuss in upcoming weeks.
Q
The resonance of an electronic circuit. "Q" actually refers to quality factor. Q is a measure of the sharpness of a resonant peak. The term Q is often used interchangeably with "bandwidth". This is not entirely correct. It is more accurate to say that Q determines bandwidth (a subtle but distinct difference). Q is most often used in reference to synthesizer filters (sometimes referred to as resonance) and equalizers, but it also applies to capacitors (a measure of efficiency, the ratio of capacitive reactance to resistance at a high frequency) and speakers (a measure of directivity). In speakers, a Q of 1 means the system sends out energy equally in all directions; a speaker with a Q of 2 radiates in a 180 degree hemisphere; higher Q's correspond to smaller angles. In EQ circuits Q is defined as the center frequency divided by the half power bandwidth. On a 1/3 octave graphic equalizer, for example, the half power point at 1 kHz is 232 Hz wide. The Q is thus 1000/232 or 4.31.
Quarter Space
When a speaker or other sound source is placed in a free field the sound it produces is able to radiate in all directions (depending, of course, on the design of the speaker enclosure). When a sound source is placed against a solid barrier, such as a wall, that same amount of energy is radiated into the space on one side of the barrier only, or into "half space." When a speaker is placed at a junction between two walls, such as in the corner of a room, it is said to be in a 1/4 space environment. This will yield an additional 3 dB of sound power level (particularly in bass frequencies) over a speaker in a half space environment and 6 dB over a speaker in the free field. For more information on this phenomenon see WFTD Half Space.
Re-Amp
The process of running an already recorded signal back through an amplifier (and possibly speakers) of some sort. With the increased popularity and flexibility of DAW systems this has become a popular technique for guitar. An engineer may record the guitar signal dry, or even directly out of the guitar itself along with or instead of any amplifiers, preamps, or effects, and then later process the raw track(s) through a guitar amp or some other preamp or processor. This is accomplished by routing the raw or dry guitar sound (or any other track, for that matter) out of an output of the DAW and into the amp - then out of the amp, usually by way of miking the speaker, back into another channel of the DAW. Sometimes this is also done with plug-ins inside the DAW itself. This final produced sound may then be recorded to another track or simply treated as a live instrument for mixing. This enables artists and engineers the maximum amount of flexibility for the sound as the piece progresses. Often times the guitar sound doesn't get finalized until the mixdown. While this technique is most commonly used for guitar it is also done for bass, keyboards, and sometimes even things like vocals or drums for special effects.
Resonant
Resonance is the tendency of a mechanical or electrical system to vibrate or oscillate at a certain frequency when excited by an external source, and to keep oscillating after the source is removed. If something tends to have resonance it is said to be resonant. Resonate is the verb form - to resonate. A bell is a good example of a mechanical resonator. When exited into vibration by being struck a bell will oscillate at its many resonant frequencies and thus produce its unique sound. All mechanical structures have some resonance at some frequencies. Resonance is a particular concern with loudspeaker manufacturers because speakers, speaker enclosures, and the listening areas they are ultimately placed in all have resonances that can cause inaccuracies in sound reproduction. An example of electrical resonance would be the oscillator in a synthesizer that is used to produce sound. A good example of both electrical and mechanical resonance is feedback in a PA system.
Resonant Frequency
The frequency at which resonance occurs. The resonant frequency determines the pitch of things like recorders and other musical instruments that rely on resonant columns of air. It also determines the pitch of feedback, another form of resonance. And it is the pitch or frequency the port of a loudspeaker may be tuned to.
Ring Out
Refers to a process of tuning a PA or monitoring system involving the intentional initiation of feedback to locate sensitive or hot frequencies. Monitor systems are most prone to feedback at frequencies where the speakers and/or open microphones have peaks in their frequency response. One can quickly find these peaks by turning up the volume on the mics in question until feedback begins. This is usually where equalization is applied to counteract troublesome frequencies - i.e. if it feeds back at 4 kHz then pull 4 kHz down on your EQ a few dB. Four or five rounds of this is usually enough to get rid of the major problems. While this technique is commonly used for stage monitoring systems, it can also prove surprisingly effective for the FOH system as well, particularly in situations where there is a heavy emphasis on vocal reproduction.
RMS
Abreviation for Root Mean Square. In electricity (including audio) this refers to the effective amount of power in an AC signal. Since AC power, by definition, is a periodic waveform with part of its period above and below "zero" it's not as easy to know how much actual power is available over time as it is with DC, where the voltage and current are (relatively) steady. The Root Mean Square method is the commonly agreed upon method of computing what some people refer to as the "heating value" of the signal - the effective voltage that would generate the same heat as a DC (Direct Current) signal, over the same time. In a circuit whose impedance consists of a pure resistance, the RMS value of an AC wave is often called the effective value or DC-equivalent value. For example, if an AC source of 100 volts RMS is connected across a resistor, and the resulting current causes 50 watts of heat to be dissipated by the resistor, then 50 watts of heat will also be dissipated if a 100-volt DC source is connected to the resistor.To determine RMS value, three mathematical operations are carried out on the function representing the AC waveform:(1) The square of the waveform function (usually a sine wave) is determined.(2) The function resulting from step (1) is averaged over time.(3) The square root of the function resulting from step (2) is found.Thus the term Root Mean Square (RMS). The RMS value of a complex signal must be read with an RMS meter to be accurate. Alternatively, the signal can be digitally sampled and the samples summed to yield the RMS value. As such, the RMS value of a complex signal can be thought of as the "area under the curve" of a signal as viewed in a wave editor software application.For a sine wave, the RMS value is 0.707 times the peak value, or 0.354 times the peak-to-peak value. Household utility voltages are expressed in RMS terms. A so-called "117-volt" AC circuit carries about 165 volts peak, or 330 volts peak-to-peak.Many speakers and amplifiers are rated in RMS value for similar reasons. We are most concerned with the amount of work to be done, which is why we use the unit of watt, but it's also meaningful to differentiate between the peak and the RMS, because the RMS determines how much real, usable power is there to be converted into air movement.
Sidefill
In stage monitoring applications sidefills are generally larger, full range speaker systems placed off to the side of the stage. The purpose of sidefills is to provide the performers a more enveloping and well balanced sound. This allows all performers to hear a good rough mix of the whole band, while still allowing for individual mixes to be provided for individuals at their position on the stage via wedge monitors. It is widely believed that the more full stage mix inspires better and more coherent performances among musicians. In past few years technology has made it possible to achieve a pretty full sounding mix at each position with relatively small wedge monitors so sidefills have begun to be used less. In very recent years in-ear monitoring systems have begun to render all stage monitors obsolete.
Signal Path
Simply the route a particular signal takes through a chain of equipment and/or electronic components on the way to its destination. When we think of signal paths in audio we are usually thinking about connecting different pieces of equipment together and routing some signal(s) through them. An example of this would be something like a microphone to mixer to speaker or recorder setup. The signal path has the signal from the microphone pass from the microphone through those (and potentially other) devices on the way to being recorded or amplified (or both). But there is also a signal path inside each piece of equipment. A mixer may be configured to route signals in different ways internally bypassing or utilizing different gain stages along the way to achieve different results. Effects processors often have highly configurable internal signal paths depending upon what they are doing.
Single-ended
When used to describe an amplifier, this term means one power tube [frequently a triode] or transistor is solely utilized to amplify the entire audio signal in the amp. Most often used in hi-fi audio, single-ended amplifiers are popular because of their simplicity of design, which follows with the concept that less is more (i.e. the fewer components in the shortest possible audio path equals higher signal fidelity). Single-ended amplifiers tend to be low output - generally less than 10 watts - and need very sensitive speakers in order to translate the signal well and genuinely sound their best.
Slope
In audio filters, slope refers to how quickly frequencies are attenuated by the filter once the cutoff frequency is passed. Slope is given as a dB/octave figure. For example in a high pass filter with a cutoff frequency of 4000 Hz, and with a slope of 6 dB/octave, for each octave (doubling of frequency) above 4000 Hz, the level of frequencies will be diminished by an additional 6 dB. Slope is determined by the "order" of the filter, or the number of poles it contains. A first order, or single pole filter will have a slope of 6 dB/octave. A second order, or two pole filter will have a slope of 12 dB/octave, and so on (slope increases by 6 dB/octave per order or pole).Creating the correct slope is very important in filter design. For example, it determines how accurately an EQ can cut or boost some frequencies without affecting others. Slope is also important in crossovers, where it is undesirable for frequencies beyond the cutoff frequency to be passed on to amplifiers and drivers (typical crossover filter slopes are in the 12-24 dB/octave range). Sometimes crossovers feature selectable filter slope so that response can be matched to particular speaker set ups.
Sound Card
An expansion board that enables a computer to manipulate and output sounds. Sound cards have become commonplace on modern personal computers and are typically associated with the consumer market. Sound cards enable the computer to output sound through speakers connected to the board, to record sound input from a microphone connected to the computer, and manipulate sound stored on a disk.
Some sound cards also support MIDI, surround sound and more. In addition, most PC sound cards are Sound Blaster- compatible, which means that they can process commands written for a Sound Blaster card, a standard in consumer PC sound.
Speaker Cabinet
A cabinet is the common term for an enclosure in which a loudspeaker is mounted. The major role of the enclosure is to prevent the negative phase sound waves from the rear of the speaker from combining with the positive phase sound waves at the front of the speaker. The result of this would be cancellation and interference patterns, causing the efficiency and sound of the speaker to be compromised.
The ideal mount for a loudspeaker is a flat board (baffle) of infinite size with infinite space behind it. Thus the rear sound waves cannot cancel the front sound waves. Given a shortage of infinite size boards, cabinets (enclosures) must use other techniques to maximize the proper output of the loudspeaker. This is called loading.
To place the loudspeaker in a large sealed box, filled mostly with foam or wadding, is commonly referred to as an infinite baffle, as it approximates the ideal mount. Following on from this is a smaller sealed box, or, an acoustic suspension enclosure. With the correct loudspeaker, this will improve the efficiency and frequency response of the speaker.
Other types of cabinets attempt to improve the low frequency response or overall efficiency of the loudspeaker by using various combinations of reflex ports, transmission lines (material or structure that forms a path from one place to another for directing the transmission of acoustic waves), and horns.
For guitar cabinets, which in the early days were little more than a speaker in a wooden box with some insulation, it was the sympathetic vibrations and resonances of the cabinet that helped to give the characteristic sound that many players still rely upon even in more modern equipment.
Speakon
A type (and brand) of multi-pin connector developed by Neutrik which is now commonly found on speakers and amplifiers intended to be used in high power mobile applications. They have become popular because they offer a very high quality reliable connection, can handle extremely high power, are very durable, and are relatively low cost compared to other similar connectors. Standard Speakon connectors come in four or eight conductor versions (though other configurations are available). The Speakon 8 has the same footprint as the EP8 connector and the Speakon 4 has the same footprint as XLR "D" type connectors.
Spider
- Any of numerous arachnids of the order Araneae, having a body divided into a cephalothorax bearing eight legs, two poison fangs, and two feelers and an unsegmented abdomen bearing several spinnerets that produce the silk used to make nests, cocoons, or webs for trapping insects.
- In audio, the assembly which holds the voice coil of a dynamic loudspeaker centered in the magnetic gap. The spider is usually a corrugated circular piece of specially coated fabric. The name comes from the early days of loudspeakers when it was made of a plastic material that resembled the legs of an arachnid.
Splay
To spread apart with diverging sides. In PA applications large loudspeaker configurations are often set up with the boxes splayed so that each cabinet points to a different part of the arena. The purpose of splaying is to provide more coverage than the inherent capabilities of the cabinets in question. Plus it helps avoid having too many drivers covering the same area, which tends to cause significant problems with phase cancellation and lobing. Manufacturers often build speakers with a trapezoidal shape to make splaying easier without taking up any more space than necessary.
Stack
In addition to methods and types of amplification, guitar amps also come in different physical configurations. Along with the Combo style, guitar amps also come in separate Head, which refers to the amplifier section and Speaker Cabinets. These allow you to use any amp head with virtually any speaker cabinet. They also break the amp into two units, making each unit lighter and easier to carry than a single combo. Combining two cabinets and a head is called a "stack."
STC
Abbreviation for Sound Transmission Class. This is a number rating that can be used to compare, in a generalized way, the acoustical isolation of different barrier materials or partition constructions. Higher numbers indicate a material will provide more acoustic isolation when used as a barrier.
The tests conducted to determine STC involves two test rooms: a ''source'' room and a ''receiver'' room. The source room will contain a full-range test loudspeaker. The receiver room will contain a microphone, which is connected to sound-measuring devices. There is a nominal opening between the two rooms - usually about 9' wide by 8' high, but can vary in accordance with the standard.
The first step is to measure the sound transmitted from one room into the other through the opening. The sound is measured in decibels (dB) in 1/3-octave bands from 125 Hz to 4000 Hz. Then the opening is plugged with the material or partition construction. This could be a single layer of barrier, such as plywood or drywall, or a complete wall with as many materials, layers, air gaps, etc. that can fit in the opening. The edges are completely sealed and sound transmission between the rooms is measured again. The sound level from the ''after'' test is subtracted from the sound level ''before'' plugging the opening. The resulting difference is known as the transmission loss or ''TL.''
Next, the TL is plotted on a graph of 1/3-octave band center frequency versus level (in dB). To get the STC, the measured curve is compared to a reference STC curve. Two criteria are used to ''match'' the curves:
1. The reference curve shall not exceed the measured TL by more than 8 dB in any 1/3 octave band, and
2. The sum of all the ''negative discrepancies'' shall not exceed 32.
(This actually sounds more complicated than it is. A simple spreadsheet can be used to calculate the STC for any range of TL values.)
Once the two above criteria are met, the value of the reference curve at 500 Hz is read as the entire STC of the material or partition type.
Stereo
Short for stereophonic. Common usage of the term stereo would have it be defined as any sound reproduction (or reproduction system) with two speakers, typically a left channel and a right channel. Stereophonic, however, refers to any system that provides the listener with an illusion of directional realism, regardless of how many channels are used. The convention of using two channels to make stereo came about in the mid 1900's mostly due to cost considerations and the relative ease of recording two channels in the groove of a phonograph record.
Sub
Short for Subwoofer, though occasionally used as an abbreviation for subgroup. A subwoofer is simply a speaker (woofer) designed to handle the very low frequencies of a speaker system. The concept has been around for many years, but only in the last 10 or 15 years has their use become widespread. With increased popularity of smaller main speakers and much more low frequency content and dynamic range in our recordings, low frequency drivers have become an important part of any speaker system. Many systems may have only one subwoofer (as opposed to the two you would expect). In fact, most of the home theater surround sound technologies (AC-3, etc.) in use today have only one mono sub output. This is based in part on the theory that very low frequencies tend to be omnidirectional so one speaker can cover an entire room. Plus it's often difficult to produce stereo separation between subwoofers, and in fact any two drivers producing the same frequency range in the same area can tend to interfere with one another as the time arrival of the sounds at the listening position causes certain frequencies to be more or less out of phase with one another, which causes uneven frequency response and even dead spots.
Surround
The flexible material that connects the out edge of a loudspeaker cone with the speaker’s frame or superstructure (normally referred to as a “basket”). Surrounds are usually made out of a flexible material such as corrugated cloth, paper, or some type of foam rubber compound that enables the speaker to move in and out efficiently while still being sturdy enough to help keep the voice coil centered in the magnetic gap. Surrounds come in a variety of styles and appearances for different applications. For example, “Roll” surrounds are characterized by one, larger rounded corrugation and are generally preferred in more delicate applications due to their increased flexibility and linear behavior when flexed.
Surround Sound
Surround sound is a multi-channel audio playback format consisting of at least three speakers (left, center and right) but more commonly consisting of five or more. The first Motion Picture shown publicly with multi-channel sound was Disney’s Fantasia in 1941 but the first commercially successful multi-channel formats would not come about until the early 1950s with the four-track CinemaScope and Todd-AO’s six-track format. While the cinema has had the largest influence on surround sound technology, it is in the home of consumers where we see the largest growth of commercial success. Today, surround sound playback systems typically consist of 5 speakers and a subwoofer which makes up a “5.1” system, similar to that found in movie theaters. A 5.1 system utilizes forward left, right and center channels, two surround channels that are commonly placed behind or to the left and right of the listener(s) and a subwoofer (the “.1”). Dolby Digital surround encoding could be found on Laser Disks as early as 1995, soon to be followed by the DVD revolution most recently. Today, DVD-Video, DVD-Audio and several other surround sound formats (including cable and VHS) have found their way into the homes of consumers.
TEF
Abbreviation acousticians use for Time-Energy-Frequency. In their work acousticians are concerned with the propagation of sound through a space. In this work one must consider the behavior of the space at various frequencies and energy levels: put a specific amount of energy at a specific frequency into a space and what happens? How much energy comes back (at all frequencies) over what period of time (direct sound versus reflections and reverberations)? It's more complex than this, but that's the basic idea. In the late 1970's Techron, a division of Crown International, made the first practical and widely accepted device for measuring TEF, called the TEF System 10. TEF machines (or TEF analyzers) provide a means to, among other things, measure energy-time curves, which is basically a fancy way of saying they can measure the acoustic energy (sound) in a space at multiple frequencies over time. These devices have been heavily used since their inception for all kinds of related applications including speaker and speaker cabinet design as well as room design or even the design of automobile interiors.
Test Tone
A tone of a certain waveform sent at a predetermined level and frequency which is used for test purposes, such as for facilitating measurements, or to help determine the optimum placement for speakers, to measure SPLs and for aligning gains and losses in an audio system. A test tone is theoretically a constant pure wave of the desired waveform. In practice, many test tones are not perfectly shaped, and the quality of the test tone produced is generally directly proportional to the quality of the device (usually some type of oscillator) being used to produce it.
Time Aligned
In speaker cabinets using multiple drivers, the sound produced by each driver can arrive at the listener's ear at different times. (i.e. in a two-way system, the sound from the tweeters arrives before or after the sound from the woofers). This results in degraded transients in the audio. A variety of factors affect this, including crossover, driver and cabinet design. Speakers that are set up to correct for timing problems (whether using electronic delays or physical methods) are said to be "time aligned".
Time Alignment
In a multiple driver loudspeaker system, it is important that the time delay inherent in each driver and its associated crossover network be the same to preserve accurate transient (see WFTD archive transient) response. In other words, the high frequencies and low frequencies much reach the listener's ear at the same time. A system which meets this criterion is said to be "time aligned." One way to accomplish this is to place the tweeter further away from the listener than the woofer, and this is done in many speaker systems. Another way is to design the crossover network to add a suitable delay to the high frequency signal before it gets to the driver.The phrase "time alignment" is also sometimes used in reference to adding delay to one or more microphones in a situation where more than one mic is being used on an instrument, and the mics are at different distances from the instrument. A good example of this is orchestral recording where several mics are employed at various distances to accurately capture the sound of the orchestra in the hall. The microphones closer to the orchestra are sometimes delayed to be more in "time" with microphones placed out in the hall."Time Alignment" was copyrighted as a trademark by a speaker manufacturer years ago and is no longer widely used as a generic term.
Transducer
For our purposes, a transducer is an electronic component that transforms one type of energy into another. Some examples: A microphone converts sound into electric current. Likewise, a speaker converts electric current into sound. Other common transducers include magnetic guitar pickups, piezo pickups, phonograph cartridges (remember those?) and tape heads. One of the main challenges we all face (whether we know it or not) is overcoming the physical limitations transducers put on our ability to reproduce the extremely wide dynamic range of acoustic sounds... deadly enemies of your gear!
Transistor
An electronic component known as a semiconductor. A semiconductor can be an excellent conductor of electricity under some conditions while it can resist conducting electricity under other conditions. There are basically two types, the bipolar transistor (also called the junction transistor), and the field effect transistor (FET). Transistors are typically designed with at least three terminals, one of which (the base) serves as a sort of control gate (for lack of a better term). A supply voltage is connected across the other two terminals and the (signal) voltage present at this third terminal serves to "turn on" the transistor to varying degrees allowing current flow. In a common amplifier circuit, for example, a large supply voltage is placed in series with the transistor and a load (such as a speaker), and then a small varying voltage (the source signal) is applied to the base. This causes the transistor to allow a varying amount of current to flow from the supply source (usually a power supply) through its load as the source signal's voltage rises and falls. This is a very, very rudimentary description of an amplifier circuit, which is the basic circuit used across a good portion of all analog audio (including equalizers, mixers, crossovers, amps, etc.). It is also worth noting that in many ways the transistor mimics its predecessor, the vacuum tube. There are some distinct and important differences to us audio people, but the basic function is the same.
Transondent
This one is sure to impress if casually thrown out at a gathering of audiocentric individuals! Transondent means transparent to sound passage - similar to transparent in reference to light. Pop filters and speaker grills are two items that should be transondent for best performance.
Tri-Amp
Similar in concept to bi-amp. Tri-amping refers to breaking an audio signal into three discrete frequency ranges prior to being amplified and delivered to speakers. The idea is to deliver audio frequencies to at least three different drivers or speakers that are optimized to reproduce the frequency ranges they get. An electronic crossover is used to break the signal into three ranges. Then each is sent to its own amplifier and speaker(s). Another benefit of tri-amping or bi-amping is that even when one frequency range overloads and distorts the other(s) remain clean, which can greatly improve overall clarity under high power/volume situations.
Truss
In our domain of concern, truss generally refers to a metal frame used to hang light fixtures from. It generally comes in three main designs - flat, box, and tri - which describe the shape created by the frame. By virtue of their construction trusses are very strong and able to carry extremely heavy loads. Most truss is now made of aluminum for weight reasons and sections can be bolted together to produce long pieces. Used extensively in concert production to form the 'roof' over the stage from which to hang everything from lighting to speakers. Even followspots can be mounted together with their operators who access their seats via circus style rope or wire ladders.
Virtual Dolby Digital
Dolby has specifically developed three types of "virtual" surround processing for computers, computer games, and video games. In "virtual" implementations, "phantom" speakers are created, as processing provides perceived sound sources in addition to the actual speaker complement. Virtual Dolby Digital is a computer format implementation of Dolby Digital. For this method, first a Dolby Digital decoder decodes the digital bit stream and 5.1 channel signals are produced. Then, a "phantom" channel is created providing a perceived center channel where none exists, and the two surround channels are processed through an additional DSP circuit and changed to "virtual" surrounds. All channels of information are provided through only two speakers. This system works best for a single listener who is centered between the left and right speakers. In the Virtual Dolby Digital implementation, some computers will decode the digital bit stream via a Dolby Digital decoder with the ability to "downmix" the 5.1 channels into a Dolby Surround encoded stereo signal. These two channels will then go through a two-channel sound card and be processed through an outboard or inboard Dolby Surround Pro Logic decoder to provide four channels of sound -- Left, Center, Right and Surround. The center channel can be switched to "phantom" mode if desired, but four speakers are needed for the left and right front and the two surround speakers at the sides or rear of the listening position.
Voice Coil
In a dynamic loudspeaker the voice coil is a winding of wire around a cylinder that is attached to the main part of a speaker. The voice coil is inserted into the "gap" created between the magnet and pole piece of a speaker magnet assembly and, when current travels through the wire, generates a magnetic field that causes the speaker to move against the permanent magnet. The alternating current in the voice coil causes alternating magnetic fields to form which interact with the permanent magnet to cause the speaker to move back and forth producing sound.
Watt
A metric unit of power defined as one Joule per second. The Joule is a unit of energy, so power equates to the rate of energy transfer, or the rate of doing work. Named after James Watt, the developer of the practical steam engine, the watt has become a common term in audio as used to describe the power handling capabilities and/or requirements of speakers, and the power delivery capabilities of amplifiers. Milliwatts (1/1000 of a watt) are often used to describe energy usage in low power electrical circuits.
Whizzer Cone
A small supplementary cone attached to the voice coil of a speaker for the purpose of producing and radiating high frequency content more effectively than the larger speaker cone. A whizzer cone is attached to the voice coil in the same place as the speaker cone; however, where whizzer cones are used it is necessary for there to be some additional flexibility in the joint between the speaker cone in the voice coil. This allows the speaker cone to become somewhat decoupled from the higher frequency motion of the voice coil so it doesn't dampen the voice coil from being able to move the whizzer cone at those higher rates. Whizzer cones have fallen out of popularity in the last couple of decades, mostly due to the added coloration of the signal produced by the necessary slop in the coupling between the speaker cone and voice coil.
Woofer
The low frequency speaker of a multi-driver speaker system. Sometimes in very low frequency applications it is called a sub-woofer. Woofers are generally larger speakers (12" - 18"), but the specific size is not a requirement or the defining feature. Rather the ability to accurately reproduce low frequencies of potentially high amplitude, which tends to require a large throw or excursion (the distance the speaker can move in and out). In any speaker system where multiple drivers are used the woofer is the one for producing the lowest frequencies.
Xophonic
An artificial reverberation device made for the home by the Radio Craftsmen in the 1950's. Basically it was a box about the size of a bookshelf loudspeaker that contained a small speaker and about 50 feet of tubing with a microphone at the other end. This produced a time delay of about 50 milliseconds and this reverb was mixed in with the original signal and radiated in the room through a separate amp and speaker. The Xophonic may have been the very first signal processor designed for home use. It was popular for a short time before falling into oblivion with the advent of stereophonic sound reproduction.
Zip Cable
The term used to refer to lower gauge, inexpensive speaker cable. This type of cable is often referred to as "lamp cord" as well. It's the type that looks the same as a low power general purpose electrical cord.
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