5.1
A subset of the Dolby AC-3 sound playback standard (a.k.a. Dolby Digital), and the specific format sound data is in that corresponds to that standard. This is the current state of the art home theater surround sound technology. It means that there are five channels of information (left, center, right, left rear, right rear) and one active sub channel (the .1 channel). It is also a major standard that is becoming part of the DVD standard, which means there will be numerous releases in this format for the next few years.

AAC - Advanced Audio Coding (also MPEG-2 AAC)
An audio compression scheme that is a continuation of the MP3 CODEC, but uses better filtering methods, noise shaping and quantization resolution to produce higher-quality audio files at smaller bit rates. AAC is designed for use in digital broadcasting systems as well as for multi-channel and surround audio (such as 5.1), and based on its capability to support up to 96kHz sampling rates and 48 channels [including LFE], AAC could be the basis for audio in multimedia for the foreseeable future. Some streaming audio formats that have adopted the AAC scheme already are Liquid Audio (.lqt), AT&T's already fading .a2b format, and the soon-to-be-realized MP4 format.

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")

Accompaniment
In music this refers to additional instrumentation that surrounds or is played along with some feature such as a solo singer, solo instrument or a speaker. For example a singer could have a piano playing along as accompaniment. Similarly, a featured pianist could have a chorus of singers singing along as accompaniment.

Acoustic 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.

AFL
AFL (After Fade Listen) is used in mixing boards to override the normal monitoring path in order to monitor a specific signal at a predefined point in the mixer. Unlike PFL (see WFTD archive "Pre-Fade Listen"), the AFL signal by definition is taken after the fader of a channel or group buss such that the level of the fader will affect the level heard in the AFL monitor circuit. AFL is sometimes also taken after the pan pot which also allows the engineer to monitor the signal with the pan position as it is in the mix. AFL is a handy way to monitor a small group of related instruments by themselves with all of their eq, level, and pan information reproduced as it is in the overall mix. An AFL circuit that includes pan information is often called "solo" (see WFTD archive "solo") or "solo in place" depending upon who builds the mixer.

Ambisonics
A British-developed surround sound system designed to reproduce a true three-dimensional sound field. Based on the late Michael Gerzon's (1945-1996; Oxford University) famous theoretical foundations, Ambisonics delivers what the ill-fated quadraphonics of the '70s promised but couldn't accomplish. Requiring two or more transmission channels (encoded inputs) and four or more decoded output loudspeakers, it's not a simple system; nor is the problem of reproducing 3-dimensional sound. Yet with only an encoded stereo input pair and just four decoded reproducing channels, Ambisonics accurately reproduces a complete 360-degree horizontal sound field around the listener. With the addition of more input channels and more reproducing loudspeakers, it can develop a true spherical listening shell. As good as some think it is, a mass market for Ambisonics has never developed due to several factors. First, the actual recording requires a special tetrahedron array of four microphones: three to measure left-right, front-back and up-down sound pressure levels, while the fourth measures the overall pressure level. All these microphones must occupy the same point in space as much as possible. So far, only one manufacturer (first Calrec, bought by AMS, bought by Siemens, sold, now Soundfield Research) is known to make such an array. Next, a professional Ambisonics encoding unit is required to matrix these four mic signals together to form two or more channels before mastering or broadcast begins. And finally, the consumer must have an Ambisonics decoder, in addition to at least four channels of playback equipment.

Aux Send
Slang for Auxiliary Send, a circuit pathway (or bus) in a mixing console that supplies an independent mix, which can be routed to an external (auxiliary) device such as an effects processor or monitor system. Most modern consoles have several aux sends on each channel so several devices can process the input to any channel or groups of channels.

Bass Management
A circuit or process that takes all the frequencies below 80Hz (according to the Dolby spec) from the main channels in a surround or stereo mix and the LFE signal and mixes them together into the subwoofer. In other words, Bass Management is the act of placing an electronic bass frequency crossover on all the channels, and redirecting those bass frequencies. Stereo or surround rooms, especially with smaller near field monitors placed on the console, can benefit from the correct integration of Bass Management and a subwoofer. With such, the engineer is now able to hear low frequency anomalies caused by room rumble, microphone stand thumping, breath pops, and other undesirable artifacts. Plus, even the least expensive Dolby Digital consumer decoder, found in millions of homes, has bass management built in, allowing the bass from all channels to be fed to a single subwoofer - which means that control rooms with proper Bass Management will be able to make sure that their mixes translate well into consumer systems.

Bi-Amp
When a single audio signal is dived 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.

Binaural
A system of recording with a plastic replica of the human head, with microphones placed in the ears, replicating as near as possible human hearing functions regarding phase, directionality etc. This signal information is absent from ordinary microphone pickups. Signals from the two mikes placed in each ear of the dummy head are kept entirely separate all the way to the two drivers of the final listener's stereo headphones. The result is a convincing preservation of the 360° soundfield and localization abilities present where the dummy head was placed.

BIOS
An acronym for Basic Input/Output System. Mostly germane to PC compatible computers, this is usually an EPROM with computer program instructions in it. A computer motherboard BIOS controls how the hardware is defined and the basic functions of the computer (such as controlling the keyboard, monitor, etc.). With a SCSI host adapter, its BIOS is used to control SCSI hard disk drives and perform the boot function. If a host adapter does not have a BIOS, then hard disk drives controlled by that host adapter cannot be used to boot from (booting must be done from another source, such as floppy, IDE, or another SCSI host adapter with a BIOS). Hard drives can have their own BIOS as well, which defines their operation. The BIOS can also contain useful software utilities, and in some cases, can be reprogrammed or updated via software to accommodate new hardware. Older PC computers often have to have their BIOS updated in order to properly work with new hardware.

Bleed
In audio, bleed is the leakage of one audio source's output into another audio source's input. This can happen onstage, such as a drum or cymbal's sound bleeding into a guitar amp mic, or in the studio, such as the output from a singer's headphones leaking into the vocal mic. Some solutions to reduce bleed include: mic selection and placement - using a cardioid or supercardioid mic on a source to reject signals from other directions; use of noise gates to attenuate mic sensitivity so they don't pick up outside noise; and optimizing the gain stage of your mixer and peripherals to achieve an ideal signal-to-noise level.

Bluetooth
A short-range wireless technology that communicates via a frequency-hopping transceiver over the 2.4-gigahertz radio frequency, a space known as the Industrial, Scientific and Medical (ISM) band. Bluetooth was originally conceived as a low cost, low power, short-range technology that would replace cables on such devices as mobile phone headsets, handsets and portable computers. However, its promoters soon envisioned the creation of "personal area networks" in which computers could be wirelessly connected to printers, audio could be transmitted over short distances (for example, to the rear speakers in surround setups), and remote control of PDAs or other appliances could be easily implemented. Some people have referred to it as a sort of wireless USB, which is a pretty apt description in many respects. First conceived in 1994 by Ericsson Mobile Communications (now a part of Sony), by 1998 the Bluetooth Special Interest Group included industry giants Intel, IBM, Toshiba and Nokia. Today more than 2000 companies produce or are developing Bluetooth enabled products. Apple Computers incorporate Bluetooth compatibility that allows keyboards, mice and other peripherals to wirelessly connect to the main unit. While Bluetooth originally had a transmission range of only 10 meters, today, three power classes exist for Bluetooth devices, the most powerful allowing transmissions up to 100 meters. Bluetooth is a different protocol from Wi-Fi, but both occupy a section of the 2.4 GHz ISM band that is 83 MHz wide. Bluetooth uses a technology called Frequency Hopping Spread Spectrum (FHSS) that allows it to hop between 79 different 1 MHz-wide channels in this band whenever it encounters interference from other transmissions.

Body Pack
In the world of wireless performance a body pack is the device a performer wears somewhere on his or her body that houses the electronics that handle sending a signal to a remote receiver or, as in the case of personal monitoring systems, receives a signal from a remote location. Typically body packs hold a battery and some combination of electronics that do the transmitting or receiving, and amplifying. Some wireless systems do not require a body pack as all of these electronics can be housed right inside of a microphone or a small plug that can be connected directly to a guitar or other musical instrument.

Bus
In audio (not transportation) terms, a Bus is a point in a circuit where many signals are brought together. For example: Most electronic items have a Ground Bus where all of a device's individual ground paths are tied together. In mixers, we have Mix Busses, where multiple channels' signals are brought (or blended) together; Aux Busses, where feeds from channels are brought together to be routed to an external processor or monitor send, etc. In general, the more busses a mixer has, the more flexible the routing capabilities of that mixer will be.

Cans
"Cans" is recording studio slang for headphones. In the recording community, as it is in all fields that use equipment with long technical names, certain commonly used items find their names shortened for convenience sake, or coolness factor. Headphones, which are mainly used for monitoring during recording, became shortened to phones, and later to "cans", which is a reference to the old trick of running a string through two cans (like coffee cans) to make a crude telephone. Recording studios are famous for technical jargon turned slang. (See most of our Words For The Day)

Circumaural
Used in reference to headphones. "Around the ear". Circumaural headphones encircle the ear, and provide a good seal. Typically, circumaural phones use a "closed" design, and provide good audio isolation.

Click Track
A metronomic "pulse" heard in monitor headsets by the musicians (or conductor in film scoring) during the performance of music. The purpose of a click track is the same as any metronome: to guide the musicians temporally for the sake of timing consistency or some other timing concern. In film scoring this would be to have hits and other cues occur at the proper time in the film. Traditionally click tracks have been recorded to tape (hence the usage of the word "track" in the name), but in modern production this is increasingly rare. Click tracks are quite often generated by computer software (such as MIDI sequencers) and played back in real time through some MIDI sound source. However, in many instances for the sake of convenience, and as a fail-safe method they may also be recorded to the multi-track being used.

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.

Component Video
A video signal where some or all of the individual components that make the signal are sent down separate wires (as opposed to composite video), either in the form of a multi-pin D-Sub type cable or a five way cable terminating to five BNC connectors (there are other types, but these two cover the majority of it). For example, in a computer monitor you may find that the three primary video colors (Red, Green, and Blue) are each sent separately, and luminance (brightness) information and video sync are separate from that, hence the five wires (it can even be separate further into horizontal and vertical video sync). In some applications "component" signals are still composite signals of another kind. Formats such as the 4-pin S-Video, the 2-RCA luma/chroma standard, or the 3-BNC YUV standard will have some combining of information, such as the sync signal(s).Regardless of the kind of cable used, modern analog computer displays have separate signal and ground wires for at least the red, green, blue, HSync and VSync signals. This separation allows the cables to carry much higher frequencies than would be possible if they were entirely or partially composited with each other. These higher frequencies allow for the high resolutions that computer displays must support. For comparison, a computer outputting a 640 x 480 resolution image with a 60 Hz interlaced refresh rate (similar to broadcast TV) has a "dot-clock" frequency of approximately 12 MHz. (Dot-clock represents the timing between adjacent screen pixels and is the highest frequency component of any computer's display-generation circuitry.) At 800 x 600 resolution (also 60 Hz interlaced), that dot-clock frequency increases to approximately 35 MHz. A modern workstation's display using 1600 x 1200 resolution at 85 Hz non-interlaced requires a dot-clock frequency of at least 220MHz.(Special thanks to inSync reader David Charlap for some of the computer specific information presented here.)

Confidence Monitoring
Listening directly from a recording medium while recording to ensure the program material is being recorded correctly. Many analog recorders have a playback head trailing the record head, allowing you to hear the material directly after it has been recorded. Professional DAT recorders usually have four heads for confidence monitoring (it takes two to record and two more to play back at the same time), as do a number of the modular digital multitrack (MDM) recorders. Most DAW systems do not provide for confidence monitoring, but do allow you to monitor the signal just prior to being recorded to the hard drive. Confidence Monitoring is a distinction that was considered much more important in the analog days, or early digital days. Nowadays, with most digital systems, we can pretty much depend on getting the signal we hear recorded in tact.

Constant Directivity
A horn provides more sound pressure level (SPL) at a given listening area by increasing the directivity of the sound towards the listener. There is more sound at the listening area, and less sound outside of that area. By analogy, think of focusing a beam of light (from a flashlight or torch). A widely focused beam spreads the light around, reducing the intensity at any one point. However, a narrowly focused beam provides much more light intensity at the center, and much less in the surrounding area. Properly designed horns can also act as a waveguide that actually serves to spread higher frequency sounds out in a much more consistent manner than would otherwise happen. Round horns and radial horns tend to change their angles of spread (their directivity, measured by the directivity index, or DI) as the frequency changes. This means that high frequencies, for instance, might be more highly directed, and therefore sound louder to someone in a central location than to someone else outside of the center (but still within the horn's low-frequency area of enhancement). To cope with this problem, the constant directivity (CD) horn was invented. The design goal of the CD horn is to provide the same SPL at all frequencies within the designed coverage angles. The term "Constant Directivity" is a trademark of ElectroVoice but has become somewhat of a catchall phrase to describe constant-beamwidth horns. In 1975, Electro-Voice introduced a single-cell horn that consisted of three-stages. The design incorporates a hybridized hyperbolic/exponential throat section coupled to a conical, vertically flared, radial bell section. Flanges that correct for midrange beaming caused by edge diffraction are comprised of a second, wider conical, vertically flared, radial bell-section. As with classic radial horn designs, the sidewalls are straight, but in two flange sections. Having constant beamwidth in both the vertical and horizontal directions, and an unprecedented high directivity index, these horns became the model for virtually all-new horn designs for the next decade. Additionally, they horn loaded the driver well, and as a result sounded very good.

Contact Pressure (Headphones)
The wearing comfort of a set of headphones is determined not only by its weight but also by the force with which the earpieces are pressed onto the ears. This force is given in newtons (N), whereby 1 N corresponds to the compressive force that a mass of about 100 g exerts on a solid surface due to the effect of gravity. The DIN Standard 45500 Part 10 limits the maximum permissible contact force to 5 N. Values of between 1.3 and 4 N are common, although lower values apply for open headphones. Higher values can be found in the case of closed headphones. Here, a higher contact pressure is required in order to achieve sufficient sealing, which is important for the reproduction of low frequencies.

Control Panel
Basically, this is just what it sounds like: a panel to control something. The usage of the term gets confusing to people in how it is applied to computers, but it's pretty simple. In computing devices, a control panel is a software program designed to give the user control over some specific part of the operation of the machine. This could be a basic function like monitor resolution, or more involved functions relating to standard and optional hardware or software that may be installed on a particular system.

D'Appolito
A loudspeaker configuration developed by and named for Joe D'Appolito, in which a high frequency driver, or tweeter, is positioned between two midrange or low frequency drivers that each cover the same frequency range. Depending on the exact implementation the speakers can be positioned with a vertical and/or horizontal orientation. In either case the two midrange drivers serve a couple of purposes: they combine to create a larger effective woofer or midrange driver size, and they also serve to control the dispersion of the tweeter. The tweeter's output is somewhat corralled or contained by the sound coming from the midrange drivers in a similar way to how two parallel surfaces control dispersion. There are some variations on the design where two same sized woofer/midrange drivers may cover slightly different frequency ranges, however those aren't considered true D'Appolito designs. (Input from an inSync Reader:) The D'Appolito design specifies a third order crossover network.  The tweeter is coordinated with the woofer so that at the selected crossover frequency, the drivers all have similar horizontal dispersion.  (This is not easily accomplished because many drivers behave badly at the extremes of their range.) The advantage of doing it all correctly is one of the most seamless blending of drivers possible.  The result is an absence of any sudden change in directivity with frequency.  This may not mean much for monitors where there is a limited listening area, but in a typical room where a large percentage of the sound is reflected by the room, the effect is dramatic.

Decca Tree
A stereo miking technique. A Decca Tree configuration is characterized by having three omnidirectional microphones in a "T" shaped setup. Two of the microphones are positioned about two meters apart. The third microphone is positioned between the first two, but about 1.5 meters forward (closer to the source) of them. This configuration is sometimes used for orchestral recordings and film scoring due to its natural sound with good separation. It is useful in film because the image doesn't usually cause problems with Dolby or other surround processes. In many cases the Neumann M50 (or now, the newer TLM50) is used as the center microphone because of its unique directional characteristics and smooth sound.

Decorrelation
A process in which an audio source signal is transformed into multiple output signals with waveforms that are different from each other but which sound similar. Distributing these signals to different loudspeakers - such as a surround system - creates the sensation of space and helps listeners identify the location of sounds. In nature decorrelation is a product of the delay, reverberation and filtering properties of any room or space. In the studio, effects processors -reverb, delay, chorus, flanger, comb filter, etc. - produce decorrelated output. There is a long history of home and studio devices that "stereoize" monophonic signals, and they typically worked by decorrelating the output channels. Vocalists have often been recorded twice on separate tracks so that the small variations in the two performances create decorrelation. The term was originally applied to a technique used in home THX systems to create a more diffuse, full surround sound environment by splitting up the mono surround channel of matrixed surround sound systems (the analog Dolby Surround or Dolby Pro Logic) into two channels and feeding them alternate information. This process, combined with the THX specification of dipole speakers, ensures that the mono surround signal is not localized or located at a specific speaker. Discrete digital surround sound formats (Dolby Digital and DTS) feature stereo, discrete, full-range surround sound channels that don't require decorrelation.

Dipole
In physics, a pair of equal and opposite electric charges or magnetic poles that are separated by a small distance. This term has been adapted to cover audio and video concepts in two different ways. In audio a dipole loudspeaker contains two drivers, usually directed 180 degrees in opposition to each other and wired in opposite phase to each other. Dipole loudspeakers are often found in home theater surround systems where they serve as rear (and sometimes side) satellites. Their donut shaped dispersion pattern can be effective for enhancing the sensation of envelopment that is an important part of the surround experience. In radio and television, a dipole antenna is an aerial half a wavelength long consisting of two rods connected to a transmission line at the center. The most common example of this is the "rabbit ears" antenna that is often used to pick up local television broadcasts. Many wireless monitor and assistive listening system transmitters use dipole antennas.

Discrete Surround
There are two primary ways to deliver a surround sound signal. The older analog matrix technology works by mixing (matrixing) multiple channels into two main channels; the matrixed information then has to be decoded into the original channel configuration. The newer digital discrete technology keeps all the channels separate from start to finish. Discrete surround sound is superior to matrix surround. Surround effects are clearer and more distinctly positioned. The most common matrix surround sound formats are Dolby Surround (left, right, and surround channels); Dolby Pro Logic (left, center, right, and surround); and the newer Dolby Pro Logic II and Dolby Pro Logic IIx, which are designed to simulate surround sound from two-channel sources. The most popular discrete surround sound format is Dolby Digital, which can reproduce up to 5.1 channels: left front, center front, right front, left surround, and right surround, plus a separate low frequency effects (LFE) channel. Dolby Digital can also reproduce mono (1.0) and stereo (2.0) soundtracks. DTS is a competing 5.1 format that offers slightly better sound due to higher data sampling rates. Dolby Digital EX adds one or two rear channels behind the normal surround channels, for 6.1- and 7.1-channel configurations. Matrix technology is used for the rear channels. DTS ES is a similar 6.1/7.1-channel format, using matrix technology for the new rear channel(s). DTS ES Discrete is different in that it adds a single rear channel on a discrete track. DTS offers slightly wider dynamic range than Dolby Digital, although this technology is used less frequently on home DVDs.

Divergence
A term used in surround sound mixing. When left, right, and center channels are available, a sound can be placed in front of the listener by mixing it entirely to the center channel, or by splitting it equally between the left and right channels. Compared to sending the track directly to the center channel, mixing the track to the left and right channels creates the impression of an extended sound source. Whether a narrow or wide source is desired depends on the situation, and many surround panners provide a so-called divergence control, which adjusts the left/center/right panning parameters to control the portion of front-placed sounds mixed to the center channel.

Dolby
Short for Dolby Laboratories, this word is often used to generically refer to noise reduction, for which they are famous. Dolby Labs was founded in 1965 when they developed their first noise reduction systems. Since then they have been a huge contributor to the betterment of recorded sound with many advancements in technology. Today they are known for noise reduction systems, stereo and surround sound encoding technologies, recording headroom extension, and many developments for theatre film sound reproduction and digital media sound reproduction. We will cover more of the specific Dolby formats in the future.

Dolby E
A multichannel coding system developed by Dolby where up to eight channels of high-quality audio plus Dolby Digital metadata can be distributed via an AES3 pair, or recorded onto two audio tracks of a digital VTR (Video Tape Recorder). Dolby E is primarily for use within the broadcast and post-production infrastructure. Audio never reaches the consumer in Dolby E form; it is encoded with Dolby Digital just prior to final transmission. To help differentiate their functions, Dolby E is referred to as a distribution coding system, and Dolby Digital as an emission coding system. Dolby E encodes up to eight audio channels plus metadata into a two-channel bitstream with a standard data rate of 1.92 Mbits/sec (20-bit audio at 48 kHz). With multichannel programming, a "5.1+2" configuration is typically used, with six of the eight channels carrying a 5.1 mix and the other two an Lt/Rt (matrix surround-encoded) or stereo two-channel mix. The system can also be used to carry a 5.1 mix plus two mono tracks (5.1+1+1), three stereo mixes (3x2), six mono channels (6x1), and so on.

Dolby Pro Logic
Dolby's second generation licensed home surround system. A major advantage of Dolby Pro Logic over the preceding system (Dolby Surround) is the use of an active center channel with its own speaker. Conventional stereo systems create a phantom center channel, which is effective for viewers seated directly in front of the television screen. However, for viewers seated off center, the dialog can appear to come from off center. But with Dolby Pro Logic and the use of an appropriately placed center channel loudspeaker, the dialog always appears to come right from the screen, allowing the main left and right stereo speakers to be widely spaced for a good spread on music and effects. Dolby Pro Logic decoders also decode surround information which is typically fed to a pair of surround speakers slightly behind and to the left and right of the listener.

Dolby Stereo
After introducing the use of Dolby A-type noise reduction to the film industry, Dolby's next major contribution was Dolby Stereo. This contribution allowed movie makers to put 4 channels (hmm... most people call that quad) of sound information on motion picture release prints using matrix technology, and gave theaters the ability to replay this 4-channel format for the movie going public. The four channels were actually left, right, center, and surround. This was the precursor to (and actually the theater version of) what became known as Dolby Surround.

Dolby Surround
Dolby Surround is an early home embodiment of Dolby Stereo. Video production companies are licensed to make VHS tapes and laserdiscs that contain the same 4-channel matrix encoded information that was contained on the original motion picture release. Consumer electronic companies are licensed to make consumer surround decoders that reproduce these 4 channels in the home. It has a passively derived center channel and is not to be confused with Pro Logic, which we will cover soon.

Dolby Virtual Speaker
An algorithm created by Dolby that attempts to reproduce the dynamics and surround-sound effects of a precisely placed 5.1-channel speaker system from a consumer electronics device or personal computer equipped with as few as two speakers. The algorithm at the heart of Dolby Virtual Speaker technology is based on psychoacoustic parameters that include an understanding of sound from both a technical and an experiential perspective. Dolby Virtual Speaker technology uses biological, psychological, and physical understanding to create the "impression" of additional speakers positioned exactly at the recommended locations for a Dolby Digital sound system with five actual speakers. In other words, audio channels are processed through filters that simulate the sonic signature of a speaker located within an acoustic space. Dolby Virtual Speaker technology was launched in fall 2002 to the PC industry, and is currently available on select software DVD players from CyberLink, InterVideo, and Nvidia, as well as models from leading PC OEMs (including Sharp, NEC, Sony, Fujitsu, and Hitachi).

Dual Concentric
A term used to characterize certain loudspeakers. The word concentric indicates a common center. Loudspeakers where the woofer and tweeter share a common center point are known as dual concentric (sometimes called coaxial, though this is not as specifically precise). Dual concentric speakers have the advantage of all sound emanating from one point (they are also called "

DVD
Latest info says "DVD" no longer stands for anything! It used to mean "digital versatile disc" - and before that it meant "digital video disc." A new type of 12-centimeter (4.72") compact disc (same size as audio CDs and CD-ROMs) that holds 10 times the information. Capable of holding full-length movies and a video game based on the movie, or a movie and its soundtrack, or two versions of the same movie - all in sophisticated discrete digital audio surround sound. The DVD standard specifies a laminated single-sided, single-layer disc holding 4.7 gigabytes, and 133 minutes of MPEG-2 compressed video and audio. It is backwards compatible, and expandable to two-layers holding 8.5 gigabytes. Ultimately two discs could be bounded together yielding two-sides, each with two-layers, for a total of 17 gigabytes. There are three versions: DVD-Video (movies), DVD-Audio (music-only) and DVD-ROM (games and computer use). The DVD-Audio standard is still being defined. Meanwhile a fourth member has joined the family: DVD-RAM defines specs for a rewritable system, opening the door for recording.

DVI
Abbreviation for Digital Visual Interface. DVI is a specification created by the Digital Display Working Group (DDWG) to convert analog video signals into digital signals and to accommodate both analog and digital monitors on a single connector where appropriate. There are three different DVI configurations. DVI-A, designed for analog signals, DVI-D, designed for digital signals, and DVI-I the mutually integrated specification, designed for both analog and digital signals. This connection is not compatible with the old VGA style connectors. It is an entirely different connector that is beginning to show up on display cards and monitors as of 2001. The original signal is pure digital but when transmitted to a DVI interface through the appropriate port, the digital signal is converted to analog if asked to do so by the program or by the monitor.DVI Also stands for Digital Video Interactive, a now rarely used compression/decompression technique developed by RCA, Intel, and GTE that made it possible to store digital graphics, audio, and full-motion video on a CD-ROM, and to decompress and display these forms of data singly or in combination. Due to the vast amount of data throughput required it was difficult for early systems to handle full motion video on the fly. DVI overcame these problems with a hardware CODEC using specialized processors to compress and decompress the data. A competing hardware and software CODEC, known as MPEG, has become much more widely used in recent years. Intel has developed a software version of the DVI algorithms, which it markets under the name Indeo.

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.

Envelopment
A term used to describe the degree to which an audio signal is perceived as being all around the listener. The term "envelop" literally means to enclose or cover completely. In audio production envelopment has been adopted to characterize a property of surround sound mixes. For example, a 5.1 encoded DVD video or DVD-Audio of a live concert is likely to incorporate more than the sound of the artist in front of a listener. It would also include the sound of the audience and additional room ambience beside and behind the listener, and in some cases the listener is placed on stage with the artist(s) with instruments coming from all sides. Envelopment is a result of panning and routing signals to multiple speakers in a surround system. In a sense, the "opposite" of envelopment is localization.

ESS
Early Sound Scattering (ESS) is a design for control rooms where the characteristic reflections are so random that they have no character to impose on the listening space. An ESS control room is one that features a highly diffusive front end (including the monitor walls), which scatters the early reflections using Schroeder-type diffusers. The body of the room is absorbent, with most of the low frequencies damped by membrane panels. These rooms can be made fairly "live" compared to older, acoustically "dead" or absorbent control rooms, with a flat frequency response and good stereo imaging, both of which remain stable all the way to the rear corners of the room.

Extended Surround
Star Wars: Episode I was the first of a number of films using an additional rear channel routed to the array of speakers along the back wall of a cinema. In the cinemas, this back channel is not a discrete channel, but is matrixed into the left and right surround channels, much as the center front channel was matrixed into the left and right front channels in earlier matrix optical surround formats. This matrixed back channel is embedded in the soundtrack printmaster, so finds its way into all cinema digital sound formats. DTS uses the name "ES" on its cinema decoder while Dolby calls the same process "Surround EX". Either set of letters stands for Extended Surround.

Fold-Down
The process of taking a certain number of audio channels and restructuring them in fewer channels. The fold-down process is most commonly used to reproduce surround sound mixes in a stereo format. When starting with a 5.1 mix the process involves integrating signals that were originally sent to the subwoofer and taking the center channel signal and creating a "phantom center." In addition, any audio sent to rear channels must be integrated into the stereo mix. This is often the tricky part, as signals that have been decorrelated to produce a front-to-rear sense of space must be carefully handled to avoid phase cancellation in the stereo mix. This process is also sometimes referred to as downmixing.

Fool

1. One who is regarded as deficient in judgment, sense, or understanding.
2. Someone who thinks he can match wits with Sweetwater's inSync editor.
3. A member of a royal or noble household who provided entertainment, as with jokes or antics; a jester.
4. A dessert made of stewed or puréed fruit mixed with cream or custard and served cold.
5. Spud Webb taking a charge from Shaquille O'Neal.
6. One who has been tricked or made to appear ridiculous; a dupe.
7. Taking the gig mixing monitors for a 100 piece bottle blowing band.
8. One who knowingly drives a Crown PSA-2 amplifier to full output into a 20 watt 100 ohm resistive load to "see what will happen."

Free Lossless Audio Codec (FLAC)
A compression/decompression program that is specially designed to store and play back music files. FLAC can reduce the size of a typical audio file by up to 50%. It supports streaming audio for playback of very large files, and also supports 24-bit audio. The key word in FLAC is "lossless," which means that audio is encoded in FLAC without losing any of the frequency response, dynamic range or timbre of the original signal. This is in contrast to common audio compression schemes such as MP3 or AAC, which discard, or "lose," portions of the digital audio information in order to create the smallest possible file. FLAC is often chosen over several other lossless codecs, largely because it is offered as open source code, licensed without charge to any software or hardware developer that wishes to employ it. FLAC also supports Windows and Mac operating systems as well as others: UNIX, Linux, BeOS, OS/2 and Amiga. A FLAC encoder has the following stages: Blocking: The input is broken up into many contiguous blocks. In FLAC, a block is one or more audio samples that span several channels, and may vary in size. The optimal size of the block is usually affected by many factors, including the sample rate, spectral characteristics over time, etc. Though FLAC allows the block size to vary within a stream, the reference encoder uses a fixed block size. Interchannel Decorrelation: This step combines similar audio data found on multiple audio channels. In stereo audio streams, the encoder will create mid and side signals based on the average and difference (respectively) of the left and right channels. The encoder will then pass the best form of the signal to the next stage. Prediction: The encoder tries to find a mathematical description (usually an approximate one) of each block in the signal. This description is typically much smaller than the raw signal itself. Since both the encoder and decoder know the methods of prediction, only the parameters of the predictor need be included in the compressed stream. FLAC currently uses four different predictor types, depending on the nature of the signal currently being encoded (from silence to 6 channels of surround audio at full dynamic range) and allows these to change from block to block, or even within a block, as needed. Residual coding: If the predictor does not describe the signal exactly, the difference between the original signal and the predicted signal (called the error or residual signal) is encoded. If the predictor is effective, the residual signal will require fewer bits per sample than the original signal. FLAC was designed to be "decoder friendly;" in other words, it plays back (decodes) files almost instantly, while encoding a file takes a little more time. Still, the encoding time is faster than the real time playback of the original file. A number of software music players and hardware devices support FLAC. This list constantly grows and changes, so your best source of information about FLAC-compatible hardware and software is the developer's website: http://flac.sourceforge.net/index.html. Musical groups such as Metallica, Phish and Primus offer FLAC versions of their live performances and album material on their official websites.

Frequency Response / Frequency Range
From inSync reader Kevin T. comes the following question (which qualifies as both a WFTD and a TTOTD): What is the difference between frequency response and frequency range as it pertains to studio reference monitors?Kevin, first of all, thanks for the question! According to the Unabridged inSync Master Dictionary (which we make up as we go...): Frequency Range is the actual span of frequencies that a monitor can reproduce, say from 5 Hz to 22 kHz.Frequency Response is the Frequency Range versus Amplitude. In other words, at 20 Hz, a certain input signal level may produce 100 dB of output. At 1 kHz, that same input level may produce 102 dB of output. At 10 kHz, 95 dB, and so on. A graph of all the frequencies plotted versus level is the Frequency Response Curve (FRC) of the monitor.When you see a Frequency Response specification for a monitor, the manufacturer is telling you that for a given input signal, the listed range of frequencies will produce output within a certain range of levels. For example: 20 Hz to 20 kHz ± 3 dB. For these frequencies, the monitor will output signals that are within a 6 dB (± 3 dB) range. This does not mean that the speaker won't reproduce frequencies outside this range, it will! But frequencies outside the range will be more than 3 dB off from the reference level. For further information, see also May 5th's inSync Word For The Day, "Flat Response", available in the inSync Archives.

Front Loaded
A speaker cabinet design characterized by the speaker being mounted to the front of the cabinet or baffle. This configuration is very popular with studio monitors, guitar and bass cabinets, stereo speakers, and many types of PA enclosures. The other prominent design used over the years is horn loading, which is where the speaker is set back in the throat of some type of horn-like configuration. Each has its own strengths and weaknesses and is appropriate in different circumstances.

Gain/Fader Riding
Also known as "Riding the Faders," gain riding is the act of constantly monitoring and adjusting gain as necessary during the recording process to prevent overloading the recorder. This is usually performed on the faders of a recording console. Gain riding in essence duplicates the action of a compressor without adding an extra device in the signal path. However, since the sound of compression has become the sound of popular music, gain riding has all but fallen out of use for that purpose. Gain riding is, however, still used as a creative tool. In fact, it is in part responsible for the signature drum sound of the Power Station recording studio, which became popular during the '80s. Power Station engineers would ride the faders on snare tracks so that the hit on beat two would be louder than the hit on beat four. By emphasizing the natural pulse of 4/4 tempo, wherein beat two is naturally played louder than beat four, musical drive is created. Ironically, this is a compensation for the loss of dynamics introduced by compression.

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.

Headphones
A headphone is an electromagnetic transducer designed to be worn on the human head for the purpose of audio listening/monitoring, and as distinct from an earphone, or system worn in the ear. Headphones (plural) would be a pair - one for each ear. They are usually based on the principle of electromagnetic induction used to convert the electrical energy output of a headphone amplifier into acoustic energy, or sound. There are two main distinctions regarding headphone design: open-back and closed-back, generally referred to as open and closed headphones. With closed headphones, the ear is completely sealed off from outside noise (pressure chamber principle). Typical features of closed headphones are the acoustically sealed housing and the ring-shaped (circumaural) pads that completely surround the ear. The sealing around the ear has a decisive influence on the sound reproduction of closed headphones. If it is insufficient, the quality of the bass sounds will deteriorate. For this reason, the contact pressure of closed headphones is higher than that of open headphones. Sound engineers often use closed headphones, which allows them to concentrate on the music without disturbance from outside noise. The problem of sealing does not exist with open headphones. In this design, the space behind and in front of the diaphragm lets sound through. Therefore, open headphones allow music to pass straight through the diaphragm without being "muffled", thus resulting in a more transparent and natural sound image. The distinguishing features of open headphones are their small size and low weight. These in turn make them extremely comfortable to wear, and no discomfort is felt even after prolonged periods of listening.

Holophonics
An acoustical recording and broadcast technology claimed to be the aural equivalent to holography, hence the name. Holophonics is an encode process that occurs during the recording session using a special listening device named "Ringo." It is claimed that "playback or broadcast is possible over headphones or any existing mono or stereo speaker system, with various levels of spatial effect."

Hub
In computing a hub is a device where several devices are connected together, a place of convergence where data arrives from one or more directions and is forwarded out in one or more other directions. This may be many computers on a network, or many devices to one computer. A passive hub serves simply as a conduit for the data, enabling it to go from one device (or segment) to another. So-called intelligent hubs include additional features that enable an administrator to monitor the traffic passing through the hub and to configure each port in the hub. Intelligent hubs are also called manageable hubs. A third type of hub, called a switching hub, actually reads the destination address of each packet and then forwards the packet to the correct port.

I/O
Abbreviation for Input/Output. Strictly speaking any device that does anything has input and output. A seesaw, for example, utilizes the energy from children's legs (the input) to rock back and forth (the output) on a fulcrum. But the term is mostly used in electronics, especially as it pertains to computers or any kind of logic functions, but also with audio and video equipment. Computers have all sorts of I/O, from serial ports, to SCSI, to monitor and keyboard ports. Audio and video equipment is obviously all designed with the ability to get signals in and out as well. These inputs and outputs, when spoken about collectively, are sometimes called I/O for short.

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.

In Line Mixer
An audio mixer configured to be able to monitor multitrack tape returns through the same channels that are used for inputs from microphones, line input sources, etc (see WFTD In Line Monitoring). This is in contrast to a configuration known as a split mixer, which has separate inputs dedicated to tape sends and returns. In line mixers have the advantage of being able to be smaller and less expensive, since each channel does double duty. It can often be accomplished with a couple more knobs and switches on each channel strip. This can potentially be a drawback since resources such as EQ, aux sends, etc, may have to be split between the input signal and the tape return signal, however, in many practical applications this limitation isn’t considered a problem since a resource like EQ will be used on the input source during tracking and overdubs, and then can be devoted to the tape return on mixdown since the mic/line input portion of the channel won’t be active.

In-Line Monitoring
A mixing board architecture. Many newer large format mixing consoles have an additional, (often smaller) fader section designed into each channel strip. This can directly feed either the recorded signal being fed to the multitrack recorder or the monitor mix. In the standard monitor mix mode, this small fader is used to adjust the monitor level of the associated tape track. In the "Flipped" mode, it is used to control the signal being sent to tape while the main fader can be used to control monitor mix levels. This enables multitrack levels (which aren't often changed while recording) to be located out of the way, while the more frequently used monitor levels are located at the more accessible main fader position. Mixing consoles that do not use an inline configuration usually employ what is known as a "split" format, where the tape sends and returns are on completely separate channel strips from the main channel inputs. Both designs have their strengths and weaknesses. Most modern consoles use the in-line approach.

Insulator
A substance that has a very high resistance to conducting electricity. Insulators are used keep electrical signals from shorting out or otherwise coming into contact with one another. In a typical electrical power cord, for example, you will find wires (the conductors) surrounded by one or more insulators. The insulators protect the two wires from each other, then there is usually another insulator around the whole thing.

ITU 775 Surround
ITU 775 stands for International Telecommunications Union, Operational Bulletin No. 775 in which recommendations are given for a multi-channel surround standard for "5.1" speaker positions. The ITU-775 setup is sometimes referred to as "3/2 format," indicating a division between a 3-speaker frontal sound stage and a 2-speaker rear "surround". To arrive at this standard, 20 speakers were placed in an anechoic room to find the critical angles for the best speaker placement. For the surround (rear) speakers, a compromise had to be found between 90ş and 135ş. Whereas 90ş (directly on either side of the sweet spot) was the best placing for ambience or "envelopment", 135ş turned out to be best for "surround-placement" or localization in the rear, hence a compromise at 110ş. In the reference loudspeaker arrangement for mode 3/2, it is recommended that the loudspeakers be placed on the arc of a circle. In those cases where the front speakers must be placed on a straight line, for example, when the center speaker cannot be placed behind the screen, it was recommended that the sound signals be appropriately delayed so that all signals reach the listener's ears simultaneously. It was further recommended that all the front loudspeakers be driven by discrete audio signals. In those cases where the center loudspeaker cannot be placed behind the screen, it must be placed above or below the screen.

Joystick
For anyone who plays video or computer games a joystick is a common household word. In audio and music production it is a controlling device that can move along two different axes simultaneously. Similar in concept and purpose to a modulation wheel (or other continuous controller) and a fader or pan pot, a joystick divides one input signal among four output channels. Some keyboards have had joysticks instead of separate modulation and pitch bend wheels (or sliders) to allow the user access to both controllers simultaneously via one mechanical interface. In modern audio production the joystick is starting to become a replacement for the pan pot. This is because the proper positioning of sounds in a 5.1 mix (for example) requires more than just left to right pan positioning. It requires, at minimum, a combination of left/right and front/rear positioning, which is most easily done with a joystick. Most software dealing with surround sound will offer some type of graphical interface based on the two axes provided by a typical joystick. This usually takes the form of a virtual grid where each sound can be positioned anywhere along either axis.

Jumper
In the music technology game this word classifies as jargon. It is most used to describe a circuit connection in circuit boards that is made by attaching a (usually) small piece of wire between two points. Jumpers are usually used to either modify existing circuits or to provide a way for the end user to make easy changes. Jumpers are often designed so end users can move or remove them to change the configuration of equipment. For example, it is often possible to change a mixing board's auxiliary send from taking its signal out before the fader (commonly used for monitor mixes) to taking it after the fader (commonly used to drive signal processing). Another example would be changing a patch bay from full-normal to half-normal operation. A third example would be changing the SCSI ID on a hard drive mechanism. Jumpers come in varying shapes, sizes, and configurations and are more or less easy to change depending upon the intentions of the manufacturer of the equipment.

LED
Abbreviation for Light Emitting Diode. It is not an acronym. You do not pronounce it as "lead." It is pronounced EL - EE - DEE. An LED is an electronic component that glows when current passes through it. LED's are found in all sorts of electronic equipment these days from watches to laser disc players (in fact the laser that reads the disc is usually an LED). The lights that glow on top of your keyboard are almost surely LED's. The numerical readout and meters on your DAT machine are almost surely LED's. LED's do not always produce visible light. Infrared LED's are used in wireless remote control devices as well as things like wireless headphone systems.

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.

LFE
Abbreviation for Low Frequency Effects, a term used in surround sound mixing. Low-frequency effects are mixed to a separate so-called LFE channel in modern movie sound production. The LFE channel carries non-essential effects enhancement - such as the low-frequency component of an explosion - often at higher levels than the other channels. The idea behind a separate LFE channel is to provide the extra low-frequency headroom needed to put low-frequency signal components on equal psychoacoustic footing with midrange signal components which require less energy for the same perceived loudness.

Limiter
A limiter is a dynamics processor very similar to a compressor (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!

Load
In electrical terms a load is something that dissipates power and does some work. The work done may take many forms, including generating heat as almost always happens as a side effect of work being done. Without a load no power can be transferred. A speaker is the load for a power amp. In order for current flow to occur a complete circuit must exist. In order for the circuit not to be a short-circuit (a decidedly bad thing) a load must be present to the power the amp. The power amp drives power through the circuit by way of increasing the voltage at its outputs and as a result the load (speaker) draws current and does work. In this case two major forms of work occur: The speaker moves and generates sound, and heat is produced. Any device you plug into an electrical outlet can be considered a load (toaster, light bulb, etc). Plug in too many devices drawing too much current and you will "load down" the power delivery system (another bad thing). In order to protect against this power delivery systems have fuses and circuit breakers to break the circuit when current flow gets too high. Many power amps employ current limiting devices in their output stages to limit current flow without interrupting the audio. It's sort of a self regulating protection system (back in the old days the amp just blew up). An important thing to understand is that a load will DRAW from an available pool of power all of the current it needs to operate at the given voltage. This is somewhat simplified, but in principle remains fundamentally true for all electrical systems. A speaker's impedance rating is an indication of what kind of load it presents to an amplifier. An appliance's current or amperage rating is exactly the load it will place on the electrical system. The reason a speaker cannot be rated in exact terms of current usage is because the voltage and frequencies presented to it constantly change. Impedance is a way of approximating a speaker's resistance to a varying voltage and frequency signal.Also related to us is acoustical loading. The efficiency of a loudspeaker depends to some extent on the acoustic load placed on it by the way it couples to a cabinet and the surrounding structures. A speaker placed in the throat of a horn, for example, will see a higher acoustic impedance than a speaker placed in a free space.

Loudspeaker
A transducer that converts electrical energy into sound energy, providing the audible sound in equipment such as public address systems, studio monitors, guitar or bass amplifiers, radios, televisions, and home stereos. A standard dynamic loudspeaker consists of a voice coil, a magnet, a diaphragm and a cone. The electrical energy output of a power amplifier is transmitted as voltage over a wire to the voice coil. The current flowing through the voice coil produces an electromagnetic field that reacts with the stationary magnet in the speaker assembly. The voice coil is attached to a diaphragm, which in turn is attached to the cone. The magnetic fluctuations cause the diaphragm and thus the cone to move, moving air and radiating sound. There are other types of loudspeaker technology, the best known being electrostatic speakers. These differ from dynamic loudspeakers in that they consist of a thin sheet of electrically conductive film suspended between two wire screens. A high-voltage charge is applied to the film and it is alternately attracted to one screen and then the other. This creates motion, which again radiates sound. Another type of loudspeaker are servo drive loudspeakers. These employ servo driven motors attached to the speaker cone in place of the magnet/wire assembly. This type of speaker is generally only used in subwoofer applications, and even then only rarely.

Matrix Surround
An analog approach to surround sound in which third and sometimes fourth channels of sound are folded into the two front stereo channels and later partially decoded in a reverse operation. Examples are SQ, QS, and Dolby Surround. Only a small portion of the original additional-channel information can be recovered, as opposed to Dolby Digital and DTS, where each of the six discrete channels can be totally recovered without mixing with the others.

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.

Meridian Lossless Packing (MLP)
An audio encoding scheme using technology developed by Meridian Audio that is the industry-accepted standard for compressing audio data on DVD-Audio disks for transfer and decoding by a DVD-A player. MLP is a "lossless" compression codec. DVD-Audio can play back surround sound audio (in 5.1 format) at 24-bit/96kHz resolution. However, DVDs have a maximum data transfer rate of 9.6MB/second. So a 5.1 surround track at 24-bit/96kHz, which requires a data transfer rate of about 13.8MB/second, would get bogged down unless it is compressed. MLP uses a complex combination of filters, entropy coding, prediction and buffers to compress (encode) that signal so it can be transferred at the 9.6MB/second rate, and then decode (uncompress) it for playback by a DVD player. The MLP process also reduces the size of audio files by 30 to 50%, thus allowing a full 74 minutes of surround audio to fit on a DVD-A. The DVD Forum, the industry group that sets standards for all types of DVD production, selected Meridian Lossless Packing as the specified standard for audio compression on DVD-Audio discs and players. MLP can also be used to simply get longer stereo recording times onto the disc, whether the files are at 96kHz or 192kHz sampling rates. Since Meridian Audio is primarily a manufacturer of high-end home theater hardware, they licensed MLP to Dolby Laboratories for management. Dolby handles further licensing of MLP to DVD player manufacturers and software publishers who incorporate it into their DVD encoding programs.

Mini Plug/jack
One of those terms that could mean almost anything, but in practice generally refers to a 1/8 inch diameter plug (or jack) used in smaller audio visual interconnects. The connector may be TRS or TS, as well as some other configurations. This is the size of most of the Walkman style headphone connectors.

Monitor
This term has several meanings as applied to audio and video technology. As a verb, to "monitor' means to listen to a sound source such as a recorded track or a mix. In a recording environment, monitors are the loudspeakers used to play back the live signals and recorded tracks of a project. Monitor also refers to a special mix (monitor mix) that is provided to the talent, usually through headphones, to give them a reference to the music they are performing. This is sometimes called a cue mix. In sound reinforcement, monitors refer to the system of loudspeakers and/or in-ear systems that transmit an often-custom mix of the audio program back to the performers. In computer usage, a monitor is the CRT or flat-panel LCD display screen that provides visual images of your programs and activities.

Near Field
The sound field very close to the audio source is called the near field. "Very close" in this case means less than one wavelength at the frequency(s) of interest. Near field is a phrase we hear thrown around a lot these days due to the popularity of monitoring systems commonly known as Near Field Monitors. Technically, however, near field monitors aren't truly used in the near field most of the time. Even at a relatively low frequency like 400 Hz the near field stops less than three feet from the speaker, beyond which one is typically in some cross between an approximation of a Free Field and Far Field (or Mid Field). The idea behind near field monitors is that the listener is so much closer to the speakers than the surrounding walls, floor, and ceiling that their effect on the sound is minimized as stated in the Inverse Square Law. Thus the term near field can be appropriate as a reference to being relatively near the listener compared to other objects in a room, including the larger studio monitors.

Normal

1. Corresponding to the usual state, not out of the ordinary.
2. Something the inSync team is NOT accused of being (Can't figure that out; we don't think being nocturnal, doing strange things to guitars, lusting ferociously after electronic gear, and living in caves lit only by the blue phosphorescent glow of computer monitors is so strange. Besides, the resident sloths, bats and owls like it...)
3. In patchbays, a normal is an internal connection from the top row of jacks, to the bottom row. Normalling allows connections that are normally in effect to exist without the need for inserting a patch cable in the front of the bay. For example, the stereo outs of a mixer are generally connected to the inputs on a stereo mixdown deck. By connecting the mixer's outputs to the top back row of a normalled patchbay's jacks, and the mixdown deck to the bottom back row, a connection is made internally in the bay, and does not require extra patch cables.

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.

Open Air
When referring to headphones, the term open air means that the headphone remains open to the outside. Open air headphones are usually more comfortable than their closed air counterparts, but they do allow sound to leak in both directions. This means that sounds from the outside can get it, potentially making it difficult to hear the headphone signal, which is of particular concern with drummers or anyone who is monitoring near loud sound sources. They also allow sound to escape to the outside, which can become an issue in recording sessions. For example, a singer's headphone mix may leak into the vocal mic enough that it can be audible in the final recording, or at least to the extent that it effects the tonality of the final mix. For these and other reasons, closed air headphones are often used in recording situations; however, open air headphones are often preferred for general listening and/or audiophile listening applications.

PAL
An acronym that stands for many things. The most relevant to us is the Phase Alternate Line (or Phase Alternation Line). The standard for color television broadcast throughout much of Europe. The United States uses the NTSC standard, which is used in all of North America and many other parts of the world. PAL has good color transmission and sends an analog signal at 625 lines of resolution, 25 interlaced frames per second, whereas NTSC delivers 525 lines of resolution at approximately 30 interlaced frames per second. The two formats are incompatible with one another, but there are video adapters that enable computer monitors to be used as television screens to support both NTSC and PAL signals.

Pan (Panning)
Comes from the term panoramic, which pertains to large visual scenes that can completely surround a subject. In film work panoramic shots require a camera to be "panned" across the landscape (or whatever the subject is). This terminology was adopted when two-channel (stereo) audio first arrived on the scene. In audio a pan control is used to position an audio track somewhere between the left and right loudspeaker in the stereo soundfield. A pan control generally works by simply reducing the level of a track in one channel, which makes it appear louder in the opposite channel. Modern designs are more sophisticated in their approach, but the basic concept has stayed the same: turn the pan pot to the left and that track comes out of the left speaker.

Peak Hold
On non-mechanical (LED) indicators, Peak Hold allows the meter to continue displaying the highest signal level for a certain amount of time or until it is exceeded by an even higher peak. This is very useful, as it gives clear indications of where and how hot peaks are, but still allows monitoring of the current signal level. Knowing where the peaks are allows easier adjustment of dynamics processing, as well as more accurate input and output level settings on other gear.

Phantom Channel
A special mode in many surround sound systems that reproduces the effect of a center channel through a left and right stereo speaker setup. The mode is designed for users who wish to experience surround listening, such as with Dolby Surround, but who do not yet have a center speaker to reproduce the discrete center channel information. Basically the audio that would normally be sent to the center channel is added to the audio in the left and right speaker channels. This produces a mono image centered between the two speakers, almost as if a real center channel speaker were there.

Phono Plug
A small inexpensive coaxial connector used for interconnection of many audio devices, especially consumer devices. Not to be confused with the phone plug, which was developed and used by Bell labs for telephone patch cables, the phono plug was first used by RCA to connect phonograph tonearms to their amplifiers, hence the name Phono Plug. As a result of it being common in RCA brand equipment, it is also widely known as an RCA plug. Phono plugs are not renowned for their durability and longevity, but they are small and easy to use. The coaxial configuration (a center "hot" conductor surrounded by a "ground" connection that is a consistent distance from the hot) also makes them an affordable solution for high frequency transmission of signals like video and digital audio so long as proper insulators are used that will maintain the proper impedance through the connector. S/PDIF formatted digital audio data is often transmitted on coaxial cable terminated with phono plugs.

Pickup
Essentially a simple mechanism, a Pickup is based on a magnetic field that induces a current in a coil in that field. The pickup converts string vibration into electrical energy, usually by means of a permanent magnet, or six magnetic polepieces, surrounded by a wire coil. There are two basic types of pickup, single coil and humbucking.

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.

Pixel
Short for Picture Element. The pixel is the smallest element that is used to build an image, whether it is displayed on a video screen, computer monitor, printed photo, or newspaper. A complete monitor image is made up of thousands of pixels. The pixel is often used as a unit of measurement for image size and resolution. The number of pixels (width and height) in an image defines its size, and the number of pixels in an inch (or other quantifiable measurement) defines the resolution of the image. The more pixels in an image the better its resolution.

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.

Potting
The term "potting" refers to the sealing of pickup coils in a solid material. Potting stabilizes the components of the pickup so that they cannot move relative to each other. This eliminates vibration-induced signals that make a pickup act like a microphone causing unwanted feedback. Potting can also protect the inner coil from corrosion. The best technique for potting also includes "coil immersion." Coil immersion is allowing a solid (wax) to be absorbed into the coil. Wax is used because it works well, is inexpensive, and it makes it possible to work on the pickup later. A correctly potted pickup coil will have the wax absorbed throughout the coil as well as the surrounding parts such as magnets, pole-pieces, and metal covers. This eliminates movement of parts inside the pickup.

Pre Fade/Post Fade
Refers to functions that happen either before (Pre Fade) or after (Post Fade) the main fader of a channel in a mixing board. For example, the Pre Fade Listen (PFL) is taken before (pre) the fader by definition. In certain situations it can be advantageous to monitor or take signals out of the board before or after the signal has passed through the fader. When driving headphone mixes or stage monitors, for example, it is common to use a pre fade aux send, so that the levels can be set independently of the fader position. It's like having two separate mixes at the same time. Plus, you avoid unexpected levels being sent to the artists because you made a change for the main mix. On the other hand, it is customary to use a post fader signal to drive effects units such as reverbs, where it can be advantageous for mix of the signal sent to the reverb to be the same as the overall mix. Pull down a particular part in the main mix, and you also pull down its signal going to the reverb unit. Most of this is a matter of individual preference based on different circumstances. There are a number of other functions in a mixing board that may take place pre or post fade (inserts, EQ, dynamics, etc.) depending upon the board in question.

Pre-Fade Listen (PFL)
In a console, pre-fade listen is a one of several possible means of overriding the normal monitor signal routing for various purposes. PFL generally sends a signal to monitor outputs regardless of the setting of that channel's fader, and simultaneously mutes the other channels. In other words, PFL allows you to solo a channel even if the fader is pulled all the way down. Note that on most consoles, this affects monitors only, and does not interfere with main, tape, or aux outs. In broadcast situations, PFL is often referred to as "cueing".

PrecedenceEffect
Also known as Haas effect. Refers to how we locate sounds based upon time arrival differences between our two ears. Not only does it effect our perception of where the sound is coming from, but it also effects our perception of the volume. The same sound can be presented to each ear at the same volume, but we will hear the one arriving first as louder. This effect can be so drastic that you can be fooled into thinking one of your monitor speakers isn't outputting any sound simply because you are sitting a few milliseconds closer to the other one.

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.

Progressive Scan
A video term that describes a method of displaying images in which every horizontal line is drawn on the screen in a single pass to create a complete frame or single full-screen video image. This is different from interlaced video, in which each video frame is created by drawing two fields, one of which is made up of the odd numbered lines and the other the even numbered lines. Traditional television video uses interlaced scanning. With the advent of digital video and high definition video, progressive scan technology has become much more common. Computer monitors have used progressive scan (calling it “non-interlaced”) for quite some time.

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.

Recording Console
At its simplest level, an audio device used to add (combine or sum) multiple inputs into one or two outputs, complete with level controls on all inputs as well as routing and monitoring capabilities designed to get signals on and off of one or more recording machines. From here signal processing is added to each of the inputs and outputs until behemoth monsters with as many as 80 (or more) inputs are created - at a cost of around 10-20 kilo-bucks per input for fully digitized and automated boards. At these price points a mixer becomes a recording console. (Sometimes referred to as a Desk or Board.)

Repro
Short for Reproduction, the repro function in tape recorders allows for the playback of all tracks, including those that are currently being recorded. The way this works is through the use of a separate repro head. The repro head of a tape machine is generally only capable of playback (not recording) and is optimized as such. In the earlier days of tape recorders the repro head was the only head on a tape machine capable of full fidelity playback. It usually has its own set of corresponding electronics in the machine that can also be optimized for playback. Also important is the positioning of the head, which is after the other tape heads along the tape path. Since the repro head is last in the series, the engineer can monitor off of this head even while recording on another head, which means it's possible to literally monitor the actual results of the recorded (and played back) signal on the tape. This is sometimes also referred to as confidence monitoring. The only caveat is that the signal one hears is delayed by the length of time it takes a particular point on the tape to travel from the head where it was recorded over to this separate repro head. This distance is usually not much more than an inch and a half, so if the tape were moving at 15 ips the delay would be about a tenth of a second.

Resolution
There are many definitions, but the relevant one for our purposes is that resolution is a measurement of the fineness of detail captured in a representation of something. This could pertain to the level of detail captured in a photograph or displayed on a computer monitor. It could even relate to video frames and time code: 30 frames per second is more resolution than 24 frames per second. We most commonly speak about resolution in terms of digital audio and how much resolution a digital audio system has. In digital audio resolution is affected by the sampling rate and the bit depth of the recording: 24-bit audio is higher resolution that 16-bit audio, and a 48 kHz sample rate is more resolution that a 44.1 kHz sample rate.

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.

RGB
An abbreviation for "Red, Green, Blue." The RGB color model is an additive method of creating colors by utilizing red, green, and blue light combined in various ratios. The very idea for the model itself and the abbreviation "RGB" come from the three primary colors. Primary colors are based on the physiological response of the human eye to light. The human eye contains photoreceptor cells called cones, which normally respond best to yellowish-green, green, and blue light. The color yellow, for example, is perceived when the yellow-green receptor is stimulated slightly more than the green receptor, and the color red is perceived when the red receptor is stimulated significantly more than the green receptor. Although the peak responsiveness of the cones does not occur exactly at the red, green and blue wavelengths, those three colors are described as primary because they can be used relatively independently to stimulate the three kinds of cones. One common application of the RGB color model is the display of colors on a cathode ray tube or liquid crystal display such as a television picture tube or a computer monitor. Each pixel on the screen can be represented in the computer's memory as independent values for red, green and blue. These values are converted into intensities and sent to the CRT or LCD display. By using the appropriate combination of red, green and blue light intensities, the screen can reproduce many colors between its black level and white point. Typical display hardware used for computer monitors uses a total of 24 bits of information for each pixel. This corresponds to 8 bits each for red, green, and blue, giving a range of 256 possible values, or intensities, for each color. With this system, approximately 16.7 million discrete colors can be reproduced.

S.M.A.R.T
An acronym for Self-Monitoring Analysis and Reporting Technology, S.M.A.R.T. was developed by a number of major Hard Disk Drive Manufacturers in a concerted effort to increase the reliability of drives. It is a technology that enables the computers to predict the future failure of hard disk drives. Through the S.M.A.R.T. system, hard disk drives incorporate a suite of advanced diagnostics that monitor the internal operations of a drive and provide an early warning for many types of potential problems. When a potential problem is detected, the drive can be repaired or replaced before data is lost. S.M.A.R.T. monitors the disk's performance, bad sectors, calibration, CRC errors, disk spin-up time, distance between the head and the disk, temperature, features of medium, heads, motor or servo mechanism. Armed with a failure prediction, the user or system manager can back up key data, replace a suspect device prior to data loss, or avoid undesired downtime. Glyph's current line of hard drives feature S.M.A.R.T.

SACD (Super Audio Compact Disc)
SACD is one of several emerging new standards for high-resolution audio on compact discs. It was developed by Sony and is based on a licensed technology called Direct Stream Digital, which was developed by Sony and Phillips and is theoretically capable of sample rates up to 2.8 MHz. The SACD format allows for playback of multi-channel audio and a bandwidth of 100 kHz at over 120 dB dynamic range while retaining compatibility with existing compact disc technology. There are several subformats in the works (single layer, dual layer, etc.) that are optimized for different tasks, but Sony claims that all SACD discs have fully uncompromised audio quality. That is, no data compression, and no computer generated surround mixes from stereo data or vice versa. The potential success of this format in the mainstream is currently under scrutiny amidst other developments such as DVD Audio, but there are a significant number of titles available on the Sony label with promised support from other record labels.

Samarium-Cobalt Magnet
Samarium Cobalt (SmCo5) is used in making a new permanent magnet material and has the highest resistance to demagnetization of any known material. Samarium itself is a rare earth metal, with a bright silver luster. Cobalt is a tough lustrous silver-white magnetic metallic element that is related to and occurs with iron and nickel and is used especially in alloys. Other uses of Samarium include: Carbon-arc lighting for the motion picture industry (together with other rare earth metals), doping CaF2 crystals for use in optical lasers, as a neutron absorber in nuclear reactors. It is also used for headphones, and now guitar pickups. (Recall that guitar pickups are magnetic coils.) Samarium Cobalt Magnets can achieve very strong fields for their size (second to neodymium), which has far reaching implications for transducer design.

Samarium-Cobalt Magnet
Samarium Cobalt (SmCo5) is used in making a new permanent magnet material and has the highest resistance to demagnetization of any known material. Samarium itself is a rare earth metal, with a bright silver luster. Cobalt is a tough lustrous silver-white magnetic metallic element that is related to and occurs with iron and nickel and is used especially in alloys. Other uses of Samarium include: Carbon-arc lighting for the motion picture industry (together with other rare earth metals), doping CaF2 crystals for use in optical lasers, as a neutron absorber in nuclear reactors. It is also used for headphones, and now guitar pickups. (Recall that guitar pickups are magnetic coils.) Samarium Cobalt Magnets can achieve very strong fields for their size (second to neodymium), which has far reaching implications for transducer design.

Semi-Open
A headphone design that falls somewhere between closed ear and open air designs. Many headphones that are called open air are really semi-open, as a true open air design does not stop any outside sound from getting in. Semi-open designs will muffle or partially stop outside sounds from getting to your ears, and as such can serve as a workable compromise to open air or closed ear designs, giving the listener the comfort of an open air design while helping to maintain some isolation.

Shotgun Microphone
A type of microphone characterized by an extremely directional polar pattern. Shotgun mics may be condenser or dynamic, but are almost always built with a long (8 to 24 inch) tube protruding from the front. This tube has a series of holes or slots along the side, which act as a phase canceling device for sounds coming from the rear of the microphone. Sounds coming from directly in front of the mic enter each of the holes or slots in succession and therefore add in phase by the time they reach the diaphragm. Sounds from the rear enter in reverse order and thus are out of phase when they reach the diaphragm, resulting in little or no output. The longer the tube the more directional the microphone becomes. These properties make them ideal for pinpointing and capturing the audio of something from far away without capturing as much of all the ambient (or surrounding) sound. Shotgun mics are sometimes called Line Microphones.

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.

Solo
A function commonly found on mixing consoles, soloing a channel is the opposite of pushing a mute switch; solo mutes all channels EXCEPT the one being soloed. In general, solo only affects signals in the control room monitors, or headphones on a live console. It does not mute signal being sent out other outputs. This allows the engineer to listen to individual signals while not interfering with other mixer functions (feeding recorders or PA amplifiers, etc.).

Sound Card
An expansion board that enables a computer to manipulate and output sounds. Sound cards have become commonplace on modern personal computers and are typically associated with the consumer market. Sound cards enable the computer to output sound through speakers connected to the board, to record sound input from a microphone connected to the computer, and manipulate sound stored on a disk. Some sound cards also support MIDI, surround sound and more. In addition, most PC sound cards are Sound Blaster- compatible, which means that they can process commands written for a Sound Blaster card, a standard in consumer PC sound.

Source
In audio, source refers to (you guessed it) the source of the audio. This would seem to be obvious, but in the context of how it applies to audio and video equipment it sometimes isn't obvious. It mostly applies to how signals are monitored. You can determine whether you monitor the signal coming in from the source or at some later point inside a device. For example, the source button on a cassette deck might switch you between monitoring the signal coming off the tape to the signal being applied at its inputs (the source).

Stage Monitor
A speaker that is typically placed within and pointed at the performance area to aid in a performer being able to hear critical elements for performance. Where a PA speaker is designed to provide sound reinforcement intended for the audience, stage monitors are designed in principle to provide sound reinforcement intended for the performers on stage. Stage monitors come in many shapes, styles and forms. Perhaps the most common is the stage wedge, which gets its name from its shape. Stage wedges tend to sit on the floor and the speaker itself is then set at an angle pointing up toward the performers on stage (some are designed with a 30 degree angle, for instance). Some bands prefer to have a separate stage wedge for each member where other bands prefer a general wash of sound provided by a few stage wedges across the front of the stage. Side fill, another application of stage monitors, is intended to perform as its name would incline in that they are typically placed to the side of the stage and are intended to "fill" the stage with sound. Side Fill stage monitors might be an active or passive PA speaker on a stand, stage wedges placed on the floor or on a table, or even studio reference monitors on stands. Just about anything can, and has been used for this application. In-Ear monitors are ear-bud style earphones that are fed their signal by use of a wired or wireless transmission. Similarly speaking, many drummers have been known to use closed headphones on stage for monitors, but usually when they are playing to a click track of some sort.

Start Time
A value that is set in synchronizers, sequencers, or other equipment that is to be synchronized with some form of time code. The start time tells the device where to start its function in relation to the incoming time code. For example, on a sequencer that is being synchronized to a video tape the start time would be the time code value (usually SMPTE time code) where you want to sequence to begin playing. This is normally a few seconds before the beginning of the program material on the video. When that value of time code arrives the sequencer starts its playback (or recording as the case may be). The sequencer will usually continue to monitor the time code and adjust its speed so it stays in sync as the tape plays. Typically time code is striped to a whole tape from beginning to end so as to provide the maximum flexibility with start times and synchronization equipment later.

Stem
In audio/video/film production a stem is a group of audio information, not unlike a submix. In fact they are often created using subgroups from a mixer in just the same way as a submix. In film and some video production many different stems are put together during the final stages of mixing (mixdown) to form the final soundtrack. Stems may include foley tracks, music tracks, sound effects tracks, dialog, location sound, etc. Each of these stems is a submix of dozens of individual components each carefully and often individually prepared to enhance the overall experience of the film. Nowadays individual stems are often prepared in full 5.1 or 7.1 surround, which means that "a stem" may actually be comprised of as many as eight separate channels/tracks of information.

Sub
Short for Subwoofer, though occasionally used as an abbreviation for subgroup. A subwoofer is simply a speaker (woofer) designed to handle the very low frequencies of a speaker system. The concept has been around for many years, but only in the last 10 or 15 years has their use become widespread. With increased popularity of smaller main speakers and much more low frequency content and dynamic range in our recordings, low frequency drivers have become an important part of any speaker system. Many systems may have only one subwoofer (as opposed to the two you would expect). In fact, most of the home theater surround sound technologies (AC-3, etc.) in use today have only one mono sub output. This is based in part on the theory that very low frequencies tend to be omnidirectional so one speaker can cover an entire room. Plus it's often difficult to produce stereo separation between subwoofers, and in fact any two drivers producing the same frequency range in the same area can tend to interfere with one another as the time arrival of the sounds at the listening position causes certain frequencies to be more or less out of phase with one another, which causes uneven frequency response and even dead spots.

Subframe
In audio and video timecode operations, a Subframe is a subdivision of one SMPTE or MTC frame. Subframes are generally considered to equal 1/100 of a frame. There is no actual component this small in a timecode word, but the subframe information can be interpreted by monitoring the phase sync of the individual bits making up the signal. In some equipment a slightly more coarse resolution of 1/80 of a frame is used. While often referred to as a subframe, this 1/80 of a frame is more accurately known as a bit, as it refers to any of the 80 bits of individual data that are transmitted as part of a standard timecode word. There are also other derivations of the subframe in use.

Supraaural
Used in reference to headphones. Supraaural phones rest "on the ear", rather than enclosing the ear. Supraaural phones typically are lightweight, and because they do not seal around the ear, tend to not provide good isolation.

Surround
The flexible material that connects the out edge of a loudspeaker cone with the speaker’s frame or superstructure (normally referred to as a “basket”). Surrounds are usually made out of a flexible material such as corrugated cloth, paper, or some type of foam rubber compound that enables the speaker to move in and out efficiently while still being sturdy enough to help keep the voice coil centered in the magnetic gap. Surrounds come in a variety of styles and appearances for different applications. For example, “Roll” surrounds are characterized by one, larger rounded corrugation and are generally preferred in more delicate applications due to their increased flexibility and linear behavior when flexed.

Surround Sound
Surround sound is a multi-channel audio playback format consisting of at least three speakers (left, center and right) but more commonly consisting of five or more. The first Motion Picture shown publicly with multi-channel sound was Disney’s Fantasia in 1941 but the first commercially successful multi-channel formats would not come about until the early 1950s with the four-track CinemaScope and Todd-AO’s six-track format. While the cinema has had the largest influence on surround sound technology, it is in the home of consumers where we see the largest growth of commercial success. Today, surround sound playback systems typically consist of 5 speakers and a subwoofer which makes up a “5.1” system, similar to that found in movie theaters. A 5.1 system utilizes forward left, right and center channels, two surround channels that are commonly placed behind or to the left and right of the listener(s) and a subwoofer (the “.1”). Dolby Digital surround encoding could be found on Laser Disks as early as 1995, soon to be followed by the DVD revolution most recently. Today, DVD-Video, DVD-Audio and several other surround sound formats (including cable and VHS) have found their way into the homes of consumers.

Sweet Spot
The optimal listening position for a pair (or more) of loudspeakers. The Sweet Spot provides the best listening conditions for tonal balance, stereo separation, detail, and overall imaging. Locating and maintaining the Sweet Spot for studio monitors has historically been an important ingredient in the recording process as it is the place where the best overall judgements about the sound can be made. The Sweet Spot is generally a few feet in front of, and in between a pair of stereo speakers. Many schools of thought say the Sweet Spot is found by positioning the tip of your nose and the high frequency drivers of each speaker at the tip of an imaginary equilateral triangle. Your actual position may vary, however, due to the effects of other objects in the near field. The top of a typical mixing console is one of potentially many items that can change the quality and location of the true Sweet Spot. Some speakers are known to have a wider and more forgiving Sweet Spot than others as well. The best way to reliably find and optimize the Sweet Spot in your set up is through continued listening and tweaking of speaker and listener placement.

Talkback
A feature offered on recording consoles, talkback is an in-board intercom system, allowing the engineer and producer in the control room to talk to musicians in the studio. Normally, there is either a built-in microphone for this purpose, or there is a dedicated talkback mic input. This mic/input is routed only to the cue/studio monitor sends, preventing feedback problems with the control room monitors.

Tape Delay
A type of delay or echo processor that uses analog recording tape to achieve the effect. Back in the “old days,” producers and engineers created delay and echo effects using tape machines. Basically a signal would be routed to a separate tape recorder (from the one being used for the performance) that was set to monitor off of the repro head. The slight delay that occurred between when the signal was presented and when it finally came off the repro head provided a delayed signal back to the main recording. The delay time could be adjusted by changing the speed of the tape machine used for the delay. Feedback or multiple echoes could be generated by routing the delayed signal back into the machine, and other, more exotic effects could be created by changing the speed of the machine while signals were passing through it. When applied subtly chorus would result, but other more dramatic effects were easy as well. This technique became so popular that engineers began to devise ways to make the tape continuously loop on a machine so the tape would never “run out.” Manufacturers eventually stepped in with products to make it easier. The Echoplex and Roland’s Space Echo Series were some of the more popular devices. They had features and controls that were optimized for the intended purpose and used proprietary tapes that were already looped and in small cartridges. These machines were tremendously popular because they were easy to use and portable. Besides being great delay/echo units they were also many musician’s first foray into looping effects. There were some significant downsides though. It was tape. The recording quality wasn’t always great, depending upon the condition of the heads and how worn the tape was. Even on a perfectly tuned machine the quality of the signal would degrade after several regenerations. In other words, on a repeating echo, the quality would change (degrade) with each repeat because this involved playing the first echo out, recording it again, playing it, recording that signal again, etc. The degradation happened fairly quickly (as opposed to the pure looping effects, which were one recording that just played back over and over). Eventually analog and later digital devices were created that were in many ways better for doing delay effects. They were cheaper, more reliable, required virtually no maintenance and eventually sounded better in terms of the delayed signals being more like the original and not degrading. But by that time so many musicians and producers were accustomed to the tape delay sound that it was missed. Developers of digital delays had to put filters in to roll off high frequencies as echos repeated and so forth. Even with a modicum of parameters designed to help digital devices sound more like their tape-based ancestors, the difference was still significant. In more recent years the technology of modeling has allowed much more accurate emulation of these old techniques. When you see “tape delay” as one of the algorithms in your digital box or software, it’s probably referring to the emulation of these old sounds, and many of their inconsistencies. Still some purists prefer the sound of the old tape units. This is not unlike the arguments that exist for analog recording today: some of the deficiencies of analog recording actually produce a sound many people find desirable.

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.

TRS
Abbreviation for Tip Ring Sleeve. This is the descriptively accurate term used to describe 1/4" (or 1/8") balanced connectors. A TRS plug can be found at the end of most headphone cords if you want to know what one looks like. They look like a standard 1/4" plug with an extra section in them. The three sections of the shaft are called the Tip, Ring, and Sleeve (a "standard" 1/4" connector just has a tip and sleeve). TRS connectors are used wherever it is desired to have two conductors plus a ground (shield) in one plug. Common uses are as a way to connect balanced equipment (where the TRS plug has a positive, negative, and ground connection), or stereo unbalanced equipment (left and right are on the Tip and Ring, with a common ground) like headphones, or as an insert for your mixer or other processor (Tip or Ring is the send with the other being used as the return and again ground is common).

Tweeter
The highest frequency transducer of a multi-driver speaker system. Considered the opposite and complement to the woofer or lower frequency drivers, Tweeters come in many different shapes, sizes, and styles. Some are simply very small versions of typical voice coil based speakers, while others may use piezo elements or other more elaborate techniques. Sometimes the high frequency driver in a speaker system isn't referred to as the tweeter, but instead a horn or simply high frequency driver. This is generally only the case when a compression driver is mounted to a horn. The reason for this is historically that the compression driver/horn combination was used more for upper midrange frequencies and there was often still a device (or devices) more like a stereo system's tweeter with very small elements reproducing frequencies above that if they were needed. Quite often in large sound system PA applications this very high frequency reproduction was not utilized because it was deemed either unnecessary or too difficult to achieve in large venues. It could require a dozen or more little tweeters to keep up with the output of one good compression driver mounted to a horn, and even then the dispersion characteristics were radically different, which caused lots of phase and interference problems in some listening positions. Nowadays compression driver and horn technology has advanced to a degree that in large PA systems true tweeters can be avoided while achieving good fidelity overall. However in stereo systems and many near field studio monitor systems you will not find compression drivers and horns in favor of tweeters, which most listeners agree produce a softer and more pleasant sound, not to mention the potential for increased linearity at very high frequencies.

Virtual Dolby Digital
Dolby has specifically developed three types of "virtual" surround processing for computers, computer games, and video games. In "virtual" implementations, "phantom" speakers are created, as processing provides perceived sound sources in addition to the actual speaker complement. Virtual Dolby Digital is a computer format implementation of Dolby Digital. For this method, first a Dolby Digital decoder decodes the digital bit stream and 5.1 channel signals are produced. Then, a "phantom" channel is created providing a perceived center channel where none exists, and the two surround channels are processed through an additional DSP circuit and changed to "virtual" surrounds. All channels of information are provided through only two speakers. This system works best for a single listener who is centered between the left and right speakers. In the Virtual Dolby Digital implementation, some computers will decode the digital bit stream via a Dolby Digital decoder with the ability to "downmix" the 5.1 channels into a Dolby Surround encoded stereo signal. These two channels will then go through a two-channel sound card and be processed through an outboard or inboard Dolby Surround Pro Logic decoder to provide four channels of sound -- Left, Center, Right and Surround. The center channel can be switched to "phantom" mode if desired, but four speakers are needed for the left and right front and the two surround speakers at the sides or rear of the listening position.

Virtual Dolby Surround
Similar to Virtual Dolby Digital, Virtual Dolby Surround is another way to use the Dolby Digital bit stream from a DVD-ROM. The first step is for the Dolby Digital decoder to down mix the 5.1 channels into a Dolby Surround encoded two-channel format. The center channel is provided by a "phantom" mode. And the mono surround channel is further processed by additional DSP circuitry and turned into "virtual" surround.

Voicing
A term that has several applications in the music and audio world. When applied to pianos, voicing refers to the methods of obtaining a particular quality of tone from an individual instrument. This function is performed so that the tone of each note is uniform throughout. The adjustment involves the hardening or softening of the felt on the hammers that strike the strings, often achieved by filing down worn felts or replacing them entirely. If you encounter a piano on which certain notes seem "brighter" or "duller" than the others, the chances are good that it needs to be voiced. Voicing is generally done at least every few years to revitalize worn felts. In musical arranging and orchestration, voicing is the process of assigning different notes of a chord or melody to different combinations of instruments in the ensemble. This allows each note to have its own timbre and dynamic level, or "weight," in the overall sound. For example, In a C Major triad (C-E-G) having the "E" played by trombone, cello and bass clarinet will produce a distinctly different timbre than assigning it to French horn, trumpet, first violin and piccolo. In synthesizers, "voicing" is an industry term for applying the architecture of a particular synth to create sounds (presets) that reflect that synth's sonic capabilities. This skill requires a unique combination of technological and musical knowledge. There have been a few "stars" in the synth voicing business, among them Joe Ierardi and Jennifer Hruska (responsible for many signature Kurzweil K2xxx sounds), Dave Smith (besides creating the legendary Sequential Circuits synthesizers he was responsible for voicing the Korg Wavestation, an update of his Prophet-VS) and Eric Persing (a chief sound designer for Roland as well as creator of dozens of independent sound libraries). Finally, voicing is sometimes used to describe the equalization and balancing of a speaker/amplifier system (as in studio monitors or a theater sound system) to achieve the optimum sound reproduction capability.

Voltage Regulator
As the name implies, a device that regulates or controls the voltage in a system. The idea is it can take input from a source of voltage that may vary (such as the output of a typical electrical outlet) and output a constant and steady voltage. High quality voltage regulators can significantly improve the performance and reliability of an audio or video system. In order to regulate voltage a regulator must be able to monitor both input and output voltage and have a method of changing the output. There are many ways to accomplish this. The most common ones involve some sort of transformer where the regulator can switch between different "taps" off of the windings for the different voltages. Even among those there can be vast differences in how this is accomplished. The number of taps used and the timing of the switching can both have a dramatic impact on its overall effectiveness for a given application. For audio it is best to use units that switch taps only at zero crossings to prevent noise and spikes.

Zeppelin
A special type of microphone shock mount and wind screen assembly. It essentially consists of a sort of skeleton/shock mount surrounding a microphone over which some type of foam or muff type material is placed. The purpose is largely the same as a conventional wind screen only these can do an even better job of isolating the mic from the elements while also providing a more natural sound. Typically these are used with shotgun microphones, partly because these mics then to be extremely sensitive to mechanical (handling) noise as well as wind. Furthermore, shotguns are often used in applications where one is already trying to capture a signal that's difficult to get so any extra noise is a real problem. The name comes from the old dirigible as the appearance (and in some respects construction) is similar. They are also referred to as blimps for similar reasons.

Zero Latency
Latency is the time a message takes to traverse a system. For music recorded via computer, latency is major concern. A human playing an instrument, for example, needs nearly instantaneous feedback from that instrument in order to play it correctly. While this is generally not a problem with non-digital instruments, audio routed through a computer always has some delay in the signal path. Latencies higher than 100 ms make working with real-time music programs or instruments impossible, and many musicians find much lower latencies objectionable. While virtually every digital process involves some latency (just converting a signal to digital and back to analog takes some small amount of time) there are some systems where it is much more of an issue than others. Historically host based computer recording systems (ones that don't rely on dedicated audio processing hardware, but use the computer's CPU for instead) have been the worst offenders. A TDM based Pro Tools DAW, for example, has virtually no latency because the computer is merely acting as a host while most of the audio processing is done on the DSP cards residing in the computer. Out of the need for low-latency interconnects, Steinberg created ASIO, a protocol designed for low-latency transmission (on the order of a few ms) of digital instrument and other music data. The term 'Zero Latency Monitoring' was introduced in 1998 by RME with the DIGI96 series of audio interfaces and refers to the technique of routing the input signal directly to the output on the audio card. This has become one of the most important features of modern, host based hard disk recording. Progress is continually being made in lowering the latency of these systems. With ASIO Direct Monitoring (ADM, since ASIO 2.0), Steinberg has not only introduced Zero Latency Monitoring to ASIO, but also extended it substantially. ADM also allows for monitoring the input signal via the hardware in real-time. Over and above that, ADM supports panorama, volume and routing, which requires a mixer (i.e. DSP functionality) in the hardware though. Thus it is possible to copy a routing through a software mixer into the hardware in real-time, so that the sound difference between playback and monitoring is very small. In total, ADM renders a substantial step towards 'mixer and tape recorder inside the computer'. There are similar advancements being achieved with other brands. On the whole zero latency monitoring is a reality now, but there are still some compromises to be made in terms of workflow to achieve it. The only easy way around this is still to go with more costly solutions until processing speeds allow the power and flexibility of dedicated systems to be truly replicated with host based systems.

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