3:1 Rule of Microphone Placement
When using two microphones to record a source, normally you will get the best results by placing the second mic three times the distance from the first mic that the first mic is from the source. Confusing? An example: If the first mic is 1 foot from a source, the second mic should be placed 3 feet from the second mic. Using the 3:1 Rule will minimize phase problems created by the time delay between mics.
This rule originated when engineers were miking multiple sources in the same vicinity. The same principle applies. If you are recording two different sources of sound, their respective microphones should be at least three times further apart than they are close to their respective sources. Keep in mind that rules are meant to be broken; you may prefer the sound created by ignoring the 3:1 Rule - experiment and let your ears be your guide!
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.
2:1 Rule of Ambience
To capture an equal amount of room ambience, a cardioid microphone must be placed twice as far from a source as an omnidirectional pattern microphone. Keep this in mind the next time you are trying to capture some of a room's natural sound when recording!
1/4 wave
Refers to wavelengths of audio or electromagnetic radiation (i.e. radio waves). One quarter of a wave denotes some dimension having a relationship with a signal such that it is 25% (or 1/4) as large as the space required for the entire wave of the signal. For example, in an RF (Radio Frequency) or wireless system 1/4 wave antennas are common. A 1/4 wave (typically pronounced "Quarter Wave") antenna's length is 1/4 as long as the wavelength of the carrier frequency used by the system. There are also 1/2 wave antennas, and so on. Similar relationships exist in the field of acoustics (see WFTD Quarter Space), though they aren't usually this specific (audio systems generally have to respond to a wide range of frequencies) and it's not as common to hear things referred to in this manner.
A-weighted
A standard for noise measurement that takes into consideration the human ear's sensitivity to certain frequencies (see Fletcher-Munson Curves). This is expressed as part of noise specifications and can be denoted by adding the letter 'A' to the spec - i.e. 15dBA.
Absorption
In acoustics (as opposed to paper towels), the opposite of reflection. Sound waves are "absorbed" or soaked up by soft materials they encounter. Studio designers put this fact to work to control the problem of reflections coming back to the engineer's ear and interfering with the primary audio coming from the monitors. The absorptive capabilities of various materials are rated with an "Absorption Coefficient," which is a measure of the relative amount of sound energy absorbed by that material when a sound strikes its surface. (See also WFTD "Anechoic")
Absorption Coefficient
Represented by the Greek letter Alpha, a measure of the relative amount of energy that will be absorbed when a sound hits a surface. Absorption coefficients are always a value ranging from 0 to 1 that when multiplied by the surface area in question yield a percentage of sound that will be absorbed by that surface. This percentage is in units known as Sabins, after the Harvard professor and acoustician, Wallace Sabine. An absorption coefficient of 1 means that all acoustic energy striking the surface will be absorbed and none reflected. A coefficient of 0 means that all the energy is reflected. The latter condition is virtually impossible and the former condition is rare. The absorption coefficient varies by frequency because most materials have different absorption characteristics at different frequencies. Acousticians use absorption coefficients to help determine the RT-60 or reverberation time of rooms, and as such many common building materials have been measured and results published for their absorption coefficients.
Acoustic Treatment
Acoustically treating a room is necessary in audio production due to the fact that very few "spaces" have the physical qualities that make for accurate monitoring or desired recording. There are many things that can be done to a space before and during construction to optimize its acoustic behavior. These include the shape of the space, its isolation, and the surface materials. Once a room is already constructed, Acoustic Treatment mostly tends to consist of treating the surfaces. There are two primary elements to consider: absorption and diffusion. Acoustic foam is well suited to alleviate slap and flutter echo, the two most common problems in rooms not specifically designed for music recording and performance. In fact, foam can turn even the most cavernous warehouse or gymnasium into a suitable acoustic environment. Diffusion keeps sound waves from grouping, so there are no hot spots or nulls in a room. In conjunction with absorption, diffusion can effectively turn virtually any space into one that is appropriate and useful for the purpose of recording or monitoring sound with a high degree of accuracy.
Anechoic
Literally, without echoes. Anechoic refers to the absence of audio reflections. The closest thing to this situation in nature is the great outdoors, but even here there are reflections from the ground, various objects, etc. It is almost impossible to create a truly anechoic environment, as there is no such thing as a perfect sound absorber. At high frequencies, it is possible to create near-anechoic conditions, but the lower the frequency, the harder this is (Absorption is wavelength dependent. As an example, a 100 Hz wave is about 10 feet long; the absorber must be at least 1/2 a wavelength deep to function properly. It quickly becomes impractical to create a large enough space with enough material in it to absorb low frequencies).
It is not desirable to create anechoic or near-anechoic conditions in a recording studio. The total absence of reflections skews perception, and will not result in good recording or mixing decisions. Anechoic chambers are used for testing and spec'ing microphones and loudspeakers, as well as for a variety of other audio measurements.
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.
Bass trap
A device used to help acoustically tune a room. Enclosed spaces all have resonant frequencies based upon the various dimensions of the space. As a room becomes energized with sound certain frequencies will build up or be cancelled at various locations around the room based upon its shape and dimensions. A bass trap is a low frequency sound absorber used to reduce the effects of standing waves in a room. They are usually placed in corners or along wall joints where low frequency energy tends to build up. The absorption qualities of bass traps prevent low frequencies from interfering with each other throughout the rest of the room, which results in much more accurate response in the listening area. Bass traps come in many shapes and sizes and employ a variety of construction techniques. Some are tuned to kill a narrow band of frequencies while others are designed to cover a broad range.
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.
C-Weighting
A type of weighting curve designed into filters for equipment measuring sound output levels. The C-curve is basically "flat," with -3 dB corners of 31.5 Hz and 8 kHz, respectively. It is designed to loosely correspond to how humans perceive sound at higher volume levels as indicated by curves such as the Fletcher Munson Curves.
CCIR 468-weighting
A weighting filter used in sound level measurements. This filter was designed to maximize its response to the types of impulsive noise often coupled into audio cables as they pass through telephone switching facilities. Additionally it turned out to correlate particularly well with noise perception, since modern research has shown that frequencies between 1 kHz and 9 kHz are more "annoying" than indicated by A-weighting curve testing. The CCIR 468-curve peaks at 6.3 kHz, where it has 12 dB of gain (relative to 1 kHz). From here, it gently rolls off low frequencies at a 6 dB/octave rate, but it quickly attenuates high frequencies at ~30 dB/octave (it is down -22.5 dB at 20 kHz, relative to +12 dB at 6.3 kHz).
CCIR ARM-weighting or CCIR 2 kHz-weighting
CCIR ARM-weighting or CCIR 2 kHz-weighting - A type of weighting filter for sound level measurements. This curve derives from the CCIR 468-curve we've covered in WFTD previously. Dolby Laboratories proposed using an average-response meter with the CCIR 468-curve instead of the costly true quasi-peak meters used by the Europeans in specifying their equipment. They further proposed shifting the 0 dB reference point from 1 kHz to 2 kHz (in essence, sliding the curve down 6 dB). This became known as the CCIR ARM (average response meter), as well as the CCIR 2 kHz-weighting curve. (See: R. Dolby, D. Robinson, and K. Gundry, "A Practical Noise Measurement Method," J. Audio Eng. Soc., Vol 27, No. 3, 1979)
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).
Coupling
In electronics, coupling refers to ways of connecting circuits or subsystems to one another. For example, gain stages of an amplifier may be directly coupled or may have capacitors or transformers in line. The capacitors and transformers eliminate a direct connection, but still provide coupling that allows the signal to be transmitted from one stage to the next. There are many different types of electronic coupling.
Critical Distance
When dealing with acoustics, critical distance is the point at which the volume of a sound source is equal to the volume of reflections from that source off of other surfaces. Control of the volume and timing of these reflections is an important part of creating an accurate listening environment.
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.
Decay
In audio, Decay is the manner in which sound ceases. Any acoustic signal or waveform envelope of an electronic musical instrument can be said to have a few components such as attack, internal dynamics, sustain, release and decay which help define the character of the signal or waveform envelope. The nature of the Decay of any signal or waveform envelope varies based on factors such as time duration and amplitude of the Decay.
Decay Time
The time it takes for the sound pressure level of reverberations to drop in level by 60 dB (one millionth) from their original strength. This is sometimes also called "reverb time." Carefully setting the decay time allows you to have the mix be as "wet" as you desire, without making things muddy or unclear...
Decibel
We've all used the term "decibel" hundreds of times, but what does it REALLY mean? A decibel (named for Alexander Graham Bell) is a tenth of a bel, and is used as an expression of power. Here's where the confusion arises: A decibel isn't a measure of ANYTHING; it is a ratio of two power levels. Because of the way our ears perceive volume, these ratios follow a logarithmic curve, expressing them as a decibel keeps things easier to deal with. Here are a few convenient decibel figures worth remembering: One decibel is commonly taken as the smallest volume change the human ear can reasonably detect. Doubling the POWER of an amplifier results in a 3 dB increase, which is a "noticeable" volume increase. Doubling the VOLUME of a sound is a 6 dB increase (you may occasionally see 10 dB listed as the "double-volume" figure, 6 dB is the more mathematically correct number). By doing the math, you can see that truly doubling your volume actually requires 4 times the amplifier power! Keep these figures in mind the next time you are comparing the specs of two pieces of equipment...
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.
Diffraction
A phenomenon in the propagation of waves where the direction of a wave front (either sound wave or electromagnetic [light] wave) is altered when passing by an object or through a small aperture in a large surface. At shorter wavelength relative to the obstacle, sound (and light) will tend to reflect off the surface more and bend around it less (which partially explains why you can hear, but not see at a concert when someone is standing in front of you). Waves will also bend to fill an opening behind a surface (which partly explains why you can hear someone talking in the next room through an open door even though you can't see them).
Diffusion
Diffusion is the process of spreading or dispersing radiated energy so it is less direct or coherent. A Diffuser is a device that does this. The plastic covers over fluorescent lights in many office environments are diffusers. They make the light spread out in a more randomized way so it is less harsh. In audio, diffusion is a characteristic of any enclosed (or partially enclosed) space. It is caused by sound waves reflecting off of many complex surfaces. For example, a flat concrete wall produces a pretty distinct echo when sound reflects off of it. However a brick wall, while still pretty reflective, tends to diffuse the sound reflections and produces a much less distinct echo. This is due to both the surface of the brick itself and the mortar between the bricks (more specifically the edge diffraction of the joint between the two). All surfaces will of course differ and it is usually a variety of surfaces that create the most randomized diffusion of sound. Diffusion is a very important consideration in acoustics because it minimizes coherent reflections that cause problems. It also tends to make an enclosed space sound larger than it is. Diffusion is an excellent alternative or complement to absorption in acoustic treatment because it doesn't really remove much energy, which means it can be used to effectively reduce reflections while still leaving an ambient or live sounding space.
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).
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.
Far Field
In terms of sound radiation from a source the far field is the area beyond the near field boundary. Exactly where this boundary occurs is the subject of some debate, but engineers basically agree that it is very close to the source, or within one wavelength at a particular frequency of interest. The far field is actually comprised of two different sub-fields. These are the free field and the reverberant field. The main difference between these two fields is that sound in the free field will theoretically follow the Inverse Square Law of dissipation (-6dB each time the distance from the source is doubled), whereas sound in the reverberant field is sufficiently altered by the reverberations that the overall level will not fall as rapidly.
Fletcher-Munson Curves
Fletcher and Munson were researchers in the '30s who first accurately measured and published a set of curves showing the human's ear's sensitivity to loudness verses frequency. They conclusively demonstrated that human hearing is extremely dependent upon loudness. The curves show the ear to be most sensitive to sounds in the 3 kHz to 4 kHz area. This means sounds above and below 3-4 kHz must be louder in order to be heard just as loud. For this reason, the Fletcher-Munson curves are referred to as "equal loudness contours." They represent a family of curves from "just heard," (0 dB SPL) all the way to "harmfully loud" (130 dB SPL), usually plotted in 10 dB loudness increments. Though the Fletcher-Munson Curves are by far the most widely known contours there have been others defined in recent years that some engineers think are more accurate.
Flutter Echo
A condition that occurs in acoustic spaces when two parallel surfaces reflecting sound between one another are far enough apart that a listener hears the reflections between them as distinct echoes. The audible effect is in many cases a sort of "fluttering" sound as the echoes occur in rapid succession. In smaller rooms it can take on a sort of tube-like hollow sound, as the echoes are closer together.
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.
Gobo
Short for "Go-Between." A gobo basically forms a type of barrier: sometimes this can be between a light source and an area to be lighted where you want to keep the light off of part of it, or it can be to form a barrier for sound such that a particular sound source is shielded from a microphone during recording. Gobos are often used in recording studios for just this purpose. Say you have an acoustic guitar and a drum set in the same room. In order to help reduce the amount of drums bleeding into the acoustic guitar mic sound, absorbent panels, or gobos, are place between the drums and the guitar mic.
Haas Effect
At Sweetwater Sound this is the phenomenon that occurs when our Director of Human Resources, Kristine Haas, enters a room carrying pink slips. The more commonly known usage of the term is in audio and pertains to the psychoacoustic phenomenon of sound source localization. If a sound source is presented to our ears at the same level, but one arrives just a few milliseconds later, our hearing mechanism will judge the sound to be coming from the side of the head where the earliest sound arrived. How far to one side or the other depends on the difference in time between the sound arriving at each ear. This is true for arriving sounds up to about 25 milliseconds of delay, after which it will begin to sound like two distinct sounds. This phenomenon is used in all kinds of audio production techniques to help position various instruments around the stereo (or three-dimensional) soundfield without creating imbalances in the levels of the left versus right signals.
The effect is also called the precedence effect and means that if there are two sources of sound, as is often the case with PA systems or studio monitoring systems, the sound will be localized to the speaker that provides the earliest sound. The other speaker will not be heard at all it some cases.
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.
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.
Imaging
The ability to localize a sound in a stereo field or mix is called imaging. Several things will affect the ability of a speaker system to image accurately: How matched the speakers are in construction and level (volume), exactly matched phase, and the interaction of the speakers with the listening environment will all be critical in determining imaging. Assuming that the first items are determined by the speaker manufacturer and your system set up, careful acoustic treatment in your room can often make the largest difference in the clarity, stereo spread, and imaging of your studio monitoring system.
Infrasonic
Refers to sounds or signals whose frequencies are below the normal human hearing range, generally considered to be 20 Hertz (see WFTD archive Hertz). The lowest audible frequency is not easy to absolutely define as it depends strongly on level. Some experiments have found that hearing can extend down to 10 Hz at very high levels. Sometimes the term "subsonic" is wrongly used to mean infrasonic. Subsonic actually refers to the speed of sound propagation through a medium and has noting to do with frequency or pitch.
Intelligibility
A generally subjective assessment of how understandable something is. In music and acoustics this refers to how well we can discern specific information in a particular listening space. A space designed to have a high degree of intelligibility, such as a recording studio, will enable the listener to pick out subtleties in program material that will be masked in a less refined space, such as a gymnasium. Speech Intelligibility is a significant area of study in acoustics for obvious reasons. Here there are some well defined guidelines and objective criteria that can be used to design and rate not only the space itself, but a sound reinforcement system placed within it.
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).
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...
Isolation booth
Isolation rooms and smaller iso-booths are acoustically sealed areas built into and (hopefully) easily accessible from the main studio area and/or control room. These areas provide improved separation between loud and soft instruments such as guitars and vocals.
ITD
Short for Initial Time Delay. ITD is the gap in time between the arrival of direct sound and the first sound reflected from a surface of the room to the listener.
Impulse
An impulse is a signal or sound that has a very short (vanishingly short) duration. A true mathematical impulse has zero duration and infinite amplitude, but still a finite amount of energy.
Impulse response
In essence, the way a particular device responds to an impulse. For example, the reverberation of a room can also be thought of as its impulse response. A great deal of information about a device can be determined by how it reacts to an impulse. The frequency response, phase response, and transient response are all tied to this specification, though this specification itself is rarely seen on a spec sheet.
Kick Drum Tunnel
A makeshift device occasionally used by recording studios to capture a more ambient kick drum sound. It basically involves building a cavern or tunnel that extends from the front of a kick drum several feet out into the room. By placing a mic at the end of this tunnel, in addition to (or instead of) a normal mic inside the kick drum, you can capture a more roomy sounding kick, but without picking up the other drums and sounds in the room too much. Normally when the technique is employed the front head is removed from the drum, but this isn't necessary. A kick drum tunnel is simple to build. You take a couple of boom mic stands, extend their booms about two or three feet parallel to the ground and a few feet in front of the kick drum (this is your framework). Place your mic about a foot off the floor at the end of this structure and then cover the mic stands with blankets.
Lede
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.
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.
Localization
Localization refers to how we determine where in space a given sound originates. Less obviously, It also refers to the ways in which we perceive the environment in which sounds happen. Humans localize preconsciously, quite easily, and with considerable accuracy, though some of the various mechanisms we employ are either not fully understood or are in some dispute.
Loudness
Loudness is a subjective (even psychological) measurement associated with a given SPL and frequency or frequencies. A unit of loudness, called the phon, is equal to a difference in sound intensity of one decibel. The number of phons of a given sound is equal to the number of decibels of a 1kHz tone judged by the listener to be equally loud. Group judgements of whether two sine wave tones sound equally loud usually show fairly varied opinions among different individuals. Judgements on how much louder one tone is than another require previous conditioning or training, yet still yield results that can fluctuate greatly from individual to individual and from occasion to occasion. Tones of the same SPL but with different frequencies are generally judged as having different loudness. SPL is thus not always a good measure of loudness, if we attempt to compare tones of different frequencies.
Machine Room
A room dedicated for the housing of mechanical devices, normally for the purpose of isolating them from areas where humans work. This may be due to noise or heat, or other environmental considerations. As it applies to audio studios, this is the room where you might place tape machines, computers, decks and other devices that produce audible machine noise. By placing these devices in a space other than your recording and mixing environment, you are freeing your creative space from the noise that accompanies them - thus allowing focused recordings and mixing. You can also provide separate, and more suitable, ventilation for them without disturbing the main environment. Machine Rooms are found in forms such as expensively finished rooms in professional studios, bedroom closets in home studios and everything in between.
MDF
An acronym for Medium Density Fiberboard. MDF is an engineered wood product made from mechanically refined wood fibers combined with resin, which are bonded together under heat and pressure. The durable homogeneous construction of MDF resists warping, cracking and splitting - offering unparalleled design flexibility where intricate shaping and finishing are required. Some of the more common uses of MDF include furniture, cabinetry, millwork, store fixtures and laminate flooring, however it is used in construction of studio monitors, PA speakers and other forms of pro audio equipment.
Median Plane
The name sometimes given to an imaginary line equidistant from two speakers in a left/right studio monitoring setup. If you draw an imaginary line between the center of each speaker and then draw another line of the same length from the middle of that line, but perpendicular to it toward the listening position, the spot where that second line terminates is where your head should be for optimal listening. That second line defines the median plane, effectively separating the listening space into left and right halves. This is another way of saying the listening position should form an equilateral triangle between the listener and the two speakers. This is where you will get the best phantom image in the center as well as the best recreation of the overall soundstage.
Modality
See also Mode. The state of being modal. In acoustics this pertains to the modal properties of a space, where certain frequencies are accentuated or canceled by the resonances and sound reflections as based on its dimensions. In music, it pertains to the patterned arrangement of a scale.
Mode
Several definitions apply to us.
In electronic or mechanical devices, a mode is a state of operation where a defined set of actions can be accomplished. In an effects processor, for example, a mode may be the editing mode. In this mode you can get inside the parameters and alter the way the device processes sound. Many electronic devices have keys or buttons that can perform different operations depending upon which mode they are in.
In music theory modes are fixed patterns or arrangements of the diatonic tones of an octave. Also a patterned arrangement of a scale such as Major, Minor, Dorian, Locrian (sometimes jokingly referred to as the Devil's mode), etc.
In acoustics, a mode is one of the possible resonant frequencies of a vibrational system. The geometry of a room, for example, will cause it to have a number of modes. These result in standing waves at those frequencies, which will have the effect of reinforcing or attenuating some frequencies in that space. Modes can be predicted mathematically so one of the practices of acousticians is to construct spaces so room modes get distributed so as not to reinforce or attenuate any one section of the frequency spectrum.
NC Curve/Contour
NC stands for Noise Criterion and refers to the quiescent or ambient background noise present in an acoustic space such as an auditorium or room. Curve, or contour, refers to the way in which our ears are sensitive to noise, which essentially follows the guidelines outlined by the Fletcher-Munson Curves, or other similar studies. In a nutshell this means that the human auditory system is not equally sensitive to noise at all frequencies. Further, as the noise level changes these relative sensitivities change with respect to one another. NC curves were developed to take all this into consideration, thus providing a reasonably objective way in which to document and communicate ambient noise levels in rooms. This is important because quite often the majority of noise in most auditoriums is caused by the ventilation system, where most of the noise is at relatively low frequencies - frequencies at which human hearing is relatively insensitive. There are ratings given for various levels across the spectrum that take these curves into account. So a room with a certain amount of noise at 100 Hz will rate significantly better than a room with the same amount of noise at 1 kHz. Typical ratings range from NC-15 to NC-70. For example, a room said to meet the NC-15 requirement would be so quiet that the average listener would not perceive any background noise at all, yet there could be noise at 30 dB SPL below 80 Hz. An NC-20 room is noticeably noisier, but still considered very quiet, while a room rated above NC-25 or 30 is generally considered too noisy for critical listening.
Node
The opposite of a anti-node. When standing waves occur, there are positions in space relative to the wave, called nodes, at which there is no movement at all. The wave interferes with itself to create this instance of opposition (e.g., a wave reflecting off of a wall and back into its own path). Nodes are spaced one-half wavelength apart. On either side of a node is a vibrating antinode. The antinodes alternate in the direction of the wave's displacement so that the wave at any instant resembles a graph of a sine wave. On a guitar string the nodes are the places on the string that best produce harmonics - particularly at the 12th fret. Touching the antinodes damps the sound.
Also node refers to connection points along a cable. For example, a ribbon cable may have connectors at each end AND in the middle somewhere. These points are often referred to as nodes.
NRC
Abbreviation for Noise Reduction Coefficient. This is a specification often used to indicate the effectiveness of acoustic absorption materials. Generally it is arrived at by averaging the Sabine absorption coefficients of a material in the octave bands between 125 Hz to 4 kHz. The higher the number, the more sound is absorbed. The NRC is thus a sort of general result when compared to the detailed information that can be gleaned from good absorption coefficient data across different frequencies. For example, many common household materials such as curtains or carpet are very effective absorbers at 4 kHz, but don't do much at all to lower frequencies. NRC values don't include this detail.
Oblique Room Mode
Generally speaking, a room mode is essentially a "bump" or anomaly in a room's frequency response caused by the room's dimensions and the way those dimensions cause soundwaves to interact with each other resulting in resonances and/or cancellations at certain frequencies.
Oblique Room Modes involve all six surfaces - four walls, the ceiling and the floor. They are about one quarter as strong as the Axial modes, and half as strong as the tangential modes. Oblique modes are the ones that most often become overly excited.
Off Axis
Refers to an audio source that is not directly in front of a transducer, especially a microphone. This results in off-axis coloration; a distortion or change in the frequency response of the reproduced audio signal. Often this coloration is put to good use. For example, many engineers intentionally set up mics on guitar amps so that they are slightly off access to control the amount of high frequencies captured. A microphone will generally produce the "truest" results if it is used on-axis (oriented directly in front of the sound source).
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 polarpattern. 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.
Period
In any repeating phenomenon the time it takes for one repetition is known as the period. Repeating waveforms, such as sine waves, are called periodic waveforms for this reason. Example: A sine wave at a frequency of 100 Hz repeats 100 times per second. Its period is how many seconds it takes it to repeat, which in this case is .01 seconds, or one-one hundredth of a second.
Phase Cancellation
Phase describes where in its cycle a periodic waveform is at any given time. The relationship in time of two or more waveforms with the same or harmonically related periods gives us a measurement of their phase difference. Phase cancellation occurs when two signals of the same frequency are out of phase with each other resulting in a net reduction in the overall level of the combined signal. If two identical signals are 100% or 180 degrees out of phase they will completely cancel one another if combined. When similar complex signals (such as the left and right channel of a stereo music program) are combined phase cancellation will cause some frequencies to be cut, while others may end up boosted.
Phase and phase difference is a real-world issue in areas such as electrical wiring of audio equipment, signal path, and microphone placement during the recording process. Phase reversal can be a serious compromise of sound quality or a special effect affecting the perceived spaciousness of the sound depending on the context of its occurrence.
Phon
A subjective unit of apparent loudness. A phon is defined as the sound pressure level of a pure sine wave tone that sounds equal in loudness to some other sound(s) in question. Because of the ear's complicated response versus frequency characteristics (see Fletcher Munson Curves) you can't accurately relate the perceived loudness of sounds directly to their sound pressure level as read on a sound level meter. The phon allows us to relate this perceived loudness of sound to objective measurements of that loudness.
Pink Noise
Random noise with equal energy per octave. Our ears perceive this as sounding relatively "flat" in frequency response (since pink noise is based on octaves rather than individual frequencies, there is no increase in energy in the high octaves). Because of this, and because Real Time Analyzers (RTA) tend to look at octave or 1/3 octave ranges, pink noise is very useful for measuring the frequency response of audio equipment, as well as for determining room response for sound reinforcement applications.
Pinna Effect
The pinna is the flap of skin surrounding our ears. Reflected sound off the pinna combines with the direct sound into the ear to create high frequency comb-filtering effects (typically above 6kHz). These effects change as a function of angle of arrival, so that each angle of arrival has a unique sound quality. Our brain uses this quality as one of the ways to localize sound at each ear individually. The effect seems most persuasive in the vertical realm, so it is reasonable to hypothesize that we localize horizontally mostly by time difference while in the vertical axis the pinna effect is used more.
Point Source Monitor
A type of studio monitor or loudspeaker system in which sound only radiates from one location. One type of point source system would be one that has only one loudspeaker. In order to produce a high fidelity "full range" signal - one that can adequately cover the human range of hearing - more than one driver is generally required. These are normally positioned across the face of a loudspeaker system, which causes different parts of the frequency range to emanate from slightly different spots. An example of a point source monitor would be a coaxial design where the tweeter sits in the center of the woofer, or on top of the center of the woofer - the full range of sound all comes from one place. The advantage of a point source design can be minimal phase cancellation of common frequencies reproduced by both drivers due to an overlap of energy around the crossover point due to path length differences from the two (or more) devices to your ear.
Polar Pattern
Depending on their design and construction, microphones respond to sound coming from different directions with varying degrees of sensitivity. A plot or graph of this response is called a polar pattern (sometimes polar response curve). Looking at a mic's polar pattern will tell you how directional it is, how well it will reject sound from certain directions, etc. It is important to note that polar patterns are frequency dependent. Typically, low frequency response will be almost omnidirectional; the polar pattern will be come more directional as frequency rises.
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 Gain
D1 = Distance between microphone and loudspeaker
D2 = Distance between the loudspeaker and the furthest listener
D3 = Distance between the source and the furthest listener
D4 = Distance between the source and the microphone
NOM = Number of open microphones
There 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.
Precedence Effect
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.
Pre-Delay
Pre-delay is a parameter found in reverb processors. It refers to the amount of time between the original dry sound, and the audible onset of early reflections and reverb tail. Carefully adjusting the pre-delay parameter makes a huge difference in the "clarity" of a mix. For example, a longer pre-delay will move the reverb tail out of the way of the vocals, making them much more present and understandable.
Pressure Wave
This term is not as scientifically grounded as it is descriptive, but we do hear it used to describe sound propagation quite a bit. When a sound first occurs there is always an initial wavefront or pressure that is generated in the air. Changing air pressure is how sound is heard by the ear and also how sound is able to move through the air. There are waves of high and low pressure that correspond to the frequency(s) and volume of the sound. The phrase "pressure wave" is usually used to describe the "initial" high pressure zone created by the onset of some sound. For example: If a drummer hits a drum, the movement of the drum head when first struck creates an area of high pressure around the drum that then moves the surrounding air molecules, and so on until it reaches the ear. This is the initial pressure wave. It is followed by other waves of higher and lower pressure that correspond to the sound of the drum.
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.
Rarefaction
In sound waves this is the opposite of compression. An area of decreased air pressure caused by a sound wave. In a graphical depiction of a cyclical waveform rarefaction occurs when the wave is in the bottom segment (the anti-node).
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.
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.
Reverb
The remainder of sound that exists in a room after the source of the sound has stopped is called reverberation, sometimes mistakenly called echo (which is an entirely different sounding phenomenon). We've all heard it when doing something like clapping our hands (or bouncing a basketball) in a large enclosed space (like a gym). All rooms have some reverberation, even though we may not always notice it as such. The characteristics of the reverberation are a big part of the subjective quality of the sound of any room in which we are located.
Our brains learn to derive a great deal of information about our surroundings from the sound of a room and it's reverberation. Consequently it is necessary to have the proper type and amount of reverberation on recordings in order for them to be aesthetically pleasing or to sound natural to us. This can be accomplished with careful microphone placement, but it is often necessary to employ artificially created reverb.
To create reverb, a device known as a reverb unit is employed. Reverb units have historically come in many shapes and sizes, and have used many different techniques to create the reverberation. These days most of the reverb units employed throughout the world are digital, where the sound of the reverb is generated by a computer algorithm and mixed with the original signal. We will be discussing other types of reverb units in the future.
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.
RT60
An abbreviation for Reverb Time -60dB. It is an expression used to more specifically state what a given reverb (see WFTD archive Reverb) time is. The reverberation decay time in a large empty concert hall may be as much as 15 to 20 seconds. This means that in practice it takes that long for the reverberations of a sound to decay into the ambient noise of the hall (which includes thousands of ongoing reverberations). Raise the ambient noise level in the hall by 20 dB (perhaps by turning on a heating system) and the reverb time will sound shorter because as it decays it gets lost in the sound of the heating system. The purpose of the RT60 specification is to provide an objective measure of reverb time. The spec says that reverb time is defined as the time it takes the reverb to go down in volume by 60 dB, or to 1 millionth of the original volume.
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.
Soffit
In architecture a soffit is the underside of a structural component, such as a beam, arch, staircase, or cornice. In recording studios soffits are sometimes created along the front walls of a control room or studio space as an area to place large studio monitors. These are often recessed into the front wall of the room such that their baffles are flush with the wall surrounding them. They may be supported by being attached to the wall, flown from the ceiling joists above, or on stands attached directly to the foundation. In fact, some facilities pour separate foundations specifically for soffit mounted speakers to achieve maximum decoupling from the control room or studio. Generally soffit mounted speakers are tuned for the specific room with equalizers and other tools. In some cases rooms have been designed with specific speakers in mind beforehand. In either case the monitors used are not those designed for near field applications.
Soundstage
Typically known as a room or studio, usually soundproof, where audio production for film and video is done. It has also become a word occasionally used to describe the virtual acoustic space produced in a recording or sound playback system. When you listen to a CD of an orchestra and you can tangibly hear the acoustic space of the recording and where the instruments are in it, the created space or sonic signature is known as the sound stage.
Sound Pressure Level
The acoustic volume or loudness of sound, measured in decibels. SPL is a function of a signal's amplitude. Aside from the usual (and justified) warnings about hearing damage from high SPLs, it is worth noting that because of the way our ears function, sounds will appear to have a different timbre (or tone) to us at different SPL levels. This is important to keep in mind, especially when mixing in a studio environment. Be sure to check your mixes at a variety of volume levels to ensure that the mix is accurate. The old rule of thumb is that if a mix sounds good at a low SPL, it will sound great at higher levels...
Sound Transmission Loss
Sound Transmission Loss (STL) represents the amount of sound, in decibels (dB), that is isolated by a material or partition in a particular octave or 1/3 octave frequency band. Example: 1/2" drywall has an STL at 125 Hz of 15 dB.
Comparing material or partition performances for applications like recording studio isolation and sound proofing should involve comparing the STLs of each in the different bands, as opposed to just the more generic STC ratings of a material. If both materials or partitions are measured in accordance with the STL/STC standard, ASTM E90, then the comparisons being made will be "apples to apples." It should be noted that real-world performance is not going to provide the same level of STL that is achievable in the laboratory. However, the relative performance of one material or partition versus another typically holds true in real-world construction. i.e., if the lab measures one partition better than another, it should hold true for a real partition built in your studio. Even though an actual field test of a concrete wall might reveal a field STC (FSTC) that is 5 points lower than the lab test, it is still better - relatively speaking - than a simple, single-leaf, uninsulated drywall partition in the same configuration.
Spread
In mastering, the time given between each song is called the Spread. As much an art as the rest of the mastering process, often the spread between songs is not a set value of 2, 3 or 4 seconds. Many mastering engineers and producers prefer to use the spread to help define the overall feel or experience of a project by timing the spread to the tempo feel of the song that just ended and in some cases the song that's about to begin. So, if the tempo of a song is 120 bpm, the spread would correspond with that tempo so that the down beat of the next song is in tempo with 120 bpm from the end of the previous song. The number of beats used in the spread is a subjective decision based on the general flow of the album. This is simply one method for determining the spread, but the general idea is to use the spread to further amplify the intended feel or experience of the project and can result in varied spaces between different songs.
Spread is also sometimes referred to as the width of a stereo image. The "stereo spread" of a recording refers, in a general way, to how far left and right various sounds are panned. What is usually being described is the subjective assessment of how "wide" the recording is in terms of the perceived soundstage. Some signal processors employ a spread parameter that may use any of a variety of techniques to manipulate a signal so its width or stereo spread can be increased or decreased.
Standing Wave
A phenomenon where a sound is reflected back and forth between two parallel surfaces, such as two side walls in a room. Technically they are created by "room modes" or "eigentones," which are modes of vibration of air in the room. The sound waves interfere with one another to produce a series of places where the sound pressure level (SPL) at some frequencies is high, and another series of places where they are low. The places are sometimes called peaks and nodes. A standing wave exists in a room where a frequency is such that the distance between any two surfaces is equal to one half of its wavelength. For a given distance there will be many frequencies that will generate standing waves, each a multiple of the fundamental frequency whose wavelength is related to the dimension in question. Standing waves are always detrimental to the acoustics of a room, but can be avoided by careful room design, or minimized by absorbing the frequencies where they build up, which is usually along walls or in corners.
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.
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.
Temporary Threshold Shift
TTS for short, this is an upward shift in the threshold of human hearing. It is usually caused by being subjected to a loud sound. The human auditory mechanism is remarkably tolerant of abuse, and it has several ways of protecting itself from damage when exposed to very loud sounds. One of these is to reduce its sensitivity, causing the hearing threshold to shift upward. As the name implies it is a temporary condition, but continual exposure to loud sound will cause it to become permanent. This is known as permanent hearing loss. You only get one set of ears for life, so take care of them. Make it your New Year's resolution.
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.
Threshold of Feeling
A term in audiology to describe the levels and frequencies at which humans can perceive sounds through the nerve cells in the skin. This sensitivity decreases rapidly at frequencies above 1000 Hz. Deaf people are able to perceive music in this way and, for example, dance to the bass rhythm in the floor.
THX
Basically, THX is a set of standards to which components and systems can be designed to meet. Initially developed by Lucasfilm, with much input from George himself, THX was intended to define a minimum standard of equipment and quality control that theaters would need to meet in order to become 'certified' as THX approved. The idea was to raise the overall quality and consistency of audio in movie theaters. And Lucasfilm had the clout to make such a certification mean enough from a marketing point of view that theaters wanted to support it and be certified. This also spawned an explosion of THX approved mixing rooms and dubbing stages. Subsequently it has been adopted as a standard in hi-fi equipment with many manufacturers licensing and producing certified THX components for home theater systems.
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.
Tinnitus
Tinnitus (pronounced tin - I - tes) is the medical term for the perception of sound when no external sound is present. It is often referred to as "ringing in the ears," although some people hear hissing, roaring, whistling, chirping, or clicking. Tinnitus can be intermittent or constant-with single or multiple tones-and its perceived volume can range from subtle to shattering. The exact physiological cause or causes of tinnitus are not known. There are, however, several likely sources, all of which are known contributors to tinnitus: noise-induced hearing loss, wax build-up in the ear canal, certain medications, ear or sinus infections, jaw misalignment, cardiovascular disease, certain types of tumors, thyroid disorders, and head and neck trauma. Of these factors, exposure to loud noises is by far the most probable cause of tinnitus. Up to 90 percent of all tinnitus patients have some level of noise-induced hearing loss.
Transient
A non-repeating waveform, usually of much higher level than the surrounding sounds or average level. Good examples of transients include the attack of many percussion instruments, the "pluck" or attack part of a guitar note, consonants in human speech (i.e. "T"), and so on. Due to their higher-than-average level and fleeting nature, transients are difficult to record and reproduce, eating up precious headroom, and often resulting in overload distortion. Careful use of compression can help tame transients and raise average level, although over-compression will result in a dull, squashed, flat sound to the signal.
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.
U-Boat
Aside from being the name given to German submarines in WW-I and WW-II, U-Boats are a product manufactured and sold by Auralex corporation that are designed to provide acoustic isolation in room construction. They are configured as a U-shaped channel made out of a rubber compound. Framing materials rest in the channel, which sits on a solid surface (usually a concrete slab), thereby helping to decouple it and provide isolation from vibrations and low frequency acoustic energy from adjoining spaces. Combined with eliminating contact with other surfaces, a good amount of acoustic energy transfer from room to room can be prevented. When a room or space is constructed in this way, it is said to be "floated" or "floating," meaning there is no (or minimal) contact with the adjoining spaces.
Ultrasonic
Having frequencies above the range of human hearing, commonly considered to be 20 kHz. Not to be confused with "supersonic," which means faster than the speed of sound. Ultrasonic frequencies in the signal path can sometimes cause distortion in audio components that are in the human hearing range and quite audible.
Vocal Booth
Typically a part of a recording studio portioned off from the rest and isolated (to some degree) in terms of sound transmission. Vocal booths have typically been used to provide a space to record solo instruments (such as, but not limited to voice) without the recording being compromised by other elements in the studio. An obvious example would be recording a vocalist live while an ensemble is playing. But applications vary widely. Many times a vocal booth (which could more aptly be called an isolation booth) may be used for a guitar or bass cabinet when recording drums and other basic tracks for a song. In other applications a vocal booth may be in or adjacent to a control room and used to record voice overs for video or commercials. In this case there may not be any large studio space at all. In any case vocal booths are typically made out of materials that provide a good deal of sound isolation from outside and provide mostly sound absorption on the inside.
Wavefront
Technically a surface of constant phase in a wave's motion through some medium. It can be thought of as a line or surface within a two or three dimensional medium through which waves are passing, where the disturbances are all in phase. For light you can visualize this is light waves emanating from some light source and radiating outward. The front of that wave is one of the wavefronts. In audio the same principle applies.
Wavelength
Denoted by the Greek lower-case lambda symbol (l), the distance between one peak or crest of a sine wave and the next corresponding peak or crest. The wavelength of any frequency may be found by dividing the speed of sound (approximately 1100 feet (or 34 meters) per second at sea level) by the frequency. Thus the wavelength of a 60hz sine wave would be approximately 18.3 feet. Knowing wavelengths of sounds is very important when designing or working with acoustic spaces such as studios, control rooms, and speaker enclosures.
Weighting
Often times when laboratory measurements are taken of audio gear, the literal, "true" figures obtained do not reflect the anomalies introduced by human perception. In these cases, the specs obtained may be mathematically modified, or "weighted" to take into account the way our ears work. A good example is "A-weighting", a curve applied to sound pressure levels to more accurately reflect our loudness perception. Other types of weighting compensate for the ear's frequency response, etc.
Weighting Filter
A special filter used in measuring loudness levels, and consequently carried over into audio noise measurements of equipment. The filter design "weights" or gives more attention to certain frequency bands than others. The goal is to obtain measurements that correlate well with the subjective perception of noise. (Technically termed psophometric (pronounced "so-fo-metric") filters, after the psophometer, a device used to measure noise in telephone circuits, broadcast, and other audio communication equipment. A psophometer was a voltmeter with a set of weighting filters.) Weighting filters are a special type of band-limiting filters designed to correspond to the way we hear or some other specific criteria.
White Noise
Random noise with equal energy per frequency is called white noise. It tends to sound very bright and "hissy" due to our ears frequency response curve. (Each ascending octave contains twice as many frequencies as the next lower one, so there is a significant "build up" of energy in the higher octaves.)
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