Audio Glossary

An A-Type plug is a type of electrical plug commonly used in North America, Central America, and parts of South America, Asia, and the Caribbean. It’s characterized by two flat parallel pins, with or without a grounding pin (depending on the variant). The A-type plug is designed for use with corresponding type A electrical outlets. It’s typically used for low-power devices like smartphones, laptops, and small appliances.

A-Weighting is a type of frequency weighting commonly used in audio measurements and equipment. It’s a method to adjust or filter the measured sound levels to approximate the sensitivity of the human ear to different frequencies. The A-weighting curve emphasizes frequencies in the midrange while reducing the contribution of low and high frequencies, reflecting the fact that human hearing is less sensitive to very low and very high frequencies.

This weighting is often used in sound level meters and other audio equipment to provide measurements that better correspond to how humans perceive sound. It’s particularly useful in environments where the noise contains a mix of frequencies and you want to assess its impact on human hearing accurately. A-weighted measurements are denoted with the unit “dBA” (A-weighted decibels).

Analog to Digital conversion. The A/D conversion process involves taking samples of the analog signal at regular intervals and assigning numerical values to represent the amplitude of the signal at each sample point. These numerical values are then stored digitally for processing, storage, or transmission.

A/D conversion is a fundamental process in digital audio recording, playback, and processing. It allows analog audio signals to be stored, manipulated, and transmitted using digital technology, offering benefits such as improved fidelity, easier editing, and the ability to apply various digital audio effects.

Absolute phase refers to the timing relationship between different audio signals or between different components within a single audio signal. It’s about the alignment of the waveform’s peaks and troughs across the audio spectrum.

Simply put, absolute phase determines whether the waveform of an audio signal is in sync or out of sync with a reference point. This reference point is often considered to be the point of origin or starting point of the waveform.

It is important to note that absolute phase is a topic of debate among some audio engineers. While some argue that it’s critical for preserving the integrity of the audio signal and ensuring accurate reproduction, others contend that it’s less important than factors like frequency response, distortion, and dynamic range.

Practically speaking, absolute phase discrepancies are typically only noticeable in certain situations, such as when mixing or mastering audio, or when using stereo equipment with highly accurate reproduction capabilities. In everyday listening environments, the impact of absolute phase is often negligible compared to other factors influencing audio quality.

Absorption refers to the process by which sound energy is converted into other forms of energy, typically heat, within a material. This conversion reduces the reflection of sound waves and helps to dampen or attenuate sound within a space.

Absorption materials are commonly used in acoustic treatments for rooms, studios, theaters, and other environments where controlling reverberation and echo is important. These materials are designed to absorb sound waves rather than reflect them back into the space, thereby improving the acoustics and reducing unwanted noise.

Various materials can be used for sound absorption, including acoustic foam, fiberglass panels, fabric-wrapped panels, perforated panels, and specialized acoustic tiles. The effectiveness of an absorption material depends on factors such as its thickness, density, porosity, and surface characteristics, as well as the frequency range of the sound it’s intended to absorb.

An accelerometer is a sensor used to measure acceleration forces, including the force of gravity. It’s commonly used in various applications to detect changes in velocity, orientation, or vibration.

Accelerometers typically consist of one or more small, sensitive elements that respond to changes in acceleration by generating electrical signals proportional to the acceleration they experience. These signals can then be processed and used to determine the device’s movement, tilt, or vibration.

In audio, accelerometers are not typically used directly to capture or process sound signals like microphones or other audio sensors. Instead, accelerometers can be employed in audio equipment and devices in various ancillary roles. Here are a few examples:

1. Vibration Monitoring: In high-end audio equipment, accelerometers might be used to monitor and mitigate vibration. Vibrations can negatively impact audio quality by causing mechanical noise or interfering with delicate components. By using accelerometers to detect and analyze vibrations, audio equipment can adjust internal components or activate damping mechanisms to reduce unwanted noise and vibrations, thus improving audio performance.
2. Motion-Activated Controls: Some audio devices, particularly portable or wearable ones like headphones or smart speakers, may incorporate motion-activated controls. Accelerometers can be used to detect gestures or movements, allowing users to control playback, adjust volume, or perform other functions by tapping, shaking, or tilting the device.
3. Spatial Audio Processing: In virtual reality (VR) or augmented reality (AR) audio systems, accelerometers can be used in conjunction with other sensors to track the movement and orientation of the user’s head or body. This information is then used to adjust the spatial audio processing, providing a more immersive audio experience that corresponds to the user’s movements within the virtual environment.

This refers to a microphone used to capture specific sounds or instruments in a recording or performance, adding emphasis or detail to the overall audio mix.

In audio engineering and recording, different microphones are often employed to capture various elements of a performance or sound environment. While some microphones, such as those used for vocals or as overheads for drums, may capture the primary elements of the sound, accent mics are used to capture additional detail, specific instruments, or particular sonic characteristics.

For example, in a live music recording, accent mics might be used to capture the sound of individual instruments like a guitar amplifier, a specific percussion instrument, or a solo instrument within a larger ensemble. In a studio recording setting, accent mics might be used to capture unique sounds, effects, or textures that complement the main elements of the mix.

The choice of accent mic and its placement can significantly influence the overall sound and texture of a recording or performance, allowing audio engineers and producers to tailor the mix to achieve the desired artistic or sonic goals.

Acoustics is the branch of physics that deals with the study of sound, including its production, transmission, propagation, and reception. It encompasses a wide range of phenomena related to the behavior of sound waves in various mediums, such as air, water, and solids.

Key aspects of acoustics include:

1. Sound Waves: Acoustics examines the physical properties of sound waves, including their frequency, wavelength, amplitude, and velocity. Sound waves are mechanical vibrations that propagate through a medium, such as air, as variations in pressure.
2. Sound Sources: Acoustics studies the generation of sound by vibrating objects or sources, such as musical instruments, speakers, and vocal cords. It explores how different types of sources produce sound waves with distinct characteristics.
3. Propagation: Acoustics investigates how sound waves travel through different mediums and environments, including the effects of reflection, refraction, diffraction, and absorption. Understanding sound propagation is essential for predicting how sound behaves in architectural spaces, outdoor environments, and underwater.
4. Room Acoustics: Room acoustics focuses on the interaction between sound waves and enclosed spaces, such as concert halls, recording studios, and classrooms. It examines factors like reverberation, resonance, and diffusion, which affect the quality and clarity of sound within a room.
5. Noise Control: Acoustics addresses the mitigation and control of unwanted noise, including noise pollution from sources such as transportation, industrial machinery, and HVAC systems. It involves techniques such as sound insulation, soundproofing, and noise barriers to reduce the impact of noise on human health and the environment.
6. Psychoacoustics: Psychoacoustics explores the psychological and physiological aspects of sound perception, including how humans perceive pitch, loudness, timbre, and spatial location. It investigates factors like auditory masking, sound localization, and the perception of musical harmony and rhythm.

Acoustics has applications in various fields, including engineering, architecture, music, telecommunications, medicine, and environmental science. By understanding the principles of acoustics, researchers and engineers can design better sound systems, improve building designs, enhance communication technologies, and create more pleasant and comfortable environments for human activities.

An acoustic amplifier or acoustic amp is a type of amplifier designed specifically to amplify the sound of acoustic instruments, such as acoustic guitars, violins, mandolins, and acoustic-electric guitars.

Unlike electric guitar amplifiers, which are optimized for amplifying the signal from electric guitars and typically include features like distortion and overdrive, acoustic amplifiers are tailored to preserve the natural tone and characteristics of acoustic instruments and typically have more of a flat frequency response.

An acoustic echo chamber is a space designed to produce reverberation and echo effects for audio recordings or live performances. It’s typically a room with hard, reflective surfaces like walls, floors, and ceilings, which reflect sound waves, creating a rich and prolonged reverberation effect. Musicians, audio engineers, and producers use acoustic echo chambers to add depth and spaciousness to recordings, particularly for vocals and instruments. These chambers were more prevalent in earlier recording techniques but are still used today, either physically or emulated digitally.

When referring to a speaker system, these systems that have built-in amplification, requiring an external power source to operate. Active speakers have a built-in amplifier, which eliminates the need for a separate amplifier unit.

When referring to guitars, some electric guitars and basses use active pickups, which require a power source (often a battery) to operate. Active pickups offer benefits such as higher output levels, improved signal-to-noise ratio, and tonal versatility compared to passive pickups.

In recording studios, active studio monitors are speakers with built-in amplifiers that are used for accurate audio monitoring during recording, mixing, and mastering processes.

Active circuitry typically refers to electronic components or systems that actively manipulate audio signals to achieve desired effects or functions. These circuits are often found in devices such as audio processors, amplifiers, equalizers, and effects units. Unlike passive components, which do not require an external power source and primarily modify the signal without additional energy, active circuitry relies on external power to operate and can actively boost, filter, or shape the audio signal.

Active device typically refers to any electronic component or equipment that requires an external power source to operate and actively manipulates audio signals. Active devices are contrasted with passive devices, which do not require power and mainly interact with signals through passive means like resistance, capacitance, or inductance.

An active loudspeaker, also known as a powered speaker, is a speaker system that incorporates built-in amplification and active signal processing. Unlike passive speakers, which require an external amplifier to drive them, active loudspeakers have amplifiers built directly into their enclosures.

An actuator typically refers to a device or component that converts electrical signals into physical movement or mechanical action. Actuators play various roles in music production, performance, and instrument design, often contributing to the manipulation of sound or the interaction between musicians and their instruments.

ADAT Lightpipe is a digital audio interface protocol that allows for the transmission of multiple channels of digital audio over optical fiber cables. It was developed by Alesis as part of the ADAT (Alesis Digital Audio Tape) format and has since become a widely used standard in professional audio equipment.

The ADAT Lightpipe protocol uses a single optical fiber cable to transmit up to eight channels of digital audio at a time. It operates at a sample rate of 48 kHz and a resolution of 16 bits per channel, which is standard for most digital audio recording and playback applications.

ADAT Lightpipe interfaces are commonly found on audio interfaces, digital mixers, and other professional audio equipment. They allow users to expand the number of input and output channels on their audio systems by connecting external devices that support the ADAT Lightpipe protocol.

One of the key advantages of ADAT Lightpipe is its simplicity and ease of use. It provides a convenient way to transmit multiple channels of digital audio over a single cable, making it ideal for applications where space and cable management are concerns. Additionally, because it uses optical fiber cables, ADAT Lightpipe connections are immune to electromagnetic interference, ensuring reliable and high-quality audio transmission.

Additive synthesis is a method of sound synthesis that builds complex sounds by combining multiple individual sine waves, known as partials or harmonics. In additive synthesis, each partial is generated at a specific frequency and amplitude, and the combination of these partials creates a rich and diverse spectrum of timbres.

ADSR, which stands for Attack, Decay, Sustain, Release, is a fundamental concept in sound synthesis and audio envelope shaping. It refers to the four distinct stages that characterize the change in volume (or amplitude) of a sound over time.

1. Attack: The Attack stage is the initial phase of a sound where the volume gradually increases from zero to its maximum level after a key is pressed or a sound is triggered. The duration of the Attack stage determines how quickly the sound reaches its peak volume. A shorter attack time creates a more immediate onset, while a longer attack time results in a gradual buildup.
2. Decay: After the Attack stage, the sound enters the Decay stage, during which the volume decreases from its peak level to a predefined sustain level. The decay time parameter controls how quickly the sound decreases in volume. A shorter decay time produces a faster decay, while a longer decay time results in a slower fade.
3. Sustain: The Sustain stage occurs after the Decay stage and represents the period during which the sound maintains a constant volume as long as the key or trigger is held. The sustain level parameter determines the amplitude level at which the sound remains constant during this stage.
4. Release: Finally, when the key is released or the trigger ends, the sound enters the Release stage, during which the volume gradually decreases from the sustain level to zero. The release time parameter controls how quickly the sound fades out after the key is released or the trigger ends. A shorter release time produces a quicker fade, while a longer release time results in a more prolonged decay.

AES commonly refers to the Audio Engineering Society. The Audio Engineering Society (AES) is an international professional organization dedicated to the advancement of audio technology and the science of sound.

AES10, also known as MADI (Multichannel Audio Digital Interface), is a standard for the digital transmission of multiple channels of audio over a single cable. Developed by the Audio Engineering Society (AES) and standardized as AES10, MADI provides a means of transmitting up to 64 channels of uncompressed digital audio between audio devices such as mixing consoles, digital audio workstations (DAWs), routers, and other professional audio equipment.

AES11 is a standard defined by the Audio Engineering Society (AES) that specifies the synchronization of digital audio equipment using embedded audio signals. Specifically, AES11 addresses the synchronization of digital audio clocks through the use of embedded digital audio signals, such as those found in AES3 (also known as AES/EBU) or S/PDIF formats.

AES17 is a standard set by the Audio Engineering Society (AES) that specifies the measurement of digital audio equipment. Specifically, AES17 provides guidelines and recommendations for conducting measurements of the analog-to-digital (ADC) and digital-to-analog (DAC) converters used in digital audio devices.

See “AES/EBU”

AES42 is a standard for digital audio transmission over XLR connectors using the AES/EBU protocol. It was developed by the Audio Engineering Society (AES) and is primarily used for the transmission of digital audio between microphones and digital audio equipment.

AES42 allows for the transmission of both audio data and control signals over a single XLR cable. This enables features such as remote control of microphone parameters, such as gain and polar pattern, as well as powering the microphone through the same cable using Power over Ethernet (PoE) technology.

AES/EBU stands for Audio Engineering Society/European Broadcasting Union. It is a digital audio interface standard developed jointly by the Audio Engineering Society (AES) in the United States and the European Broadcasting Union (EBU). The AES/EBU standard specifies the format for transmitting digital audio signals between professional audio equipment.

AES/EBU typically uses balanced, shielded twisted-pair cables terminated with XLR connectors for transmission. It supports both PCM (Pulse Code Modulation) and non-PCM formats, making it versatile for various audio applications.

After Fade Listen (AFL) is a feature commonly found in audio mixing consoles and digital audio workstations (DAWs). It allows an engineer or operator to monitor a specific audio signal from a channel strip after it has been processed by the channel’s fader.

Aftertouch typically refers to a feature found in electronic musical instruments, particularly keyboards and synthesizers. Aftertouch, also known as pressure sensitivity or pressure response, allows a performer to modulate the sound of a note after it has been played by varying the pressure applied to the keys or other control surfaces.

An algorithm refers to a set of instructions or procedures used to analyze, synthesize, process, or manipulate audio signals. These algorithms can be implemented in various types of digital signal processing (DSP) systems, software applications, or electronic devices to achieve specific audio-related tasks.

Aliasing refers to a phenomenon where higher frequencies in a signal are incorrectly represented as lower frequencies due to undersampling or improper sampling rates during digitization or signal processing.

Ambient miking is a technique used in audio recording where microphones are strategically placed to capture the natural ambient sound of a space, rather than focusing solely on the direct sound source. It aims to capture the reverberation, room tone, and spatial characteristics of the environment, adding depth and realism to the recorded audio.

An ampere, commonly abbreviated as “amp,” is the SI unit of electric current, measuring the rate of flow of electric charge through a conductor. One ampere is defined as the amount of current that flows through a conductor when one volt of electric potential is applied across it, resulting in a one-coulomb charge passing through the conductor per second.

An amplifier, often abbreviated as “amp,” is an electronic device used to increase the amplitude or power of an electrical signal. It takes a weak input signal and outputs a stronger version of that signal, typically to drive speakers, headphones, or other transducers. Amplifiers are fundamental components in audio systems, ranging from small headphone amplifiers to large power amplifiers used in concert sound reinforcement.

Amplitude refers to the intensity or volume of a sound, often measured in decibels (dB) or perceived loudness. It represents the magnitude of the fluctuations in air pressure produced by a vibrating object, such as a musical instrument or vocal cords.

Analog recording involves the process of capturing and storing sound using analog technology, typically on magnetic tape or vinyl records. It relies on the continuous variation of electrical signals that directly correspond to the fluctuations in air pressure generated by musical vibrations, preserving the nuances and warmth of the original sound.

Anharmonic refers to the deviation from the harmonic series, where the frequencies of overtones are not integer multiples of the fundamental frequency. Anharmonic phenomena are often encountered in complex vibrating systems or non-linear oscillators, where the relationship between frequency components is not strictly harmonic. These deviations from harmonicity can lead to the production of dissonant or irregular sounds, contributing to the richness and complexity of acoustic phenomena.

Anharmonic distortion refers to the generation of frequencies that are not integer multiples of the fundamental frequency in a sound signal, typically resulting from non-linear behavior in audio equipment or systems. This distortion can introduce harmonic components that are not naturally present in the original signal, altering its timbre and potentially introducing unwanted artifacts into the sound.

An anti-alias filter is a type of low-pass filter used in digital audio systems to prevent aliasing artifacts during analog-to-digital conversion. It attenuates frequencies above the Nyquist frequency (half the sampling rate) to ensure that only signals within the desired frequency range are accurately sampled. By removing high-frequency components that could fold back into the audible range as aliases, anti-alias filters help maintain the fidelity and integrity of the digitized audio signal.

See “Audio over IP”

An app refers to a software application designed to analyze, manipulate, or simulate sound waves and their properties. These apps often incorporate features such as spectral analysis, sound pressure level measurement, and room acoustic modeling to aid professionals in various tasks like sound engineering, architectural acoustics, or noise pollution assessment. They serve as convenient tools for both researchers and practitioners in the field of acoustics, offering versatile solutions for sound-related challenges.

ATL stands for “Acoustic Transmission Line,” a design concept used in the construction of loudspeaker enclosures. Unlike traditional box enclosures, ATLs utilize a labyrinth-like pathway within the enclosure to control and manipulate sound waves, reducing unwanted resonances and improving bass response. This innovative approach allows for more efficient and accurate reproduction of low-frequency sound, resulting in clearer and more immersive audio experiences.

Attenuate refers to the process of reducing the intensity or amplitude of sound waves as they propagate through a medium or encounter obstacles. This reduction in amplitude can occur due to factors such as absorption, scattering, or reflection, leading to a decrease in sound energy. Attenuation is crucial in various applications, including soundproofing, noise control, and telecommunications, where minimizing unwanted sound transmission or signal loss is essential.

An audio chain is a sequence of interconnected audio devices or components that work together to capture, process, and reproduce sound. It typically includes elements such as microphones, preamplifiers, mixers, signal processors, amplifiers, and speakers, each contributing to the overall audio signal path. The quality and characteristics of each component in the audio chain profoundly influence the fidelity and tonal characteristics of the final sound output.

Audio data reduction is the process of compressing audio signals to reduce their file size while retaining perceptual quality. This compression is achieved through various algorithms and techniques, such as lossy or lossless compression, to remove redundant or less essential information from the audio stream.

Audio frequency refers to the range of frequencies within the audible spectrum, typically perceived by the human ear, spanning from approximately 20 Hz to 20,000 Hz. These frequencies correspond to the pitch or tone of sound waves and are essential for discerning various aspects of auditory perception, including melody, rhythm, and timbre.

Audio Random Access (ARA) is a technology that facilitates seamless integration between digital audio workstations (DAWs) and audio plugin software. It allows plugins to communicate directly with the DAW, enabling features such as real-time audio editing, automatic tempo detection, and instant access to audio regions for processing.

Audio scrubbing is a technique used in digital audio editing to navigate and preview audio recordings by manually moving through the waveform at variable speeds. This process allows users to locate specific sections or fine-tune edits with precision by listening to the audio playback in real-time. Audio scrubbing is commonly used in audio editing software for tasks such as identifying errors, synchronizing sound effects, or aligning musical elements.

Audio Video Bridging (AVB) is a set of standards for transmitting audio and video data over Ethernet networks with guaranteed quality of service (QoS). It enables synchronized, low-latency streaming of multimedia content, making it suitable for applications such as live performances, conferencing systems, and professional audio/video production environments.

An audiophile is an individual who has a passionate and discerning appreciation for high-quality audio reproduction, often pursuing the highest fidelity in sound reproduction systems and recordings.

An auditory area refers to a specific region or section of the brain responsible for processing auditory information, including sound perception and interpretation. These areas, such as the primary auditory cortex and associated auditory regions, play crucial roles in recognizing speech, detecting sound patterns, and distinguishing between different frequencies and timbres.

Auto-Tune is a pitch-correction software used in music production to adjust the intonation of vocal performances. It works by analyzing and modifying the pitch of individual notes, helping singers achieve a more polished and in-tune sound, though its distinct effect can also be used creatively for stylistic purposes.

An autolocator is a device used in audio recording and post-production to mark and navigate specific points in a recording session or project. It allows users to quickly locate and access desired sections of audio material for editing, mixing, or playback, enhancing workflow efficiency and organization.

Automation in audio production refers to the process of controlling various parameters within a digital audio workstation (DAW) or hardware equipment over time without manual intervention. It allows users to program changes in settings such as volume, panning, effects, and plugin parameters to create dynamic and evolving soundscapes, enhancing the expressiveness and precision of audio projects.

An Aux Return, short for auxiliary return, is an input channel on a mixing console or audio interface designed to receive signals from auxiliary sends or effects processors. It allows users to blend processed audio signals back into the main mix, providing control over the level of effects such as reverb, delay, or chorus in the overall sound mix.

Auxiliary equipment in audio refers to additional devices or tools used to complement primary audio systems or processes, often enhancing functionality or providing specific features. This category encompasses a wide range of equipment, including signal processors like compressors and equalizers, effects units such as reverbs and delays, as well as utility devices like DI boxes and headphone amplifiers.

Auxiliary sends, often abbreviated as auxes, are dedicated output channels on a mixing console or audio interface used to route signals from individual channels to external devices or effects processors. They enable users to create separate mix buses for effects such as reverb, delay, or chorus, providing control over the amount of processed signal blended back into the main mix.

See “Audio Video Bridging (AVB)”

A B-type plug usually refers to one of the variations of electrical plugs used for power sockets in different regions. The specific type of B plug can vary depending on the country or region’s standards.

A back electret is a type of condenser microphone design where the electret material, which provides permanent electric polarization, is positioned behind the diaphragm. This configuration allows the microphone to function without external power for polarization, making it a common choice in consumer electronics and communication devices.

Background noise refers to any unwanted sound present in a recording or transmission that is not part of the intended audio signal. It typically includes ambient sounds from the environment or electronic hiss and hum from recording equipment, often requiring noise reduction techniques for cleaner audio reproduction.

Baffles refer to physical structures or materials strategically placed within a space to manage sound reflections and control acoustics. They are designed to absorb or diffuse sound waves, reducing echoes and reverberations that can affect the clarity and quality of audio recordings or playback. By manipulating the direction and intensity of sound propagation, baffles help optimize the acoustic environment for improved listening experiences in settings like recording studios, theaters, or concert halls.

Balanced describes the even distribution of sound energy across different frequencies and spatial dimensions within a listening environment. Achieving balance involves optimizing the acoustic properties of a space, such as absorption, reflection, and diffusion, to minimize unevenness or resonance that may distort the perceived sound

Balanced cables consist of three conductors: two carrying identical signals but with opposite polarities, and a third serving as a ground. This configuration effectively cancels out electromagnetic interference, resulting in cleaner signal transmission and reduced noise, particularly over longer cable runs.

Balanced wiring involves using a cable with three conductors: two carrying the audio signal with equal amplitude but opposite phase, and a third serving as a ground. This configuration minimizes noise and interference, allowing for cleaner signal transmission over longer distances, especially in professional audio setups.

Bandwidth refers to the range of frequencies within a signal or a system that can be effectively transmitted or processed. It is typically measured as the difference between the highest and lowest frequencies of a signal or the frequency range over which a device or system can operate efficiently.

A bank typically refers to a group or set of related parameters or settings within a device or software interface, such as a digital audio workstation (DAW) or synthesizer. Banks allow users to organize and manage multiple presets, effects, or sound parameters efficiently, facilitating quick access and manipulation during the production or performance process.

Bass refers to the low-frequency range of sound, typically found in music or audio productions, characterized by deep tones that provide richness and depth to the overall sound profile. It’s often produced by instruments such as bass guitars, double basses, synthesizers, or electronically manipulated sounds to anchor and enhance the rhythm and harmony of a piece.

Bass response refers to the ability of a speaker, audio system, or musical instrument to accurately reproduce low-frequency sounds within a given range. It’s measured in hertz (Hz) and indicates how well the system can reproduce bass frequencies, impacting the depth and richness of the overall audio experience.

Beaming refers to a phenomenon in loudspeaker design where sound waves become directional at higher frequencies, causing dispersion patterns to narrow. This narrowing effect can result in reduced sound quality and coverage, particularly in larger venues or spaces where uniform sound distribution is essential.

A beat is a fundamental unit of rhythm in music, representing a regular pulse or tempo that provides the underlying framework for a piece. It serves as a reference point for musicians to coordinate timing and execution, crucial for maintaining cohesion and groove within a musical composition.

Beat mapping is a process used in audio editing and music production to align audio tracks with a consistent tempo or beat grid. By analyzing the rhythmic structure of the audio, beat mapping allows for precise synchronization of elements such as drum loops, vocal recordings, or MIDI sequences to ensure they align seamlessly with the established tempo of the project.

Beatmatching is a DJ technique used to synchronize the tempos of two tracks so that they play in harmony, ensuring seamless transitions between them. It involves adjusting the speed of one track to match the tempo of the other, typically done by ear or with the aid of specialized equipment like pitch sliders or software.

A beta version is a preliminary release of a software product, made available to a limited audience for testing purposes before the final version is officially launched. It’s often used to gather feedback and identify bugs or issues that need to be addressed before the full release.

Bi-timbral refers to a synthesizer or sound module’s capability to produce two distinct timbres or sounds simultaneously, often allowing the user to play different sounds on separate parts of a keyboard or MIDI controller. This feature is commonly used in music production to layer or split sounds, providing versatility and depth to compositions.

Blending refers to the process of seamlessly combining multiple sound sources or tracks to create a cohesive and balanced mix. It involves adjusting levels, EQ, dynamics, and spatial properties to ensure that each element contributes harmoniously to the overall sound.

Blumlein refers to a stereo microphone technique developed by Alan Dower Blumlein in the mid-20th century, utilizing a coincident pair of bidirectional (figure-of-eight) microphones to capture a natural stereo image with excellent channel separation. This technique, often used in recording studios and live sound applications, captures both the intensity and phase information of the sound source, resulting in a rich and immersive audio experience.

Breathing refers to the subtle audible inhalations and exhalations that occur during vocal recordings, particularly noticeable in quiet or sensitive passages. Engineers often manage or minimize breathing sounds through careful microphone placement, editing techniques, or using tools like noise gates to maintain the clarity and professionalism of vocal tracks.

Bucking refers to the process of reducing or counteracting an undesired effect or signal. In electrical engineering, it can specifically describe the method of decreasing voltage or current in a circuit to mitigate noise or interference.

Bulk dump refers to the process of transferring large amounts of data or settings between digital devices, typically via MIDI (Musical Instrument Digital Interface). It allows for the mass transfer of parameters such as instrument patches, sequences, or configurations, facilitating efficient setup and synchronization in digital audio and musical equipment.

A cable assembly refers to a complete unit composed of one or more cables, connectors, and possibly additional components like shielding or strain relief. It is designed to facilitate the transmission of audio signals between different audio devices or components, ensuring reliable connectivity and signal quality.

A cable sheath refers to the outer layer of material that encloses and protects the internal conductors and insulation of a cable. It provides mechanical strength, durability, and insulation to the cable, shielding it from environmental factors and preventing electrical interference.

A capacitor microphone, also known as a condenser microphone, utilizes a vibrating diaphragm and a fixed backplate to convert sound waves into electrical signals. It requires phantom power or an external power source to operate and is prized for its detailed audio capture and wide frequency response, commonly used in studio recording and broadcasting applications.

A cardioid refers to a directional microphone pickup pattern that is heart-shaped, with its sensitivity highest at the front and progressively decreasing towards the sides and rear. This pattern is commonly used to capture sound primarily from the front while minimizing pickup from the sides and rear, making it ideal for isolating a sound source in noisy environments or recording setups.

A cartridge refers to the transducer component of a phonograph (turntable) that converts the mechanical motion of a stylus tracing a record groove into an electrical signal. It contains magnets, coils, or other elements that generate this signal, which is then amplified and reproduced as sound.

Chamber reverb is a type of artificial reverberation effect created by simulating the acoustic characteristics of a reverberant chamber or room. It adds depth and spaciousness to audio recordings by blending reflections of sound waves, mimicking the natural reverberation that occurs in physical spaces.

A channel path refers to the complete signal flow or route that audio or video signals take from input to output through a specific channel of a device or system. This path includes all processing stages, such as pre-amplification , equalization, effects processing, and final output stages, ensuring that the signal is correctly handled and shaped according to the desired specifications.

A channel strip refers to a single module or section within a mixing console or audio interface that combines essential processing elements for one audio channel.

Characteristic impedance refers to the impedance that a transmission line or circuit presents to a signal at its input terminals, under ideal conditions. It is a key parameter in ensuring efficient signal transfer and minimizing reflections in high-frequency applications such as telecommunications and digital signal transmission.

A chip generally refers to an integrated circuit or semiconductor device specifically designed for processing audio signals. These chips can be found in various audio equipment such as digital-to-analog converters (DACs), amplifiers, and digital signal processors (DSPs), playing a critical role in signal conversion, amplification, and manipulation within audio systems.

A chord refers to the simultaneous playing or synthesis of multiple musical notes or tones. This can be achieved using synthesizers, samplers, or other electronic instruments to create harmonic textures and musical layers in audio compositions.

Class-A, Class-AB, Class-D, and Class-G are different amplifier designs that dictate how the amplifier handles the input signal and powers the output. Class-A amplifiers operate with the output devices conducting throughout the entire input cycle, while Class-AB amplifiers use two sets of output devices to handle the positive and negative halves of the input signal more efficiently. Class-D amplifiers switch rapidly between on and off states to deliver power efficiently, and Class-G amplifiers use multiple power supply rails to improve efficiency at different output levels. Each class offers different trade-offs in terms of efficiency, fidelity, and heat dissipation.

Clipping occurs when a signal exceeds the maximum amplitude that a system or device can accurately reproduce. This results in distortion, where the waveform is cut off or “clipped” at the maximum level, producing a harsh, unpleasant sound.

Clocking refers to the synchronization of digital audio devices to a shared timing reference, ensuring all devices sample audio signals at precisely the same rate. This synchronization minimizes timing errors and ensures accurate playback or recording of audio across multiple devices in a digital audio system.

Coincident refers to a microphone technique where two microphones are positioned closely together with their capsules aligned at the same point in space. This method captures sound sources with accurate phase coherence and minimal phase issues, often used for stereo recordings where precise imaging and localization of sound sources are desired.

Comb filtering occurs when a sound wave is combined with a delayed version of itself, creating peaks and nulls in the frequency response. These peaks and nulls resemble the teeth of a comb when viewed on a frequency spectrum, hence the name “comb filtering.”

A compact cassette, commonly known as a cassette tape, is a magnetic tape audio recording format that was widely used from the 1960s through the 1990s. It consists of a plastic case containing magnetic tape wound between two reels, allowing for portable and convenient playback and recording of audio content on cassette players and recorders.

A compression driver is a type of transducer used in loudspeakers and horns to convert electrical signals into sound waves. It achieves this by using a diaphragm that moves in response to varying electrical signals, compressing and decompressing air to produce sound.

A compressor is an audio processing device that reduces the dynamic range of an audio signal by attenuating the louder parts while leaving the quieter parts relatively unchanged. It helps to control the volume variations in audio recordings or live performances, enhancing clarity, improving perceived loudness, and ensuring that the signal remains within a desired range.

A computer is an electronic device capable of processing data according to programmed instructions, performing tasks such as calculations, data storage, and communication. It uses hardware components like processors and memory, along with software programs, to execute various functions efficiently.

A condenser microphone, also known as a capacitor microphone, operates on the principle of a capacitor where sound waves cause a diaphragm to vibrate, varying the distance between the diaphragm and a backplate.

A cone refers to the part of a loudspeaker driver that moves back and forth to generate sound waves. It is typically made of lightweight material such as paper, plastic, or composite materials, designed to efficiently convert electrical signals into audible sound by displacing air.

A console, also known as a mixing console or mixer, is a device used to combine and process multiple audio signals. It includes input channels for connecting microphones, instruments, and other audio sources, as well as controls for adjusting levels, applying effects, and routing signals to various outputs.

Contact cleaner is a chemical solution designed to remove dirt, dust, oxidation, and other contaminants from electrical contacts and connectors. It helps restore conductivity and improve electrical connections in audio equipment, ensuring reliable signal transmission and reducing noise or intermittent issues.

Control voltage (CV) refers to a steady electrical signal used to control parameters such as pitch, modulation, and amplitude in electronic music synthesizers and other audio equipment. It enables precise manipulation of sound parameters by applying varying voltages to corresponding control inputs, influencing the characteristics of generated audio signals.

Convolution in audio processing refers to a mathematical operation that combines two signals to produce a third signal. It is commonly used in digital audio effects to simulate acoustic spaces, reverberation, and other complex signal transformations by applying the impulse response of a system or space to an input signal.

Convolution reverb is an audio processing technique that uses convolution to simulate the reverberation of a physical space or acoustic environment. It achieves this by convolving an input audio signal with the impulse response of the desired space, accurately replicating the spatial characteristics and reverberation decay of real-world locations.

Copy protection refers to measures implemented in digital media and software to prevent unauthorized duplication or distribution. These measures can include encryption, digital rights management (DRM), and other technological barriers designed to limit or control access to copyrighted content.

Corner frequency, refers to the frequency at which a filter’s response begins to attenuate or change significantly. It marks the boundary where the filter’s effect on the signal becomes noticeable, whether in terms of frequency cutoff in low-pass or high-pass filters, or resonance in band-pass and band-stop filters.

CPU stands for Central Processing Unit, which is the primary component of a computer responsible for executing instructions and performing calculations. It acts as the brain of the computer, handling tasks such as running programs, processing data, and managing input and output operations.

A crossover refers to an electronic circuit or device that splits an audio signal into two or more frequency bands, directing each band to a specific speaker or driver optimized for that frequency range. This ensures that different parts of the audio spectrum (such as bass, midrange, and treble frequencies) are reproduced accurately and efficiently by the appropriate speakers, enhancing overall sound quality and clarity.

Cut and paste editing involves digitally manipulating recorded audio by selecting and removing segments (cutting) and rearranging them in a new sequence (pasting). This technique allows for precise editing of audio recordings to enhance timing, remove mistakes, or create new compositions seamlessly.

Refer back to “Control Voltage” for definition.

A Digital-to-Analog Converter (DAC) transforms digital signals into continuous analog voltages or currents. This process enables digital systems to interface with analog devices, such as speakers or analog meters.

A daisy chain is a wiring or connection method where devices are connected sequentially in a series, with each device linked to the next. This configuration allows for a simple and scalable setup, but can be prone to issues if one device fails or if the chain becomes too long.

Daisy-chain mains distribution is a wiring method where electrical power is distributed by connecting devices or outlets in a sequential series, with each device receiving power from the previous one. This approach simplifies wiring but can lead to power distribution issues or increased risk if one connection fails.

Damping refers to the process of reducing oscillations or vibrations in a system by dissipating energy, often through friction or other resistance mechanisms. It helps control and stabilize motion, improving performance and longevity in various applications such as mechanical systems, audio equipment, and structural engineering.

DARS (Digital Audio Radio Service) is a satellite-based radio service that broadcasts digital audio content directly to receivers, offering a wide range of channels and high sound quality. It provides nationwide coverage and consistent reception, regardless of geographic location or terrestrial interference.

DAT (Digital Audio Tape) is a digital magnetic tape format used for recording and playing back high-fidelity audio with a resolution superior to analog tape. It offers precise, reliable audio storage and playback, making it popular for professional audio recording and archival.

Data refers to measurements and recordings of sound characteristics, such as frequency, amplitude, and duration, collected to analyze and interpret acoustic phenomena. This information is essential for designing audio systems, assessing sound quality, and optimizing acoustic environments.

A Digital Audio Workstation (DAW) is a software application used for recording, editing, and producing audio and music, integrating a range of digital tools and effects. It provides a comprehensive environment for managing audio tracks, mixing, and mastering within a single platform.

A decibel (dB) is a logarithmic unit used to measure the intensity or level of sound, representing the ratio of a particular sound level to a reference level. It quantifies the relative strength of sound, with each 10 dB increase corresponding to a tenfold increase in intensity.

A decibel per octave (dB/octave) measures the rate at which the level of a signal decreases or increases with each octave in frequency. It is commonly used to describe the slope of filters or the frequency response of audio systems, where each octave represents a doubling or halving of frequency.

DBX refers to a brand known for its audio processing equipment, including dynamic range compressors, noise reduction systems, and other signal processors. Their products are widely used in professional audio to enhance sound quality and control various aspects of audio signals.

Direct Current (DC) is an electrical current that flows consistently in one direction, providing a steady and unidirectional flow of electric charge. It is commonly used in low-voltage applications, such as batteries and electronic devices, where stable voltage is required.

DC coupling refers to a method of connecting electronic circuits that allows both direct current (DC) and alternating current (AC) signals to pass through without blocking any component of the signal. This technique ensures that the entire signal, including its DC offset, is transmitted between stages of a system.

DC bias refers to the application of a constant direct current (DC) voltage to a circuit or component to set its operating point or adjust its performance. It ensures proper functioning of electronic devices by stabilizing their signal conditions and optimizing their response.

DC offset is an unwanted constant direct current (DC) component added to an otherwise alternating current (AC) signal, shifting its baseline from zero. This shift can affect signal accuracy and performance, often requiring correction to ensure proper signal processing and measurement.

DCC (Digital Command Control) is a system used in model railroading to control multiple locomotives and accessories independently on the same track using digital signals. It enables precise control over speed, direction, and functions of model trains, enhancing the realism and flexibility of model railroad operations.

A Digital Controlled Oscillator (DCO) is an electronic oscillator that generates a stable waveform with its frequency controlled by digital input signals. It is commonly used in synthesizers and communication systems for precise frequency modulation and signal generation.

Direct Current (DC) is a type of electrical current that flows consistently in one direction, providing a steady and unidirectional flow of electric charge. It is commonly used in batteries, electronic devices, and various low-voltage applications where stable voltage is essential.

DDL (Digital Data Link) refers to a communication protocol used for transmitting digital data between devices or systems over a network. It ensures reliable and efficient data exchange, often employed in various applications including telecommunications and computer networks.

DDP (Distributed Data Processing) refers to a system architecture where data processing tasks are distributed across multiple computers or locations rather than being centralized. This approach enhances efficiency and scalability by leveraging the processing power and resources of multiple interconnected systems.

De-emphasis is a process used in audio and telecommunications to reduce the amplitude of high-frequency components of a signal to counteract the effects of pre-emphasis applied during transmission. It helps to restore the original frequency response and improve signal clarity during playback or reception.

A de-oxidizing compound is a chemical substance used to remove or reduce oxidation, often applied to restore and maintain the conductivity of metal surfaces. It is commonly used in electrical and electronic applications to clean connectors and prevent performance issues caused by corrosion.

Decay refers to the gradual reduction in amplitude or intensity of a signal, sound, or physical process over time. In acoustics, it describes how the volume of a sound diminishes after the initial attack, while in other contexts, it can refer to the breakdown of materials or radioactive substances.

Defragmenting is the process of reorganizing fragmented data on a storage device to improve access speed and efficiency. By consolidating scattered files into contiguous blocks, it reduces the time required for data retrieval and enhances overall system performance.

Delay refers to the intentional or unintentional time gap between the input and output of a signal or system. In audio processing, it involves adding a time interval to a signal to create effects like echo or reverberation, enhancing the depth and spatial quality of the sound.

A detent refers to a mechanical feature in audio equipment that provides distinct, tactile feedback when adjusting controls, such as volume or tone settings. This helps users make precise adjustments by ensuring each setting is clearly defined and reduces the risk of accidental changes.

Direct-to-Indirect Ratio (DI) measures the balance between direct sound from a source and the reflected sound that arrives after bouncing off surfaces. A high DI indicates a clearer sound with more direct, focused audio, while a low DI suggests more influence from reflections, which can affect clarity and spatial perception.

A DI (Direct Injection) box is a device used to connect an instrument or audio source directly to a mixing console or recording system, converting high-impedance signals to low-impedance ones. It helps to preserve signal quality and reduce noise or interference during the signal transfer.

A diaphragm is a flexible membrane in a microphone or speaker that vibrates in response to sound waves, converting acoustic energy into electrical signals or vice versa. Its design and material properties significantly impact the sensitivity, frequency response, and overall sound quality of the device.

A diffusor is an acoustic device designed to scatter sound waves in multiple directions, improving the evenness of sound distribution and reducing echoes in a space. By diffusing sound, it helps to create a more balanced and natural acoustic environment, enhancing the clarity and richness of the audio experience.

Digital delay is an audio effect that records an incoming sound signal and plays it back after a set period, creating an echo-like repetition. It allows for precise control over the timing and duration of the delay, enabling effects like rhythmic echo and spatial enhancement in audio production.

A digital reverberator is an audio processor that simulates the reverberation effects of a physical space by generating complex, algorithmically created reflections and echoes. It allows for precise control over the characteristics of the reverberation, such as decay time and early reflections, to enhance the spatial quality and depth of the sound.

A DIN connector is a type of electrical connector characterized by its circular design and multiple pins arranged in a specific pattern, used for various audio, video, and data connections. Named after the German standardization organization, DIN connectors are known for their reliability and ease of use in connecting equipment like keyboards, audio devices, and older computer peripherals.

A diode-bridge compressor is an audio compressor that uses a bridge circuit composed of diodes to control the gain reduction of an audio signal, offering smooth and transparent compression. This type of compressor is known for its ability to manage dynamic range while preserving the tonal qualities of the audio.

Direct coupling refers to the connection of electronic circuits or components without the use of intermediate components like transformers or capacitors, allowing for a direct electrical path. This method improves signal fidelity and reduces distortion by maintaining a consistent impedance and minimizing signal loss.

A disk is a flat, circular storage medium used to hold and access data, such as a hard disk drive (HDD) or a diskette. It can also refer to optical media like CDs and DVDs, where data is encoded on a reflective surface for reading by a laser.

Distortion refers to any alteration or modification of an audio signal that deviates from its original waveform, often resulting in unwanted changes to sound quality. In audio equipment, distortion can occur due to overdriving components or imperfections in the signal path, affecting clarity and fidelity.

Dither is a technique used in digital audio processing to reduce quantization error and minimize audible artifacts when converting audio from a higher bit depth to a lower one. By adding a small, controlled amount of random noise to the signal, dither helps to mask quantization distortions and preserve audio quality.

Direct Memory Access (DMA) is a feature that allows hardware components to directly transfer data to or from memory, bypassing the central processing unit (CPU) to improve efficiency and speed. This method reduces the CPU’s workload and can enhance overall system performance by enabling simultaneous data handling and processing tasks.

Dolby Atmos is an advanced audio technology that creates a three-dimensional sound experience by allowing sound to move freely around the listener in a 360-degree space. It enhances immersion in movies, games, and other media by placing audio objects in a three-dimensional soundscape, rather than being confined to traditional channel-based audio formats.

Dolby HX (Headroom Extension) is a noise reduction technology designed to improve the recording quality of analog cassette tapes by extending the dynamic range and reducing distortion. It achieves this by optimizing the headroom in the tape recording process, allowing for higher fidelity and more detailed sound reproduction.

Dolby Noise Reduction is a technology designed to reduce the audible hiss and background noise in analog audio recordings by encoding and decoding noise-reduction signals. It works by compressing high-frequency noise during recording and expanding it during playback, resulting in cleaner, clearer sound.

Dolby Surround-Sound is an audio technology that creates a multi-channel audio experience by encoding four audio channels into two tracks, allowing for a more immersive sound experience in home and cinema environments. It enhances the spatial depth and realism of audio by delivering sound from multiple directions, simulating a surround sound effect even with limited speaker setups.

A dome is a curved, hemispherical structure that can influence sound distribution and acoustics within a space by reflecting and diffusing sound waves. This architectural element can enhance the auditory experience by improving sound clarity and reducing echoes or reverberation.

Double-ended noise reduction is an audio processing technique that reduces noise by applying noise reduction algorithms to both the recording and playback processes. This approach helps to minimize the impact of noise across the entire audio chain, enhancing clarity and fidelity in both recorded and played-back sound.

A drive unit is a component of a speaker or audio system that converts electrical signals into sound waves. It typically includes elements such as a magnet, voice coil, and diaphragm to produce sound.

A drum booth is a soundproof or acoustically treated enclosure designed to contain and isolate the sound of drums during recording or practice sessions. It helps to reduce noise bleed into other recording areas and control the drum’s acoustic environment for clearer and more precise audio capture.

Ducking is an audio processing technique where the volume of one audio signal is automatically reduced when another signal is detected, often used to ensure that speech remains intelligible over background music or noise. This effect is commonly employed in broadcasting and live sound to manage audio levels and improve clarity.

A dump refers to the process of exporting or transferring audio data from one device or system to another, often to create backups or to facilitate editing. This term can also apply to the act of saving or moving large amounts of data quickly and efficiently.

Duophonic is a type of audio playback that simulates stereo sound by creating a two-channel output from a single audio source, often through the use of artificial separation techniques. This method enhances the audio experience by providing a sense of spatial dimension, even when the original recording was not recorded in true stereo.

Digital Vinyl System (DVS) is a technology that allows DJs to control digital audio files using traditional vinyl turntables by converting the analog signal from the turntable into digital commands. This system combines the tactile feel of vinyl with the flexibility and convenience of digital audio, enabling precise manipulation of digital tracks.

A dynamic microphone is a type of microphone that uses an electromagnetic induction principle to convert sound into an electrical signal, making it durable and capable of handling high sound pressure levels. It is well-suited for live sound applications and instruments due to its robustness and ability to capture sound without distortion.

Dynamic range refers to the difference between the quietest and loudest parts of an audio signal or recording. It measures the range of volume levels that can be accurately captured or reproduced, influencing the clarity and depth of sound.