Sound/Record Terms 2

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.

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.

Comb Filter

A comb filter is a filter that has a series of very deep notches in its frequency response with the spacing of all of the notches at multiples of the frequency of the lowest notch (they are all harmonically related). It got its name from looking like a comb when plotted on a frequency response graph. Comb filters are produced when a signal is time delayed and added back to itself. Some frequencies will cancel and others will be reinforced, which can dramatically change the tonal color of the sound. In practice this is common problem that occurs when a stereo mix is collapsed to mono because many stereo effects, such as chorus and flanging, achieve their stereo imaging by using some form of the Haas effect. A static comb filter will make its audio sound kind of hollow or "phasey" depending upon how severe it is. Add some modulation and you have a flanger. Comb filtering is one of the main ingredients in the distinctive sound of a jet aircraft passing overhead. The difference in the time arrival to your ears of the direct sound versus the sound reflected off of the ground causes various frequencies to be cancelled or reinforced. As the plane moves these distances are all changing, thus causing the coloration of the sound to change. Again, it's the same principle used in a flanger.

Compressor

A compressor is a device that reduces the dynamic range of an audio signal. First a threshold is established. When the audio signal is louder than this threshold, its gain is reduced. The amount of gain reduction applied depends on the compression ratio setting. For example, with a 2:1 ratio, for every 2 decibels the input signal increases, the output is allowed to increase only 1 decibel. A variety of other parameters in the compressor will also affect its performance processing specific signals; attack time, release time and others are very important. There are a variety of uses and applications for compressors, the most obvious one being to control the dynamic range of a live performance so that it will fit into the fairly narrow dynamic range of recorders, etc. Other applications include making a signal's average level higher, increasing the apparent sustain on a guitar, evening out a vocal or bass guitar performance, fattening up sounds, and on and on. The list of possibilities is extensive!

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.

Gate

A dynamics device whose function is to remove unwanted audio material below a certain threshold. Some type of "gain cell" is employed (usually a VCA) that can raise or lower the volume of the audio going through the unit. When the signal falls below a certain threshold that is set the gain cell will quickly drop the audio level down to a predetermined level. This level is usually very low, or even off, but in some applications it may only be a reduction of a few dB. The reason they are called gates is because when they "close" it sounds as if the audio has suddenly stopped, or has been "gated." Now, it is possible to set many gates for slower response time so the effect is not as sudden, but often a sudden change is what is desired. Gates are often used on drum tracks to prevent bleed from other nearby drum mics, and they are sometimes used on noisy sources so when the desired audio signal stops the noise is automatically muted. The gated reverb sounds made popular by Phil Collins and Peter Gabriel in the 1980's were the result of running a reverb's decay through a gate. When the reverb level fell below a certain threshold the sound would abruptly cut off.

Ducker/Ducking

A dynamics processor/process that lowers (or "ducks") the level of one audio signal based upon the level of a second audio signal. A typical application is paging over background music: A ducker senses the presence of audio from a paging microphone and triggers a reduction in the output level of the music signal for the duration of the page signal. It restores the original level once the page message is over. Most dynamics processors (usually compressors are used) that give the user access to the detector circuit can be used for ducking. It is simply a matter of routing a copy/split of the second audio signal (the page in the example above) to the detector input such that it will trigger the gain cell to lower the level of the main signal (the music).

VCA Group

A feature found on many high end live mixing boards. A VCA group provides the same type of control over signal levels that a mute group provides for muting. Basically, VCA groups allow the sound engineer to control the volumes of several independent sources through one control fader without having to route them all through a common subgroup. It is called a VCA group because Voltage Controlled Amplifiers are used. In fact every controllable channel in the desk has its volume controlled by a VCA (as opposed to audio passing through a resistive fader) in order for this to work. Some more modern (and expensive) designs have employed a motorized fader scheme (also known as Moving Fader), but these sometimes aren't referred to as VCA groups since there may no longer VCA's involved (see the Technical Tip of the Day from 04/09/2002 for more background on that).

Low Pass Filter

A filter designed to allow low frequencies to pass while reducing the amplitude of higher frequencies. A low pass filter is synonymous with a high (or hi) cut filter.

Highpass Filter

A filter designed to let high frequencies pass while attenuating lower frequencies. A low cut filter is synonymous with a high pass filter. Sometimes the word high is written "hi" as a shortcut (as in Hi Pass Filter).

Distribution Amplifier

A fin like antenna for Wireless IEM units. Lessens signal loss in transmission from IEM transmitter to IEM receiver.

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!

Active Speaker

A loudspeaker that features built-in active electronics, such as filtering, EQ, amplification, and crossover. Active speakers afford the manufacturer the advantage of being able to optimize all of the system's components for best performance and to match the components to one another. The big advantage of active speakers for users is the convenience; one power cable and one audio cable are usually all that are required to use all the components contained in the monitor (as opposed to a system where the amps, crossovers, EQs, etc., are all in separate boxes and must be cabled together).

Threshold

A parameter found on compressors, limiters and noise gates (and a variety of other dynamics-based processors), the threshold setting determines at what level the processor will begin working. For example, on a compressor, when signal level exceeds the threshold setting, it will be compressed; below the threshold signal will be passed unprocessed. On a gate, threshold determines the minimum input level required to cause the gate to open up and pass signal; when input level drops below the threshold, the gate will be closed preventing signal from passing. Carefully setting the threshold allows you to very specifically control when processing is being applied to a signal.

Field Effect Transistor (FET)

A particular type of transistor, an FET behaves in a similar fashion to a triode (tube). There are actually several types of FETs, a common one in the pro audio world being the MOSFET (Metal Oxide Field Effect Transistor). FETs have a high input impedance, and respond in a linear fashion. This makes them ideal for condenser microphone preamps, as well as for certain power amplifier designs.

Pumping

A phenomenon associated with the use of dynamic processors such as noise reduction systems, compressors, and gates. Pumping is generally associated with breathing and is often used synonymously with that term. In context, the term pumping is also sometimes used to describe variations in the level of the desired signal that can occur as a result of the processor being unpredictably triggered by other elements of a sound that are not as apparent as the sound being processed. An example of this would be a wide band sound (like a full music track) where something that is happening in the very low end (perhaps below the frequency response of the studio monitors) is triggering the processor to make seemingly arbitrary (though often repetitive) changes to the level of the audio.

Feedback Eliminator

A product category that has become popular in the last 15 years. The Feedback Eliminator is a type of automatic, electronic equalizer. They work by finding the frequency of feedback and then tuning a precision, narrow bandwidth equalizer to it and cutting the level of that exact frequency until the feedback stops. A good feedback eliminator can go through this entire procedure in a few milliseconds, which means feedback can be stopped and controlled before it gets out of hand. It also means they can be used very effectively in live situations where feedback may occur at different frequencies over the course of a performance (for any of a dozen reasons). Like most product categories that have reached maturity, feedback eliminators come in many different configurations, brands, and price ranges. There are many different capabilities available within the general category, but for the most part the basic purpose is the same: to control feedback problems.

IEM

A special earpiece or earplug containing high quality miniature loudspeaker systems, similar to hearing aids, used for on-stage and recording studio purposes in lieu of traditional floor foldback monitors.

De-Esser

A special type of compressor that is tuned to be sensitive to sibilant sounds, or sounds with high frequencies such as the sound produced by the letter "s", hence the name de-esser. The need for de-essing arises out of a combination of the presence peak many microphones have in their frequency response to accentuate vocal recording combined with close proximity vocal work and possible added high frequency boost from equalizers and tone controls. While these things often make a vocal track have more "air" and high-end clarity, they can also add enough accentuation to certain consonants (especially the "s") that they become too pronounced. The problem can range from being slightly annoying to being bad enough to cause distortion in the signal path. Many years ago broadcast engineers figured out they could tune compressors to be more sensitive to these frequencies, which in effect produces an automatic volume control that can turn down the audio anytime one of the sibilant sounds occur. In fact, any compressor with a sidechain input can be turned into a de-esser by inserting an EQ and boosting the offending frequencies. Even more flexibility comes from using a multi-band compressor. The de-essing action no longer has to lower the overall signal level. It can just lower the level in the specific range of frequencies specified. Some modern de-essers, however, have very sophisticated circuitry and controls that are optimized for achieving results beyond what would be easy with a simple compressor with an EQ in the sidechain.

Microphone Preamp

A specific circuit, device, or section of a device (such as a mixer or an audio interface) designed to amplify the signal from a microphone. A microphone preamp must handle the relatively small and fragile output signals from all types of microphones, increasing the signal strength while maintaining the original integrity of the signal. Microphone preamp features may include pads, polarity invert/phase invert, input impedance selection, mute, phantom power, and more.

Multiband Compressor

A specific type of compressor that looks at specific frequency bands of audio and acts on them independently. For example, a multiband compressor can be set to only compress frequencies below 100 Hz, which would prevent a potential build up of low frequency content in a PA system or broadcast. Or it can be used as a mastering tool to aid in adjusting the overall spectral balance of a recording. Keep in mind this is fundamentally different than using a standard compressor with a sidechain. In that case the compressor is always acting on the whole signal, whereas a multiband compressor only acts on specific frequency ranges. One has to be careful, however, because misuse of a multiband compressor can result in skewed tonality. If you compress the high frequencies of a signal going to cassette tape (because you want to print more overall level to tape) to the point where the material ends up sounding dull you have defeated the purpose.

Clipping

A specific type of distortion. If a signal is passed through an electronic device which cannot accommodate its maximum voltage or current requirements, the waveform of the signal is sometimes said to be clipped, because it looks on a scope like its peaks have been clipped off by a pair of scissors. A clipped waveform contains a great deal of harmonic distortion (see WFTD archive harmonic distortion) and often sounds very rough and harsh. Clipping is what typically happens when an audio amplifier output is overloaded or its input over driven. Interestingly, light to moderate clipping does not usually reduce the intelligibility of some signals, especially speech. In fact, it has been shown that clipped speech is easier to understand than normal speech in noisy environments. A probable reason for this is the increased high frequency content that accompanies this type of distortion, which can make a signal stand out more among other sounds and noises. Aphex and some other companies have been using this principle for years in their "exciter" type products. By adding the right amount of distortion at the right frequencies a signal will sound almost clearer and more distinct amidst other sounds, thus standing out more in a mix.

Passband

A term used when working with filters. Passband refers to the frequency span (range) a filter passes, or the range of frequencies not attenuated by a filter. The passband is usually measured between the points where the response is 3 dB down in amplitude relative to the maximum overall level.

Haas Effect

A time delay from a sound source to each ear. Causes our brain to locate where the sound is coming from by recognizing which ear heard the sound first. applies to speaker systems in the sense where you want your speakers to be in "phase" with eachother and your audience in order to give them the clearest/best representation of the sounds you are sending them.

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.

Class B

A type of amplifier design. Class B differs from Class A in that there is no current flowing when the output devices are at idle, and as a result, they have to turn on from a zero current state when signal is present. In a push pull Class B design the output devices would each produce half of the audio waveform (one set for the positive half, and another for the negative half) and would not have any current flow when the other half is operating. Class B designs tend to have a slower slew rate and more crossover distortion but are less expensive and require less robust power supplies.

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.

Soft Knee (Compression)

A type of compression where the onset of compression is gradual. In normal or hard knee compression when the signal reaches the threshold the unit immediately begins to compress at whatever ratio is set. In some situations the compression becomes very easy to hear (which is often not desirable) as the signal amplitude moves above and below the threshold. This is usually made worse when using high compression ratios. The solution is to have the compressor gradually enter into compression at a lower ratio prior to the signal reaching the threshold. The ratio is gradually ramped up as the signal gets louder until, at some point beyond the threshold, the full compression ratio is reached. This slower onset often makes the compression much more difficult to detect. The process is called "soft knee" because of how the compression ratio looks when plotted on a graph. In normal compression the knee (which is the point where compression begins) is an abrupt angle (how steep depends on the ratio) whereas in soft knee it is more of a curve.

Closed Ear

A type of headphone design where the headphone forms some type of a seal around (or in) the ear. The purpose of closed ear headphone designs (in contrast with open air designs) is to provide isolation between the headphone signal and the outside world. This benefits users who are trying to monitor signals in loud environments. They also help keep headphone signals from leaking out and possibly corrupting a recording by leaking into mics, etc. Closed ear designs are usually not as comfortable as the better open air designs, and some users believe they don't generally sound as good, but they are a necessity for most recording studios.

Companding

A type of noise reduction used in audio equipment, a compander circuit is a combination of a COMPressor and an exPANDer. The signal is compressed before recording it to tape (which maximizes the signal to noise ratio), then expanded as the tape is played back. As the signal is expanded, tape noise tends to be "pushed down," resulting in a quieter signal.

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. Can be run great distances with less loss and can have many speakers on them as compared to typical high current speaker lines. situations where an amplified signal must be distributed over vast areas without the need for a very high SPL in any one area.

Notch

A word used to describe a very narrow band of frequencies to be cut by an equalizer. When an EQ circuit has a very high Q (narrow bandwidth) it is sometimes referred to as a notch filter. Notch filters are commonly used to suppress feedback in monitor or PA systems, and are sometimes used to remove specific types of hum and noise in recordings.

M-S Stereo

Abbreviation for Mid-Side, a method of stereo miking and recording. MS recordings capture the relative intensity of different sounds across the stereo soundfield. In order to make an M-S recording one must deploy a cardioid pattern mic facing the sound source(s) and a figure 8 pattern positioned sideways to the source. The figure 8 mic is connected to two channels of the mixer, with one channel having its polarity reversed. Each of the two signals (one of which is polarity reversed) of the figure 8 mic, when combined with the signal from the cardioid mic produces either a left or right "image" that is roughly equivalent to two cardioid mics positioned with a 90 degree angle between them. The only advantage to the MS method is the user can alter the width of the stereo image by varying the relative levels of the two microphones. There are several disadvantages, most of which are a function of having two dissimilar mics reproducing the same signal. Of course they can't occupy the exact same space either, which produces other phase and frequency response anomalies.

RFI

Abbreviation for Radio Frequency Interference. RFI is a specific type of EMI relating only to signals produced by radio and television systems. These hi frequency RF signals can find their way into our audio equipment where they can produce hums, buzzes, and occasionally even reproduce the actual audio of the radio station through the equipment. Most audio circuits are not designed to deal with RF frequencies, which is what turns them into problems, so the normal course of action is usually to prevent them from entering in the first place. This is typically accomplished by shielding, filtering, and proper grounding among other things.

Ultra High Frequency

Abbreviation for Ultra High Frequency. Similar to VHF, UHF pertains to a band or range of radio frequencies defined by the FCC (Federal Communications Commission) to be used for some television stations and a wide variety of wireless two-way communication systems. UHF picks up where VHF leaves off, having a frequency range of between 300 MHz and 3000 MHz.

Very High Frequency

Abbreviation for Very High Frequency. VHF pertains to a band or range of radio frequencies defined by the FCC (Federal Communications Commission) to be used for some television stations and a wide variety of wireless two-way communication systems. The frequency range of VHS is between 30 MHz and 300 MHz.

Low Cut Filter

Also known as a high pass filter. Basically this is a type of filter that removes low frequencies from an audio signal. Normally they are designed so they remove frequencies below a certain determined frequency (often somewhere between 20 Hz and 150 Hz). In typical designs these filters have slopes, which means there is more and more attenuation as the frequency gets lower. So right around the rolloff (or cutoff) frequency the signal may only be down 3dB to 6dB (3dB is standard), but depending upon the design of the filter, lower frequencies may be considerably more attenuated. This is usually rated in dB/octave, or decibels per octave of rolloff. If your filter is at 150 Hz it is safe to assume the signal will only be reduced by 3dB at that frequency. However, one octave below that, at 75 Hz, your signal may be attenuated 15dB, or 12dB more. This would represent a 12dB per octave rolloff, which is common.

Real Time Analyzer (RTA)

An RTA is a device which uses a number of narrow bandwidth filters connected to a display to give a visual indication of the amplitude in each frequency band. RTA's are useful for getting a reading on how a room will subjectively sound, where problem frequencies might be, and how to approach EQ'ing to correct for those problem frequencies.

Circumaural

Around the ear.

Hi-Z

As the letter Z is the commonly agreed upon abbreviation for impedance, then Hi-Z simply refers to "hi-impedance". This refers to the input or output impedance of a device (in our cases an audio device). Precisely what Hi-Z means, and how it is applied in the audio industry, is not entirely concrete. In general devices with impedances up through 600 ohms are said to be "low impedance", while devices with impedances of several thousand ohms and up are considered "high impedance". Typically we only come in to contact with these generic terms on microphones (usually low cost microphones), some direct boxes, and certain types of line inputs (on mixing boards, some tape decks, etc.). A typical guitar, for example, generally needs to be connected to a Hi-Z input. Otherwise the electronics will be "loaded down" and the sound will be significantly altered. A Hi-Z microphone, which we don't encounter very often in pro audio (we generally use low impedance mics), definitely needs to be connected to a high impedance input, and even then the cable length can't be more than 10 or 20 feet before the signal degrades.

Bridged

Configuring a 2ch amp to act as 1 amp, combing the power of both channels. the input is split into 2 signals like the amp would normally work but 1 signal is flipped in polarity and combined with the other in the output connection. bananna to speakon cable. doubles ohm load. combines each single channel wattage.

EMI

EMI (Electro Magnetic Interference) refers to interference in audio equipment produced by the equipment or cabling picking up stray electromagnetic fields. This interference usually manifests itself as some type of hum, static, or buzz. Such electromagnetic fields are produced by fluorescent lights, power lines, computers, automobile ignition systems, television monitors, solid state lighting dimmers, AM and FM radio transmitters, and TV transmitters. Methods for controlling EMI include shielding of audio wiring and devices, grounding, elimination of ground loops, balancing of audio circuits, twisting of wires in balanced transmission lines, and isolation transformers among others. Completely eliminating EMI in a system ranges from easy to nearly impossible depending upon the equipment and the environment in question.

Flange/Flanging

Flange is a time based effect that happens when a duplicate signal is slowed down and combined with the original signal. Comb filtering happens, or an extreme form of phasing between the two signals slightly out of time from eachother. originates from the metal rim on tape decks where engineers would press down on one side. now in the digital world this can be done without that.

Cutoff Frequency

In a filter, the cutoff frequency is the point where the response is 3 dB down in amplitude from the level of the passband. Beyond the cutoff frequency, the filter will attenuate all other frequencies, depending on the design of the filter. On a sweepable shelving EQ or filter, what you are "sweeping" (or changing) is the cutoff frequency. To our ears, this changes the point at which the filter is operating.

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.

Sensitivity

In audio terms, sensitivity is the minimum amount of input signal required to drive a device to its rated output level. Normally, this specification is associated with amplifiers and microphones, but FM tuners, phono cartridges, and most other types of gear have a sensitivity rating as well. In general, higher sensitivity is better (less input signal required for full output), but there are definitely situations where a device can be TOO sensitive (picture a very sensitive microphone in front of a wound-up Marshall guitar amplifier!) resulting in unwanted distortion.

Knee

In audio the term knee refers to a point where the response of some system or function exhibits a notable change. (The name comes from the human body part, because when these things are plotted on a graph the resulting line sometimes resembles a somewhat bent human leg, where the knee joint represents a change in behavior.) For example, in an audio compressor the threshold setting creates a knee at a particular level inside the compressor as it relates to the dynamic range of any passing audio. Below this knee the compressor will exhibit a different behavior than above the knee. The term "soft knee" was coined to describe circuit behaviors where the change is smoothed out over a finite range of levels.

Signal To Noise Ratio

In layman's terms this is simply a measurement of a given noise level in a device as compared to the level of the signal. Higher numbers signify a greater difference, which is better. In technical terms it is the ratio of signal power at a reference point in a circuit to the noise power that would exist if the signal were removed (its noise floor). The maximum signal to noise ratio (which in many schools of thought is equivalent to dynamic range) of a given piece of equipment can be an important thing to know. This ratio is how much absolute noise it has compared to the highest signal voltage it can pass without distortion. While signal to noise ratio is often used as a specification to characterize relative quality differences in equipment, the way in which measurements must be done, and the degree to which they can differ, makes the true objectivity of such measurements highly suspect. Factors such as how much distortion can be allowed before you say the signal has reached "maximum" as well as other kinds of noise (like modulation noise) that may only show up when signals are present are just two examples of many variables that affect objective measurements. In digital equalizers the signal to noise ratio is a function of the maximum possible sine wave signal power compared to the quantization noise (a.k.a. quantization error) power. This is a very unambiguous value in linear PCM (Pulse Code Modulation) systems, but in other types of PCM systems the quantization noise (or quantization error) depends strongly on the level of the audio being recorded so it is very difficult to nail down the actual signal to noise ratio. It is sometimes useful to be able to compare S/N Ratio differences between equipment in certain applications, but it is more important to just understand the concept. Signal to noise ratio concerns us every time we pass audio (or video or data) though anything, and knowing what factors in our setup (such as gain structure) affect it is a fundamental part of building clean, quiet systems and mixes.

Crosstalk

In multi-channel audio systems, crosstalk is signal bleeding or leaking from one channel to another. Mixers, tape recorders, and many other pieces of gear are all susceptible to this problem. In most modern gear, crosstalk is not a major concern, but be aware that older gear can have significant amounts of bleed between channels!

LEDE - Live End, Dead End

LEDE is a trademarked term for a particular acoustic design. In an LEDE studio, the area around the monitors is deadened, or made absorbent acoustically. The remainder of the room (behind the listener) is made "live" or reflective. The main principle is that the arrival of reflections at the console is in a specific order: 1. direct sound from the monitors; 2. First studio reflection (from the recording room, through the mics and monitors); 3. First control room reflection (off the back wall, assuming it is 10 feet or so behind the engineer). The idea is that by staggering these arrivals, the control room reflections don't interfere with monitoring recorded studio acoustics.

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.

Bandwidth

Literally, bandwidth is a frequency span. Beyond that definition, its meaning will depend somewhat on context. For example, the bandwidth of a bandpass filter is the upper cutoff frequency minus the lower cutoff frequency (cutoff frequency being the filter's -3 dB point). The audio bandwidth is generally given as 20 Hz to 20,000 Hz, although there are harmonic components of audio that extend far above the 20k point. In most situations where bandwidth is given as an audio spec, the wider the frequency range the better. Be sure that when comparing bandwidth on different devices, that the same spec is being expressed. For example, some effects devices cite their bandwidth spec based on the dry, or unprocessed signal, while others give the bandwidth of the actual processed sound. These difference between these two specs (both listed as "bandwidth") can be substantial!

Neodymium

NE - O - Dim - E - um. a stronger magnet alternative for microphone/speaker voice coils. boost signal 6db.

Constant Q

On many equalizers changing the gain of a frequency band also changes the Q, which effects the slope of the EQ curve and how many adjacent frequencies are effected to what degree. This is also the normal way for a simple filter design to work. Some manufacturers employ what they call constant Q designs with the idea that the equalizer behaves in a more predictable fashion as gain changes are made at various frequencies. The Q of an equalizer 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. If the half power bandwidth of this EQ remains 232 Hz wide throughout its cut and boost range it can be said to be a Constant Q equalizer. Engineers debate whether this is really useful and whether it sounds as good as more conventional designs.

PZM

Pressure zone microphone. commonly known as boundary mics. an electret is mounted to a metal back plate which causes a hemispheric polar pattern and 6db increase in sensitivity. great for piano micing.

Bandpass

Refers to filters. It is basically the opposite of band-limit. A bandpass filter allows a specific range of frequencies through while attenuating all others.

Saturation

Saturation refers to the maximum amount of magnetism a magnetic tape can hold. Attempting to add more magnetism to the tape's oxide particles will result in distortion. Normal record levels do not generally approach saturation as distortion will be introduced before saturation is reached, especially in the low frequencies. High frequencies normally do not saturate as they have a tendency to self-erase during recording. Engineers using analog tape often make use of tape saturation as an effect. By carefully controlling record levels, compression, warmth, and fatness can be added to a signal - all part of the much-heralded analog "mystique".

Sibilance

Sibilance refers to the high frequency components of certain vocal sounds, especially "s" and "sh". Sibilance lives in the 5 to 10 kHz frequency range, and can cause problems if over-emphasized in a recording. While it is possible to use a graphic or parametric EQ to correct for sibilance, this is often an unsatisfactory approach. Often the overall track will begin to sound dull before the sibilance is corrected. A better solution is to use a dedicated de-esser, or use an EQ in the sidechain input on a compressor to perform de-essing (see "sidechain" in the inSync Word For The Day archives for more on this). Since a de-esser dynamically corrects for sibilance (only processes where necessary), the resulting track will sound much more natural.

Slew Rate

Slew rate is the ability of a piece of audio equipment to reproduce fast changes in amplitude. Measured in volts per microsecond, this spec is most commonly associated with amplifiers, but in fact applies to most types of gear. In amplifiers, a low slew rate "softens" the attack of a signal, "smearing" the transients and sounding "mushy." Since high frequencies change in amplitude the fastest, this is where slew rate is most critical. An amp with a higher slew rate will sound "tighter" and more dynamic to our ears.

XY Stereo

Stereo micing 90 degrees. capsules must meet at the same point or one on top of the other to avoid any phase problems due to the distance between the capsules. The stereo image is produced by the off-axis attenuation of the cardioid microphones. While A-B stereo is a difference-in-time-stereo, the XY stereo is a difference-in-level stereo. But as the off-axis attenuation of a typical cardioid microphone is only around 6dB at 90 degrees, the channel separation is limited, and wide stereo images are not possible with this recording method.

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.

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.

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).

Headroom

The difference between the normal operating level of a device, and the maximum level that device can pass without distortion is the headroom. Music generally has wide variations in dynamic range; without enough headroom, you'll find your gear clipping (distorting) far too frequently! There are a variety of other places where it is desirable to have large amounts of headroom (i.e. when mixing signals together); in general the more the better!

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.

2-bus

The main left/right stereo output bus on a mixing console.

Expander

The opposite of a compressor. Where a compressor takes a given dynamic change and reduces it, an expander increases it, making changes larger. Expanders are used to "un-do" compression in some circuits (companding). More commonly, expanders are used for noise reduction. In this application (downward expansion), a threshold is set at a level below desired audio signals, but above the noise floor. When signal drops below the threshold, expansion is applied, pushing signal even further down, reducing the level of noise. For example, an expander might be set up with a 1:6 ratio. This means that for every 1 dB of input level change the expander sees, it will output a 6 dB change. When a signal drops below the threshold by 2 dB, the output of the expander will drop by 12 dB, similarly dropping the level of any background noise floor. (See also "Compressor" and "Ratio" in the WFTD archives).

THD (Total Harmonic Distortion)

The ratio of the power of the fundamental frequency at the output of a device versus the total power of all the harmonics in the frequency band at the output of the device. Basically, all electronic audio devices introduce some distortion to audio passed through them. The simplest form of this distortion is the addition of harmonics to the outputted signal. THD represents the sum of all the harmonics added by a device as a percentage of the level of the signal being measured. The closer THD is to zero, the more "transparent" a device should sound (all other things being equal, which they never are...). Various devices contribute differing types of harmonic content to a signal, this is part of what can give them their distinctive sounds. For example, tubes add different harmonics than transistors, different circuit designs emphasize different harmonics, etc. When engineers talk about the "sound" of a piece of equipment, this is a part of what they are referring to.

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.

RMS (Root Mean Square)

The square root of the mean of the square. RMS is (to engineers anyway) a meaningful way of calculating the average of values over a period of time. With audio, the signal value (amplitude) is squared, averaged over a period of time, then the square root of the result is calculated. The result is a value, that when squared, is related (proportional) to the effective power of the signal. Unfortunately, calculating the RMS value of anything but a simple sine wave (.707 of peak) is very difficult. The further a signal gets in harmonic content from a sine wave, the less accurate RMS values will be. For a dynamic signal like most music, it is nearly impossible to get even close to a true RMS value. Note: RMS Power is actually a misnomer, since the RMS of a signal is a really just a value used to calculate average, or continuous, power.

Pole

This word has a number of mathematical definitions, some of which relate to audio, and particularly the design and implementation of filters in audio. Capacitors and/or inductors are often integral components of (analog) filter design due to the way in which they interact with varying frequencies of periodic energy. For example, it is possible to create a low pass or high pass filter by placing a single capacitor in a circuit, assuming there is some resistance (a load) elsewhere in the circuit. This is known as an RC circuit (Resistance and Capacitance). In filter design it is understood that a single RC circuit - a circuit with one capacitor and one resistor - is a one "pole" filter. A two pole filter has two RC circuits, and so on. A one pole filter will generally provide a high or low pass roll off in the neighborhood of 6 dB per octave (accompanied by some phase shift). For example, in a high pass filter the voltage of the signal will be cut in half each time the frequency is reduced by an octave, which corresponds to a reduction of 6 dB according to the formula 20log (V1/V2). A two pole filter will have a steeper roll off of 12 dB per octave. The more poles in a filter the steeper the roll off. In a band pass filter more poles equates to a higher Q. If one were to create a resonant filter, as is common in synthesizers (and wireless communication systems (radio, TV, etc.)), more poles would equate to a higher degree of resonance around a more narrow range of frequencies. All of these concepts are clearly related, but applied in different ways. Digital circuits and software can emulate the behavior of poles through certain types of algorithms; however, software can also be written which will mathematically reduce the amplitude of frequency ranges in a manner quite different from "conventional" filters, which can allow for more control over certain aspects of a complex signal and produce a different sonic characteristic.

Diversity Receiver

Used to improve reception of RF signals. Uses two separate, independent antennas. the diversity receiver continually compares the signal strengths from each antennas and quickly switches back and forth to whichever has the strongest signal.

Bi-Amp

When a single audio signal is divided into two frequency ranges and then sent to two separate amplifiers, which in turn drive separate loudspeakers it is said to be bi-amped. A crossover network is used to divide the audio into ranges that are more suitable for the drivers that will be used to drive them. It also allows the amplifier(s) to be chosen or designed with a more specific set of criteria in mind. Bi-amping, Tri-amping, and beyond have been used in sound reinforcement systems for years and it has become quite common in active studio monitors as well.

Skin Effect

When more current travels across the outside of a conducting wire than the inside core.

3:1 Rule of Microphone Placement

When using 2 microphones to record a source. Place the 2nd microphone 3 times the distance away from the source as the 1st mic.

Binding Post

an electrical terminal commonly found on an amplifier output. accepts banana plugs, alligator clips, and bare wire.

Frequency Agile

equipment that can operate on more than one frequency.

Sidechain

inSync reader Shawn E. in Tokyo wonders what a sidechain is, and how it is used - A sidechain (sometimes called a key input, or a detector input) is a control input used to trigger a compressor or gate with an external signal. Let's look at a common example, ducking: When recording voice-overs, the background music bed is run through a compressor, which is set so that it is not normally operating on the input signal. The voice-over announcer's mic signal is split so that it feeds both the mixer's input, and a sidechain input on the compressor. When the announcer speaks, their voice goes to the sidechain, where it tells the compressor to start working, turning down the background music. When they stops speaking, the sidechain tells the compressor to stop working, and the music comes back to its uncompressed level. Other uses? Try using a kick drum to trigger a gated bass synth for extremely tight rhythms, or insert an EQ'd signal into a sidechain, making a compressor more or less sensitive to certain frequencies (de-essing is a good example of this), many other applications are possible - feel free to experiment!

Coincident

stacked stereo micing technique getting capsules as close as possible. either XY cardioid 90 apart. or Mid-Side two figure 8 capsules 90 off axis from eachother.

Breathing

the change in level of background noise with noise reduction