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Pro audio glossary
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A/D converter (ADC)
An analog-to-digital converter (ADC) is an electronic device that converts analog audio signals to digital data - ones and zeros. Analog sources may include microphone feeds for vocals and instruments, as well as stereo signals from mixers or other processors. The data can then be stored on a DAW (digital audio workstation) for editing, or sent to a digital recorder.
The name of a digital audio transfer standard developed by the AES (Audio Engineering Society) and the EBU (European Broadcast Union). AES/EBU is the most common alternative to the S/PDIF standard, and the most common AES/EBU interconnect is a balanced 110-ohm twisted pair cable with XLR connectors.
AGC (Automatic Gain Control)
A circuit that automatically adjusts the level of incoming audio signals to optimize the recording level of the receiving device. Often found in low-cost audio recorders to avoid the complication of manual adjustment.
AIFF (Audio Interchange File Format)
A file format for storing digital audio data. It supports a variety of bit resolutions, sample rates and channels of audio. Like the WAV file format, AIFF is really a "container" or "wrapper" for the underlying digital audio information, which is PCM. AIFF is often associated with Apple, but is also widely used in software that processes digital audio data.
In acoustics, the tendency of certain materials to absorb (as opposed to reflect) sound waves. A material's absorption coefficient is a value between 0 (total reflection) and 1 (total absorption), which varies according to the sound's frequency and angle of incidence.
A type of speaker enclosure that uses a sealed box to provide tight, accurate bass response. It gives up some efficiency to provide bass that is typically more accurate and controlled than that of a ported bass reflex design. Acoustic suspension speakers may require more amplifier power than than a bass reflex speaker to play at the same volume level.
An active device requires power to operate, as opposed to a passive device that doesn't need power. Examples include "active crossovers" and "active monitors."
A type of distortion that can occur during the digital sampling and recording of analog audio signals. If the frequency of the analog signal exceeds one-half the sampling rate, spurious signals and harmonics not present in the original signal may be created. Careful design and filtering before the sampling stage can reduce aliasing to a minimum.
The acoustic qualities of a listening space, including the background noise and perceived sense of space. Adding echo or reverb to music can enhance its ambience, giving listeners a sense of what kind of room it's being played in.
A measure of electrical current. An ampere — often referred to as an amp — is the unit of measure for the rate of current flow past a certain point and in a given amount of time, through an electrical conductor. Using water flow as a metaphor, amperage would represent the water volume, while voltage represents water pressure.
An electronic device that increases the voltage, current or power of a signal.
An analog is a representation of something. Analog audio signals use changes in voltage to represent changes in sound pressure. On vinyl records, groove depth is an analog for sound pressure levels. On analog tape, changes in magnetism are an analog for changes in sound pressure. This continuous representation of sound is a key difference between analog and digital systems, which use “quantized,” or sampled representation.
The initial part of a sound. On a compressor/limiter, a control that affects how that device will respond to the attack of a sound.
To reduce or make quieter.
A system that memorizes then plays back the positions of all faders and mutes on a console. In a DAW (digital audio workstation), the automation can also record and play back other parameters, including sends, returns, panning, and plug-ins.
A type of "twist and lock" connector often used with coaxial cables and antennas in wireless systems. Also used in audio and video to carry clock signals. A BNC male connector has a pin that connects to the primary conducting wire and then is locked in place with an outer ring that turns into a locked position.
BPM (beats per minute)
A standard measure of tempo.
A type of audio signal where the signal-carrying circuit uses two conductors, each of which carries the same signal potential but with the polarity of one reversed with respect to the other. Any noise that is induced into the circuit will be common to both "legs" and on arrival at the destination, is cancelled out by combining the out of phase signals. Balanced connectors are generally 3-pin XLR or 1/4" TRS phone plug.
A connector designed primarily for connecting speaker wire to the binding posts on the back of power amplifiers and speakers.
As it applies to audio, a range of frequencies. For example, a 10-band graphic equalizer has 10 “bands” of frequencies which can be independently adjusted.
An electronic device or circuit that allows signals between two specific frequencies to pass, but that attenuates signals at frequencies outside those band points. Bandpass filters that have amplifiers for boosting the levels of signals in the accepted frequency range are known as active filters. Devices that do not amplify are referred to as passive filters.
The range of frequencies that can be produced – or reproduced – by a transmission medium or piece of equipment. The bandwidth of human hearing is generally considered to be 20 Hz to 20,000 Hz, although there are harmonic components in audio that extend above 20,000 Hz.
Barrier strip (a.k.a. terminal strip)
A series of screw terminals arranged in a line, to which other devices are connected. Today these connectors are mostly found on amplifiers or crossovers that are going to be installed into permanent installations.
A circuit that utilizes the subwoofer in a 5.1 or 7.1 system to provide bass extension for the main speakers. The bass manager steers all frequencies below a set crossover point (typically 80 Hz) to the subwoofer, along with the LFE signal.
A type of speaker enclosure that includes a "tuned" port or passive radiator to increase and extend bass response (by releasing some of the energy created by the inward movement of the woofer cone). Bass reflex designs are more power-efficient than acoustic suspension designs — they'll play louder when driven with the same amplifier power. But they may sacrifice some bass accuracy in exchange for the added bass output.
Bit depth (a.k.a. word length)
Describes the number of bits a digital device uses to process audio. Bit depth refers to the dynamic range that can be captured during recording. The number of possible "levels" that can be recorded at 16-bit is 65,536 (2 to the 16th power), while this figure jumps to 16,777,216 using 24-bit hardware.
In audio, a bus is a point in a circuit where many signals are brought together. For example, a mixer assigns input signals to a bus, and the bus carries all the signals, mixed together, to an output. There's a left main output bus, and a right main output bus. There may also be an auxiliary bus, a monitor bus, or an effects bus that carry these designated mixes to where they need to go. The more buses a mixer has, the more flexible its routing and control capabilities will be.
The portion of a microphone that converts acoustic energy to electrical energy. Often includes shock mounts, acoustic isolators, protective covers and electronic circuitry in addition to the basic transducer. Also called an element.
Cardioid pickup pattern of the AKG C214 microphone.
A microphone pickup pattern with strong sensitivity to sound presented to the front of the microphone, good but reduced sensitivity to sources on the sides, and good rejection of sound coming from the rear. Cardioid mics are popular for both studio and live use.
The particular frequency to which a given filter-band in an equalizer is tuned to. Graphic EQs typically have a number of fixed center frequencies, while parametric EQs are able to sweep the center frequency, providing a greater degree of flexibility.
When a gate rapidly turns on and off due to fluctuating signal dynamics.
An effect that modifies the sound of a single instrument to simulate a group of the same instruments.
Class A amplifier
A type of amplifier circuit where the output device is always on for both parts of a complete audio waveform. Class A amplifiers are large, heavy, inefficient, and run very hot, due to the amplifier constantly operating at full power. Class A amps produce very low distortion and are mostly found in high-end equipment.
Class AB amplifier
A type of amplifier circuit that combines aspects of both Class A and Class B operation. If an amplifier operates in Class A for only a portion of its output, and has to turn on additional current in the devices for the remainder of its output, it is operating in Class AB.
Class B amplifier
A Class B amplifier circuit differs from Class A in that there is no current flowing when the output devices are at idle, and as a result, they have to turn on from a zero current state when signal is present. Class B amps tend to have more crossover distortion than Class A amps but are less costly to build because they can use less robust power supplies.
Class D amplifier
While Class A, B, and AB amplifiers are variations of "linear" amp designs, Class D amplifiers are a fundamentally different type known as a "switch mode" amplifier. This type of amp operates very efficiently because the output transistors switch on and off at very high speeds. Class D amps can be much more compact and lightweight than conventional amps because they don't require robust power supplies or heat sinks. Because Class D amplifiers operate in an on/off mode, people often mistakenly think that the "D" in Class D stands for "digital." This is not the case as there is no digital coding of the signal — Class D amps are analog-based.
Clipping occurs in analog and digital audio circuits when the incoming signal exceeds what a particular device can accommodate. Viewed on an oscilloscope, it results in the flattening of the signal peaks, as if the waveform had been "clipped" off. In certain analog circuits light clipping can have a positive effect, producing a pleasing distortion. In digital circuits the general rule is that clipping is to be avoided at all costs, since it produces an unpleasantly harsh sound.
Short for encoder/decoder. A piece of hardware or software capable of encoding or decoding a digital data stream. In audio, there are "lossy" codecs like MP3, that discard some of the information in the original file. There are also lossless codecs like FLAC, which provide bit-perfect accuracy in a file that is significantly smaller than the original.
A distortion produced by combining an electronic or acoustic signal with a delayed copy of itself. The result is peaks and dips introduced into the frequency response.
A compression driver is a high frequency speaker or tweeter designed to mount at the rear of an acoustic horn, which amplifies the sound and improves the speaker's efficiency. The driver's sound compresses in the horn's throat and expands with its outward flare.
A circuit that performs dynamic compression of an audio signal. By setting the ratio and threshold controls, it's possible to level out large dynamic swings. For instance, a 2:1 ratio means that if the program material rises by two decibels, the output will only rise by one decibel, once the threshold setting has been exceeded.
Type of microphone in which the capsule consists of conductive diaphragm next to a backplate. Condenser mics need power, which can be provided by batteries or a phantom power supply in the mixer. Applying an electric charge basically creates a capacitor out of the capsule. Sound waves hitting the diaphragm cause it to move in relation to the backplate, producing a variation in the capacitance of the capsule. This in turn produces a variance in the output voltage, which can then be turned back into acoustic energy.
The distance from a sound source where the listener hears the direct and reverberant sounds at equal volume.
An electrical device that divides the audio spectrum of an audio signal into smaller groups of frequencies, making it easier for downstream components to handle the load. The most common use of crossovers is in amplifier/speaker systems, which allows the separate components to function more accurately and efficiently.
The flow of electricity; the unit of measure is the ampere.
DAW (digital audio workstation)
A computer with the necessary hardware and software to digitize audio for recording, playback and editing.
DSP (digital signal processing)
Any form of manipulation performed on an audio signal while it is in digital form.
D/A converter (DAC)
A digital-to-analog converter takes a digital audio bitstream containing ones and zeros and reconstructs it back into an analog waveform. DACs come in a variety of configurations and prices ranges, and vary in how accurately they can reassemble the program material.
A power amplifier specification describing the amp’s ability to control speaker motion once signal has stopped. The effects of damping are most audible at low frequencies. An amplifier with a high damping factor will make a speaker sound "tighter" in the low end.
The process of making a digital audio or video stream more efficient for storage or transmission by using an algorithm that selectively eliminates bits of data. Examples of audio codecs that use data compression include MP3 and Dolby® Digital.
The time it takes for a signal to fall below audibility.
A logarithmic unit of measure used to express a ratio of sound power or intensity. In electronics, things like the gains of amplifiers, attenuation of signals, and signal-to-noise ratios are often expressed in decibels. When a signal's level changes by 10 dB, its power changes by a factor of 10.
Device for reducing the effect of sibilance in vocal signals. Basically, a de-esser is a frequency dependent compressor, although these days such devices are very sophisticated and optimized in the way they tackle the problem of sibilance.
An electronic device that can store an audio signal for a specified period of time and then release it, thereby delaying the sound relative to other parts of an audio program. Delay technology is at the core of time-based effects such as flangers and chorus units.
The part of a microphone that is mechanically moved by sound waves. The resulting interaction with a backplate or moving coil (depending on the microphone type) allows the conversion of sound energy to electrical energy.
A term generally used with guitars and basses, to "go direct" means to plug an instrument directly into a console or recording device without using an amplifier or microphone.
Device used to convert high-impedance unbalanced signals from stage instruments (like a bass or keyboard) into low impedance balanced signals suitable for a mixer. It also allows the signal to be carried over long lengths of cable. Often abbreviated DI for "direct insertion."
The angle of effective coverage for sound radiated from a speaker.
Any change in the shape of an audio waveform compared between two points in a signal chain. Generally the term refers to the desirable or undesirable "breaking-up" of audio (as in distorted guitars.)
The process of electronically converting a multichannel surround sound mix to stereo.
The taking of a audio file with a high sample rate, and sample rate converting down to a lower sample rate. Used in particular when a project is recorded at a high sample rate and the audio files need to be prepared for CD distribution.
The component(s) in a loudspeaker that physically create the sound: the woofer (for low frequencies), tweeter (for highs), and the midrange (for mids).
A dynamics processor/process that lowers (or "ducks") the level of one audio signal in response to a second audio signal. Typical examples include voice-overs or paging over background music.
Dynamic microphones work by having a diaphragm vibrate with the incoming sound to move a voice coil through a magnetic field, creating the electrical output signal. This is the exact opposite of the way a dynamic speaker works. Dynamic mics do not require external power, are physically robust, and can handle very high sound pressure levels (SPLs), making them a popular choice for live sound. Due to their construction, dynamic mics are less sensitive to fast transients and high frequencies than condenser mics, which are often preferred for studio use.
The ratio of the loudest undistorted signal level to the quietest, expressed in decibels.
A discrete, reflected version of a sound, heard somewhat after the original, and much louder than background reverberation.
A send or bus on a mixer, guitar amp, or other piece of gear, that is intended to be routed to an effects processor. On a mixer, effects sends are generally located after the fader (post fader) so that the moves made with the fader will also affect what is passing out of the send.
Electret condenser microphone
An electret mic is a condenser-type microphone that uses a permanent magnet to help create its signal, so it doesn’t require a battery or phantom power.
EMI (electro magnetic interference)
Audible interference produced when equipment or cabling picks up stray electromagnetic fields. This interference may be heard as hum, static, or buzz. Sources of EMI include fluorescent lights, power lines, computers, TVs, lighting dimmers, and radio and TV transmitters.
The attack, sustain, and release of a sound.
Hardware device or computer plug-in used to alter the frequency balance of an audio source. An equalizer has the ability to boost or cut specific frequency ranges based around a center frequency. Equalizers come in various forms, including parametric and graphic, active or passive.
An expander is the opposite of a compressor. Where a compressor takes a given dynamic change and reduces it, an expander increases it, making changes larger.
A type of volume or level control typically found on mixing boards and graphic equalizers. A fader works like a standard potentiometer, only instead of rotating, it slides along a straight path.
When a portion of the output signal is intentionally or unintentionally "fed back" into a device's input. One of the most common forms of feedback is the acoustic feedback that occurs when the sound from a loudspeaker gets picked by the microphone creating the sound, resulting in a squealing or ringing noise.
An electronic circuit designed to attenuate a sound source's energy at a particular frequency. Filters can be active or passive. Most filters these days are active, with amplifiers attached to them to allow the user to both boost and cut particular frequencies.
An audio process where two copies of the same signal are played together, with one variably delayed against the other. Originally created by holding a finger against a tape flange (the metal part that holds the tape on the reel). Today, the effect is produced using digital effect processors.
FOH (Front of House)
This refers to the main mixer for a venue's a sound system, usually located in the audience, as opposed to the monitor mixer which usually sits to the side of the stage.
Any device that sits on the floor and allows you to change a sound, select a channel, or turn on and off specific effects with your foot.
In audio the indication of how many cycles of a repetitive waveform occurs during one second. A waveform which repeats once per second has a frequency of 1Hz (Hertz). When applied to human hearing, frequency is often referred to as pitch.
A measurement of the frequency range that can be handled by an electronic component, loudspeaker, or signal cable.
The extent to which a circuit amplifies a signal. Usually part of an amplifier specification, its value is most often expressed in decibels.
Setting the gain of each stage in the signal path so that one stage doesn't overload the next one in line.
When multiple audio devices are used together, the gain structure of the system becomes an important consideration for overall sound quality. You have to look at which pieces are amplifying or reducing the signal, and by 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.
Gate (aka noise gate)
A dynamic device that has the ability to stop audio passing through it based on a certain threshold. Often called a "noise gate."
A type of equalizer that uses a series of slider controls to adjust the frequency bands, and the configuration of the sliders provide a "graphic" visual display of the EQ settings.
The common reference point for an electrical circuit, at zero voltage potential, that also represents the path for the electric charge to return to the power source.
Ground lift switch
A switch found on some equipment that disconnects the shield of a balanced cable from the local equipment ground. Used in situations where ground loops are a problem.
A common problem that occurs when an audio system has multiple paths and path lengths to ground. Ground loops will cause varying levels of hum, occurring at 60 Hz or some multiple of 60 Hz because that is the line frequency of AC power in the U.S. You can sometimes cure ground loops through the use of ground lift switches, but generally it is better to find the offending piece of equipment and troubleshoot it to find the reason for the ground hum.
A psychoacoustic effect where any delay signal below 40 milliseconds is indistinguishable from the source event. In other words, instead of hearing the sound and then a delay (two events), you hear both the source and the delay together as a single event. The effect is also called the precedence effect, meaning that if there are two sources of sound, as is often the case with PA systems or studio monitoring systems, the sound will be localized to the speaker that provides the earliest sound.
A harmonic is a sound wave whose frequency is a multiple of the frequency of a reference signal. Complex sounds, such as the human voice or the sound of a musical instrument always consist of a fundamental frequency along with a number of harmonics, which, together form the basic characteristic of that sound. For example, if the fundamental frequency is 25 Hz, the frequencies of the next harmonics are: 50 Hz (2nd harmonic), 75 Hz (3rd harmonic), etc.
An unwanted byproduct of passing audio through an electronic device. Since it is impossible to make a perfectly linear device (where the audio out exactly matches the audio in) harmonic distortion is always a byproduct of signal processing. The less distortion a particular piece of electronics creates, the more "transparent" it is.
Most often used to describe the amplifier in a two-piece amp/speaker combination. Typically, the head sits on top of a matched speaker cabinet.
The difference between the typical level of a signal and the maximum level before the onset of clipping or other distortion. Having lots of headroom ensures that quick bursts of louder sound (transients) will be cleanly reproduced, increasing the overall dynamic range of the system.
A unit of sound frequency, named after 19th century German physicist Heinrich Hertz. If the waveform of a sound performs one complete cycle in one second's time, that sound has a frequency of 1 Hertz.
A filter that attenuates frequencies below a certain cutoff point, while passing on frequencies above the cutoff unaffected. Sometimes called a low-cut filter.
Refers to connecting or disconnecting a device in a system while AC power is applied. Most electronic equipment is designed so that it's only safe to switch signal connections when the unit, and any attached equipment, is powered off. However, some devices are designed to be connected or disconnected while powered up. And certain connections, like USB, lend themselves to hot-swapping.
Hypercardioid pickup pattern of the AKG C414 XLS microphone.
A type of microphone pickup pattern. Similar in shape to a cardioid pattern, except that the sides have the greatest amount of rejection, as opposed to a cardioid pattern where the rear of the microphone has the least sensitivity.
The ability of speakers to reproduce spatial information so that listeners can visualize the relative positioning of individual voices and instruments.
An expression of the opposition that an electronic component, circuit or system offers to AC or DC current. Impedance contains both resistive and reactive components, although generally only the resistive part of the circuit is usually specified as Ohms. The higher the resistance, the higher the impedance.
In-ear monitor (IEM)
Earbud-like devices used by musicians and sound engineers to listen to music mixes of vocals and stage instrumentation for live performance or recording studio mixing. The most common professional stage in-ear monitors use a wireless system to send the mix to the IEMs.
Refers to sounds or signals whose frequencies are below the normal limit of human hearing, generally considered to be 20 Hertz. Some audio components include an "infrasonic filter" to remove ultra-low frequencies, which are often forms of noise like rumble and resonance. The term "subsonic" is often incorrectly used to mean infrasonic. Subsonic actually refers to the speed of sound.
A point in the signal path of a circuit where it is possible to interrupt the signal and "insert" another signal. This allows for the introduction of a piece of external equipment, so that the original signal now flows into and is processed by the external equipment before being returned back into the original circuit. Inserts are typically found on mixers, and some common applications include applying compression, noise gating, or EQ to a particular channel without affecting any other channels. An insert may also be called a "patch" or "injection point."
A device that acts as an intermediary between two or more pieces of equipment. An audio interface for a computer allows signals generated by a preamplifier for example, to be communicated to the computer software.
The interaction of two or more frequencies in a signal that results in the generation of new frequency components not present in the original signal. These new components have frequencies equal to the sum and difference of the frequencies of the original signals, along with multiples of those frequencies. This type of distortion is a challenge for loudspeaker designers because of issues relating to the back-and-forth motion of speaker diaphragms.
Jitter is the undesired deviation of some aspect of the pulses in a digital signal. Jitter can be introduced into a digital audio system at many points, including the analog-to-digital converter and digital-to-analog converter. These pulses have to maintain very precise timing, and any departure from the reference clock will cause timing jitter. The effects of jitter can be heard as low level distortion or noise, and can compromise other sonic aspects such as imaging.
Kilobit per second (kbps)
A measure of data transmission speed equal to one thousand bits per second. It's mostly used when describing streamed audio file formats like MP3. Higher audio bit rates typically correspond to higher sound quality. As a comparison, the top bit rate for MP3 is 320 kbps, while the rate for CD is 1411 kbps.
One thousand Hertz.
Refers to a point where the response of an audio system or function exhibits a notable change. Commonly used to describe the action of an audio compressor where the threshold setting creates a "knee" at a particular level inside the compressor as it relates to the dynamic range of any passing audio. Below this knee the compressor will exhibit a different behavior than above the knee. You can often choose between so-called "soft knee" and "hard knee" behavior, where the change is either smoothed out or abrupt.
LFE (Low Frequency Effects)
In modern movie soundtracks, low-frequency sound effects are mixed to a separate Low Frequency Effects channel. By having a separate LFE channel and feeding that extra low-frequency information to a dedicated subwoofer, it's possible to put low-frequency content on an equal psychoacoustic footing with midrange frequencies, which require less energy for the same perceived loudness.
Refers to the size of the diaphragm in a microphone. Microphones with 3/4" or larger diaphragms are considered to be large diaphragm microphones. Large diaphragm mics are generally more sensitive than small or medium diaphragm mics because of the increased surface area. Large diaphragm mics produce a comparatively "big" sound that is favored by many engineers for vocals.
Latency refers to unwanted time delays in audio production between when a signal enters a device or program and when it emerges. It's a common issue for digital audio workstations, where input signals must be converted to digital and then processed before reaching the monitoring stage.
Similar in principle to a compressor, a limiter is an audio processor that prevents the amplitude of an audio signal from rising above a certain threshold, regardless of what is happening to the amplitude of the source audio. Dynamics below the threshold are more or less unaffected.
Generally applies to the two line level references: balanced and unbalanced. Balanced or professional equipment operates at +4dBm or 1.23 volts, while unbalanced or consumer equipment operates at -10dBV or 0.32 volts. If two pieces of equipment need to be connected that use different line levels, then matching transformers should be used to equalize the levels. Otherwise the +4dBm signal will overdrive an -10dBV input and equally a -10dBV signal will not deliver enough level to a +4dBm input.
The resistance of a device to applied power, or the device itself.
A circuit where the output is connected back to the input.
A filter that attenuates frequencies above a certain cutoff point, while passing on frequencies below the cutoff unaffected. Sometimes called a high-cut filter.
MIDI (Musical Instrument Digital Interface)
MIDI can be used to transmit almost every aspect of a musical performance. MIDI data is instructions about how a sound will be produced, not the actual sound itself. So, data sent from one device to another could be played with a piano sound while the original information was actually a drum sequence. MIDI has expanded beyond strictly music and can be used for lighting cues among other applications. Apart from its ubiquity, MIDI's main advantage is that file sizes are relatively small compared to what an actual audio file would be. MIDI connections are made via a five-pin DIN connector.
Officially known as MPEG-1 Audio Layer 3, MP3 is a form of lossy data compression that can be used to create audio files that are fairly faithful to the original CD file while using only a fraction of the storage space. The MP3 codec relies on perceptual audio coding to remove audio content that our ears have trouble hearing anyway. MP3 is a popular format for streaming music services, and is compatible with most portable music players.
A microphone is a transducer which picks up sound waves and converts them into electric current for the purpose of transmitting or recording sound.
The signal level generated by a microphone, typically ranging between 0.001 to 0.005 volts. To make it usable for recording, the signal requires a microphone preamp to boost the signal up to line level.
A circuit, stand-alone device, or section of a device designed to amplify the low-voltage signal from a microphone up to standard line level. Microphone preamps are often built into mixers and audio interfaces.
A modern digital mixer that uses an iPad® as a control surface.
Any device that can take two or more audio signals and mix them down to a single mono or stereo signal. Although this happens on large consoles in professional studios, home studios can achieve surprisingly professional results using digital audio workstations, which combine recording, editing, and mixing functions.
A highly accurate loudspeaker designed for recording and mixing use. Monitors come in a variety of shapes, sizes, and configurations, and can be passive (requiring an amplifier) or active (built-in amplifiers).
Mono is short for monaural, which means only one audio channel, as opposed to stereo, which means two channels (usually a left and a right).
Generally describes a loudspeaker system that is designed to be close to the monitoring position. In this way the listener receives more of the direct sound from the speakers, while minimizing the effect of sound produced by reflections from walls, floors and ceilings.
As it applies to audio equipment, noise is unwanted sound that is not related to the signal. Noise sources can be external to a device, such as environmental noise sources. Or the noise can be generated internally. A component's signal-to-noise ratio measures how powerful the signal is in relation to the noise accompanying it.
The amount of self-noise generated by a piece of electronic equipment when no signal is present. Minimizing the noise floor leads to expanded dynamic range, and cleaner recordings or sound production.
An octave is a frequency ratio of 2:1. For example, there is one octave between 100 Hz and 200 Hz, and between 1,000 Hz and 2,000 Hz. To our ears, two frequencies that are an octave apart sound like the same note. An octave band consists of all of the frequencies within an octave.
Term used to describe the position of a sound source in reference to the microphone recording the source. Generally, a microphone will record best when the source is directly in front of it.
A unit of electrical resistance. Named for its discoverer, Georg Ohm.
A microphone that is turned on and turned up, and ready to be used.
Short for "public address" system, which is any sound system used to amplify speech or music. This can range from someone giving a speech in an auditorium to a band playing on a stage. A PA system may comprise a mixer, amplifier(s), speaker(s), various types of signal processors, and more.
Microphone with 3-position switchable attenuation pad.
A circuit designed to attenuate the output of a device by a given amount. For example, some microphones have so much output that they can overdrive the input stage of many mic preamps. To prevent this, mics often include a switchable "pad" on the output stage of the mic that reduces the mic's output by 10 or 20 dB. Some devices have built-in pads, but it is also possible to purchase external pads, which plug in to a device's output and reduce its level.
To pan means to move the perceived location of a sound source within a stereo soundstage. Generally works by reducing or making louder the particular sound source in either the left or right channel of a stereo output. If a source is panned hard left, then it will appear at only the left speaker, and likewise with the right side. The amounts of signal present in both speakers will determine the exact location of the sound source in between the left and right sides in the stereo field.
An equalizer whose filters contain controls over three parameters: frequency, gain/cut, and "Q," which refers to a measure of the sharpness of the resonant peak. In other words, with a narrow Q, gain or cut affects fewer frequencies adjoining the center frequency, while a wide Q will affect a greater number of adjoining frequencies.
A passive audio device doesn't have built-in power, as opposed to an active device, which does. Typical passive devices include equalizers, crossovers, and speakers.
The maximum instantaneous level of a signal. When a signal peaks beyond what a circuit can handle, distortion results.
DC voltage, usually 48V that is supplied to devices with active electronics using a standard balanced microphone cable. Common devices requiring phantom power include condenser microphones and direct boxes. 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 became known as "phantom" power. Most mixer consoles and preamplifiers provide phantom power.
Sound is made up of waves, and when you're dealing with more than a single sound source - say, recording a source using two channels - the sound waves have to stay in sync or some of the sonic energy will be canceled. This is known as phase cancellation. A complete waveform is 360 degrees. If you have two identical waveforms and one is 180 degrees out of phase, the result will be total cancellation - silence. When capturing audio with multiple microphones, even a small delay on one mic can put the summed audio out of phase.
A type of coaxial audio connector found more often on consumer equipment than pro gear. In addition to the ubiquitous red and white analog stereo jacks and plugs found on turntables, CD players, and receivers, coaxial cables with phono plugs can be used to carry digital audio. Phono plugs are often referred to as RCA connectors.
An audio cable that has bare ends rather than a specific type of connector.
Random noise with equal energy in all octaves. Our ears perceive pink noise as sounding relatively "flat." Because of that, and because real time analyzers tend to look at octave or 1/3-octave ranges, pink noise is a very useful audio source for measuring the frequency response of audio equipment, as well as for checking room response for sound reinforcement applications.
Polarity is often mistakenly used interchangeably with phase, but they're two different concepts. Phase implies a relationship with time, polarity does not. So, two signals with reversed polarity are not the same as two signals being 180 degrees out of phase, although the results can be similar. What most engineers, consoles and preamps refer to as a "phase" switch is actually a switch reversing polarity.
An electronic component used to provide variable control over an electronic circuit. Potentiometers are often used as volume controls on audio equipment - picture the typical rotary knob volume control. A potentiometer is often called a "pot" for short.
An electronic device used to amplify low-level signals, like those from microphones, before they are fed to subsequent gain stages or devices.
In a console, pre-fade listen is one of several possible means of overriding the normal monitor signal routing for various purposes. This signal is normally subject to the position of the channel fader, rising and lowering along with the fader position. However fader movement could be a distraction when sending monitoring information to players. Therefore a switch is usually placed that allows the split off to occur pre or post the fader, thus bypassing any fader movements. In broadcast situations, PFL is often referred to as "cueing."
Refers to the effect caused by a sound source being very close to a directional microphone. The net result is that low frequencies are boosted as the source comes closer.
An artifact introduced into an audio source that is caused by excessive dynamic compression. Pumping, also known as breathing, causes the program level to rise and fall depending on some frequency that is crossing the threshold
In equalizers, Q is the ratio of center frequency to bandwidth, and if the center frequency is fixed then the bandwidth is inversely proportional to Q. Translation: As you raise the Q, you narrow the bandwidth. Q is the most powerful tool offered by a parametric equalizer, allowing you to attenuate or boost a very narrow or wide range or frequencies within each EQ band.
An industry standard measurement for equipment intended to be mounted into racks. The width must be 19", while the height must be 1-3/4" or some multiple thereof. As an example, equipment height might be specified as 3 rack spaces or 3U high. There is no specification for depth.
Real time analyzer (RTA)
A professional audio device which measures and displays the frequency spectrum of an audio signal in real time. It uses a number of narrow bandwidth filters connected to a display to give a visual indication of the amplitude in each frequency band. The input signal can be provided by a built-in microphone or by a PA system. RTAs are often used by sound engineers and acousticians when installing audio systems in all kinds of listening spaces: Venues, home theaters, cars, etc.
In acoustics, a sound wave that has bounced, or reflected, off of a hard surface is termed a reflection. Reflections, particularly first reflections, which occur off of the nearest surfaces to the listener, can be problematic as they can cancel and reinforce frequencies in the direct sound coming from the sound source.
Has a few meanings, but often refers to the release time control found on noise gates and compression units. This parameter determines how long it takes the device to return to its normal state (not gated or compressed) after the signal has risen above (gate) or fallen below (compressor) the threshold.
The tendency for an object or system to vibrate at a specific frequency or frequencies when excited, with the resonant frequency being determined by the physical parameters of the object or system. For example, when music is played at high volume in a space, certain features of the room may resonate to a greater extent than other aspects of the room.
As it relates to audio signal processing, a return is the opposite of a send. Together, a send and return provide a path that a signal can take to be externally processed then fed back into the device. This type of send/return combination is often called an insert point.
The sonic effect a room adds to direct sound - echoes, reflections, diffusions, and absorptions due to the room's size, shape, material, and furnishings - is called reverberation. A reverb unit (or plug-in) electronically mimics those effects, creating a sense of space, and often giving the sound a pleasing timbre and depth.
S/PDIF (Sony/Philips Digital Interface)
A transfer protocol for digital audio, similar in concept to AES/EBU, but requiring a different electrical specification. S/PDIF typically uses either unbalanced, 75-ohm coaxial cables with RCA or BNC coaxial connectors, or fiber optic cables with TosLink connectors.
Sample (and sampling rate)
When an analog-to-digital converter captures the instantaneous amplitude of a sound source, it creates a sample. When a series of amplitude values are captured at periodic intervals, the rate is known as a "sampling rate." For example, the standard sampling rate for CD audio is 44.1kHz or 44,100 samples per second.
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 the signal coming back from the output of the processor.
Refers to the amount of signal needed at an input for a device to deliver its rated output. A higher sensitivity means the input signal is smaller for the given output and although this is generally preferred, too much sensitivity can result in a processor being overdriven by a preceding device.
A device that triggers a series of events in a particular order. Today's software sequencers are typically MIDI-based.
A type of equalizer circuit used to cut and boost a signal above or below a specified frequency, as opposed to boosting or cutting on both sides of a frequency, which is the way typical EQs operate.
A type of microphone with a long body and a series of holes running along the sides. The polar pattern characteristic is highly directional, with strong rejection of side and rear sources. Useful for picking up sounds at a distance, but also for pinpointing certain sources while ignoring much of the ambient or surrounding noise.
A high- frequency component of certain vocal sounds, especially "s" and "sh" that can cause problems while recording. A de-esser is a type of processor specifically designed to dynamically correct this problem without affecting the main vocal content too much.
Also known as key input, a side chain is a part of a circuit that splits off a portion of the main signal to derive a control signal. It can also be a secondary physical input on a processor, such as a compressor that allows control over the compressor's functions according to the side chain input. A common example is ducking when recording voice-overs.
In electrical systems S/N ratio is a measure of the ratio of the maximum signal level relative to the self-noise of the system. Usually expressed in decibels (dB.)
Measures how quickly a circuit can respond to fast changes in amplitude in the source signal. This spec applies to many kinds of gear, but is most commonly associated with power amplifiers. A high slew rate will generally translate to sound that is "tighter," cleaner, and more dynamic.
A type of cabling where multiple lines are enclosed in a larger single shield. The most common use for a snake is for live sound, where microphone leads and monitor mixes are sent back and forth between the front of house mixer and the stage. Snakes are also used in recording studios.
Sound pressure level (SPL)
A standard measurement of the acoustic volume or loudness of sound, expressed in decibels (dB).
A type of multi-pin connector developed by Neutrik® and found on many power amplifiers and speakers used for PA applications. Besides providing a very reliable connection, Speakon cables are intended solely for use in high-current audio applications. There's no way to confuse them with low-voltage microphone or instrument cables that use 1/4" or XLR connectors. Standard Speakon connectors come in four- or eight-conductor versions.
A speaker that is typically placed within and pointed at the performance area to help the performer(s) on stage to hear how they sound. By contrast, a PA speaker produces sound intended for the audience.
Standing waves occur when the sound from your speakers is reflected back and forth between the parallel surfaces in your room: the side walls, the front and rear wall, and the floor and ceiling. This effect amplifies some frequencies while canceling others out, creating areas of differing sound pressure or loudness around your room.
TDIF (Tascam Digital InterFace)
Usually pronounced TEE’-dif, this acronym refers to a protocol developed by Tascam for use in their MDM and digital mixing products for doing digital transfers of audio. TDIF connections are made via a 25-pin d-Sub connector and data is carried on a shielded cable. TDIF is one of the major formats used in pro and semi-pro products for digital transfer of more than two tracks of audio simultaneously using only one cable.
THD (Total Harmonic Distortion)
All electronic audio devices add some form of distortion to a signal passing through them. One form of distortion is the addition of harmonics not present in the original signal. THD measures the sum of all the harmonics added and expresses it as a percentage of the signal being measured. The closer THD is to zero, the cleaner and more transparent a device should sound.
TRS (Tip, Ring and Sleeve)
A type of 1/4" phone plug used in a variety of audio applications. Common uses include connecting 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.
TS (Tip, Sleeve)
A type of 1/4" phone plug used for unbalanced connections.
See barrier strip.
Pronounced “tamber,” it is the characteristic quality of sound produced by a particular instrument or voice. An oboe has a different timbre than a violin, which is why they sound different when playing the same note. Timbre is made up of all of the qualities of a sound: frequency, harmonic content, transient attack, and more.
A device that converts energy from one form to another. Some examples of audio transducers: A microphone converts acoustic sound into electric current; by contrast, a loudspeaker converts electric current back into acoustic sound.
A very short-duration signal. Due to their speed, transients are difficult to record and reproduce accurately.
The power that is lost in transmitting a signal from one point to another. Transmission loss occurs in both wired and wireless systems.
USB (Universal Serial Bus)
A standard for interconnecting computers and peripherals. USB is a plug-and-play interface found on both Windows® and Apple® computers. It allows new devices to be configured automatically upon attachment without the need to reboot, run setups, or add adapter cards.
To pass an electrical signal through a wire requires two conductors. In unbalanced circuits, one conductor carries both signal and supplies ground, unlike in balanced connections, which have a third wire dedicated to ground. For this reason, unbalanced circuits are less expensive to produce, but the down side is the cables are susceptible to noise, especially with longer cable lengths.
A device or setting which does not amplify or attenuate the signal is said to be at “unity gain.” Many processors are set up for unity gain; that is, they can be plugged into a system without changing its overall levels. In practice, unity gain is often a desired setting for maintaining gain staging, and for minimizing distortion and feedback.
Voltage is an electric charge, or potential, between two points, one being of higher relative voltage than the other is. The unit of measurement is called the “volt.” If you think of electricity as a flow of water, voltage can be thought of as the pressure, while current (amperage) would be the water volume.
Short for Waveform Audio File Formet, WAV is the standard for storing an audio bitstream on Windows-based PCs. Although a WAV file can hold compressed audio, the most common WAV format contains uncompressed audio in PCM format.
A metric unit of power defined as one Joule per second. The watt has become a common term in audio used to describe the power handling capabilities and/or requirements of speakers, and the power delivery capabilities of amplifiers.
A sound that contains every frequency within the range of human hearing in equal amounts.
The name for a type of audio connector used for sending balanced signals and microphone feeds. An XLR connector consists of three pins housed in a barrel and often having locking components. The male side is for sends, and the female is always a receiving connector.
A cable used to split or combine two signals. Another use is to send a signal out of one side of the "Y" and return through the other, as when patching into an insert point of a console.
An audio artifact resembling distortion that is sometimes heard from inexpensive digital devices when changing certain parameters. It is caused by quantization, where under digital control changing values are spaced apart and stepped, as opposed to the analog domain, where such changes are continuous. It becomes noticeable when the resolution of the digital control signal being used to change the parameter is too coarse to avoid audible stepping between successive levels.