Absorption
Absorption is the process by which a material takes in sound energy rather than reflecting it. Porous materials like foam or thick fabric are often used to absorb mid and high frequencies, reducing echoes and reverberation in a room.
Accuracy
Accuracy is a measure of how faithfully an audio system reproduces the original recording. A system with high accuracy has minimal distortion, a flat frequency response, and precise timing, ensuring the listener hears the audio as the artist and engineer intended.
Acoustics
Acoustics is the science of how sound is produced, transmitted, and received. In audio, it most often refers to the characteristics of a room or space that determine how sound behaves within it, including its reflections, reverberation, and resonances.
Acoustic Treatment
Acoustic treatment is the strategic placement of absorbers, diffusers, and bass traps to control how sound reflects and decays in a room. The goal is to minimize unwanted acoustic distortions like echoes and standing waves to achieve a more balanced and clear listening experience.
Adaptive Filter
An adaptive filter is a digital signal processing (DSP) algorithm that continuously updates its own properties based on feedback from an error signal. This self-correcting mechanism allows it to perform real-time suppression of changing acoustic problems like echoes, feedback, or resonances that evolve as conditions in a room change.
Aliasing
Aliasing is a type of distortion that occurs when a signal is sampled at a rate less than twice its highest frequency component. This process creates unwanted, false tones that were not in the original signal, which can be prevented by using a proper anti-aliasing filter before the digital conversion.
Ambisonics
Ambisonics is a full-sphere surround-sound format that captures or creates sound in terms of its directional components. This format is highly flexible and can be decoded to play back on any arrangement of speakers or through headphones for an immersive 3D audio experience.
Audio
Audio is the term for sound when it is converted into an electrical or digital format for recording, transmission, or playback.
Bass
Bass is the term for the lowest part of the audible frequency spectrum, typically ranging from about 20 Hz to 250 Hz. These frequencies provide the foundational rhythm and weight of music.
Bass Management
Bass management is the process of handling low-frequency information in a multi-speaker system, especially one with a subwoofer. It involves routing, timing, and level adjustments to ensure a clear, powerful bass response by seamlessly integrating the main speakers and the subwoofer.
Bass Range
The bass range is the spectrum of low frequencies in audio, often subdivided into sub-bass (below 60 Hz) and upper bass. It is crucial for the sense of power and foundation in music.
Bass Traps
Bass traps are acoustic absorption devices specifically designed to capture low-frequency sound energy. They are used to control the resonances (room modes) that cause uneven and "boomy" bass in a room.
Bit Depth
Bit depth is the number of bits used to represent the amplitude of each audio sample. It directly determines the maximum possible dynamic range and the level of the quantization noise floor in a digital recording.
Bloated
Bloated is a descriptive term for a sound that has an excessive and poorly defined mid-bass response, leading to a sense of thickness and a lack of clarity or precision.
Blurry
Blurry is a descriptive term for a sound that lacks focus and transient detail. It is often caused by poor timing alignment between drivers or significant smearing from room reflections.
Boomy
Boomy is a descriptive term for an excessive and resonant bass response, typically concentrated in a narrow frequency range. This is often caused by room modes and results in a "one-note" bass that sounds uncontrolled and overwhelming.
Brick-Wall Filter
A brick-wall filter is a digital low-pass filter with an extremely steep cutoff slope. It is used to prevent aliasing by sharply cutting off frequencies above the Nyquist frequency—the highest frequency that can be accurately captured at a given sample rate. This sharp cutoff comes at the cost of increased latency and potential ringing artifacts.
Bypass
A bypass is a switch that routes the audio signal around a processor. It allows for an immediate comparison between the processed and the original signal without affecting level or timing, which is critical for evaluating the effect of the processor.
Calibration Microphone
A calibration microphone is a measurement microphone with a very flat, accurate frequency response. It is used to capture a room's impulse responses and sound pressure level (SPL) to ensure that correction algorithms are based on precise acoustic data. For the most accurate room measurements, an omnidirectional microphone, which captures sound equally from all directions, is recommended.
Calibration Profile
A calibration profile is a saved set of measurement data and the corresponding correction filter settings for a specific listening environment. It allows a user to quickly recall the optimal settings for a particular room layout, speaker configuration, or listening preference.
Colouration/Coloration
Colouration is any audible alteration to the sound signal caused by a component or the room itself. A "coloured" sound is not neutral and has a distinct character imposed upon it, changing its tonal balance.
Comb Filtering
Comb filtering is an interference effect that creates a series of sharp peaks and dips in the frequency response. It is caused when a direct sound combines with a delayed version of itself, which commonly happens when sound reflects off a nearby surface like a wall or desk.
Convolution
Convolution is a mathematical operation that combines two signals to produce a third. In audio, it is used to apply the acoustic signature of an impulse response to an audio stream, making it possible to recreate the sound of a specific space or apply a corrective filter with high precision.
Correction Filter
A correction filter is a custom equalization and, sometimes, phase adjustment curve derived from acoustic measurement data. It is applied to the audio signal in real-time to counteract room coloration and speaker inaccuracies, resulting in improved tonal balance and clarity.
Crossover
A crossover is an electronic or passive network that divides the audio signal into different frequency bands. It directs the low, mid, and high frequencies to the appropriate drivers (woofers, midranges, tweeters) or amplifiers for a smoother and more efficient response.
Crossover Optimization
Crossover optimization is the process of refining the point where frequencies are divided between speakers and subwoofers. It involves fine-tuning the slope, phase, and level to create a seamless and coherent transition, ensuring a uniform bass response.
Cumulative Spectral Decay (CSD)
A cumulative spectral decay plot, also known as a "waterfall" graph, shows how sound energy at each frequency decays over time. This visualization is excellent for revealing lingering resonances from a room or speaker cabinet that are often missed by standard frequency response graphs.
DAC (Digital-to-Analog Converter)
A DAC is a device that converts a digital audio signal (a stream of numbers) into an analogue electrical signal that can be sent to an amplifier and speakers.
dB (Decibel)
A decibel is a logarithmic unit used to measure sound intensity or the level of an electrical signal. Because human hearing perceives sound pressure logarithmically, the decibel scale is a convenient way to represent loudness. An increase of 6 dB is roughly a doubling of perceived loudness.
Decay
Decay is the way a sound fades away after its initial peak. In a room, it refers to the persistence of sound due to reverberation. The character of the decay can affect the perceived clarity and spaciousness of the audio.
Decay Time (RT60)
Decay time (RT60) is the measurement of the time it takes for a sound to decrease by 60 dB in a given space. Long decay times, especially in the bass frequencies, can cause a muddy and unclear sound.
Delay Compensation
Delay compensation is an automatic or manual process that aligns the timing of multiple audio channels by offsetting any latency introduced by signal processing. This ensures all channels remain time-coherent, which is critical for preserving correct phase relationships and stereo imaging.
Diffraction
Diffraction is the bending or spreading of sound waves as they pass around an object or through an opening. This is why you can hear sound from around a corner. In speaker design, diffraction from cabinet edges can cause distortion.
Diffusion
Diffusion is the process of evenly scattering sound reflections in various directions and over time. This helps to avoid harsh, distinct echoes while preserving a sense of spaciousness and air, leading to improved clarity.
Room Correction
Digital room correction is a technology that uses digital signal processing (DSP) to analyse the acoustic behaviour of a room and apply corrective equalization and timing adjustments to counteract its negative effects, such as peaks, nulls, and reflections.
DSP (Digital Signal Processor)
A DSP is a specialized microprocessor designed to efficiently perform mathematical operations on digital signals. In audio, DSPs are used for a wide range of tasks, including equalization, filtering, compression, and room correction.
Dither
Dither is a very low-level noise that is added to an audio signal when reducing its bit depth. This process randomizes the quantization errors that would otherwise cause audible distortion, trading a small amount of hiss for the preservation of fine, low-level details in the sound.
Downsampling
Downsampling is the process of reducing the sample rate of a digital audio signal. To do this correctly, the signal must first be passed through an anti-aliasing filter to prevent distortion. This is often done to reduce file size or match the capabilities of playback hardware.
Dynamic Range
Dynamic range is the difference between the quietest and loudest possible sounds a system can reproduce without significant noise or distortion. It is influenced by the system's bit depth, headroom, and the ambient noise of the listening environment.
Envelope
The envelope of a sound describes how its amplitude (loudness) changes over time. It is typically defined by four stages: Attack (the initial rise), Decay (the drop from the peak), Sustain (the level while the note is held), and Release (the final fade out).
FFT (Fast Fourier Transform)
An FFT is a highly efficient algorithm used to break down a time-domain signal (like a waveform) into its constituent frequency components. It is the core technology behind real-time spectrum analysers, convolution engines, and many correction algorithms.
Filter Bank
A filter bank is a set of digital filters that work together to split an audio signal into multiple frequency bands. This is an essential component for implementing graphic equalizers, crossovers, and multiband room correction systems.
FIR Filter (Finite Impulse Response)
An FIR filter is a type of digital filter where the impulse response is finite (it settles to zero). A key advantage is that its coefficients can fully define both magnitude and a linear-phase response, preserving timing integrity. However, achieving fine low-frequency control can require many "taps" (coefficients), which can introduce latency.
Fletcher-Munson Curve
The Fletcher-Munson curves are part of a broader set of equal-loudness contours that illustrate how human hearing sensitivity varies with frequency and loudness. Our ears are most sensitive to mid-range frequencies, which is why audio engineers are often advised to mix at moderate levels to ensure the final balance translates well to all playback volumes.
Frequency Response
Frequency response is a measurement of a system's output level across the range of audible frequencies. Ideally, this would be a "flat" line, indicating that all frequencies are reproduced at the same level. Deviations from flat reveal coloration or limitations in the system.
Gain Staging
Gain staging is the process of managing the signal level at each step in an audio chain. Proper gain staging maximizes the dynamic range by keeping the signal well above the noise floor while ensuring it never gets loud enough to cause clipping (distortion).
Golden Ear
A "golden ear" is a person perceived to have exceptional listening skills, able to discern very subtle differences in audio quality, such as slight coloration, distortion, or spatial cues.
Group Delay
Group delay is a measure of the time lag that occurs at different frequencies as a signal passes through a system (like a filter or loudspeaker). Excessive or non-uniform group delay can smear transients, shift localization cues, and reduce overall clarity and punch.
Harmonic Distortion
Harmonic distortion consists of added frequency components at integer multiples of the original signal's frequencies. This is caused by nonlinearities in an audio system. In small, controlled amounts it can be perceived as warmth, but in excess, it becomes harsh and unpleasant.
Head-Related Transfer Function (HRTF)
An HRTF describes how the physical properties of a listener's head, outer ears (pinnae), and torso alter an incoming sound before it reaches the eardrums. These directional filtering effects are what the brain uses to localize sound and are essential for creating realistic binaural audio for headphones.
Headroom
Headroom is the buffer, measured in decibels, between the normal operating level of a signal and the maximum level the system can handle before clipping occurs. Sufficient headroom gives a mix space to accommodate unexpected peaks and allows for further processing during mastering.
High-Pass Filter
A high-pass filter is a filter that allows high frequencies to pass through while attenuating (cutting) frequencies below a selected cutoff point. This is commonly used to protect speakers from excessive low-frequency energy or to remove unwanted rumble from a recording.
IIR Filter (Infinite Impulse Response)
An IIR filter is a type of digital filter that uses feedback, meaning its impulse response can theoretically continue indefinitely. This design allows it to achieve very steep filter slopes with low latency, but it comes at the cost of introducing non-linear phase shifts, which can alter the timing of the signal.
Imaging
Imaging is the ability of an audio system to create the illusion of distinct, localized sound sources within the soundstage. Good imaging allows the listener to pinpoint the apparent location of each instrument or voice.
Impact/Slam
Impact or "slam" is the visceral, physical sensation of a powerful, fast, and well-controlled low-frequency transient, like the kick of a drum. It requires both powerful amplification and excellent transient response from the speakers and room.
Impedance
Impedance is the total opposition to the flow of alternating current in an electrical circuit, measured in ohms. Properly matching the output impedance of an amplifier to the input impedance of a speaker is crucial to avoid changes in tone or a loss of power.
Impulse Response
An impulse response is a measurement that captures how a room or an audio device responds to a very brief, sharp sound (an impulse). It contains a wealth of information about the system's frequency response, reflections, and decay characteristics, allowing for detailed acoustic modelling.
Impulse Shortening
Impulse shortening is a technique used in digital correction to limit the time length of the correction filter. This helps to reduce processing latency and audible artifacts like pre-ringing while still maintainingthe essential low-frequency control needed to correct room modes.
Jitter
Jitter is a small, rapid variation in the timing of the clock edges in a digital audio signal. If not minimized, jitter can cause a blurring of the stereo image and an increase in the noise floor, degrading the overall audio quality.
Kilohertz (kHz)
A kilohertz is a unit of frequency equal to one thousand cycles per second. In digital audio, the sample rate (e.g., 44.1 kHz) determines the highest frequency that can be recorded.
Latency
Latency is the total time delay that a signal experiences as it passes through a system, including converters, buffers, and digital signal processing. High latency can disrupt the synchronization between audio and video or make real-time performance monitoring difficult.
Line-Level
Line-level is a standardized signal strength used for interconnecting professional or consumer audio hardware, such as between a DAC and an amplifier.
Linear-Phase EQ
A linear-phase EQ is a type of equalizer that adjusts the magnitude of frequencies without altering their phase relationships. This preserves the integrity of transients but produces symmetrical pre- and post-ringing artifacts around the corrected signal.
Lossless
Lossless is a data compression format (like FLAC or ALAC) that reduces the size of a digital file without discarding any of the original information. When uncompressed, a lossless file is a bit-perfect replica of the original source.
Loudness
Loudness is the perceived intensity of a sound, which is a subjective quality influenced by frequency content and duration. It is codified in standards like LUFS (Loudness Units Full Scale) to help normalize audio levels for broadcasting and streaming.
Loudspeaker-Room Interaction
Loudspeaker-room interaction describes the combined effect that a speaker's placement and a room's geometry have on the sound that reaches the listener. Correcting the negative aspects of this interaction is the primary objective of room correction systems.
Low-Pass Filter
A low-pass filter is a filter that allows low frequencies to pass through while attenuating frequencies above a selected cutoff point. It is commonly used to send only bass information to a subwoofer or as an anti-aliasing filter.
Mid-Tone Range
The mid-tone range (or midrange) is the frequency spectrum where human hearing is most sensitive, typically from around 250 Hz to 4 kHz. This range contains the core of most melodies and vocals, and its clarity is critical for intelligibility.
MIMO (Multi-Input, Multi-Output)
MIMO is a signal processing approach that treats multiple inputs (microphones) and multiple outputs (speakers) as a single, coordinated system. In room correction, MIMO allows all loudspeakers to work together to control room acoustics, providing a more effective solution for low-frequency resonances than correcting each speaker individually.
Mixed-Phase Technology
Mixed-phase technology is a filter design approach that combines the benefits of linear-phase and minimum-phase filters. It aims to correct both magnitude (frequency response) and phase (timing) issues across a wide listening area, preserving transient integrity while minimizing latency and artifacts.
Modal Resonance
A modal resonance, or room mode, is a standing wave that occurs in a room when a sound's wavelength is directly related to one of the room's dimensions. This causes specific bass frequencies to be either strongly reinforced or cancelled out at distinct locations in the room.
Mono
Mono (monophonic) is a single-channel audio reproduction where the same signal is sent to all speakers. It is a crucial tool for checking a mix for phase issues and ensuring that it will translate well to systems like radios or voice assistants.
Multiband Compression
Multiband compression is a process that applies independent compression to different frequency bands. This allows for targeted dynamic control, such as taming a boomy bass or harsh highs, without causing the "pumping" artifacts that can occur when compressing the entire spectrum at once.
Noise Floor
The noise floor is the sum of all unwanted background noise present when no intentional signal is being played. It includes electronic hiss from components, ambient environmental sound, and quantization noise. A lower noise floor allows for the perception of more delicate details.
Notch Filter
A notch filter is a type of band-stop filter with a very narrow bandwidth. It is designed to surgically remove a single, specific frequency, such as electrical mains hum or a problematic room resonance, with minimal impact on the surrounding frequencies.
Null/Cancellation Zone
A null or cancellation zone is a location in a room where sound waves from various sources arrive out of phase and cancel each other out, leading to a significant drop in volume or "dead spot." DSP solutions can mitigate these by applying targeted, inverse-phase corrections.
Nyquist Frequency
The Nyquist frequency is the highest frequency that can be accurately captured at a given sample rate and is equal to half of that sample rate. Attempting to sample frequencies above the Nyquist limit results in aliasing.
Ohm
The ohm is the standard unit of electrical resistance or impedance, represented by the symbol Ω. It is a fundamental unit in audio for ensuring proper matching between speakers and amplifiers.
Oversampling
Oversampling is a technique where the sample rate of a signal is temporarily increased within a digital signal processor. This is done to make filter design easier, reduce certain types of distortion, or improve headroom before the signal is downsampled back to its original rate.
Overdubbing
Overdubbing is the process of recording a new performance while simultaneously listening to previously recorded tracks. This technique allows solo artists or bands to layer multiple parts on top of each other, building up dense and complex arrangements.
Parametric EQ
A parametric equalizer is an equalizer that provides simultaneous control over three parameters: the centre frequency, the gain (boost or cut), and the bandwidth or "Q" (how wide or narrow the adjustment is). This allows for highly precise spectral corrections.
Peak Limiter
A peak limiter is a very fast-acting dynamics processor designed to prevent the peaks of a signal from exceeding a set threshold. Its primary use is to protect speakers from damage and to avoid digital clipping.
Perceptual Targeting
Perceptual targeting is an approach to audio correction that is based on the principles of human hearing. Rather than aiming for a purely mathematical target, it accounts for psychoacoustic phenomena to achieve a result that sounds more natural to the listener.
Phase
Phase describes the timing relationship between two or more periodic waveforms. When waveforms are "in phase," they reinforce each other. When they are "out of phase," they cancel each other out, which can dramatically alter the tone and stereo image.
Phase Response
Phase response is a graph that shows the phase shift of a system at different frequencies. A desirable phase response is linear, meaning all frequencies are delayed by the same amount. Erratic shifts in the phase response can blur the stereo image and smear transient details.
Pink Noise
Pink noise is a random signal that has equal energy per octave. This means its energy decreases as frequency increases, which corresponds more closely to human hearing than white noise. It is useful for equal-loudness testing and for calibrating audio systems.
Pinna Gain
Pinna gain refers to the acoustic amplification that occurs due to the shape of the pinna (the outer ear). The folds and ridges of the ear selectively boost certain high frequencies, which provides crucial directional cues to the brain.
Preamp
A preamp (preamplifier) is a circuit designed to boost a weak electrical signal, such as one from a microphone or a turntable, up to the standard line level. A good preamp achieves this with minimal added noise or coloration.
Pre-Ringing
Pre-ringing is an artifact that can occur in linear-phase filters, where a faint oscillation appears just before the main transient of a sound. If audible, it can be perceived as a smearing of the attack of sharp sounds.
Psychoacoustics
Psychoacoustics is the scientific study of how the brain and auditory system interpret sound. This field informs the design of perceptual audio codecs (like MP3), spatial audio technologies, and the development of equal-loudness contours.
Q Factor
The Q factor of a filter describes the sharpness of its resonance. In an equalizer, a high Q value means the filter will boost or cut a very narrow band of frequencies, while a low Q value will affect a much broader range.
Reference Curve
A reference curve, also known as a target curve, is a desired frequency response profile that a correction system aims to achieve. This target is not always perfectly flat; it often includes a gentle bass boost and a slight treble roll-off to match listener preferences and create a tonally balanced sound.
Reflection
A reflection is a sound wave that has bounced off a surface, such as a wall, floor, or ceiling. The timing and strength of reflections determine a room's acoustic character.
Resolution
Resolution refers to the ability of an audio system to reveal fine, low-level details in a recording. A high-resolution system can clearly distinguish subtle textures, ambient cues, and the delicate decay of notes.
Reverberation
Reverberation is the dense collection of overlapping reflections that persists in a space after the original sound source has stopped. It conveys the size and surface texture of the room, but excessive reverberation can muddy the sound and reduce clarity.
Room Correction
Room correction is a process that uses digital analysis and filtering to compensate for the acoustic problems caused by a room's resonances and speaker placement. The goal is to even out the frequency and phase response for a more accurate listening experience.
Room Mode
A room mode is a low-frequency resonance that is directly tied to a room's dimensions. These modes create a predictable pattern of peaks and nulls (loud and quiet spots) throughout the room, which correction software is designed to mitigate.
Sample Rate
The sample rate is the number of times per second that a continuous analogue audio signal is measured (sampled) to convert it into a digital signal. Higher sample rates can capture higher frequencies but also result in larger file sizes.
Schroeder Frequency
The Schroeder frequency marks the approximate transition point in a room's acoustics. Below this frequency, the sound is dominated by distinct room modes, while above it, the sound field is more diffuse and statistical (reverberation). This guides correction strategies.
Sibilance
Sibilance is an unpleasant, high-frequency hissing sound that can occur on "s," "sh," and "t" sounds in vocals. It is often caused by microphone placement, over-equalization, or compression.
Smart Acoustics
Smart acoustics is an approach to audio processing that optimizes performance by analysing all three key elements of the listening experience: the source (the audio content), the system (the playback hardware), and the space (the listening environment).
Software Defined Audio
Software defined audio is an approach where core audio processing functions, traditionally handled by dedicated hardware, are instead performed by flexible and updatable software. This allows for greater innovation and customization.
Sound
Sound is a vibration that propagates as an acoustic wave through a medium such as air. It is the raw, physical phenomenon that audio technologies aim to capture and reproduce.
Sound Processing
Sound processing refers to the manipulation of an audio signal to alter its characteristics. This includes a wide range of actions, from basic volume and tone controls to complex effects like reverberation, compression, and room correction.
Soundscape
A soundscape is the total acoustic environment of a location, including all the sounds that are present. It can refer to a natural environment, an urban setting, or the immersive sonic world created by a film or game.
Soundstage
The soundstage is the perceived three-dimensional space created by an audio system in front of the listener. A wide and deep soundstage gives the illusion that the sound extends beyond the physical location of the speakers.
Spatial Room Impulse Response
A spatial room impulse response is a multi-channel measurement that captures not only the magnitude and timing of reflections but also their spatial characteristics. This detailed data allows advanced correction algorithms to optimize immersive audio playback.
Spectrogram
A spectrogram is a visual representation of sound that plots frequency on the vertical axis, time on the horizontal axis, and the amplitude (or intensity) of the sound as a color or brightness. This allows for the detailed analysis of sound patterns like transients, sibilance, or the decay of room resonances.
STFT (Short-Time Fourier Transform)
An STFT is a signal processing technique that involves performing a sequence of Fourier transforms on small, windowed sections of a signal. This provides time-varying spectral data, making it useful for real-time analysis or adaptive filtering applications.
Stereo Image
The stereo image is the perceived spatial placement of sounds between the left and right channels of an audio system. It is created through differences in level, timing, and spectral content between the two channels.
Sweet Spot
The sweet spot is the primary listening position where the audio system's performance is optimal. At this location, the stereo image is most focused, and the frequency response is most balanced.
Tap Count
The tap count is the number of coefficients used in a Finite Impulse Response (FIR) filter. A higher tap count allows for more precise control over low frequencies and steeper filter slopes, but it comes at the cost of increased processing power and latency.
Target Curve
A target curve is the desired frequency response that a room correction system is programmed to achieve. The target curve can be any shape, often with a gentle bass boost and treble roll-off, designed to provide a sound that is tonally balanced and pleasing to the human ear in a typical listening environment.
Texture
Texture describes the perceived surface quality of a sound. It relates to the interplay of micro-details, harmonics, and transient information that can make a sound feel rough, smooth, grainy, or liquid.
THD + N (Total Harmonic Distortion + Noise)
THD+N is a measurement that expresses the sum of all harmonic distortion components plus all noise as a percentage of a system's output relative to its input. A lower number indicates a cleaner, more accurate system.
Timbre
Timbre (pronounced "tam-ber") is the characteristic quality or "color" of a sound that distinguishes different types of musical instruments or voices. It is determined by the harmonic content and dynamic characteristics of the sound.
Time-Domain Correction
Time-domain correction is a correction strategy that focuses not just on how loud frequencies are (frequency domain), but on when they arrive at the listener. It works by shifting or reshaping the impulse response to reduce the smearing caused by reflections, resulting in improved clarity and transient response.
Tonality/Musicality
Tonality refers to the overall tonal balance of a sound, relating to its perceived warmth, brightness, or neutrality. Musicality is a more subjective term describing a quality that makes the audio engaging, coherent, and enjoyable to listen to, rather than merely accurate.
Transfer Function
A transfer function is a mathematical representation that describes how a system alters the magnitude and phase of a signal that passes through it. It is a central concept in designing filters for audio correction.
Transient
A transient is the brief, high-energy attack at the very beginning of a sound, such as a drum hit or a guitar pluck. The accurate reproduction of transients is crucial for preserving the sense of punch, impact, and realism in music.
Treble Range
The treble range refers to the highest frequencies in the audible spectrum, typically from about 4 kHz to 20 kHz. These frequencies contribute to the sense of air, detail, and brilliance in a sound.
Upmixing
Upmixing is a process that analyses the cues in a stereo audio signal to synthesize new surround or height channels. By decorrelating ambient information, it can create an immersive, multi-channel playback experience from legacy two-channel content.
Virtual Room Correction
Virtual room correction is a technology that uses dynamic filtering to simulate ideal room conditions without the need for physical acoustic treatment. It can adapt its equalization and phase adjustments in real-time as the listening position or audio content changes.
Virtual Surround
Virtual surround is a processing technology for headphones or speakers that uses head-related transfer functions (HRTFs) and room modelling. Its goal is to trick the brain into perceiving sound as coming from around, above, and behind the listener, creating an immersive experience from a limited number of physical speakers.
Waterfall Plot
A waterfall plot is a 3D visualization that combines frequency response with decay over time. This makes it particularly effective for spotting acoustic issues like resonances and for evaluating the effectiveness of damping.
Waveform
A waveform is a two-dimensional graph that plots the amplitude of a signal over time. By zooming in on a waveform, one can see details like transients, polarity, and the zero-crossing points, which is useful for precise audio editing.
White Noise
White noise is a random signal that has equal energy at every frequency across the audible spectrum. It is useful for certain types of acoustic measurements, for masking distracting sounds, or for synthesizing wind-like sound effects.
Windowing
Windowing is a process where a mathematical envelope is applied to a block of measurement data before it is analysed with an FFT. This technique helps to reduce an artifact known as spectral leakage, which improves the frequency resolution of the analysis.
Y-Adaptor
A Y-adaptor is a cable that either splits a single audio signal into two identical paths or combines two signals into one. While useful, using one can potentially upset the impedance balance between the connected components.
Zero-Crossing
A zero-crossing is the point in time where a waveform passes through zero amplitude. Performing edits or creating loops at these points is a frequent practice in audio editing as it minimizes the chance of creating audible clicks or pops.
Zero-Latency Filter
A zero-latency filter is typically a minimum-phase or IIR filter design that introduces a processing delay so small that it is imperceptible to humans. This makes it ideal for applications like live sound reinforcement or real-time monitoring where any delay would disrupt the performance.