The Tech-T Glossary

Part 3

A - F

G - L

M - N - O - P - Q - R - S

T - Z

M

MADI: MADI stands for Multi-channel Audio Digital Interface and is an AES specification for digital interconnection between multitrack recorders and mixing consoles. It is an international standard (AES-10) offering 56 audio channels on a single coaxial cable or fiber-optic interface.
Magneto-Optical: A technology that utilizes the optical properties of a magnetized platter upon which data is stored magnetically, but read optically, using a laser. A laser heats up the magnetic material. An electromagnet then polarizes the material while it is still hot. When the magnetic surface cools, the polarity is set. Magnetic polarity affects light polarity, so a low power laser can then read the information back by determining the polarity of the reflected light. An example is the mini-disk systems.
Maximum SPL (Sound Pressure Level): A common specification for microphones, max SPL indicates the highest sound pressure level a mic's electronics can handle before the onset of distortion. Normally, this spec is referenced to 0.5% distortion at 1 kHz. Keep in mind that the presence of an attenuator switch on the mic may allow an increase in the volume level the mic can absorb before distorting.; Obviously, this is an important spec for many applications - if the mic is going to spend its life in front of a screaming Marshall stack, or in a kick drum, it must be able to adequately deal with the volumes it will be seeing.
MDM: Abbreviation for Modular Digital Multitrack. It pertains to any digital multitrack tape machine that is designed to work in conjunction with other like machines such that together they effectively build a "machine" with more tracks and/or capabilities. The Alesis ADAT and Tascam DA series machines are the most noteworthy examples, but others have been built by Fostex, Sony, Panasonic, and Studer. Compatible machines of different brands can be "linked" together under the modular environment, but otherwise they can only be synchronized together much the same way any dissimilar recorders traditionally have been.
Mic Level: The level (or voltage) of signal generated by a microphone. Typically around 2 millivolts. Compare this with the two normal line levels (1.23 volts and .316 volts), and it becomes apparent just how much amplification is going on in a microphone preamp, and why it is essential that preamps be of as high quality as possible!
Microphonic: Sometimes components in an audio device become or are sensitive to vibration, and convert that vibration into audio signals. Tubes are a common culprit. When tapped, a microphonic tube will output noise. Other components can also become microphonic, although solid-state components are less susceptible. Audio cables can also become microphonic, adding noise to a signal when moved, or vibrated.
MIDI: Musical Instrument Digital Interface. The machine protocol (language). MIDI was developed back in the early 1980's as a standardized protocol for communication between electronic musical instruments and peripherals. It allows MIDI devices to transmit and receive almost every aspect of a musical performance. Today MIDI is used in all sorts of applications, including synchronization, sequencing, lighting control, automation systems, more. There are many different types of MIDI messages that are used in MIDI for various applications. A typical MIDI connection is made with a MIDI cable, which has a 5-pin DIN type connector of which only three pins are used (except in some special applications).
MIDI Clock: A MIDI timing reference signal used to synchronize pieces of equipment together. MIDI clock runs at a rate of 24 ppqn (pulses per quarter note). This means that the actual speed of the MIDI clock varies with the tempo of the clock generator (as contrasted with time code, which runs at a constant rate). Also note that MIDI clock does not carry any location information - the receiving device does not know what measure or beat it should be playing at any given time, just how fast it should be going.
MIDI Implementation Chart: MIDI implementation refers to the specific MIDI messages and signals a piece of gear can recognize; a MIDI implementation chart is therefore a listing of the messages a particular device can transmit and recognize. This can be very useful when attempting to determine if a device can send and/or receive various types of channel or system messages. Normally found in the back of the device's manual, its MIDI implementation chart will consist of a list of available MIDI messages, whether the device incorporates those messages, and any special notes or limitations on how it deals with those messages. For example, the chart will list the MIDI channels and modes, note numbers, and continuous controllers the device can respond to. Support for aftertouch, velocity, pitch bend (often with bit resolution), and program change will be indicated. Also listed will be recognition of system exclusive, system real time (clock commands), system common (song position, song select, etc.) and aux messages (local on/off, all notes off, active sensing, and so on).
MIDI Machine Control (MMC): A part of the MIDI spec that allows MIDI devices to control hardware devices, MIDI Machine Control is commonly used to send transport control messages to hardware recorders. Play, Stop, and Locate are examples of MMC messages. Note that MMC does not include synchronization information, although MIDI sync info could also be sent to/from the device that MMC is addressing. MMC allows you to centralize control of your studio from a MIDI source (often a sequencer). A common scenario: Pressing play on a MIDI sequencer sends an MMC play message to a connected multitrack recorder, which begins playing. As the deck plays, it generates MIDI Time Code (MTC) which the sequencer then synchronizes to (chases). When "stop" is pressed on the sequencer, the deck also stops, and ceases to send out MTC. When MTC stops, the sequencer stops chasing. Locating to a point within the sequence will cause the deck to fast forward or rewind to the corresponding location on tape.
MIDI Mode: One of several ways in which a device can respond to incoming MIDI information. There are two parts to each mode, one defining whether it is monophonic or polyphonic, and the other determining if it is multitimbral or not. Four modes are included in the MIDI spec, and two others, Multi Mode and Mono Mode (for MIDI guitar) were developed later.; Omni On/Mono - Device responds to MIDI data regardless of channel, and is monophonic. This mode is rarely, if ever, used.; Omni Off/Poly - Device responds to MIDI data only on one particular channel, and is polyphonic. This is the normal mode for most keyboards that are not functioning multitimbrally.; Omni Off/Mono - Device responds to MIDI data only on one particular channel, and is monophonic.; Multi Mode - Used by many devices for multitimbral operation. An expanded version of Mode 3, Multi Mode allows the device to respond to several independent MIDI channels at once, with each being polyphonic.; Mono Mode - Used for MIDI guitar applications, Mono Mode is an expanded version of Mode 4, allowing for six Omni Off/Monophonic channels to be used at once, one for each string of the controller. This allows for better tracking, independent pitch bend per channel, and a separate sound or patch assignment per channel.
MIDI Time Code (MTC): A form of time code representing real time in Hours: Minutes: Seconds: Frames: Subframes, and transmitted over MIDI. MTC can also be described as a way of sending SMPTE time code over MIDI cables. Like all forms of time code, MTC is designed to allow various pieces (in this case MIDI-equipped) of equipment to synchronize together.
Modulation: Literally, modulation is change. In music technology, the term normally applies to a control signal being used to change some aspect or parameter of another signal. For example, a regularly repeating sine waveform might be added to a pitched note to produce vibrato, or a control voltage might be used to change (modulate) a filter cutoff frequency. A whole category of synthesis (and radio broadcasting), FM (frequency modulation), is based around using the frequency of one signal (the modulator) to change the frequency of another audible signal (the carrier). Likewise, AM radio works because of amplitude modulation, or using one signal's volume to modulate another signal.
Modulation Noise: Noise which is present only in company with a signal. In analog recorders the recording process has a certain "granularity" due to the fact that the magnetic characteristics of the tape are not completely uniform which causes an irregularity in the recorded signal that sounds like noise. In digital audio systems there is also an "uncertainty" in the level of the signal because of quantization in the A/D converter. This uncertainty sounds like added noise and is not present if the signal is not present.
MPEG: An acronym for Motion Picture Experts Group. They manage the standards for encoding audio and video in digitally compressed forms. With the development of so many new distribution channels for audio and video data the importance of standardized compression schemes is at an all time high. There are several different types of MPEG compression in use today, and within each of those there are different levels of compression. Some compress more than others do. The type and severity of MPEG compression specifically used will depend largely upon the intended distribution channel for the data (DVD, CD-ROM, Internet, etc.).
Mult: Short for "multiple", mult refers to the parallel wiring of the jacks in a patchbay. Several jacks are wired together so that input to one of them will feed all the others; a mult is a passive splitter or "y" connection.
Multisession: A CD-R is multisession if information can be added to a disc that has already been written to once (note that data is being added to the "end" of the CD, old data is not being erased, rewritten, or removed). This is also referred to as "Orange Book" standard. This is a fairly common format when CD-Rs are used for CD-ROM archival, and for Kodak format Photo CDs. In the audio world, many stand-alone CD recorders initially write CDs as multisession discs. Then when all the audio required has been written to the disc, it is "fixed up", or has a table of contents add. This essentially converts it to a standard Red Book audio CD, readable in regular CD players.
Multitimbral: A synthesizer or sampler is multitimbral if it is capable of producing more than one type of sound or timbre at a time. Usually this is described as the number of "parts" a unit can play at once. For example, a Kurzweil K2500 is 16-part multitimbral, meaning it can produce 16 different sounds at once (a sound being defined as a single patch or preset; part one might be piano, part two strings, part three trombone, part four flute, and so on. Generally these parts are assigned to different MIDI channels for independent control). This is distinct from the amount of polyphony, or number of actual notes the unit can simultaneously generate. Using the K2500 example again, a 16-part multitimbral K2500 can produce up to 48 notes of polyphony distributed dynamically across those 16 multitimbral parts.

N

NAB: Abbreviation for the National Association of Broadcasters. NAB is the organization which establishes various standards for radio and television broadcasting, analogous to the BBC in England. Many NAB standards have trickled down to the pro and consumer audio industry as well, especially in the domain of tape recording.
Negative Feedback: If some of the output of an amplifier is made to be out of phase, and mixed back with the amp's input signal, it will partially cancel the input, reducing the gain of the amplifier; this is called negative feedback.; But, because it contains and therefore cancels any distortion introduced by the amplifier, negative feedback also has the effect of improving the linearity of the amplifier. Negative feedback can also lower output impedance, increasing damping factor, and can sometimes be made to flatten frequency response.; The key to negative feedback amplifiers is careful design. Too much phase shift and the amp will be unstable, and too much feedback will cause Transient Intermodulation Distortion.
Noise Floor: The noise floor of a device or system is the amount of noise generated by the device itself with no signal present, it is measured in decibels. All electronic devices will generate a certain amount of noise, even a piece of wire! Minimizing the noise floor leads to expanded dynamic range, and cleaner recordings or sound production.
Noise Shaping: All digital recording systems introduce quantization errors and noise. Noise shaping averages quantization errors, reducing them for low frequency sounds. This changes the timbre of the quantization noise, lowering low frequency content and emphasizing high frequency noise. A conventional filter can then be used to reduce the high frequency noise. The result to our ears is less perceived noise.
Normal: 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. Also see fully normal.
Normalize: A DSP (Digital Signal Processing) function. Normalizing increases the gain of an audio file until its loudest point (or sample) is at maximum level. There are several benefits to this: First, the overall signal level is now higher, which makes subsequent gear in the audio chain perform better. Second, the signal is now taking advantage of the full resolution of the D/A converters; this also minimizes quantization noise in some cases.
NOS: Like the ORTF method, NOS, which stands for Nederlandshe Omroep Stichting (Netherlands Broadcasting System) is a stereo miking technique. The NOS method is to place two cardioid microphones 30 cm (11.811023622 inches) apart and angled at 90 degrees from one another. This method produces more ambience than a strict coincident placement of mics, and fewer phase problems than widely spaced pairs of mics. Try the NOS method when recording ensembles or group performances, as well as on acoustic instruments. The center image of the recording will be nice and strong, but with a good amount of subdued room sound blended in as well.
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.

O

Off-Axis: Refers to an audio source that is not directly in front of a transducer, especially a microphone. This results in off-axis coloration; a distortion or change in the frequency response of the reproduced audio signal. Often this coloration is put to good use. For example, many engineers intentionally set up mics on guitar amps so that they are slightly off axis to control the amount of high frequencies captured. A microphone will generally produce the "truest" results if it is used on-axis (oriented directly in front of the sound source).
Omnidirectional: Literally, from all directions. In audio, microphones are said to be omnidirectional if they can detect sound equally from all directions. Speakers are omnidirectional if they radiate sound in all directions equally; this tends to be the case with subwoofers and low frequency drivers. Low frequencies, in general, tend to be omnidirectional, versus high frequencies which tend to "beam" or be very directional.
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. 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.
Opto-isolator: An electronic component that contains a light source, usually in the form of an LED, and a light sensing device. When current flows, the LED emits light, which is detected by the light sensor. Opto-isolators are used to replace switches and relays. The big advantage of opto-isolators (also called opto-couplers) is that they have no electrical connection, helping cut down on ground loops.
Orange Book: CD-WO or CD Write Once. This is the spec detailing physical and optical characteristics for recordable or writable CDs, whether audio or CD-ROM.
ORTF: A stereo recording method created by the French national broadcast system to simulate the directional perspective of human ears. Similar in approach to the more conventional X-Y configuration, two microphones are placed in front of a sound source. The mics are spaced 17 cm (about 6 3/4") apart, at an angle of 110 degrees. The ORTF (Office de Radiodiffusion-Television Francaise) method provides good mono compatibility and stereo imaging, but captures little of the room's ambience (this may or may not be a good thing, depending on the room you are recording!) Try this mic setup the next time you are recording a small ensemble, choir, orchestra, or even a solo acoustic instrument, it works quite well.
Oscillator: An electronic device which generates a periodic signal of a particular frequency, usually a sine wave, but other waveforms (square, sawtooth, triangle) are often used. Oscillators are common in audio devices such as synthesizers and test signal generators. Early synthesizers used oscillators as the basic component for all of the sounds of the machine. All of the filters and envelopes modified the sound created by the oscillator to produce the desired sound. Nowadays most keyboards produce sounds by playing back samples recorded on chips or by more modern synthesis techniques such as Physical Modeling, FM, LA, or any number of other methods that have been employed in the past 10 years.
Overtone: Similar in concept to a harmonic. Overtones are tones produced by an instrument (or sound source) that are higher in frequency than the fundamental. They may or may not coincide with the frequencies of a harmonic series (harmonics), but they usually do. The difference is that harmonics are always musically related to the fundamental in that they are integer multiples of it. Overtones of a sound are often exactly the same as its harmonics except the first overtone is considered the second harmonic because the first harmonic is the fundamental. Overtones are also sometimes called partials.
Oversampling: Oversampling is used during the analog to digital (A/D) and digital to analog (D/A) conversion processes in a digital recorder, sampler or playback device. Essentially, the sampling rate of the converter is multiplied to a very high rate (i.e. 4x oversampling puts the rate at 176.4 kHz). This accomplishes two things: First, it allows the anti-aliasing and anti-imaging filters on the converters to be much more gentle, which reduces phase distortion. Second, in a 4x oversampled system, it results in a 6 dB drop in noise (other rates result in more or less noise reduction).

P

Pad: A sustaining, "wash" or fill sound, usually used as harmonic background material in a musical arrangement. Arrangers often speak of using a "string pad" during a passage; this would be a section of strings playing long, sustained chords behind the melody. With the advent of samplers and synthesizers, other types of sounds have also become common as pads; just about any sound that can sustain can be used as a pad these days!
Parametric Equalizer: A type of equalizer having several "parameters" for control of various filters that can be applied to audio signals. Parametric equalizers are most widely used in situations where very fine control over the audio signal is desired. In order for an equalizer to be parametric it must at least have control over Q, and frequency. In most cases each of these controls are on rotary potentiometers, but there are a few graphic style parametric equalizers on the market. Some equalizers have selectable frequencies that can be adjusted, but no Q control. These are known as quasi-parametric or sweepable equalizers.
Passive: A passive audio device is one which does not use amplification circuits. By nature of design, passive devices do cause a loss in power (insertion loss). Because they do not contain amplifiers, and are "cut-only" or "subtractive" in operation, passive devices tend to not add noise or distortion to a signal (although noise may be added in compensating for insertion loss). Typical passive devices include direct boxes, splitters, equalizers and crossovers.
Patch Mapping: A function of some MIDI devices, patch mapping allows an incoming MIDI Program Change message to be assigned to call up any of the receiving device's internal program numbers. For example, an incoming MIDI Program Change message with a value of 44 might be mapped to call up internal program number 95, MIDI Program number 67 might call up internal program number 14, and so on. There are a variety of reasons for using this function. Just a few:; On devices which have more than 128 patch locations, but don't respond to bank select messages, internal programs higher than #128 can be accessed using mapping.; The most basic reason: convenience! Rather than copying patches around to put them in the order you want or need, simply use patch mapping to establish the desired order, and call them up through MIDI.
PCI (Peripheral Component Interconnect): A high-performance (by current standards) computer expansion slot designed by Intel. PCI allows for 32- or 64-bit bus specification. PCI is described as high-bandwidth and processor-independent data path between the CPU and high-speed peripherals. The PCI spec allows for the capability to transfer up to 132 megabytes per second at a bus clock speed of 33 MHz (although the current rates being claimed by manufacturers are more commonly in the 30 Mb/sec range). This speed makes it especially suitable for high data rate applications like digital audio and video. PCI slots are found in the current generations of both PC and Macintosh personal computers.
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.
Peak Program Meter (PPM): An alternative to VU meters, Peak Program Meters have fast rise times (30 times faster than VU meters) and a much slower fallback or decay time. Peak Program Meters respond to peak levels rather than average levels. This makes them especially useful in situations where distortion or overload is a significant concern, as in digital applications. Because other meters, (i.e. VU meters) respond more slowly, giving an average level reading, they are not as useful for indicating maximum levels or peaks. Popular in Europe, PPMs are found in mechanical, LED, and/or plasma forms in a variety of equipment types.
Phase: Audio waveforms are cyclical; that is, they proceed through regular cycles or repetitions. Phase is defined as how far along its cycle a given waveform is. The measurement of phase is given in degrees, with 360 degrees being one complete cycle. One concern with phase becomes apparent when mixing together two waveforms. If these waveform are "out of phase", or delayed with respect to one another, there will be some cancellation in the resulting audio. This often produces what is described as a "hollow" sound. How much cancellation, and which frequencies it occurs at depends on the waveforms involved, and how far out of phase they are (two identical waveforms, 180 degrees out of phase, will cancel completely).
Phase Invert: A switch found in the input sections of mixing consoles and mic preamps. The term "phase invert" is actually a misnomer, since what the switch really does is invert the polarity of the signal in that input (correct usage would be "polarity invert"). Its intended use is to correct for balanced lines and mics that are wired backwards. In some cases toggling the phase invert switch may make a sonic difference if signals are out of phase, but doing so will also put that signal out of polarity with the others.
Physical Modeling Synthesis: A type of sound synthesis performed by computer models of instruments. These models are sets of complex equations that describe the physical properties of an instrument (such as the shape of the bell and the density of the material) and the way a musician interacts with it (blow, pluck, or hit, for example).
Pigtail: A pigtail is the end of an audio cable which simply has bare wires rather than any type of connector. Pigtails are used to connect wires to binding posts and screw terminals.
Pink Noise: Random noise with equal energy per octave. Our ears perceive this as sounding relatively "flat" in frequency response (since pink noise is based on octaves rather than individual frequencies, there is no increase in energy in the high octaves). Because of this, and because Real Time Analyzers (RTA) tend to look at octave or 1/3 octave ranges, pink noise is very useful for measuring the frequency response of audio equipment, as well as for determining room response for sound reinforcement applications.
Polarity: In electronics, two points that have opposite electric potentials (one is positive, the other negative). This is not the same as being 180 degrees out of phase (although the results can be similar). Phase implies a relationship with time, polarity does not. What most engineers, consoles and preamps refer to as a "phase" switch is actually a switch reversing signal polarity.; Polarity is important when interfacing equipment, particularly speakers - you don't want one cone moving in while the other moves out. Some designers feel that maintaining "absolute polarity" (no polarity reversal in a signal chain) throughout a signal path is important.; While tests don't indicate that the ear can hear which polarity is correct, they do show that it may be possible to detect a difference between normal and inverted polarity signals. (Try it for yourself in a critical listening environment: Play a signal though a single speaker, then reverse the speaker wires and play the same signal again - remember to switch the wires back when you are finished!)
Polar Pattern: Depending on their design and construction, microphones respond to sound coming from different directions with varying degrees of sensitivity. A plot or graph of this response is called a polar pattern (sometimes polar response curve). Looking at a mic's polar pattern will tell you how directional it is, how well it will reject sound from certain directions, etc. It is important to note that polar patterns are frequency dependent. Typically, low frequency response will be almost omnidirectional; the polar pattern will be come more directional as frequency rises.
Pole Piece: A shaped piece of high permeability metal, usually soft iron, which serves to concentrate and direct the magnetic field of a permanent magnet to maximize efficiency of devices like loudspeakers, magnetic cartridges, and cutterheads. Pole pieces are needed because magnets are hard (expensive) to make in the complex shapes that can be needed. In layman's terms, the Pole Piece is the part of the speaker magnet assembly that the voice coil slips over. It is the center round piece.
Polyphonic: When discussing musical instruments, the ability to play more than one note simultaneously. All instruments have a finite number of notes they can produce at one time. For example, a six string guitar has a maximum of 6-note polyphony. A synthesizer might be 32-note polyphonic, and so on. The more notes of polyphony an instrument can produce, the more capable it is of playing complex arrangements and chords. If the polyphony of the instrument is exceeded, it must "steal" the notes it needs from others that are already sounding. For example, a synthesizer might steal the last note requested from the first one hit; the first note stops, and the new one begins to sound. Some synths and samplers use sophisticated algorithms for voice stealing, others allow you to pre-allocate a given number of voices to a particular MIDI channel, and so on.
Pop: A bassy thump or "explosive" sound heard in a vocal mic (this is called a "plosive"). Pops often occur when the vocalist pronounces words with "p", "t", "b", etc. sounds in them. These consonants can create a puff of air that strikes the microphone diaphragm, creating a thump in the audio signal. In general, windscreens will help with pops to an extent, but a pop filter will be more effective. Be careful that the pop filter you choose is transondent, and serves only to break up the plosive's effects.
Pop Filter: A pop filter is used with microphones to shield the diaphragm from sudden bursts of sound which can cause a popping effect. The shield is transondent and does not interfere with the movement of sound towards the microphone. Pop filters, which usually look like a 6" to 8" circle of mesh material, are commonly seen in recording studios situated between 1" and 8" in front of a microphone.
Potentiometer: An electronic component that used to provide variable control over an electronic circuit. Usually controlled by a rotary knob which can be turned by hand, a volume control is a good example of this. A potentiometer is often called a pot for short.
PPQN (Pulses Per Quarter Note, sometimes Parts Per Quarter Note): The timing resolution of a MIDI sequencer. PPQN indicates the number of divisions a quarter note has been split into, and directly relates to the ability of the sequencer to accurately represent fine rhythmic variations in a performance, or to recreate the "feel" of a performance. Older sequencers were capable of 96 PPQN (sometimes even less), which often resulted in a stiff "quantized" feel to the music (even if it hadn't actually been quantized). Current versions can reach 768 PPQN or even higher resolutions, which is more than adequate for most musical applications. Note that the resolution of the sequencer is especially important at slower tempos. If your sequencer is limited to a lower resolution, one trick is the double the tempo of the song, then perform the parts in half time. This effectively results in a doubling of resolution.
Pre-delay: Pre-delay is a parameter found in reverb processors. It refers to the amount of time between the original dry sound, and the audible onset of early reflections and reverb tail. Carefully adjusting the pre-delay parameter makes a huge difference in the "clarity" of a mix. For example, a longer pre-delay will move the reverb tail out of the way of the vocals, making them much more present and understandable.
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".
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 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.
Program Change: Also known as Patch Change, a type of MIDI message used for sending data to devices to cause them to change to a new program. Program Changes messages are channelized so they will only affect a device on a specific MIDI channel. These commands are used in all sorts of MIDI applications ranging from simply changing patches on a synth or reverb to controlling lighting systems. Software sequencers that appear to have the programs of your keyboard in them by name are in fact using program change commands that are known to pull up those programs in your keyboard.
Proprietary: This is a word that is (unfortunately) used fairly frequently in the computer and audio worlds. In the broader sense, proprietary means that a concept or product is unique to, and the property of a manufacturer or company. More commonly, proprietary refers to a manufacturer designing a product to only work with other products from that same manufacturer. For example, a manufacturer might make a synthesizer that can only save patch information to specific, specially designed RAM cards, rather than to more universal PCMCIA cards, floppy disks, or whatever. In order to save that synth's information you would be required to use the proprietary cards available only from that manufacturer. While the word "proprietary" is often given a negative connotation, keep in mind that building gear around proprietary designs and options allows a manufacturer to implement features that might not be possible if everything were standardized and generic.
Proximity Effect: An increase in bass or low frequency response when a sound source is close to a microphone. Proximity effect is distortion caused by the use of ports to create directional polar pickup patterns, so omni-directional mics are not affected. Depending on the mic design, proximity effect may easily result in a boost of up to 16 dB, usually focused below 100 Hz. Vocalists tend to like proximity effect since it fattens up their voice, but a constantly varying bass boost can wreak havoc on headroom and carefully set levels! Obviously, if a vocalist is "eating the mic" to get proximity effect, the Inverse Square Law tells us that the levels the mic sees are increasing dramatically as well - distortion can easily result, from either mic diaphragm breakup or electronic overload. (You may occasionally see proximity effect referred to as "bass tip-up")
Pulse Code Modulation (PCM): A method of encoding and de-encoding a digital signal. There are actually several varieties in use today, including linear, non-linear, floating point, and differential. These vary mainly in how they deal with quantization, and how they handle values that fall "between" the digital signals bits.
PZM: PZM (Pressure Zone Microphone), or more correctly boundary mics (PZM is a trademarked term) use a small electret capsule mounted close to a backing plate. The idea is that the mic capsule/plate is mounted to a large flat surface (or boundary). This increases the sensitivity of the mic by 6 dB (due to pressure doubling from reflected soundwaves), and gives it a hemispherical pickup pattern. The practical frequency response of the mic will depend on the size of the flat surface it is mounted to. If the surface is too small, low frequencies will not be reflected resulting in an apparent high frequency (treble) boost.

Q

"Q": The resonance of an electronic circuit. "Q" actually refers to quality factor. Q is a measure of the sharpness of a resonant peak. The term Q is often used interchangeably with "bandwidth". This is not entirely correct. It is more accurate to say that Q determines bandwidth (a subtle but distinct difference). Q is most often used in reference to synthesizer filters (sometimes referred to as resonance) and equalizers, but it also applies to capacitors (a measure of efficiency, the ratio of capacitive reactance to resistance at a high frequency) and speakers (a measure of directivity). In speakers, a Q of 1 means the system sends out energy equally in all directions; a speaker with a Q of 2 radiates in a 180 degree hemisphere; higher Q's correspond to smaller angles.
Quantization Error: Error resulting from trying to represent a continuous analog signal with discrete, stepped digital data. The problem arises when the analog value being sampled falls between two digital "steps." When this happens, the analog value must be represented by the nearest digital value, resulting in a very slight error. In other words, the difference between the continuous analog waveform, and the stair-stepped digital representation is quantization error. For a sine wave, quantization error will appear as extra harmonics in the signal. For music or program material, the signal is constantly changing and quantization error appears as wideband noise, cleverly referred to as "quantization noise." It is extremely difficult to measure or spec quantization noise, since it only exists when a signal is present.; Quantization error is one reason higher digital resolutions (longer word lengths) and higher sample rates sound better to our ears; the "steps" become finer, reducing quantization errors.
Quasi-Parametric: This term applies to equalizers. A quasi-parametric (also known as "semi-parametric") EQ will allow control over the frequency and gain of each band of equalization, but not bandwidth. The midrange EQs on mixing consoles are often quasi-parametric (sometimes referred to as "sweepable mids"). While more flexible in some ways than a graphic, a quasi-parametric EQ does not offer the "ultimate tweakability" of a fully parametric design.

R

R-DAT: Abbreviation for Rotating Digital Audio Tape as opposed to S-DAT which stands for Stationary Digital Audio Tape technology. R-DAT just refers to what we have all come to know and love simply as DAT. The "R" was originally placed in front of the name because there are some S-DAT systems out there and there needed to be a distinction between the two. R-DAT machines handle tape and record pretty much the same way a modern VCR does (so do ADAT machines for that matter). The tape moves somewhat slowly while the rotating head spins at about 2000 R.P.M. (revolutions per minute) recording the data on the tape in a helical scan fashion, which provides a much higher "effective" tape speed.
Random Access: As opposed to linear access, where data or items must be accessed sequentially (like on a cassette tape), random access allows you to randomly jump to any item on a piece of media, and retrieve or operate on it immediately (like a CD, or hard drive). The biggest difference is in the speed with which items can be accessed. By allowing you to "skip" to the item you want, random access greatly increases speed and productivity.
RAR: Abbreviation for Read After Read. Many helical scan decks recording digital data perform a read after write operation to ensure that the data written to tape reads back correctly. This is part of the error protection. In some high end models (such as the ubiquitous Sony 1630) a read after read is performed which gives double assurance of data integrity on the tape.
Ratio: In a compressor, limiter, or expander, ratio is the amount of output level change that results from a given input level change. For example, in a compressor with a 4:1 ratio, an input level increase of 4 dB will result in an output increase of only 1 dB. An expander might have a ratio of 1:4; for a 1 dB input change, a 4 dB output change will result. Limiters typically have extremely high ratios, some claiming infinity:1, which essentially means that for virtually any input level increase, there will be only a very minor output level increase.
Real Time Analyzer (RTA): An RTA is a device which uses a number of narrow bandwidth filters connected to a display to give a visual indication of the amplitude in each frequency band. RTA's are useful for getting a reading on how a room will subjectively sound, where problem frequencies might be, and how to approach EQ'ing to correct for those problem frequencies.
Red Book: The Red Book specifies the standards for audio compact discs. Jointly developed in 1983 by Philips and Sony, it contains detailed specs on physical and optical characteristics, and logical organization (table of contents, track, and audio stream formats). The Red Book is the primary compact disc specification, all other formats are derived from it.
Regeneration: Also called "feedback," among other things, regeneration is signal fed back through an effects processor to intensify, or extend an effect. For example, in a delay effect, signal passes through the processor, is delayed and sent out as an echo of the original sound. If some of the delayed signal is tapped off and sent through the delay circuit again, a second repeat will result. By increasing the amount of signal sent back through (or fed back, or regenerated) the number of repeats is increased. Another example: On a flanger, when signal is regenerated through the flanging circuit, the effect becomes more deep, or intense.
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.
RMS (Root Mean Square): The square root of the mean of the square. RMS is a meaningful way of calculating the average of values over a period of time. With audio, the signal value (amplitude) is squared, averaged over a period of time, then the square root of the result is calculated. The result is a value, that when squared, is related (proportional) to the effective power of the signal.; Unfortunately, calculating the RMS value of anything but a simple sine wave (.707 of peak) is very difficult. The further a signal gets in harmonic content from a sine wave, the less accurate RMS values will be. For a dynamic signal like most music, it is nearly impossible to get even close to a true RMS value.; Note: RMS Power is actually a misnomer, since the RMS of a signal is a really just a value used to calculate average, or continuous, power.

S

Sample Dump Standard (SDS): The MIDI Sample Dump Standard is a method of sending digital audio sample data from one machine to another via MIDI connections. Due to the bandwidth limitations of MIDI, SDS transfers can be quite slow, but are an effective way to share sample data between samplers, or between samplers and computer-based sample editing software.
Sample Rate: In a digital recorder or sampler, the sample rate is how many times per second the source material is being "sampled" or recorded. Sample rate affects the frequency response of the final recording or sample; the highest frequency that can accurately be sampled is 1/2 the sample rate. In general, the higher the sample rate, the better the sound quality. But, the best sample rate to use will depend on your application, your gear, and the amount of storage available (the higher the rate, the more storage required). CDs use a 44.1 kHz sample rate, while DAT recorders often default to 48 kHz. Multimedia applications may use rates of 22.05 kHz or even 11.025 kHz for maximum efficiency.
Saturation: Saturation refers to the maximum amount of magnetism a magnetic tape can hold. Attempting to add more magnetism to the tape's oxide particles will result in distortion. Normal record levels do not generally approach saturation as distortion will be introduced before saturation is reached, especially in the low frequencies. High frequencies normally do not saturate as they have a tendency to self-erase during recording.; Engineers using analog tape often make use of tape saturation as an effect. By carefully controlling record levels, compression, warmth, and fatness can be added to a signal - all part of the much-heralded analog "mystique".
SCMS: SCMS is an acronym (pronounced "scums") standing for Serial Copy Management System. This nefarious little "feature" was designed by the ever-paranoid record labels to prevent evil-doers from digitally cloning CDs. Usually found on the S/PDIF ports of "consumer" (was there ever REALLY such a thing?) and "semi-pro" DAT recorders, SCMS is a bit flagging system that prevents digital copies from being made of digital copies. This means you can make as many digital copies of your original DAT tapes as you like, but you cannot digitally make copies of any of those copies. Beware of SCMS if you are sending backup DAT copies of your master tapes to a CD duplicator, or a collaborator. They may not be able to digitally transfer your work to another tape or media!
Scrub: A useful function found on many tape, disc, and hard disk based recorders, scrubbing is manually moving through an audio waveform. Because audio can be scrubbed backwards or forwards at rates much slower than real time playback, scrubbing allows for precise location of edit points.
SCSI: Pronounced "scuzzy," this acronym stands for Small Computer Systems Interface. SCSI is a hardware interface incorporated into computers, disk-based digital recorders, samplers, and other microprocessor-based equipment. It allows for the easy connection of a variety of peripherals such as hard drives, removable media drives, CD-ROM drives, scanners, and more. One SCSI controller can support up to 7 peripherals, each having their own unique "id" or address. The first and last items in a SCSI chain must be terminated for proper operation. The "theoretical" maximum length of a SCSI chain is 19 feet, but in practice, the chain should be as short as possible!
Semitone: Literally, half a tone. Also known as a half step. A way of describing a musical interval of pitch. In our equal tempered scale system we use there are 12 semitones per octave.
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.
Sensitivity: In audio terms, sensitivity is the minimum amount of input signal required to drive a device to its rated output level. Normally, this specification is associated with amplifiers and microphones, but FM tuners, phono cartridges, and most other types of gear have a sensitivity rating as well. In general, higher sensitivity is better (less input signal required for full output), but there are definitely situations where a device can be TOO sensitive (picture a very sensitive microphone in front of a wound-up Marshall guitar amplifier!) resulting in unwanted distortion.
Serial Time Code: Editing devices which can be controlled by computers, have a connection called a "serial control port" or RS422 port. These devices communicate with the computer and are controlled via commands in a serial data protocol. Serial Time Code is a means of transmitting time code over the same data stream that carries this control information for the purposes of synchronization. Some of these devices have no SMPTE Time Code port, but send and receive time code via these control ports. Other devices send and receive only transport commands over their serial ports, but require a conventional time code connection in order to read time code.
SFSK: Abbreviation for Smart FSK. Smart FSK was adopted a few years after FSK and MIDI had been in common use. Professionals were all switching over to SMPTE based systems which were much more reliable and accurate than the old FSK based ones, but the cost of SMPTE readers and generators was extremely high at the time, and units that translated it to MIDI were yet another separate big expense. So a few companies, most notably at the time were TASCAM (MTS30) and JL Cooper (PPS1), adopted Smart FSK which added the ability to include location information to the FSK signal. These boxes, which translated between MIDI and SFSK and back again, were much less expensive than the SMPTE based ones and they still allowed users the luxury of starting from within a song and having their sequencer locate to the proper spot to begin playback. Also, because the system was continually able to "locate" itself the common problem of FSK devices getting one or two beats behind (due to drop outs and glitches) went away. Nowadays SFSK isn't used much because SMPTE and MTC based systems are so inexpensive, as well as being even more versatile and reliable.
Shield: In electronic terms, a shield is a conductive enclosure, protecting its contents from magnetic and electrostatic fields. Since audio conductors and circuits tend to be extremely sensitive to such fields, shields are very important to us! In cabling, shields often consist of braided copper strands wrapped around the signal conductors. The amount of coverage the shield provides is directly related to the noise and hum performance of the cable. Some cables offer a shield consisting of a thin wrap of metallic sheeting, which can offer complete coverage of the encased signal conductors. Quality shielding, while more expensive, makes a tremendous difference in the noise performance of a cable - skimping on cables is never a good idea!
Shockmount: Commonly found in two places in the audio industry, rack cases and microphone stands, shockmounts are systems designed to isolate a device mechanically from its stand or case. In rack cases, the idea is to prevent damage to sensitive gear by isolating it from shipping and transport bumps, drops and similar catastrophes. Often these cases consist of a case-within-a-case, with the inner case isolated with foam or spring arrangements from the outer. Microphone shockmounts are designed to reject vibrations transmitted through the stand or boom to the microphone. Several types are in use, one common design using a system of "rubber-bands" to suspend the mic away from its stand.
Sibilance: Sibilance refers to the high frequency components of certain vocal sounds, especially "s" and "sh". Sibilance lives in the 5 to 10 kHz frequency range, and can cause problems if over-emphasized in a recording. While it is possible to use a graphic or parametric EQ to correct for sibilance, this is often an unsatisfactory approach. Often the overall track will begin to sound dull before the sibilance is corrected. A better solution is to use a dedicated de-esser, or use an EQ in the sidechain input on a compressor to perform de-essing. Since a de-esser dynamically corrects for sibilance (only processes where necessary), the resulting track will sound much more natural.
Sidechain: 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 To Noise Ratio: Simply; a measurement of a given noise level in a device as compared to the level of the signal. Higher numbers signify a greater difference, which is better. In technical terms it is the ratio of signal power at a reference point in a circuit to the noise power that would exist if the signal were removed (its noise floor). The maximum signal to noise ratio (which in many schools of thought is equivalent to dynamic range) of a given piece of equipment can be an important thing to know. This ratio is how much absolute noise it has compared to the highest signal voltage it can pass without distortion. While signal to noise ratio is often used as a specification to characterize relative quality differences in equipment, the way in which measurements must be done, and the degree to which they can differ, makes the true objectivity of such measurements highly suspect. Factors such as how much distortion can be allowed before you say the signal has reached "maximum" as well as other kinds of noise (like modulation noise) that may only show up when signals are present are just two examples of many variables that affect objective measurements. In digital equalizers the signal to noise ratio is a function of the maximum possible sine wave signal power compared to the quantization noise (a.k.a. quantization error) power. This is a very unambiguous value in linear PCM (Pulse Code Modulation) systems, but in other types of PCM systems the quantization noise (or quantization error) depends strongly on the level of the audio being recorded so it is very difficult to nail down the actual signal to noise ratio. It is sometimes useful to be able to compare S/N Ratio differences between equipment in certain applications, but it is more important to just understand the concept. Signal to noise ratio concerns us every time we pass audio (or video or data) though anything, and knowing what factors in our setup (such as gain structure) affect it is a fundamental part of building clean, quiet systems and mixes.
Sine Wave: A continuous, cyclic waveform in which the amplitude varies according to the sine (a trigonomic function) of the time. It is unique in that it has no overtones whatsoever. Since it contains only the fundamental pitch it gives a smooth rounded tone. Test tones used to calibrate tape machines and other equipment are generally sine waves. In acoustic instruments a flute sometimes has a nearly sinusoidal output. On an oscilloscope a sine wave looks like a symmetrical wavy line.
Skin Effect: The tendency of high frequency current to travel near the outside of an electric conductor rather than through its cross section. Skin effect increases the effective resistance of a wire at higher frequencies. While skin effect is a very important concern when working at the high frequencies of radio and television, it is widely considered to be a minor issue at the relatively low frequencies used in the audio world. There are many audio purists, however, who dispute this and claim that it still does make a difference.
Slave: A machine or component controlled by another machine or component. When two devices are synchronized to one another it is necessary to have one be the master and the other the slave. The slave unit responds to commands or information from the master and is thus controlled by it. This is the basic principle behind all synchronization in audio and video. For example, if a computer system is following an analog tape machine (or video deck) it can be said to be "slaved" to it.
Slew Rate: Slew rate is the ability of a piece of audio equipment to reproduce fast changes in amplitude. Measured in volts per microsecond, this spec is most commonly associated with amplifiers, but in fact applies to most types of gear. In amplifiers, a low slew rate "softens" the attack of a signal, "smearing" the transients and sounding "mushy." Since high frequencies change in amplitude the fastest, this is where slew rate is most critical. An amp with a higher slew rate will sound "tighter" and more dynamic to our ears.
Slope: In audio filters, slope refers to how quickly frequencies are attenuated by the filter once the cutoff frequency is passed. Slope is given as a dB/octave figure. For example in a high pass filter with a cutoff frequency of 4000 Hz, and with a slope of 6 dB/octave, for each octave (doubling of frequency) above 4000 Hz, the level of frequencies will be diminished by an additional 6 dB. Slope is determined by the "order" of the filter, or the number of poles it contains. A first order, or single pole filter will have a slope of 6 dB/octave. A second order, or two pole filter will have a slope of 12 dB/octave, and so on (slope increases by 6 dB/octave per order or pole).; Creating the correct slope is very important in filter design. For example, it determines how accurately an EQ can cut or boost some frequencies without affecting others. Slope is also important in crossovers, where it is undesirable for frequencies beyond the cutoff frequency to be passed on to amplifiers and drivers (typical crossover filter slopes are in the 12-24 dB/octave range). Sometimes crossovers feature selectable filter slope so that response can be matched to particular speaker set ups.
SMDI: SMDI, (pronounced "smiddy") stands for SCSI Musical Data Interchange. SMDI allows samples to be transferred from some sampling keyboards to a computer equipped with SCSI. The benefit of SMDI over the MIDI sample dump standard is speed. It can take hours for large samples to transfer to a computer over MIDI. With SMDI this happens in minutes. In order to do SMDI data dumps you must have an appropriate software application on the computer, a SCSI port on both the computer and sampler, and a sampler that is compatible with the software program. Of course, then there are the other gyrations you must go through getting the two devices to cooperate with each other on the SCSI bus.
SMPTE: Society of Motion Picture and Television Engineers. One of many versions of time code.
Soft Clipping: Clipping is the "squaring off" of an audio waveform that occurs when the signal level in a device exceeds that device's capacity to accurately reproduce it. Soft clipping rounds off the edges of the clipped waveform, making the sound easier to listen to, and less damaging to high frequency drivers.
Soft Knee (Compression): A type of compression where the onset of compression is gradual. In normal or hard knee compression when the signal reaches the threshold the unit immediately begins to compress at whatever ratio is set. In some situations the compression becomes very easy to hear (which is often not desirable) as the signal amplitude moves above and below the threshold. This is usually made worse when using high compression ratios. The solution is to have the compressor gradually enter into compression at a lower ratio prior to the signal reaching the threshold. The ratio is gradually ramped up as the signal gets louder until, at some point beyond the threshold, the full compression ratio is reached. This slower onset often makes the compression much more difficult to detect. The process is called "soft knee" because of how the compression ratio looks when plotted on a graph. In normal compression the knee (which is the point where compression begins) is an abrupt angle (how steep depends on the ratio) whereas in soft knee it is more of a curve.
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.).
Song Position Pointer (SPP): A type of MIDI message held in a MIDI sequencer or synchronizer telling a connected device how many 16th notes have elapsed since the beginning of a song. Used in conjunction with MIDI clocks, this allows operation similar to true time code (which provides location as well as speed information). Song Position Pointer is used by sequencers and synchronizers which do not support MIDI Time Code (MTC) or SMPTE time code as a means of locating to a point in a song other than the beginning or the end. For example, a sequencer synched to a tape deck can use Song Position Pointer to start at the correct bar and beat even if the deck hasn't been rewound to the beginning of the song, or has been fast-forwarded from where it was stopped.
Sound Pressure Level (SPL): The acoustic volume or loudness of sound, measured in decibels. SPL is a function of a signal's amplitude. Aside from the usual (and justified) warnings about hearing damage from high SPLs, it is worth noting that because of the way our ears function, sounds will appear to have a different timbre (or tone) to us at different SPL levels. This is important to keep in mind, especially when mixing in a studio environment. Be sure to check your mixes at a variety of volume levels to ensure that the mix is accurate. The old rule of thumb is that if a mix sounds good at a low SPL, it will sound great at higher levels.
Spaced Omni: A method of stereo recording where two omnidirectional microphones are placed several feet apart in front of the sound source. This system was used by Harvey Fletcher in 1933 in the first demonstration of stereophonic reproduction of an orchestra. Because the omni pattern will pick up room ambience as well as the desired sound source, mic placement is critical in balancing room sound with direct sound. And, as with any stereo miking technique, phase must also be considered when placing the mics. Spaced omnis are excellent where a natural, "real" sound is desired.
S/PDIF: A format for interfacing digital audio equipment together, S/PDIF (Sony/Philips Digital Interface Format) is considered a consumer format, and is largely based on the AES/EBU standard. In fact, in many cases the two are compatible. There are, however differences between the two formats, particularly in the channel status and user bits.; S/PDIF typically uses either unbalanced, high impedance coaxial cables or fiber optic cables for transmission. When using coaxial cables for transmission, it is normally best to keep cable lengths to a minimum, and to use the best quality 75 ohm video-type cables available.
Spider: In audio, the assembly which holds the voice coil of a dynamic loudspeaker centered in the magnetic gap. The spider is usually a corrugated circular piece of specially coated fabric. The name comes from the early days of loudspeakers when it was made of a plastic material that resembled the legs of an arachnid.
Splatter: An onomatopoeia (word that sounds like what it is). A type of extreme distortion in an audio signal caused by hard clipping of the waveform from a device being overloaded. As mentioned above, the word splatter pretty well gets across what it sounds like. While it can happen in all sorts of audio devices the use of the word comes mostly from distortion caused by overmodulation in AM transmitters where splatter has additional implications.
Standard MIDI File (SMF): A standardized file format for saving MIDI sequences independent of the platform they were created on. Standard MIDI Files allow musicians with completely different types of computers or sequencers to exchange MIDI sequences. There are two types, Type 0 (single track), and Type 1 (multitrack). Each type contains the same information, but on a Type 0 all MIDI channels are combined into one track (MIDI channel assignments and other information are not lost) while on a Type 1 each track is kept separate.
Standing Wave: A phenomenon where a sound is reflected back and forth between two parallel surfaces, such as two side walls in a room. Technically they are created by "room modes" or "eigentones," which are modes of vibration of air in the room. The sound waves interfere with one another to produce a series of places where the sound pressure level (SPL) at some frequencies is high, and another series of places where they are low. The places are sometimes called nodes and anti-nodes. A standing wave exists in a room where a frequency is such that the distance between any two surfaces is equal to one half of its wavelength. For a given distance there will be many frequencies that will generate standing waves, each a multiple of the fundamental frequency whose wavelength is related to the dimension in question. Standing waves are always detrimental to the acoustics of a room, but can be avoided by careful room design, or minimized by absorbing the frequencies where they build up, which is usually along walls or in corners.
Star Ground: A type of grounding scheme used in some studios to prevent ground loops. It requires isolating each piece of gear from AC ground (using a ground lift adapter) and running a separate ground wire from the chassis of each piece (including the racks themselves) back to the main studio ground (we call this "Technical Earth"). This "Tech Earth" gets connected back to a main AC ground, and/or a large copper rod driven 18 feet into the ground. Thus every piece of gear still has AC fault protection, but no earth grounds are tied together. Technically limited ground loops can still exist in the studio signal wiring, but the path length differences are minimized to an extent that it isn't likely to be a problem. Star Grounding is a time consuming and complex wiring scheme, but is generally very effective at preventing ground loops and works great in conjunction with other measures such as telescoping shields. Occasionally you'll still find some piece of gear that requires audio transformers to eliminate ground loops. Generally you'll find that with today's equipment, you really don't have to go as far as all this. Telescoping shields, balanced lines, and careful consideration to signal cable wiring with today's equipment is often good enough.
Stopband: The frequency range attenuated or not passed by a filter is called its stopband (as opposed to its passband, where signal is let through unprocessed). A filter can have more than one stopband. For example, a bandpass filter has a passband with a high stopband above it, and a low stopband below it.
Strap: In audio this term usually applies to multing two signal paths together. It is often specifically used in patch-bay jargon and means to connect two patch points together for the sake of making a common signal connection. Straps are typically used to connect a top and bottom patchbay jack's normal switches for normalling. The way you strap the normalling switches determines whether a patchbay is configured as fully normalled or half normalled. These straps are a built-in, hidden feature of many commercially available "plug-and-play" patch bays. Sometimes the user can reconfigure them, often by moving jumpers inside the patch bay. In professional custom installations, however, the individual straps (normals) must be wired by hand just like the rest of the patch bay.
Subcode: On digital media, there is a region of the tape where audio data is not recorded. Rather, this area has subcode written to it. Subcode can consist of a variety of different types of non-audio information; track number, indexing, and timing information such as track length and elapsed time, is found there. Digital tape has a higher capacity for subcode information, and such proprietary information as text (song titles), or timecode can be found written to subcode.
Sustained Transfer Rate: This spec details the speed at which a drive sends and receives data. Sustained transfer rate is the total time required for system processing, head switching, and seek. This spec is the most accurate reflection of a drive's true, real world performance. Be careful when comparing drives; in many cases, manufacturers will only display their drive's burst transfer rate, or maximum transfer rate (a much higher figure reflecting only the movement of data into RAM). While this is a useful spec for many applications, it does not reflect the requirements of digital audio or video.
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.......
Synchronization: In audio terms, synchronizing, or synching, is the process of making two devices operate together as one. One device will be the "master", and tell the second "slave" device when to start, when to stop, and how fast to play. Originally, synching devices primarily meant locking two multitrack tape recorders together to allow for more tracks, or locking audio and video decks together when adding sound to picture. Today, synchronization also encompasses locking recorders to computers, various digital devices' clocks to each other, MIDI to SMPTE, and a variety of other possibilities. Synchronizing wildly different technologies together can be a complex process.
System Exclusive: One of the categories of MIDI messages, System Exclusive (Sys Ex) is data intended for, and understood by, only one particular piece of gear. Normally, this data is used to communicate with and control parameters specific to that item. For example, all of the proprietary data in a Roland D-110 synthesizer representing RAM patches might be sent as a "sys ex dump" to a computer librarian. When the computer sends this data back out over MIDI, the only device recognizing and responding to it will be a D-110, all other synths and MIDI devices will ignore it. Other uses for sys ex? MIDI control of parameters not supported by continuous controllers, remote patch editing, patch bank select, and more - uses depend on, and can be tailored for, each specific piece of MIDI gear!