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ADAT Optical
A specific form of optical audio data transfer developed by Alesis for their ADAT machines. ADAT optical uses the same interconnects as the ubiquitous TosLink two channel format, but includes eight channels of digital audio data. ADAT optical was a key cog in the development of the original ADAT back in the early 1990s. Today it is one of the standard digital I/O connections found on many pieces of digital audio gear such as mixers and recorders.
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.
Automatic Mixer
A specialized audio mixer that senses the presence or absence of a signal from a microphone and turns it on or off without requiring human intervention. Automatic mixers can have a single channel or multiple channels.
Most automatic mixer designs are based on a threshold setting that defines the minimum SPL the mixer will recognize as a valid signal. One of two basic approaches is then used. In the first, the mixer operates as a gate and shuts off all signal from the mic, and then reactivates it once the signal level rises above the threshold. In the second approach, the mixer attenuates the signal by 15-20dB, and then gradually increases it as the source volume increases. Fans of this approach point out that it does away with the “choppy” action of gates, which instantaneously changes from full attenuation to full gain.
Automatic mixers are often found in audio-for-video recording, broadcast TV and radio, and houses of worship.
Automation
In audio production automation refers to having things programmed to happen in real time during a mixdown. In the 1970's, when big multitrack tape machines were becoming common, and overdubbing parts became a standard way of working, the process of getting a good mix became exponentially more difficult. No longer was the whole recording of a live performance where the musicians pretty much balanced their own levels. Many components were put in later and eventually it became trendy to do mixes at other studios optimized for that purpose, thereby causing the mix to have to be created from scratch. Anyone who has ever had the occasion to be one of the three or four people huddled over the mixer making adjustments during a manual mixdown can appreciate the benefits of being able to automate most of the process. Early automation systems were basic level controls. They were synchronized to the tape machine by some form of Time Code (not necessarily SMPTE) and would remember any moves the engineer made and then play the data back causing the level change to occur at the proper time (assuming the automation stayed in sync with the tape - not a given). They worked by either having motorized faders, where the motors could be controlled by the automation, or by using VCA's (Voltage Controlled Amp), which was a much less expensive and cantankerous option. VCA's, however, didn't sound as pure as the passive fader with a motor attached so most successful systems were "moving fader" based. Later the quality of the VCA based systems rose (while the cost declined) and they became popular among smaller studios, but moving fader systems are still considered the best choice for analog. Not only because they sound better, but because the tactile feedback of physically moving faders is something many engineers prefer. During the 1980's many other aspects of mixing began to be automated. Things like aux sends, panning, and eventually even EQ and compression could be put under computer control. Nowadays there are many analog mixing boards that are totally under digital control and virtually every parameter can be automated. Further, with the advent of the DAW, complete recall and automation of every aspect of a mix has become a standard.
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.
Bouncing
The process of combining several tracks together and re-recording them onto another track is called bouncing. This is normally done to free up tracks for more recording. For example, you might have three background vocals recorded on tracks 1 to 3. By combining these tracks with a mixer, and routing them to track 5 (for example) tracks 1 to 3 can be erased and used to record other materials. Keep in mind that bouncing does require that you pre-decide on levels, EQ, etc. for the tracks being bounced - once they are combined, it is impossible to adjust their relative levels (overall level and EQ of the bounced track can, of course, still be adjusted).
British EQ
Today's Word for the Day was a special request from one of our inSync readers. British EQ - Can be loosely defined as any equalizer circuit that is designed and built in a way that emulates (in design or sound or both) the classic EQ circuits from the legendary mixers that came out of England in the 1950's, 60's and 70's. Soundcraft, Amek, Neve, Trident, and many other brands achieved legendary status during those years because engineers and producers liked their sound, and in particular liked the performance and sound of their equalizers. During the 1980's and 90's less expensive products began to show up from other parts of the world. British EQ was thus coined as a marketing term that became used by many of the English companies to combat the less expensive products. They felt that by making the distinction that not all mixers and EQ circuits sound the same they could maintain market share even at higher prices. By all accounts the idea worked because there is still today quite a bit of mystique around the concept of British EQ.
Bulk Dump/Load
A MIDI function that allows the transfer of system specific data, such as sample files, program setups, mixer settings, and more between MIDI capable devices. The data is transmitted as MIDI System Exclusive messages. A bulk dump has historically been a common method of backing up and restoring all user data from a MIDI device.
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.
Channel Strip
Refers to one channel of a mixing board. Due to the layout of most mixers - channels in columns across the face with the functions of each channel arrayed from top to bottom - a channel is sometimes referred to as a strip. Over the years some mixers have been so highly regarded for their sound quality that a market was developed by marrying a channel strip to a power supply and I/O connections for stand alone use. One could plug a mic directly into the channel strip and record directly to a recorder without having to take the big mixer to the location. Even when such a mixer is available the channel strip approach is often used because it is thought that the simpler signal path of the external strip produces a more pure audio signal to record. This is similar to the popular approach of using high quality stand-alone mic preamps as the only device between mics and recorders, with the major difference being that a channel strip has many more capabilities than a simple mic amp. It may have EQ, compression, gates, and more, depending upon the unit. The idea is that it's everything you're likely to need in the signal path to make a great recording. Channel strips are so popular these days that they are generally considered to be a separate product category from preamps and other processing.
Control Room
In general this refers to a space - usually an enclosed room, or booth - where the operations of something are handled, the central control point. In radio and television production, this refers to the room that houses the equipment used to bring all the audio and video signals together into a composite signal that's broadcast or recorded. All the different cameras and microphones are fed into video switchers and audio mixers here. Similarly in theater applications, this is generally where all the audio signals are mixed, additional recorded sound effects may be added, and where the lighting is controlled (though these may be in separate control rooms). In theater it is sometimes referred to as a "Bio-Box," which comes from the Greek word "Bios," or Way of Life. In a recording studio, the control room has a similar function. It's where the engineers and producers sit and take care of making sure good signals get recorded as well as controlling, in many cases, what the band hears during a performance. Ideally, control rooms are designed to be carefully regulated in terms of sound isolation and accurate sound reproduction, as this is where the final decisions are made about how a recording will sound.
On some audio equipment - typically mixers - there are control room outputs and associated control room level (volume) and mix controls. This pertains to sending signals to the control room speakers, which are usually a specially selected set of very accurate speakers designed to enable producers and engineers to hear a true reference of the audio signals being recorded and mixed. In some cases these speakers are custom designed to properly react with the control room space. In other cases a control room space may be built with a specific set of speakers (and other equipment) in mind.
Control Surface
In the music and production world a (hardware) controller is something we use as a human interface to other elements in a system. For example, a keyboard controller is used to play keyboard parts, where the performance data is transmitted to a device that produces the sound, whether it's a rack mounted module, a software synth, or another keyboard. A control surface is conceptually a more generic form of controller. They come in many shapes and sizes with (in some cases) radically different capabilities, but the thing they have in common is that they are used to control the functions of some other device, often a computer software program. In our business the words "control surface" usually conjure up images of something looking like a mixing board. These aren't actually mixing boards, but instead devices used to control other devices, which perform the functions of a mixing board (mixing, aux sends, panning, EQ, etc.). Now that so much production is done inside of computer software, it has become increasingly important to provide tools that enable musicians and engineers easy access to a familiar set of controls in order for them to most effectively be able to do their work. As such, control surfaces in many ways mimic the look and feel of a mixing board, even though in many cases they may provide more or different capabilities. Some control surfaces are designed specifically for a specific computer or software system, while others are more generic and may work with a variety of different systems. Nowadays many stand alone mixers are really nothing more than software based mixing boards under the control of a dedicated control surface, even though the outward appearance is that of a mixer. In some cases these mixers can also be used to control other software mixers.
Control Voltage
Sometimes abbreviated CV, Control Voltage is a DC electrical signal used to manipulate the values of components in analog circuits. Control voltages are used in numerous ways in many different types of electronic circuits for all sorts of purposes. A few examples germane to music technology:
If you send a specific electrical voltage to a module of an analog synthesizer (such as an ADSR envelope generator), you can specify what you want the module to do (perhaps lengthen the decay time).
In a mixer and other audio circuit that uses a voltage-controlled amplifier (VCA), a DC voltage can be used to set the gain of the VCA. This is applicable for things like compressors and gates, where the DC voltage may either be a signal proportional to the audio level the devices are acting upon, or could be from some other source. But it also applies to VCA-style mixing automation systems, where a control voltage is set by the user (via some interface, but usually faders) to determine and subsequently log the desired levels in the mixer at specific points in time.
In modern keyboards and synthesizers a control voltage pedal (or other controlling device) can be used to manipulate certain parameters, which, in the case of a pedal, would leave the player's hands free to play the keyboard. In these cases the pedal usually doesn't generate the voltage. Instead the pedal is attached to a potentiometer, which acts as a variable resistance to a circuit providing the voltage inside the keyboard. The circuit is able to "read" the position of the pedal by how it acts on the circuit and uses that information to provide data to the specified parameter.
COSM
Abbreviation for Composite Object Sound Modeling. COSM is a powerful modeling technology that Roland premiered in 1995 with the VG-8 V Guitar System, and continues in the newer VG-88 system. It enables guitarists to emulate a range of classic and modern guitars, amps, cabinets, and microphones, plus it can produce "futuristic" synth-like tones. Today COSM can be found in keyboards, digital recorders, mixers, etc. It can model rotary effects, different speaker colorations, and can even approximate expensive microphones using just an ordinary dynamic mic. Its name comes from "composite object" because its core function revolves around breaking audio producing or reproducing devices down to their component parts and creating a set of instructions to emulate how these various parts interact with each other to produce a new composite that can be dynamically controlled. Of course, that's what all modeling is, but Roland coined this name to call attention to it.
Crosstalk
In multi-channel audio systems, crosstalk is signal bleeding or leaking from one channel to another. Mixers, tape recorders, and many other pieces of gear are all susceptible to this problem. In most modern gear, crosstalk is not a major concern, but be aware that older gear can have significant amounts of bleed between channels!
DIP Switch
Acronym for Dual Inline Package Switch. A compact electric switch with a slider that may be set to "on" or "off." Because of the small form factor, arrays of multiple DIP switches can easily be incorporated into one package. This type of switch is designed to be used on a printed circuit board along with other electronic components and is commonly used to customize the behavior of devices such as mic preamps, equalizers, mixer inputs, and other specific applications. DIP switches are often used instead of jumper blocks in situations where a setting may need to be changed more than once.
Direct Box
Often abbreviated DI (for direct insertion), a device commonly used to convert high impedance (see WFTD archive Impedance) unbalanced signals from a stage instrument (like a bass or keyboard) into a low impedance balanced signals. This puts the signal at the proper voltage level for the mixer and prevents the instrument from becoming loaded down with too low impedance, which could cause tonal shifts and distortion. It also allows the signal to be transmitted over long lengths of cable. They are always used in live sound to get a signal out of an instrument on stage out to the mixing board, which can be as much as 500 feet away. A simple direct box consists only of a small transformer, but more sophisticated designs employ electronic gain stages that more closely resemble the input section of a modern instrument amp. They may also have some combination of ground lift switches, equalization switches, level matching switches, isolated line outputs, and more.
Direct Out
A patch point found on many mixing boards for taking an individual output out of a channel as opposed to routing its output through some bussing architecture. Direct outputs are not all the same though. Some take their signal right after the mic preamp and before the EQ or other gain stages, some may take the signal after the EQ, but before the channel fader, and some are taken after the fader. Many mixers provide a method of switching where the direct out signal comes from on a channel-by-channel basis.
Discrete
Constituting a separate thing, with unconnected distinct parts. In the world of audio, discrete is generally used to signify separate components. This may on the level of describing the system, where one might refer to the mixer and recorder as discrete from one another, but usually this term is used at the electronic component level. In a mic preamp, for example, discrete components is understood to specifically mean that no IC's are used. Each element of the preamp is a separate transistor (or tube) stage with its associated discrete components (capacitors, resistors, etc.). Conventional wisdom suggests that discrete designs are better, but like anything else in modern electronics this really depends on a number of other factors. All other things being equal discrete designs do give the designer more flexibility, which can obviously yield favorable results.
Double Bussing
As far as we know this term was coined by Mackie, but represents a fairly common feature found on modern recording boards. Double busing is like having "Y" cables built right into the mixer. It allows two (or more) multitrack inputs to be fed from one subgroup output. For example, on an 8-bus mixer, when you send signal to subgroup #1 output, it will also send signal to a #9 output. There is no actual subgroup #9, but there is an output that stays tied to subgroup #1 and is controlled by that fader. This allows the recording engineer a convenient way to route grouped signals to more than one destination, such as when using an 8-bus mixer with 16 or 24 track recorders. You simply put the tracks you want to record in RECORD mode, and the other tracks (not in record mode) ignore the signal.
DXi
Abbreviation for Direct X Instrument. A platform for virtual synthesizer and sampler plug-ins that integrate with sequencer programs using Windows directX drivers. These instrument plug-ins are launched from within the sequencer and can be played via an external MIDI source or from recorded MIDI tracks. They can also be subsequently patched through effects plug-ins available to audio tracks of the sequencer from its mixer window.
Effects Loop
A signal path out of one piece of gear, through an effects unit, and back into the first device. It effectively is a loop, with an effects processor in the middle. When you send a signal out of a mixer on an aux send to a reverb, and then bring that signal back to the mixer you have created an effects loop, though we rarely call it that in those circumstances. Most of the time the verbiage is used in the context of guitar or bass amps, or guitar or bass preamps. Many of these have a dedicated insert point designed to be used with some outboard processor. In most cases they work just like the insert of a mixer: plugging something in breaks the internal signal routing to send the signal through the external loop.
ESB
Abbreviation for Emagic System Bridge. The ESB is a software driver that serves as a link between Emagic's native software and Digidesign's TDM hardware. As such it allows users to be able to bring native processes (plug-ins and software synths) running on Logic Audio into the TDM mixer environment. It consists of two components: Direct TDM and EXS24 TDM.ESB TDM allows the insertion of up to 32 instances of Emagic's Xtreme Sampler 24 Bit within the Aux channels of Logic Audio's TDM mixer. The output signals of inserted EXS24 instances can be further treated, utilizing all of the possibilities of the TDM DSP environment. Each EXS24 instance is handled by the computer's CPU, and the ESB TDM routes their output signals into the TDM DSP's. EXS24 MIDI performances are recorded on TDM Auxiliary tracks and are controlled directly in Logic Audio. This eliminates the need for OMS, making playback of the EXS24 TDM sample-accurate.Direct TDM provides an additional audio engine running in parallel with DAE/TDM. It works like most native processing engines and offers up to 64 audio tracks, plug-ins, the use of VST effects and integration of Emagic or third party VST 2.0-compatible Audio Instruments. ESB provides up to eight outputs from this native audio engine, which can be streamed into Logic Audio's TDM mixer - all within the computer.
Feedback
Literally the return of a portion of the output of a process or system to the input. In our discourse (of audio and video production) we mostly encounter feedback when an open microphone is picking up sound from a nearby loudspeaker that is also being used to amplify sound from the same microphone. This forms what is known as a feedback loop. The sound of the room enters the microphone and is then amplified by the speaker. This amplified sound then becomes part of the sound of the room entering the microphone, which causes it to get amplified by the speaker again. If too much of this "feedback" occurs the signal will "run away" and quickly degrade into an oscillation at some frequency. This sound is the "squeal" we've all come to know and hate and is what we typically call feedback (though technically feedback occurred well before the squeal happened). It is also possible to produce electronic feedback. Routing the output of a mixer or effect unit back to its input is a sure way to do this. In fact, many effects are based on using this phenomenon creatively, the most obvious one being an echo with multiple repeats. Feedback and "feedback loops" are also used in all kinds of electronic circuits to achieve specific results. Old analog oscillators are based on electronic feedback.
FOH
Abbreviation for Front of House. This distinction comes from theater work where Front of House is any part of the theatre in front of the proscenium arch. In concert and PA applications it is generically used to describe anything in the audience area. Specifically the main, or house mixing position is referred to as the FOH position, which is meant to differentiate the main house mixer from the monitor mixer normally located to the side of the stage.
Foley
The art of recreating and dubbing incidental sound effects (such as footsteps) in synchronization with the visual component of a movie. Foley requires the work of foley artists (who make the sounds) foley editors (who edit them), and foley mixers (who make a composite mix or stems). Foley artists sometimes use bizarre objects and methods to achieve sound effects, e.g. snapping celery to mimic bones being broken. The sounds are often exaggerated for extra effect - fight sequences are almost always accompanied by loud foley added thuds and slaps. The process is named after Jack Foley (1891-1967), pioneering sound effect editor at Universal Studios in the 1930s.
Gain Structure
When multiple pieces of electronic audio (or video) equipment are used together, the gain structure of the system becomes an important consideration for overall sound quality. This basically refers to which pieces are amplifying or reducing the signal how much. A properly set up gain structure takes maximum advantage of the dynamic range and signal to noise ratio of each piece in the chain. No one piece is doing a disproportionate amount of the amplification unless it is a piece designed for that function (such as a mic preamp). An example of poor gain structure would be a setup where a mixer's master fader is near the bottom, while all of the individual channel faders are near the top. The resulting level out of the mixer is the same as it would be if all faders were at some mid setting, but the chances of distortion are much higher because of limited available headroom in the circuits preceding the master fader, while the S/N ratio of the final output isn't as great as it could be were the master fader at a more appropriate level. Part of assembling a system with good gain structure is making sure all the pieces can operate at the same reference level. This is where people go wrong combining -10 dBV equipment and +4 dBu equipment. It can work under the right circumstances, but sometimes the resulting gain structure severely compromises the signal to noise ratio of the final result (or in some cases causes it to be distorted). Gain structure must be considered to optimize any system where levels can be adjusted in more than one place.
Graphic Equalizer
A type of EQ that is configured to provide a graphic display of the EQ settings. Years ago equalizers were all rotary knob based. When units began to arrive on the scenes that had 15, 30, or even 45 bands (frequencies) they could EQ at once it became difficult to see what was going on at a glance. Looking at a row of 30 knobs to get an overall idea of the EQ curve is pretty difficult. So equalizers that used sliders instead of knobs were developed and quickly won the favor of engineers due to their improved ergonomics. People liked how easy it was to see the overall EQ curve at a glance, but they also just liked using sliders more than knobs (something that we'd already figured out about mixers). The ONLY thing that makes an EQ "graphic" is this configuration of being able to see the curve at a glance. Contrary to popular belief there are graphic EQ's that have the same features as parametric EQ's, including Q controls and sweepable frequencies. Most graphic EQ's, however, only give you control of cutting or boosting a pre-selected set of frequencies at a pre-selected Q.
Half Normal
In patch bay jargon this refers to a normal that is only broken by inserting a plug into one of two plugs that are normalled together. In typical patch bay design the top jack of any pair is usually not affected by having a plug inserted into it. It will still route signal down to its lower counterpart. When a jack is inserted into the lower plug, however, the normal connection is broken. This provides a convenient way to route signals to multiple destinations. For example, the output of a mixer that is normalled to the input of a DAT on the patch bay can also be simultaneously routed to another patch point. To do this one would simply run a patch cable from the patch point that is the output of the mixer (an upper jack) to the patch point that is the input to the other device (a lower jack). This connection will break the normal of what would normally be feeding that other device in favor of the mixer signal that has been patched in. Signal will now be routed to the DAT and the other device.
House Sync
Refers to a distributed synchronization signal available to all recording equipment in a studio. In modern production environments it's important to have all digital and/or video equipment in a studio synchronizing its clock or frame rate to a common, stable source. This makes synchronization among different recorders (such as video and audio machines) much easier and more consistent, plus it enables material to more easily be transferred from one to another. Of course, in digital systems good word clock must be transferred between every digital device (mixer, recorder, DAW, CD recorder, etc.) to keep the signal in tact. The same results can be achieved, however, if all devices are synchronized to a common clock. Historically house sync was just another word for video sync or black burst, which have been common in video houses for years. However, with digital equipment becoming so widespread most studios are now distributing word clock signals instead of, or at least in addition to, black burst. In studios where video and digital equipment is integrated, both the word clock and video frame rate signal (black burst) must be resolved to one another. Of course, if there are any analog tape machines then standard LTC (or sometimes VITC for video machines), usually in the form of SMPTE, must be used, and that too should be resolved to the video and/or digital clock rate. Many modern house sync generators and distribution systems now deliver digital word clock and black burst.
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.
Insert
A point in the signal path of a circuit where it is possible to interrupt the signal and "insert" another signal. Most commonly deployed in mixing boards it usually shows up as a patch point on each channel and/or bus output. The purpose is to be able to interrupt a signal in the mixer, bring it outside the mixer for some sort of special processing, and then return the processed signal to the same point from which it left. Common applications include applying compression, gating, or EQ to a particular channel without affecting any other channels or using any extra buses. Sometimes an insert is called a "patch" or "injection point."
Line Input
On mixing boards this is an input to a channel that is specifically designed for line level signals. Unlike the XLR microphone input, which is designed for low level mic signals, line inputs are usually 1/4 inch connectors, and are quite often unbalanced, though this will vary depending on the mixer. Line level signals are usually much higher than typical mic level signal and do not need as much amplification to be dealt with by the rest of the mixer. As such, on some mixers, the line inputs actually bypass the microphone preamp stage providing for a pure signal path into the board. Regardless of this, however, line inputs are always capable of handling higher level signals and high impedance signals better than the XLR mic input.
Linear
Of or relating to a line. In audio, linearity is a very important concern. A linear system is one whose function is even or, if it deviates, does so in a steady fashion. Logarithmic or exponential deviations are, by definition, non-linear. We've all heard the term "flat" used to describe frequency response. Flat is considered good. It is also linear. On a graph it looks like a straight line instead of a wavy line. A wavy line would imply non-linear frequency response, or one that is uneven. This is generally not considered a "good" quality in audio equipment, though there are plenty of exceptions. Linear does not necessarily mean horizontal though. A linear system can also be one that gradually increases or decreases in a steady fashion. Such is the output of a crossover or some EQ filters: a 12 dB per octave roll-off (which is common) is a linear decrease in level by 12 dB per octave. The actual sound level increase produced by a typical (audio taper) fader on a mixer is, however, non-linear. It is designed as a logarithmic taper to more closely match the non-linear sensitivity of human hearing. We hear the change, however, as linear in volume over their range (if they are good ones). Basically, if whatever we are interested in can be plotted on a graph as a line that doesn't curve it is considered linear.
M-S Stereo
Abbreviation for Mid-Side, a method of stereo miking and recording. MS recordings capture the relative intensity of different sounds across the stereo soundfield. In order to make an M-S recording one must deploy a cardioid pattern mic facing the sound source(s) and a figure 8 pattern positioned sideways to the source. The figure 8 mic is connected to two channels of the mixer, with one channel having its polarity reversed. Each of the two signals (one of which is polarity reversed) of the figure 8 mic, when combined with the signal from the cardioid mic produces either a left or right "image" that is roughly equivalent to two cardioid mics positioned with a 90 degree angle between them. The only advantage to the MS method is the user can alter the width of the stereo image by varying the relative levels of the two microphones. There are several disadvantages, most of which are a function of having two dissimilar mics reproducing the same signal. Of course they can't occupy the exact same space either, which produces other phase and frequency response anomalies.
MAS (Motu Audio System)
A plug-in engine developed by MOTU for use with their DAW software (Digital Performer), offering real time audio effects use and manipulation in a manner similar to the use of auxiliary sends on an outboard mixer. However, instead of using external processing, the DSP is done by the host computer and never leaves the digital domain. Supporting Macs only, MAS isn't interchangeable with any of the other plug-in engines and will only work with MAS-enabled software. MOTU also has a third-party developer program for MAS, which as a plug-in platform has developed a niche for Mac/MOTU users.
Matrix Mixer
A type of mixer that allows any input to be routed to any output. This is accomplished through a relatively elaborate bussing system whereby each output is driven by n number of level controls (where n in this case is equal to the number of input channels). With a modern matrix-mixer, not only can you assign any input to any output but you may add EQ, compression, etc. This is basically how a typical stage monitor mixer is designed. Though it is increasingly common to find matrix features on house mixers (especially those used in theater environments) to supply alternate (optimized) mixes to different areas of the venue. Very elaborate models exist with as many as 60-channels in and 24 or more output channels (and are as big as a Volkswagen).
Meter Bridge
A structure containing a bank of meters that is placed or attached to the top of a mixing board or tape machine. In some cases the meter bridge is an integral part of the mixer or machine, however, a number of mixers and tape machines provide only limited metering capabilities as a standard feature, but have optional meter bridges that can be attached for more comprehensive metering.
Midrange
Literally the middle range of something. In audio we use this term to loosely define the middle range of audio frequencies. It doesn't have a precise definition in terms of which frequencies are included, and is consequently not used that often in pro audio discussions where specific information is required, but it is acceptable as a way to get across the general idea. It's not uncommon to find mixers that have a three-band equalizer on each channel, for example. The bands are often labeled low, mid (range), and high and represent long standing paradigms of two and three band EQ circuits on hi-fi systems pro audio equipment.
Mix-Minus
A specialized matrix-mixer where there is one output associated with each input that includes all other inputs except the one it is associated with. (The output is the complete mix, minus the one input.) In this manner, the simplest mix-minus designs have an equal number of inputs and outputs (a square matrix). For example, if there were 8-inputs, there would be 8-outputs. Each output would consists of a mix of the seven other inputs, but not its own. Therefore Output 1, for instance, would consist of a mix of Inputs 2-8, while Output 2 would consist of a mix of Inputs 1 & 3-7, Output 3 would consist of a mix of Inputs 1,2 & 4-7, and so on. Primary useage is large conference rooms, where it is desireable to have the loudspeaker closest to each microphone exclude that particular microphone, so as to reduce the chance of feedback.
Mute Group
Most professional mixing boards provide some method of muting individual channels and/or subgroups. An extension of this feature is the mute group, which allows the user to set up master mute buttons that will enable the mute function on specific groups of channels. For example, the user could have the board set up so that pressing "Mute Group" #1 causes the background vocal mic channels, rack tom channels, horn mic channels, and the banjo mic channel to mute. This is much more convenient (and faster) than hunting down each of those items and muting them one by one. And since they are surely routed through different subgroup paths in the mixer this is the only convenient way to get to them all at once. Effective use of mute groups gives a sound engineer a basic level of automation when mixing, which paves the way for more attention to more important matters.
NOM
An acronym of Number of Open Mics, believed first created in 1967, or 1968, by Bill Snow after he retired from Bell Labs and went to work at Altec Lansing Research. Dan Dugan, the father of the automatic microphone mixer and Altec Lansing, the manufacturer of his first design, popularized its use. In Dan's original design, the automatic mic mixer, like human operators, turned the gain down on unused mic channels and turned the gain up on active channels, all the while ensuring that the overall level remained roughly constant. As a rough approximation, each doubling of the number of open mics (NOM) cuts the gain by 3dB, i.e., as more mics are opened up the mic mixer reduces overall gain. If not, as mics open and close, the reverberation and ambient noise fluctuates unacceptably. NOM attenuation techniques work to provide the gain, stability, and low noise qualities of a single open mic with the benefits of multiple mics. The NOM acronym is still used in some sound reinforcement circles.
Normal
- Corresponding to the usual state, not out of the ordinary.
- 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...)
- 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.
Op Amp
Short for Operational Amp, a circuit component used in all sorts of equipment. Though they are technically considered amplifiers they are quite often used in circuits that do not obviously "amplify" signals. Examples would be equalizers, crossovers, compressors, mixers, microphones, keyboards, effects and many, many, many more (the list is endless). Op amps acquired their name from early uses in analog computers (computers perform operations, get it?). They can exhibit very high gain and are extremely easy to build into audio circuits. Nowadays they are available in integrated circuit chips, each of which may have many op amps inside. In some cases they are literally a dime a dozen.
Parameter
Literally a factor that determines a range of variables. In our domain we come across parameters all the time. Any function of any device we can set or modify can be thought of as a parameter. This would include the volume of your guitar amp or the pan position of a channel in your mixer. More specifically, however, the word parameter has been used in things like keyboards, synthesizers, effects processors, and other software driven devices to call attention to specific aspects of any program or patch that can be modified. For example, even on simple non-programmable reverb units quite often the user can adjust the parameter known as Reverb Time. More complex and flexible devices typically have more user adjustable parameters. Some devices over the years have been thought of as too complex because too many parameters have to be understood by the user in order to make anything meaningful happen. This is one reason why in today's sophisticated equipment we rely so much upon the factory presets to get us started and why many users never go beyond them.
Patch Bay
A device used to make it easier to patch into various points of an equipment setup. Patch bays are comprised of several jacks that are wired to the equipment in a studio or live system. All of the 'patch points' one needs access to in the system show up at the patch bay, where they can easily be routed one place or another by plugging short lengths of wire between the points on the bay. The idea comes from the old telephone days where phone operators manually patched connections together so your mom could talk to aunt Mable. Similarly when you want to insert a compressor on that out of control bass track you can simply do it at the patch bay instead of having to crawl around under or behind the mixer and tape deck. Patch bay points can be wired in a variety of configurations including balanced and unbalanced, and with functionally different modes of operation such as Full Normal, Half-Normal, Open, Mult, and others.
Peak
Generally the highest point. In audio this refers in various ways to the maximum audio signal. A sine wave has two peaks per cycle, which represent the maximum or peak amplitude or voltage (one is maximum in the positive direction while the other is maximum in the negative). Complex musical signals have peaks, which represent the loudest sections or moments. Transients in musical material are also referred to as peaks, though they are really a specific type of peak that has a very short duration of time. On a waveform display such as an oscilloscope audio peaks often look like the peaks of a mountain.Peaking is a word that is sometimes used to describe audio that has gone beyond some specified reference. For example, when the peak LED illuminates on a mixer or recorder the signal can be said to be peaking. This is closely associated with overloading, distortion, and clipping.
Phantom Power
A DC (direct current) voltage, usually 48 volts, applied to pins 2 and 3 referenced to pin 1 of an XLR microphone connector that can be used to power transducers with active electronics. Condenser microphones require a pre amp close to the very high impedance (See WFTD archive "impedance") diaphragm which requires power to operate. Back in the '50's and '60's this power was often provided by a separate power supply that came with the microphone. Later manufacturers began to provide a source for this power at the microphone input to mixers or pre amps. Since the power is carried on the same wires that carry the audio signal, and since most dynamic microphones and other passive devices are not affected by this DC voltage it was known as "phantom" power. The theory was that only devices that needed it would be wired in such a way that they would use it. Nowadays almost all condenser microphones and active direct boxes are able to use phantom power when it is present on a microphone cable. Consequently most mixing board manufacturers include this feature in their products.
Print
Most computer users (or people who have used a pencil and a piece of paper) know what this means, but it has a specific meaning for audio production as well. Printing something in audio and video refers to recording it, as in "printing to tape." The context in which this comes up is centered around sources and signals that may not normally get recorded to the multitrack tape in a project. An example of this could be sequenced MIDI parts that are often synchronized and flown in to a project as virtual tracks. Another example is effects that are normally returned to an auxiliary channel on the mixer and mixed in with the recorded tracks. Sometimes it is useful to actually record these things to the multitrack tape (or disk in the DAW world). Let's say you are moving a project to another studio for some overdubs, but don't want to carry the entire keyboard rig there and mess with getting all the instrument levels set in another room. You could just print a rough mix of all the keyboard parts to tape and use that as a reference for the overdubs.
R-Buss
Presumably short for Roland Buss. R-Buss is a connectivity scheme (or bus scheme) developed at Roland for interfacing their digital mixer and recorder products with one another. The idea is for one cable to carry all necessary signals between units so an R-Buss cable can carry eight channels of digital audio with word clock as well as transport and location information. Two 8-track Roland recorders can thus be connected to each other with one R-Buss connection, and they can each be connected to a compatible Roland digital mixer with one R-Buss connection each. Roland also manufactures interfaces for ADAT and other popular MDM and DAW formats so they can be integrated with a Roland mixer using R-Buss.
Rack Rail
Aside from rack rails, dragsters, and tracks used in railroads the term rail is used in reference to power supplies, though its usage does come from our common sense understanding of the word. A power supply rail is in some devices (especially older ones) literally a metal rail that passes through the device to deliver power to the components. A voltage is applied to the rail from the power supply and various components are connected to it for their power. Usually there is more than one rail, and each one delivers a different voltage and has different components tied to it. If you crack open an old power amp or mixer you stand a pretty good chance of seeing a couple. In most modern and/or low power devices there aren't any actual rails, but the concept lives on. When you hear the phrase 'power supply rail' what is being communicated is some type of bus with a set voltage applied to it for the sake of powering components in the device. The connection may be delivered by wires or traces on a circuitboard.
Recap (or Recapping)
In audio (as opposed to tires) this refers to the process of replacing capacitors in equipment, usually a mixing board. Analog mixers employ hundreds or even thousands of capacitors throughout, and the quality of many of them can have a direct impact on the sound quality of the mixer. There are a number of different types and configuration of capacitors, some of which age better than others or otherwise have different characteristics that make them more or less desirable to use in a given design. In an older mixing board it is possible that some of its capacitors will begin to break down and become unstable. This can produce a wide variety of symptoms including phase shift, filtering, and distortion from DC leakage.Aside from the obvious need to "recap" a mixer as a result of its capacitors beginning to fail, some audiophiles claim that certain types or brands of capacitors can make a huge difference in the audio quality of any mixer, and will therefore recap "perfectly good" mixers to achieve the desired results. This is a practice that dates all the way back to the first production mixers to be built and still continues to this day and is where the mystique of "recapping" a mixer originates. The idea being that it is hot-rodded or otherwise improved from stock designs can add big dollars to the sale price of such a mixer, or to the hourly rate of studio time.
Receiver Image
A second transmission frequency that a superheterodyne receiver will respond to. The image frequency is two times the IF frequency either above or below the carrier frequency, depending upon whether the receiver design uses "low side" or "high side" injection. An RF signal on the "image" frequency of the receiver will produce a difference signal in the mixer just as valid as the intended IF signal created by mixing the oscillator with the carrier.
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.)
Ring Modulator
A type of audio mixer combining two audio signals, and outputting their sum and difference. The frequencies found in the original signals are not passed through to the output. For example, if two sine waves (single frequency waveforms containing no overtones) are inputted, one with a frequency of 1000 Hz, and the second at 400 Hz, the ring modulator will output two frequencies: 600 Hz and 1400 Hz. With more complex waveforms (which contain many more overtone frequencies) ring modulators produce a clangorous, "metallic" result often used for special effects, in synth programming, and so on. One popular use has been to process vocals, which produces sci-fi sounding "robotic" voices.
Scene
A "snapshot" of parameter settings for some device that can be saved and retrieved later. In many devices these may be called snapshots, configurations, or setups. The "scene" terminology comes from the theater industry, where technical issues are often oriented around how they apply to performances. For example, lighting controllers often use the scene terminology because it both refers to scene as it relates to vision (something you see), and because different sections of a performance are broken in to scenes. Different scenes generally require different lighting conditions, so lighting controllers (and lighting directors) have historically used scenes to denote their different setups. Mixers also use this terminology quite a bit for similar reasons. A mixer may store different scenes or configurations in memory, particularly with digital mixers, or mixers that have some digital control. A scene can be quickly recalled, re-establishing all of its settings instantly. This information includes parameters such as fader levels, effect sends levels, effect types, panning and more.
Scribble Strip
A portion of the front panel or user interface of some device allocated for handwritten notes. A great example pertinent to the audio industry is the area on audio mixing boards designed to accommodate notes about what is on a given channel. This space is usually a strip (hence the term “scribble strip”) that runs the width of the mixer just above or below the faders. This is the space sound engineers use to name the channels (bass, kick, snare, GTR, etc.) for quick identification. On more modern mixers and control surfaces scribble strips have become electronic displays. The names of the channels are saved with setups and can be recalled accordingly. The virtual equivalent of scribble strips are also part of computer based DAW systems.
Send
An output on an audio device used for routing signal to an external device, such as a reverb, delay, or other processor. Typically, sends are paired with returns, which accept signal coming back from the output of the processor. The more sends a mixer has, the more flexibility you will have in routing signals around your studio.
Session
Originally derived from the term "recording session," referring to the period of time when all the musicians were in the studio, the engineer was at the mixing console and the tape was rolling. Multitrack recording fragmented the original sense of this term by breaking complete "sessions" into individual segments in which only vocals, or drums, etc. were recorded at one time. Since then this word has been given different meanings by different audio software and hardware developers. Here are just a few examples.
In Pro Tools, a "session" refers to the master document that Pro Tools creates when you start a new project. The session file contains maps of all elements associated with a project, including audio files, MIDI data, and all your plug-in effects, edit and mix information. Included in the session file is a play list, which defines groups of regions arranged on each audio or MIDI track.
In the CD world a "multisession" CD is one on which individual tracks or files have been burned to the CD at different times - also known as track at once burning. The Orange Book CD standard defines this process. But a multisession CD can be "fixed up" - with a table of contents written - once all necessary material is recorded on it. If it's an audio CD it can then conform to the Red Book CD standard - which is what all commercially produced CDs conform to.
On the Mackie dXb series of digital mixers, "session" is applied to a collection of fader and knob settings, plug-in effects instantiations, and any and all automation data performed by the mixer during one song, or one recording period. Note that no actual audio data is included in this definition. Mackie uses a "template" to describe preset initial fader, knob and effects values that allow you to begin a session with your preferred settings already in place.
Sidechain
inSync reader Shawn E. in Tokyo wonders what a sidechain is, and how it is used - A sidechain (sometimes called a key input, or a detector input) is a control input used to trigger a compressor or gate with an external signal. Let's look at a common example, ducking: When recording voice-overs, the background music bed is run through a compressor, which is set so that it is not normally operating on the input signal. The voice-over announcer's mic signal is split so that it feeds both the mixer's input, and a sidechain input on the compressor. When the announcer speaks, their voice goes to the sidechain, where it tells the compressor to start working, turning down the background music. When they stops speaking, the sidechain tells the compressor to stop working, and the music comes back to its uncompressed level. Other uses? Try using a kick drum to trigger a gated bass synth for extremely tight rhythms, or insert an EQ'd signal into a sidechain, making a compressor more or less sensitive to certain frequencies (de-essing is a good example of this), many other applications are possible - feel free to experiment!
Signal Path
Simply the route a particular signal takes through a chain of equipment and/or electronic components on the way to its destination. When we think of signal paths in audio we are usually thinking about connecting different pieces of equipment together and routing some signal(s) through them. An example of this would be something like a microphone to mixer to speaker or recorder setup. The signal path has the signal from the microphone pass from the microphone through those (and potentially other) devices on the way to being recorded or amplified (or both). But there is also a signal path inside each piece of equipment. A mixer may be configured to route signals in different ways internally bypassing or utilizing different gain stages along the way to achieve different results. Effects processors often have highly configurable internal signal paths depending upon what they are doing.
Snake
An interesting word. Aside from the obvious reptilian reference it also is used to characterize individuals who are not trustworthy, plumbing equipment, and as an economic reporting method for currency, among other things... and those are just the nouns. In the audio visual industry snake refers to sets of cables that are all contained within one larger cable. The most common application we see is the need to get many audio sources from a stage to the FOH where the signals are mixed and prepared to be amplified through a PA system. Rather than run dozens of individual cables over this long length it is more convenient to use a snake. At the stage end there is generally a box with XLR connectors where each source is connected. At the mixer end there is generally a breakout where the individual wires or lines can be plugged into the console. Similarly in the studio, snakes are used to connect many inputs and outputs between the mixer and multitrack tape machines. The list of examples is endless. Historically snakes have always been configured so the individual wires of each line are contained individually within the larger cable. These may be segmented in groups and, occasionally, may share a common ground wire, but each signal has its own wire. Now we are starting to see digital snakes where many signals are multiplexed together and transmitted on just one or two wires. At the time of this writing these systems are quite expensive, but as more and more audio is handled in the digital domain it is likely that these types of solutions will become more cost effective.
Snapshot Automation
Snapshot automation means simply recalling the set-up of the mixer. This includes the status of volumes, pan, mutes, EQ, aux, effects and dynamics settings etc., plus in some cases, the ability to fade the volumes of one snapshot into another. Most automated consoles and DAWs will be able to store a number of automation snapshots in their internal memory that can be called up at any time during a mix as part of the normal automation process. Snapshot automation is useful for a complex mix that requires numerous changes to happen instantaneously, such as audio post-production for film, where scenes change constantly. It is also sometimes used to store initial setup conditions that an engineer may prefer to use when beginning different types of sessions.
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.).
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.
Subgroup
In audio, a subgroup is a group of sources (channels, tracks, etc.) that are combined into one bus on a mixer. The term is very closely related to, but not the same as, submix. Subgroups are created for a variety of reasons, most of which have to do with convenience. For example, it is very handy (especially when mixing live) to have sets of common sources grouped together into a subgroup so they can all be controlled with one fader. It is common to create drum sub, a keyboard sub, a backing vocal sub, etc.
Submix
In audio a submix is a grouping of instruments or tracks that are mixed or inserted into the "main" mix as a composite signal (usually as one or two inputs). Submixes are used in many different types of situations and can take many different forms. Most mixing boards have a number of subgroups that can be used to help the engineer organize different kinds of sources. Drums may be on one subgroup (sub, for short), while vocals are on another. This can make it much easier to control groups of instruments without having to make identical changes across many channels of a mix. Submixes are also handy when there aren't enough channels or recording tracks to handle all of the sources. In those cases a separate mixer is employed to "pre-mix" a group of sources down to a one or two channel composite that can then be brought in to the recording or main mix.
Sum/Summing
In audio, summing refers to combining two or more signals together. This can be as simple as using a Y cable or as relatively complex as a multi-channel audio mixing board. In analog mixers, signals are usually summed together by sending them from their source - usually somewhere after the channel fader - to a common bus, which in turn feeds another gain stage. Consequently, the integrity of the bus, the devices sending signals to it, and the device it feeds are all an important part of the overall sound quality. It is assumed that signals should be summed in phase and with minimal distortion. In digital audio systems, summing is done mathematically. For each sample, the numeric values of the signals being combined are added together, and the resulting value represents the combined or summed signal. Prior to this, the individual signals undergo any necessary DSP processes that may be set up on their channels in the system. These could range from simple volume adjustments (analogous to fader position in analog systems) to complex compression, normalization, or reverberation routines.
Summing Resistor
A component inserted into the signal path of a mixer or other device to prevent unwanted interaction ("crosstalk") between individual input signals at a summing stage. Resistors are made of material that opposes the flow of current, and a summing resistor is no different; its name merely denotes its specific application. The way it works is that all of the individual signals are sent through a summing resistor and then, on the other side of the resistor (which provides the isolation) the signals are simply combined at a common point (a bus), where they become one (mixed) signal that can be amplified and further dealt with. While commonly used at summing points in mixers, summing resistors are also employed in external summing devices.
Template
Generally something that establishes or serves as a pattern or gauge, such as a thin metal plate with a cut pattern that is used as a guide in making something accurately in woodworking. In our discourse this normally refers to a computer document or file having a preset format that is used as a starting point for a particular application so that the format does not have to be recreated each time it is used. This could be a loan amortization template for a spreadsheet program; a memorandum template for a word processing program, a mixer configuration for Pro Tools, a MIDI setup for sequencing software, a basic two zone layer for a keyboard controller, or any of dozens of other applications. Anytime one is working with a device that is programmable and has many different parameters templates can save a lot of time configuring new sessions, mixes, setups, and so on. An overlay that fits over all or part of a keyboard or other type of hardware control panel and has labels describing the functions of each key within a particular application is also known as a template.
Terminal Strip (a.k.a. barrier strip)
A series of connections, usually screw terminals, arranged in a line to permanently connect multiple audio lines to such devices as recording equipment, mixers, or outboard gear. Back in the '70's much of the high end recording equipment only had terminal strips on the back for connection. Since this equipment was primarily used in pseudo-permanent professional facilities it was considered the most cost effective and reliable way of connecting things. Nowadays most pro equipment is fitted with some combination of 1/4" (TS or TRS), XLR (see WFTD archive XLR) and possibly still terminal strip connections. Equipment sold primarily for use in permanent installations (contracting) such as stadium or theater sound is still frequently equipped with terminal strips only.
Touch Sensitive
A generic term for that which is sensitive to and has a response to touch (presumably human). It is commonly used to denote a keyboard instrument that will respond to the performance dynamics of the player or the manner in which faders on a mixer might respond to the touch of your hand.
In keyboards, this minimally refers to a sensitivity to the velocity with which a key is struck. The idea is to give synthesizers and other keyboard instruments a performance feel similar to that of a piano. While touch sensitivity is usually used to control the volume of the instrument - defined by key velocity - it can also be used to change filter settings or many other parameters relating to the sound.
In a mixing automation system, it defines the manner in which a motorized fader responds to writing automation movement. When a finger touches a fader that is "touch sensitive" the fader responds to the electrical energy found in the finger by becoming active and ready to write automation - or fader movement.
Transformer
A transformer is a device consisting of two or more coils of wire wound on a common core of magnetically permeable material. The number of turns in one coil divided by the number of turns in the other is called the turns ratio. An alternating voltage appearing across one coil will be inducted into the other coil multiplied by the turns ratio. Some transformers are designed to operate at 60 Hz (see WFTD archive "Hertz") and to handle large amounts of current. They are called power transformers, and are found in almost all electronic equipment to change our 110 volt line voltage to one or more voltages more suitable for operating the device. Audio transformers are designed to operate at audible frequencies, and are used to step audio voltages up or down to send signals between devices such as microphones, tape recorders, mixers, and all types of other electronic equipment. Transformers are also sometimes used in audio to provide isolation between two audio circuits. Because the two coils of wire never electrically touch one another a transformer provides a certain amount of isolation that can help prevent ground loops and other problems that can crop up in complex audio systems.
Transistor
An electronic component known as a semiconductor. A semiconductor can be an excellent conductor of electricity under some conditions while it can resist conducting electricity under other conditions. There are basically two types, the bipolar transistor (also called the junction transistor), and the field effect transistor (FET). Transistors are typically designed with at least three terminals, one of which (the base) serves as a sort of control gate (for lack of a better term). A supply voltage is connected across the other two terminals and the (signal) voltage present at this third terminal serves to "turn on" the transistor to varying degrees allowing current flow. In a common amplifier circuit, for example, a large supply voltage is placed in series with the transistor and a load (such as a speaker), and then a small varying voltage (the source signal) is applied to the base. This causes the transistor to allow a varying amount of current to flow from the supply source (usually a power supply) through its load as the source signal's voltage rises and falls. This is a very, very rudimentary description of an amplifier circuit, which is the basic circuit used across a good portion of all analog audio (including equalizers, mixers, crossovers, amps, etc.). It is also worth noting that in many ways the transistor mimics its predecessor, the vacuum tube. There are some distinct and important differences to us audio people, but the basic function is the same.
Trim
Found on most mixers, trim controls provide the initial level setting for each channel's input gain. In most cases, trim adjusts gain of the microphone preamp, but it may also apply to line level signals. Optimizing this gain stage will make a tremendous difference in the mixers signal to noise ratio and in gain staging later in the signal chain.
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).
Virtual Studio Technology
A plug-in effects processing engine for DAW software that was created by Steinberg. The VST engine works with the VST-enabled host software to allow audio and MIDI effects processing, synth and sampled sound modules, and mastering tools to run in real time with multitrack digital audio, operating in a fashion similar to how the auxiliary sends on an outboard mixer work, in that you can assign plug-ins to a channel (or channels) of audio or MIDI in the host VST audio recording/editing software. VST has become arguably the most proliferated plug-in engine for DAW software, mainly because of Steinberg's support for third-party development and also due to the fact that VST plug-ins are developed for both the Mac and Wintel platforms (note that they are not interchangeable with one another; they are also not compatible with any other plug-in engine). Aside from the wealth of high-quality retail plug-ins, there are also shareware and freeware VST plug-ins offered in a plethora of locales around the web.
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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