The abbreviation for Analog-to-Digital conversion. Since computers “think” in discrete steps, in order to convert analog audio signals to the digital domain, it’s necessary to describe the continuous analog waveform mathematically as a succession of discrete amplitude values. In an analog-to-digital converter, this is accomplished by capturing, at a fixed rate, a rapid series of short “snapshots”—samples —of a specified size. Each audio sample contains data that provides the information necessary to accurately reproduce the original analog waveform. For more information, please refer to this article.

Absolute Phase occurs when the positive pressure on a microphone equals positive pressure in the reproduced signal.

As it relates to acoutics, absorption refers to what happens when sound waves encounter a surface that limits reflectivity and prevents them from bouncing back into the room or passing through the surface. For more information on how absorbtion can be used for acoustic treatment, please refer to this article.

The acoustic horn is used to increase the overall efficiency of the driver. Most commonly used for high-frequency reproduction, the usual configuration consists of a compression driver that produces sound using a small metal diaphragm that is vibrated by an electromagnet. The compression driver assembly is then attached to the base of the horn’s throat. Well-designed horns, like the Pivot X110 horn used in the PreSonus ULT-series loudspeakers, can better control the high-frequency response throughout their coverage angle for improved off-axis response.

Also known as the “bass reflex port,” an acoustic port redirects the inward pressure produced by the outward movement of the speakers. The backward motion of the diaphragm pushes sound waves out of the port and boosts the overall sound level. Ported speaker designs are much more efficient because the power moving the driver produces two sound waves instead of one

The physics of sound. Specifically, the study of the properties and behaviors of sound waves.

The Alesis ADAT modular digital multitrack tape recorder allowed users to record eight tracks of digital audio simultaneously. The ADAT Optical interface protocol, commonly referred to as “ADAT Lightpipe,” was developed to stream eight channels of 16-, 20-, or 24-bit digital audio at 44.1 kHz or 48 kHz, allowing multichannel digital transfers between ADAT digital recorders and other digital audio devices over a single fiber-optic cable. The ADAT Lightpipe format has been adopted by many audio manufacturers because it's a compact way to transfer multichannel digital audio data between devices.

For more information, please review this article.

Developed by the Audio Engineering Society and the European Broadcasting Union, AES/EBU (officially known as AES3) is a 2-channel format that can carry audio signals at up to 192 kHz. AES/EBU employs a 3-pin XLR connector, which is the same connector used for most professional microphones. A single cable carries both channels of audio data.

For more information, please review this article.

The height of a wave form as is relates to the zero line crossing. This generally equates to the overall signal strength or volume of the sound.

AFL sends the channel or subgroup signal to the solo bus post-fader so that you can control the level of the soloed signal with the fader.

A studio monitor high-frequency driver design that employs an ultra-light material that is inlaid with an aluminum circuit that functions as a voice coil. This material is folded like an accordion and moves like a bellows, launching sound waves from the two surfaces rushing toward one another. For more information, please see this article.

As its name implies, an Aux mix allows you to create an alternate, or “auxiliary,” mix that can be routed to an output separate from the Main bus. Aux buses have many applications, the two most common of which are creating monitor mixes and inserting external effects processors into the mix. When you turn up the aux send level on a channel, its signal is sent to the corresponding aux output at the level you choose. In this way, the same channel can be used to create several alternate mixes to the main mix.

An audio interface is an essential component to any modern recording studio. Its main function is to convert analog audio to digital audio and vice versa, so you can record to and play back from a computer. But more than just the converter that translates what you are hearing into information your computer can understand, an audio interface is the central hub around which your studio in connected. Your audio interface may also include any or all of the following: a MIDI interface, microphone preamps, instrument preamps, onboard monitoring functions, and even onboard plug-in processing, to name just a few of the features available to modern recordists.

PreSonus makes a wide range of audio interfaces that are designed to suit just about any use case you may have. These interfaces range from small compact devices like the AudioBox USB 96 to a professional 64-channel mixer with integrated recording and DAW control like the StudioLive 64S. Let’s go through some of those options and why you would select one over the other depending on how you’d like to record in your studio.

AVB (Audio Video Bridging) is an extension to the Ethernet standard designed to provide guaranteed quality of service, which simply means that audio samples will reach their destinations on time. AVB allows you to create a single network for audio, video, and other data like control information, using an AVB-compatible switch. This enables you to mix normal network data and audio network data on the same network, making it easier to create both simple and complex networks. Numerous audio companies have adopted it, and more companies are adding it all the time.

For more information, please review this article.

An AVB controller can be a talker, a listener, or neither. These devices handle routing, clock, and other settings for AVB devices using AVDECC.

For more information, please review this article.

AVB Listeners are the destinations for the streams sent out by the AVB Talkers.

Fore more information, please review this article.

AVB Talkers act as the source for an AVB stream, sending out audio onto the network.

For more information, please review this article.

This is the network hub to which every Talker and Listener must be connected. At its most basic level, an AVB Switch analyzes and prioritizes traffic on the network. It should be noted that just like there can be multiple talkers and listeners on the same AVB network, there can also be multiple AVB Switches.

For more information, please review this article.

A predefined, limited range of frequencies with a fixed upper and lower threshold. For more infomation on how bandwidth relates to equalization, please review this article.

The frequency range from 60 Hz to 250 Hz. This range contains the fundamental notes of the rhythm section.

The act of powering the two components of a loudspeaker (the high-frequency driver and the low-frequency driver) with two separate amplifiers. By separating the frequencies before they hit the amplifiers, a biamped system removes one of the major sources of intermodulation distortion. The resulting sound is more open and clear and less fatiguing.

A polar pattern is the three-dimensional space around the microphone capsule where the capsule is most sensitive to sound. Also known as “figure-eight,” bidirectional microphones pick up audio from both the front and the back.

The on/off status of each switch in a computer is represented as 1 or 0, a system known as binary. Thus, a string of binary digits—bits —is used to describe anything a computer does, including manipulating and displaying text, images, and audio. Computers can manage entire strings of these bits at a time; a group of 8 bits is known as a byte; one or more bytes compose a digital word. Sixteen bits (two bytes) means that there are 16 digits in a word, each of them a 1 or 0; 24 bits (three bytes) means that there are 24 binary digits per word; and so on.

The number of bits in a word determines how precise the values are. Working with a higher bit depth is like measuring with a ruler that has finer increments: you get a more precise measurement. When the values are in finer increments, the converter doesn’t have to quantize as much to get to the nearest measuring increment.

For more information, please refer to this article.

A buffer is a region of memory storage used to temporarily hold data while it is being moved from one place to another.

For more information, please review this article.

The most common microphone polar pattern is called “cardioid.” The word “cardioid” is derived from the Greek word for “heart.” A cardioid polar pattern means the microphone primarily picks up sound within an area that is roughly heart-shaped when shown on a two-dimensional graph. (In three dimensions, the pickup region looks more like an apple.) Cardioid mics primarily are sensitive to sound on one side or end of the microphone—that is, they are unidirectional—and reject sound from the sides and rear of the mic.

Center divergence allows you to pan each channel across three positions, from left to center to right and back again, or through any position in between. This powerful feature provides a true LCR panning experience and provides more precise panning placement for each channel in your LCR system, affording more clarity and greater system efficiency.

Center Divergence works in conjunction with the pan control, setting the prominence of each channel in the Center bus as the pan position approaches center.

For more information, please review this article.

A modulation effect created by mixing a source signal with one or more pitch-shifted copies that is then modulated by an LFO. See this article for more information.

This amplifier type uses output transistors to conduct the full waveform. Because Class A amplifiers can provide very low distortion, they are popular in audiophile applications; however, this amplifier type is too inefficient for pro audio applications.

As its name indicates, this is a hybrid between Class A and Class B designs. At low levels, it functions just like a Class A, keeping the crossover distortion low. At higher levels, it changes to work as a Class B. The advantages of this design can best be heard in high-frequency reproduction. PreSonus AIR-series loudspeakers use a Class AB amplifier to power the high-frequency driver for this reason.

This design conducts half of the waveform in separate sets of positive and negative signals. This efficient design suffers from crossover distortion where the positive and negative devices meet, especially at low signal levels.

This amplifier controls the output using pulse modulation of the input signal. The pulses control the voltage and current flow at the output. Class D amplifiers are very efficient and lightweight, making them popular for mobile loudspeakers. This is the most common amplifier type used in active pro audio loudspeaker designs.

As its name indicates, a cluster system consists of a group of point-source loudspeakers being used in conjunction with one another. This is a flexible option to create a custom coverage pattern and to provide a higher SPL output. However, like a distributed loudspeaker system, a loudspeaker cluster requires a skilled system designer to achieve the desired result.

A coaxial speaker like the one used in the PreSonus Sceptre® S6 and S8 places the high-frequency driver in the center of, and on the same axis as, the low-frequency driver, which is similar to the way the human ear works. Coaxial designs offer a symmetrical response both horizontally and vertically. This means a wider “sweet spot” that is more consistent throughout the room. Properly designed coaxial speakers can also offer a seamless crossover transition because of their symmetrical response.

The coil of wire that is attached to the base of a cone driver. It provides the movement to the cone by reacting to the magnetic field as electrical current passes through it. In other words, it is the motor that moves the cone driver. In general, the larger the voice coil, the stronger the movement will be, resulting in a tighter bass response. For more information on loudspeaker components, please review this article.

The ratio sets the compression slope, which is a function of the output level versus the input level. For more information on Compressors, please refer to this article.

A type of amplifier in which gain is dependent on the signal level passing through it. The set maximum level of a compressor allows to pass through causes automatic gain reduction above the predetermined signal level, also known as the "threshold." Punch, apparent loudness, and presence are just three of the many terms used to describe the effects of compression. For more information on compressors and dynamics processing, please review this article.

Sets the speed at which the compressor acts on the input signal. A slow attack time allows the beginning component of a signal (commonly referred to as the initial transient) to pass through, uncompressed, whereas a fast attack time triggers compression immediately when a signal exceeds the threshold. For more information on Compressors, please refer to this article.

Fixes the attack and release times of a compressor to a preprogrammed curve. For more information on Compressors, please refer to this article.

Sets the length of time the compressor takes to return the gain reduction back to zero (no gain reduction) after crossing below the compression threshold. Very short release times can produce a choppy or “jittery” sound, especially when compressing instruments that have a lot of low-frequency components, such as a bass guitar. Very long release times can result in an overcompressed, or “squashed,” sound. For more information on Compressors, please refer to this article.

The use of one signal source to trigger the compressor threshold for another. For more information, please refer to this article.

When the signal’s amplitude (level) exceeds the threshold setting, the compressor engages. As the threshold is lowered, compression begins at a lower amplitude (volume) and further limits the dynamic range of the signal. For more information on Compressors, please refer to this article.

This is the most common type of microphone you’ll find in a studio. A condenser capsule consists of a thin membrane, or diaphragm, in close proximity to a solid metal plate. The diaphragm must be electrically conductive so when sound waves hit it, the diaphragm moves back and forth relative to the solid backplate and creates an electrical signal. Condenser microphones come in small- and large-diaphragm varieties. In general, the larger the diaphragm, the more sensitive and more delicate the microphone. Because of this, large-diaphragm condenser microphones are typically used on vocals, acoustic guitars, and as room mics, while small-diaphragm condensers are used for close-miking instruments and as overheads for drums.

Named for their diaphragm shape, most loudspeaker cone drivers are paper or composite with two surrounding components and are typically responsible for low-frequency reproduction. For more information on loudspeaker components, please review this article.

Combining the uniform frequency response of a point source design with the uniform SPL coverage of a line array, constant directivity loudspeakers like the PreSonus CDL12 provide a "best of both worlds" approach for mid-sized mobile and large installed systems by supporting both single speaker and vertical array configurations

An internal crossover separates the frequencies coming into a speaker and distributes them appropriately to the woofer and tweeter. This helps the speaker run more efficiently and reproduce the frequency spectrum more reliably. External hardware crossovers allow the user to send a specified range of frequencies to each loudspeaker in a system.

A type of network topology that connects devices serially. This configuration requires that at least one device be equipped with an onboard network switch. For more information on Network Topologies, please refer to this article.

The time (in milliseconds) between the source signal and its echo. See this article for more information.

Low-frequency waves are powerful enough to cause the walls, ceiling, and even the floor to flex and move. This is called “diaphragmatic action,” and it dissipates energy and strips away the low-end definition. Therefore, if a room’s walls and floor are made of solid brick and concrete that don’t vibrate much, the bass response is going to be much more powerful than in a room where the walls are normal sheet rock construction and the floors are hardwood. For more information, please refer to this article.

Sound-wave diffusers are designed to break up standing waves by reflecting the waves at different angles. For more information, please refer to this article.

As it relates to audio interfaces, direct monitoring provides the user with ability to listen to the audio interface inputs on the device itself before the signal is transported to the driver. For more information, please review this article.

In a distributed audio network, each musician could have their own node on the network, potentially. Multiple networked stage boxes can be spread around the stage, making the analog cable runs as short as possible to minimize signal degradation. Take this concept a step further, and multiple sources can be spread throughout a large facility, each sitting on the network to be sent to many mixers on the network, not just the one at front-of-house.

For more information, please review this article.

A distributed loudspeaker system uses multiple smaller format speakers spread throughout the coverage area. These systems often supplement the main front-of-house system to provide coverage for areas that may be outside the main listening area. A distributed system can be challenging to configure because it requires carefully designed delay settings, level matching, and tuning.

In contrast to compression, which decreases the level of a signal after it rises above the compression threshold, downward expansion decreases the level of a signal after the signal goes below the expansion threshold. The amount of level reduction is determined by the expansion ratio. This type of expansion reduces the level of a signal when the signal falls below a set threshold level. This is most common used for noise reduction. For more information on Downward Expansion and Dynamics Processing, please refer to this article.

A speaker component that produces sound waves within a limited frequency range.

This type of expansion is the opposite of compression. It increases the the dynamic range by emphasizing the peaks of an audio signal, amplifying the signal level that rises above the set threshold. For more information on Expansion and Dynamics Processing, please refer to this article.

A dynamic microphone essentially works like a loudspeaker in reverse: A coil is attached to the back of a membrane inside a powerful magnet. As the membrane moves with the audio waves, the coil moves with it inside the magnetic gap. This movement induces a small voltage into the coil, converting sound into an electrical signal. Because of their design, dynamic microphones are durable and can accept very loud sound pressure levels. Typical applications are live vocals and close-miking drums or guitar amplifiers in the studio.

Dynamic range can be defined as the ratio between the loudest possible audio level and the lowest possible level. For more information, please refer to this article.

A filter or group of filters that adjusts the volume level of a frequency, or range of frequencies, within an audio signal. For more information on Equalizers, please refer to this article.

An elliptical waveguide is designed to shape the high-frequency wavefront of a soft dome tweeter, controlling the dispersion pattern so that it is as uniform and smooth as possible. By minimizing early reflections, the response becomes more consistent both on- and off-axis. For studio monitors like the PreSonus Eris E5XT and E8EXT, this facilitates widening the horizontal dispersion to provide a wider sweet spot while maintaining a tightly focused vertical plane. The result is greater detail and depths that reveal subtle transients, delicate reverb trails and a consistent listening experience in any mix environment.

For more information on studio monitors, please review this article.

Both data and audio networks rely on a set of standards for Ethernet cabling infrastructure to ensure that network performance is both reliable and consistent. These standards include specifications for the cable construction itself, as well as specifications for the termination of cabling and physical connections to devices. More information on Ethernet cables can be found in here.

Expanders increase the dynamic range of a signal after the signal crosses a threshold. There are two basic types of expansion: dynamic and downward. For more information on Expanders and Dynamics Processing, please refer to this article.

A modulation effect created by mixing two identical signals together and delaying one by a constantly varying time. See this article for more information.

With two parallel reflective surfaces in a room, such as opposing walls or the floor and ceiling, there is always the possibility of successive, repetitive reflections that are equally spaced in time. Known as “flutter echoes,” these reflections can produce a perceived pitch or timbre that colors the audio heard in the room and reduce intelligibility. Diffusors can break up flutter echoes by reflecting the sound waves in different directions so that the repetitive reflections are eliminated. For more information, please refer to this article.

This is the frequency range that a loudspeaker can reliably reproduce. Operating range limit measurements are not standardized and can be specified at -3 dB, -6 dB, or even -10 dB. As a general rule of thumb, a -3 dB measurement will be the most accurate representation of how a speaker will actually perform. A -10 dB measurement typically exaggerates a loudspeaker’s low-end capability.

The process of setting the correct amount of gain for each component in an interconnected audio system. Setting the optimal gain at each stage in an audio system is essential to minimize noise and optimize the performance of each component. See this article for more information.

In general, most graphic EQs have between 7 and 31 bands. Professional sound-reinforcement graphic EQs generally have 31 bands, and the center frequency of each band is spaced 1/3 of an octave away from the center frequency of the adjacent bands, so that three bands cover a combined bandwidth of one octave. 2/3-octave designs (15 bands or fewer) are also common on affordable compact mixers, guitar and bass amplifiers, and other applications were less precision is required. In traditional graphic EQ designs, the center frequency of each band is fixed. For more information on Graphic EQs, please review this article.

Applies gain reduction to the signal immediately after the signal exceeds the level set by the threshold. For more information on Compression, please refer to this article.

The frequency range from 4 kHz to 16 kHz. This frequency range extends to the upper limits of human hearing. The range from 4 kHz to 6 kHz is generally refered to as "Presence". The range from 6 kHz to 16 kHz adds or removes "Brilliance."

The part of the speaker that is responsible for high- and upper mid-frequency reproduction. Sometimes referred to as the "tweeter."

Also known as a low cut filter, a high pass filter attenuates all frequencies below a set threshold, while letting every frequency above the set threshold passes through uneffected. For more information, please refer to this article.

A polar pattern is the three-dimensional space around the microphone capsule where the capsule is most sensitive to sound. Hypercardioid microphones are considered more directional than cardioid microphones because they are less sensitive at the sides of the pick-up pattern, even though they do pick up a slight amount of audio from the rear.

As its name implies, an input delay delays the input signal at the source. An input delay has many uses. On small stages where the guitar amp and the kick and snare mics can be clearly heard in the vocal mic, an input delay can “move up” the backline. Delaying the backline so that the close mic’d signals and the bleed in the vocal mic align with one another at the mixer will decrease comb filtering that blurs the mix. This will tighten the overall mix and give it more clarity and punch.

In large venues, the bottom snare mic can be aligned with the top mic, or the bass cabinet mic can be aligned with the direct line to create a more coherent signal. This is especially useful to prevent phasing problems.

For more information, please review this article.

In addition to a device’s MAC address, every NIC has a user-definable addressing layer to make it easier for network managers to configure their local network. Called the Internet Protocol or ‘IP’ address, this is normally 4 bytes long (IPv4) consisting of the network number and a host address. The division between the two is also 4 bytes long and called the subnet mask. For more information, please review this article.

The use of a specified frequency to trigger the threshold of a noise gate or a compressor. For more information, please refer to this article.

The ability to audition the trigger signal of a Key Filter for a compressor or a noise gate. For more information on Compressors, Noise Gates, and other dynamics processing, please refer to this article.

Latency is the time it takes for the sound you are generating to come back to your headphones, and many things impact it. But basically, everything between your guitar, for example, and your headphones, needs a little time to perform its function, and that time adds up.

For more information, please review this article.

Similar to a compressor, a limiter is an amplifier that limits the upper dynamic range of a signal to a specific threshold. Unlike a compressor, which works gradually to reduce the signal, the limiter prevents virtually any increase in gain at the upper end of the dynamic range. For more information on limiters and dynamics processing, please refer to this article.

A line array is one of the most popular loudspeaker systems for installations and large touring productions. It consists of multiple loudspeakers suspended in a straight or curved vertical contour. Line arrays have become popular because of their ability to provide an optimal listening experience throughout a large audience area because of their consistent SPL performance.

A type of digital comrpession that analyzes the signal before it is processed to accurately place the attack time from the onset of the threshold being exceeded. For more information on Compressors, please refer to this article.

The frequency range from 250 Hz to 2 kHz. Boosting the range from 250 Hz to 500 Hz will accent ambience in the studio and will add clarity to bass and lower frequency instruments. The range between 500 Hz and 2 kHz can make midrange instruments (guitar, snare, saxophone, etc.) “honky,” and too much boost between 1 kHz and 2 kHz can make a mix sound thin or “tinny.”

The speaker component that produces the low-mid and low frequencies. The larger the size, the lower the frequency range will extend.

Low-pass shelving filters pass all frequencies below a specified cutoff frequency, while attenuating all the frequencies above the cutoff. For more information on Equalizers, please refer to this article.

Every NIC must have a Media Access Control (MAC) address programmed by the manufacturer. Every MAC address is unique, and the allocation of MAC addresses to networking manufacturers is strictly managed by the IEEE standards organization. For more information, please review this article.

When compressing a signal, gain reduction usually results in an overall attenuation of level. The gain control allows you to restore this loss in level and readjust the volume to the precompression level (if desired). For more information on Compressors, please refer to this article.

On the simplest level, a matrix mix is a mix of mixes. A matrix mix allows you to combine any bus on your mixer and sum them together. Some mixers, like the StudioLive Series III also provide the capability to blend input channels with bus outputs.

MTM configurations such as the PreSonus Eris E44 and E66 feature two midrange drivers that cover the same frequency range, with a high-frequency driver nested between them. Because the two woofers cover the same frequency range and are placed so that their acoustic centers are less than one wavelength apart, the combined signal of the two drivers propagates forward as a single waveform. This provides a much more dynamic output than their relatively small size would normally afford. The two woofers also work to partially contain the dispersion of the tweeter, minimizing phase displacement. This results in smoother frequency response and an ultra-wide, detailed stereo soundstage.

Maximum SPL is measured at one meter using continuous broadband pink noise. This number does not mean that the speaker is reproducing every frequency at that range, so it’s important to evaluate potential loudspeakers with your ears, not just the cut sheet.

A NIC is built into a computer, digital mixer, networked stage box, etc., allowing these devices to communicate with other devices on a digital network.

A node is any device connected to a network. This could be a computer, an iOS or Android device, your mixer, etc. For more information on networking audio devices for remote contol, please review this article.

Any part of the network that is separated from other parts of the network by a router, switch, or bridge. For more information on networking audio devices for remote contol, please review this article.

Network switches bring all the cables together into a central hub and enable the correct routing of information throughout the network. For more information, please review this article.

The way in which each node connects to a network. For more information on network topologies, please review this article.

A noise gate helps to reduce unwanted sounds by only allowing the signal to be heard once it has exceeded a certain amplitude. For more information on Noise Gates and Dynamics Processing, please refer to this article.

The gate attack time sets the rate at which the gate opens. A fast attack rate is crucial for percussive instruments, whereas signals such as vocals and bass guitar require a slower attack. Too fast of an attack can, on these slow-rising signals, cause an artifact in the signal, which is heard as a click. All gates have the ability to click when opening but a properly set gate will never click. For more information on Noise Gates, please refer to this article.

Hold time is used to keep the gate open for a fixed period after the signal drops below the gate threshold. This can be very useful for effects such as gated snare, where the gate remains open after the snare hit for the duration of the hold time, then abruptly closes. For more information on Noise Gates, please refer to this article.

The gate range is the amount of gain reduction that the gate produces. Therefore, if the range is set at 0 dB, there will be no change in the signal as it crosses the threshold. If the range is set to -60 dB, the signal will be gated (reduced) by 60 dB. For more information on Noise Gates, please refer to this article.

The gate-release time determines the rate at which the gate closes. Release times should typically be set so that the natural decay of the instrument or vocal being gated is not affected. Shorter release times help to clean up the noise in a signal but may cause “chattering” in percussive instruments. Longer release times usually eliminate “chattering” and should be set by listening carefully for the most natural release of the signal. For more information on Noise Gates, please refer to this article.

The gate threshold sets the level at which the gate opens. Essentially, all signals above the threshold setting are passed through unaffected, whereas signals below the threshold setting are reduced in level by the amount set by the range control. If the threshold is set fully counterclockwise, the gate is turned off (always open), allowing all signals to pass through unaffected. For more information on Noise Gates, please refer to this article.

Also called “coverage angle,” “nominal dispersion,” and “dispersion pattern,” this measurement will show you how wide or narrow a loudspeaker’s horizontal and vertical coverage patterns will be. For more information on loudspeakers, please review this article.

A polar pattern is the three-dimensional space around the microphone capsule where the capsule is most sensitive to sound. Omnidirectional microphone pick up audio uniformly in all directional, making this pattern ideal for measurement microphones, like the PreSonus PRM1.

The performance range from the noise floor to the maximum operation level.

A fully parametric EQ offers continuous control of the bandwidth, which determines the range of frequencies affected, or control over the Q, which is the ratio of the center frequency to the bandwidth. For most purposes, a Q control accomplishes the same thing as a bandwidth control but the two are not identical. For more information on Parametric EQs, please refer to this article.

Also known as “Step Sequencing,” Pattern Sequencing lets you create a beat using a grid.

For more information, please review this article.

This is the maximum power that an amplifier can deliver or a passive speaker can momentarily handle without being damaged. It's important to mention that pushing a speaker to its peak power load for longer than a few seconds can damage a speaker.

Plug-ins are software applications that run within a DAW. Most commonly, plug-ins refer to effects, like reverb and EQ.

A point-source loudspeaker is what most people think of when discussing loudspeakers. Available in two- and three-way designs, point-source loudspeakers are fairly easy to set up because they provide very good coherence as compared to multi-speaker designs. While ideal for mobile applications, small venues, and floor monitoring, point-source speakers may not be capable of achieving the sound pressure level necessary for larger venues.

For more information on loudspeakers, please review this article.

While not technically a network in and of itself, a Point-to-Point configuration relies on networking protocols in order for two devices to transmit data to one another. This is the direct connection of one network device to another. Please refer to this article on Network Topologies for more information.

A polar pattern is the three-dimensional space around the microphone capsule where the capsule is most sensitive to sound. The most common microphone polar pattern is called “cardioid.” Other variations include: Hypercardioid, Supercardioid, Omnidirectional, and Bidirectional.

PFL sends the channel or subgroup signal to the solo bus before it reaches the fader so the fader does not affect the soloed signal.

A variation on the semi-parametric EQ is the quasi-parametric EQ, which typically provides full frequency and gain adjustment but only two or three Q settings. For more information on Quasi-Parametric EQs, please refer to this article.

In acoustics, reflectivity refers to a sound wave bouncing or "reflecting" off a hard surface. For more information, please refer to this article.

A ribbon microphone is a special type of dynamic microphone that uses a narrow strip of foil instead of a separate membrane and coil, so that the ribbon itself is moving inside the magnetic gap. Since the ribbon is much lighter than a separate membrane and coil assembly, ribbon microphone are much more sensitive, making them great for studio use. Please note that most ribbon microphones are very fragile and need to be handled with great care.

The RMS power rating is the amount of continuous power that an amplifier can output or a passive speaker can handle. As a general rule of thumb, this rating is roughly half of the peak power rating.

The Nyquist-Shannon sampling theorem states that in order to accurately reconstruct a signal of a specified bandwidth (that is, a definable frequency range, such as 20 Hz to 20 kHz), the sampling frequency must be greater than twice the highest frequency of the signal being sampled. If lower sampling rates are used, the original signal’s information may not be completely recoverable from the sampled signal .

If the sampling frequency is too low, aliasing distortion can result. Aliasing is a major concern when using analog-to-digital conversion. Improper sampling of the analog signal will cause high-frequency components of the signal to be aliased with genuine lower-frequency components. If this happens, the digital-to-analog conversion will create an incorrectly reconstructed signal.

For more information, please refer to this article.

On a semi-parametric EQ, the gain and frequency are adjustable but the Q and bandwidth are fixed at a preset value. For more information on Semi-Parametric EQs, please refer to this article.

A shelving EQ attenuates or boosts frequencies above or below a specified cutoff point. Shelving equalizers come in two different varieties: high-pass and low-pass. Low-pass shelving filters pass all frequencies below a specified cutoff frequency, while attenuating all the frequencies above the cutoff. A high-pass filter does the opposite, passing all frequencies above the specified cutoff frequency while attenuating everything below. Usually, the frequencies beyond the cutoff are rolled off, following a predetermined curve, not cut off sharply, as with a “brickwall” filter. For more information on Equalizers, please refer to this article.

A high-frequency driver design employed by studio monitors. The round shape of this type of design radiates sound with a wide dispersion pattern to create a larger sweet spot. The larger the diameter of the dome, the wider the sweet spot. Usually made from a high-quality textile, like the silk, these designs are known for their smooth and refined sound. For more information on studio monitors, please see this article.

The onset of gain reduction occurs gradually after the signal has exceeded the level set by the threshold. For more information on Compression, please refer to this article.

When mixing live, or when recording multiple musicians at once, it is often necessary to quickly listen to just one instrument or group. The solo bus can be used with the monitor bus (if available) to provide the front-of-house engineer with a way to isolate channels in a dedicated mix, without affecting what the audience and musicians hear.

In general, there are three different types of solo options: AFL, PFL, and SIP.

• AFL (After-Fader Listen). AFL sends the channel or subgroup signal to the solo bus post-fader so that you can control the level of the soloed signal with the fader.
• PFL (Pre-Fader Listen). PFL sends the channel or subgroup signal to the solo bus before it reaches the fader so the fader does not affect the soloed signal.
• SIP (Solo In Place). This is also known as “destructive solo.” When channels are soloed in this mode, every channel that isn’t soloed will be muted, and only the soloed channels will be sent to their assigned outputs. While useful for dialing in dynamics during soundcheck, this mode is dangerous during a live show.

Also known as “destructive solo,” when channels are soloed in this mode, every channel that isn’t soloed will be muted, and only the soloed channels will be sent to their assigned outputs. While useful for dialing in dynamics during soundcheck, this mode is dangerous during a live show.

“Sample Multiplexing” or S/MUX is used to transmit high-bandwidth digital audio using lower-bandwidth technology, such as ADAT Lightpipe. S/MUX works by joining two or more digital audio channels to represent a single higher-bandwidth channel. By using S/MUX technology, you can stream 8 channels of digital audio at 88.2 kHz or 96 kHz over the same Lightpipe connection originally designed to stream 16 channels of 44.1 kHz or 48 kHz audio.

For more information, please review this article.

S/PDIF (Sony/Philips Digital Interface) was co-developed by Sony and Philips to transfer stereo digital audio. It is essentially a consumer version of AES/EBU, and as with AES/EBU, a single cable carries both channels of the stereo audio signal. Digital Audio Tape (DAT) machines were among the first devices to be equipped with this protocol but is has since become popular in consumer audio products such as DVD players, as well as in semi-pro and professional audio products.

For more information, please review this article.

The process of setting the output level of a loudspeaker or studio monitoring system so that a specific metered audio level in a DAW or on a mixer equals a predetermined SPL from the speaker. Proper calibration can help reduce unwanted noise and maximize the reference capabilities of different speaker types. See this article for more information.

The measurement of the acoustic pressure of a sound relative to a reference value. Measured in decibels (dB), SPL is generally used to define how loud a sound is.

A method of speaker calibration that ensures when the output meters in a DAW or on a mixer register 0 dB, the SPL at a set position is 85 dB (or another predetermined SPL level). See this article for instructions.

When a room’s width or length correlates directly to the length of a waveform at a specific frequency, a standing wave can occur where the initial sound and the reflected sound begin to reinforce each other. The reflective wave travels back along the same path, bounces off the other wall, and the cycle repeats. For more information, please refer to this article.

A type of network topology that relies on a central network switch to which all network nodes are connected. For more information on Network Topologies, please refer to this article.

The lowest range of human hearing from 16 Hz to 60 Hz. These very low bass frequencies are felt, rather than heard, as with freeway rumbling or an earthquake.

A subgroup allows you to combine multiple channels into a single bus so that the overall level for the entire group is controlled by a single fader and can be processed globally on the output bus to add dynamics processing like noise gate, limiter, and compression, as well as inserting an overall EQ curve. This global processing is in addition to the processing available for each channel.

A subwoofer is designed to reproduce low- and sub-low-frequency content only. When carefully tuned to the full-range system, a subwoofer will naturally extend the low end to create a fuller listening experience. A properly calibrated three- or four-way system can improve your listening environment by offloading much of the bass-frequency reproduction to the sub, letting the low-frequency component of the full-range system focus on the low mids.

A polar pattern is the three-dimensional space around the microphone capsule where the capsule is most sensitive to sound. Supercardioid microphones are a little less directional than their hypercardioid cousins, but have the advantage of offering a slightly smaller rear lobe.

USB-C™ is an industry-standard connector that is designed to transmit both data and power for multiple protocols – unlike USB-A and USB-B connectors (both 2.0 and 3.0) which are designed for their respective USB protocols only. For more information, please review this article.

Variable feedback, or regeneration, produces multiple decaying repeats. Increasing the feedback value increases the number of echoes, as well as the resonance that is created as one echo disappears into another. For more information on Delay Effects, see this article.

As their name indicates, virtual instruments are software plug-ins that run within a DAW.

Using a multitrack recording of a previous performance to set mix levels, EQ, and dynamics with or without every musician present. For more information, please review this article.