David Howard, Jamie Angus — coustics and Psychoacoustics

1883 published claims from this book.

• A 0–30 ms gap before first reflections cues spatial size while avoiding echoes
• A 10 dB change roughly doubles or halves perceived loudness for complex sounds
• A 16-bit CD word yields a maximum signal-to-noise ratio of 96 dB
• A 5 dB threshold change on retest triggers full retesting and best-value recording
• A bare string radiates almost no sound; the body is what makes music audible.
• A bare vibrating string is nearly inaudible; the instrument body amplifies it
• A basic grasp of voice acoustics helps people synthesize desired sounds
• A bell’s perceived pitch arises from virtual pitch even when its prime is absent.
• A bowed string alternates constant-velocity grip with rapid snapback motion
• A bowed string cycles between slow grip and fast release phases
• A bowed string moves in two straight segments joined at a moving corner
• A brass bell reflects sounds selectively by frequency
• A brass mouthpiece has a measurable popping frequency near 850 Hz.
• A cappella just intonation makes a group’s overall pitch drift across key modulations
• A cent is one twelve-hundredth of an octave, or one hundredth of a semitone.
• A chord is defined by the intervals between its simultaneous notes
• A clamped drum edge always forms one circular displacement node.
• A closed cylindrical resonator filters sound to odd harmonics
• A complete ear-canal seal improves low-frequency transmission while feeling physically intrusive
• A complex wavenumber encapsulates dispersion and can drive propagation calculations directly
• A composer’s experimental soundscape can become an established tradition years later
• A concentrated pure tone damages hearing more than broadband sound at equal level
• A conical pipe closed at the narrow end behaves like an open-open pipe acoustically
• A constant-directivity horn cannot reproduce low frequencies, so a separate woofer is required.
• A critical distance exists where reverberant sound begins to dominate the direct field
• A crowd-control tone was repurposed into a ringtone adults cannot hear
• A data source is most efficient when all its symbols are equally probable
• A defined frequency marks the transition between modal and diffuse acoustic behavior
• A delay gap under 20 milliseconds makes a hall feel intimate
• A filter resolves a harmonic it is centered on even when bandwidth exceeds spacing
• A filter’s bandwidth sets how fast its output can change over time
• A filter’s inherent time response is determined by its bandwidth
• A filter’s transition region can audibly affect frequencies up to an octave away
• A finger hole’s acoustic effect depends on its size relative to pipe diameter
• A fixed frequency ratio always yields equal logarithmic distance at any pitch
• A fixed musical interval’s dissonance varies with register, worsening as notes lower
• A free-edged cymbal plate produces inharmonic, mixed modes
• A full orchestra’s sound can blanket a soloist’s spectrographic trace.
• A full-scale sine wave yields 1.76 dB more signal-to-noise than uniform signals
• A gentle bass rise in reverberation adds perceived warmth to acoustic music
• A hammer strike excites every mode an instrument can support simultaneously
• A high central cluster’s coverage angles should match the trapezoidal audience footprint.
• A large low-absorption area can outweigh a small high-absorption one in a room
• A line array’s −6 dB angle depends on wavelength relative to array length
• A local bore constriction can tune individual pipe modes separately
• A looping rhythm in tempo octaves with faded extremes sounds endlessly accelerating
• A loudspeaker’s physical size sets a hard floor on the frequencies it can direct.
• A loudspeaker’s size sets a floor below which it cannot be directive
• A mean opinion score averages many listeners’ ratings into one quality number.
• A medium’s density and elasticity relate sound pressure to particle velocity.
• A melody remains recognizable from a single non-octave harmonic alone
• A minimum number of cycles is required before a definite pitch is perceived
• A mole of any substance always contains the same count of molecules
• A mouthpiece’s popping frequency reveals how it extends effective pipe length.
• A narrowband tone damages the ear more than the same energy spread wide
• A noise-shaping quantizer works by feeding extracted quantization error back through a filter.
• A note needs a minimum cycle count to be heard as pitched
• A note’s onset, not its steady state or decay, most strongly determines perceived timbre.
• A pipe closed at one end produces only odd-numbered harmonics.
• A polar pattern’s squared magnitude equals the transform of its autocorrelation
• A principal organ pipe’s first two harmonics dominate its spectrum
• A proper ear seal improves low-frequency transmission in passive devices
• A quarter-wavelength resonator tube boosts a bar’s fundamental loudness
• A rectangular pulse transforms into a sinc-shaped spectrum.
• A rectangular room supports infinitely many modal frequencies set by half-wavelength fits
• A reed’s oscillatory persistence width is set by its material and mounting
• A register hole kills one resonance while sparing another.
• A reverberant field only exists once sound has reflected many times.
• A reverberant field requires roughly twenty or more reflections to exist.
• A room can be acoustically dead yet not anechoic, with early reflections surviving.
• A room mode’s decay time depends on its path length between reflections.
• A room’s critical frequency depends on absorption and mode bandwidth, not just room dimensions
• A room’s lowest modal frequency is its first axial mode along the longest dimension
• A room’s reverberant field can vary by 14 dB across the audible frequency range.
• A room’s reverberant field has its own frequency response that colors any distant-miked recording
• A sequence with delta-function autocovariance produces uniform-magnitude scattering.
• A Shepard tone creates an endlessly ascending pitch while staying within one octave
• A single active speaker creates an interaural time delay that listeners localize.
• A single critical frequency marks the modal to diffuse transition
• A single melodic line can be heard as several distinct voices by frequency grouping.
• A single modulation-based index predicts speech intelligibility across most forms of degradation.
• A single nondirective speaker over a wide audience produces unacceptable volume variation.
• A single performed melodic line can be heard as multiple simultaneous parts
• A single sound source near a reflecting surface creates interference via an image source
• A single sound source near a reflective boundary creates a virtual second source
• A single source near a wall creates a virtual image source that interferes
• A small listening room’s first reflections mask the recorded acoustic space
• A small loudspeaker radiates like a bare bulb rather than a focused beam.
• A smaller listening zone needs smaller absorption patches
• A sound can rise in pitch forever without ever rising in frequency
• A sound wave’s pressure component arises from the force driving molecular displacement.
• A sound’s perceived warmth depends on specific frequency balance
• A sound’s time-domain rate of change determines its frequency-domain bandwidth
• A source repeating one symbol conveys zero total information
• A source sending one symbol repeatedly conveys zero information despite infinite potential self-information.
• A source’s changing directionality with frequency alters perceived timbre via the reverberant field
• A speaker array on the boundary can synthesize and preserve a recorded sound field
• A speaker array’s directional pattern is set by its spacing and element properties
• A speaker near hard surfaces produces phantom image sources that color its sound
• A speaker’s coverage angle roughly doubles for each octave below its minimum directivity frequency.
• A spectrogram displays a sound’s spectrum plotted against time
• A spectrogram preserves timing of acoustic changes in a note
• A spectrum’s relative harmonics become real frequencies when scaled by the fundamental
• A spherical-cap transducer with Legendre shading holds constant beamwidth above a cutoff frequency.
• A square wave is built from odd-multiple sine components
• A standing wave is still moving at normal speed while appearing stationary
• A steady-sounding note still has amplitude envelope variation within it
• A stopped cylindrical pipe supports only odd-numbered modes
• A stopped flue pipe sounds its fifth harmonic before its fundamental on attack
• A stopped flue pipe’s odd-mode support creates its characteristic chiff at onset
• A stopped pipe reflects compression from the closed end and rarefaction from the flue.
• A stopped pipe sounds an octave lower than an open pipe of equal length
• A stopped pipe’s cycle takes twice as many snapshots as an open pipe’s
• A string’s nth mode has n+1 nodes and n antinodes
• A stringed instrument’s pluck position can be read from harmonic amplitude dips.
• A struck bell’s overtones enter at different times, shaping perceived timbre.
• A struck bell’s spectral components begin at staggered times rather than together
• A symbol’s information content rises as its probability of occurring falls
• A symbol’s self-information falls to zero as its probability approaches certainty.
• A tangential room mode is defined by half-wavelengths along two wall dimensions
• A transducer’s physical size relative to wavelength determines its directivity.
• A trumpet mouthpiece extends effective pipe length, especially lower ones.
• A tuning fork behaves as two bars clamped at one end
• A vibrating bar’s higher modes fall outside the musical scale and harmonic spacing.
• A-weighting is recommended for all sound levels despite fitting low levels best
• A440 tuning and equal temperament dominate modern keyboards but not historical or organ practice
• Above resonance, wall mass blocks sound, but coincident resonances leak high frequencies
• Absolute category-rating tests average listener ratings into a mean opinion score.
• Absorbers treating one frequency band also lower reverberation time at other bands
• Absorbing first reflections cascades to attenuate the whole reflected sound field
• Absorbing reflections is easier to build but shortens reverberation time
• Absorption at each reflection governs how fast reverberation decays
• Absorption at each reflection makes sound energy decay exponentially over time
• Absorption coefficients reduce reflected intensity by their proportion
• Accentuating individual harmonics makes them perceptually break away as separate notes
• Accurate square waves require infinitely many harmonics
• Achieving intelligible speech reinforcement in real acoustics is genuinely difficult.
• Acoustic design now shapes everyday spaces from offices to wind farms
• Acoustic diffusion needs surface bumps sized between an eighth and a half wavelength
• Acoustic illusions may become musically useful when electronic sound control is exact
• Acoustic illusions reveal how hearing analyzes sound and can become compositional tools.
• Acoustic illusions reveal perception mechanisms and shape music performance
• Acoustic instrument notes are never truly steady, even in their sustain phase.
• Acoustic instrument sounds constantly change even during steady-state portions
• Acoustic length scales are set by wavelength, not absolute distance.
• Acoustic measurements require environmentally controlled conditions
• Acoustic refraction parallels optical refraction via material-dependent wave speed
• Acoustic standing waves are termed modes with numbered order by frequency
• Acoustic wavelength size relative to objects governs how sound interacts with them
• Acoustic waves reflect at discontinuities in tract area
• Acousticians measure lengths in wavelengths because object effects scale with wavelength.
• Acoustics explains why pitch drifts, letting conductors make informed tuning choices
• Acoustics increasingly shapes everyday life through sound-aware design choices
• ACR tests average five-point ratings into a mean opinion score
• Active noise cancellation contaminates the wanted signal when phase alignment is imperfect
• Active noise cancellation degrades the wanted signal and weakens at high frequencies.
• Active noise cancellation fails at high frequencies due to phase shift limits
• Adaptive quantization and noise shaping distinguish lossy from lossless coders
• Adding a harmonic-reinforcing organ stop boosts all multiples of that harmonic
• Adding a mouthpiece lowers higher modes more than lower ones.
• Adding absorbent-free volume to a hall raises its reverberation time.
• Adding an organ stop reinforces that harmonic plus all its integer multiples
• Adding harmonic-series members of an unsounded note creates perceptual ambiguity about pitch location.
• Adding more loudspeakers can worsen speech intelligibility by shrinking critical distance
• Adding sound levels differs depending on whether sources are correlated or not.
• Adding sound powers yields identical totals for correlated and uncorrelated sources
• Adding uncorrelated sound sources raises level as the square root of their number
• Adding valve pipe lengths does not add linearly to frequency drops, forcing compromise tuning
• Adjacent equalizer bands interact, making careless boosting degrade the result.
• Adjacent harmonics in the natural series produce progressively smaller musical intervals
• Adjacent harmonics of a fundamental produce every familiar musical interval
• Adjacent natural harmonics create smaller intervals as harmonic number increases.
• Advertised constant-directivity specs are often meaningless across most speech frequencies.
• Advertised constant-directivity specs often cover a band irrelevant to speech intelligibility.
• Advertised speaker directivity specs can be meaningless for the speech band.
• Age-related hearing loss affects high frequencies most and strikes men more than women
• Age-related hearing loss directly causes increased preferred treble
• Aging and loud sound exposure both erode hearing from high frequencies downward
• Air absorption can be approximated as an equivalent absorption area scaled to room volume.
• Air absorption is negligible in small rooms but overbrightens artificial reverberation
• Air absorption must be modeled or artificial reverb sounds overbright
• Air absorption of high frequencies makes distant voices sound duller
• Air and tap resonances align with open-string fundamentals that differ across instrument sizes
• Air and top-plate resonances dominate violin low-frequency response
• Air leaks and ventilation paths defeat sound isolation more than walls do
• Air possesses compressibility that substitutes for a Young’s modulus
• Air reed oscillation is driven by pressure pulse reflections at the pipe mouth
• Air reed pitch is set entirely by resonator length and pipe openness.
• Air transmits sound through springiness even though it lacks a Young’s modulus
• Air’s springiness needs an equivalent measure since it lacks Young’s modulus
• All common Western scales are built on the octave ratio of two to one
• All complex sounds, including noise, are superpositions of pure sinusoids
• All information-bearing sound must vary over time
• All musical instruments and voices share a source-filter acoustic structure
• Almost every musical sound contains at least two frequency components
• Alternating hair direction makes bowing work equally in both directions
• Alternating octave tones between ears produce a fixed high-right low-left percept
• Alternating timbres can split a single melodic line into separate perceived streams
• Ambisonic sound fields degrade gracefully as channels are removed
• Ambisonic systems synthesize a sound field through coordinated pressure and velocity, not channel-to-direction mapping
• Amplitude diffusion gratings have no low-frequency limit, only a high-frequency one
• Amplitude gratings diffuse less well than phase gratings but work at lower frequencies
• Amplitude stereo produces directional images only when both ears hear both loudspeakers
• Amplitude variation can arise purely from pitch change interacting with fixed resonances.
• An 16.8 kHz sine wave is used commercially as a teen deterrent
• An acoustic reflex desensitizes the ear to protect cochlear hair cells
• An audience’s breathing and perspiration dynamically alter a hall’s high-frequency acoustics
• An auditory brainstem implant aids lip reading and treats neurofibromatosis-related hearing loss.
• An electrolaryngograph measures vocal fold vibration via neck electrodes
• An even time-domain function yields a purely real Fourier transform.
• An Indian temple bell’s energy shifts between modes over seconds after striking
• An Indian temple bell’s low tone grows louder over seconds after striking
• An instrument’s output spectrum equals its input spectrum weighted by the body’s resonance response
• An instrument’s timbral category is not fixed but depends on playing technique
• An N-sample DFT yields exactly N discrete frequency values.
• An on-axis speaker produces no level variation from its own directivity.
• An open pipe’s fundamental period equals twice its length divided by wave speed
• An open tube end reflects sound with its energy almost entirely in air velocity
• An open-open pipe’s modes form a complete harmonic series at integer multiples
• Angling room walls changes resonance frequencies but cannot eliminate them
• Antiphase waveforms cancel when added together
• Any change in surface reflectivity, not just bumps, causes sound diffusion
• Any complex tone is decomposable into sine wave components displayed as a spectrum
• Any measurable data can be rendered as sound to enable rapid pattern exploration
• Any measurable data can be sonified to explore long timescales rapidly
• Any measurable data can be turned into sound to reveal its patterns.
• Any periodic signal decomposes into harmonic sinusoids at integer multiples of a fundamental frequency.
• Any periodic waveform can be built by summing harmonic sinusoids with chosen amplitudes
• Any sound wavefront can be synthesized from elemental spherical waves
• Any waveform can be approximated by summing orthogonal sine waves
• Any waveform can be exactly modeled by combining sine waves
• Aperiodic sounds emerge when continuous in-between frequencies cancel periodic repeats
• Application requirements dictate whether time or frequency domain coding is chosen
• Applied forces in plates generate bending waves rather than pure shear waves
• Applied forces launch bending waves because they produce both lateral and longitudinal displacements
• Applying progressive phase shifts steers a beam array off-center
• Array far-field patterns obey Fourier transform theorems, so directional elements spatially filter aliased lobes.
• Array spacing beyond half a wavelength causes spatial aliasing with multiple lobes
• Articulation is the fraction of speech energy audible per critical band.
• Ascending and descending circles of fifths produce different pitches
• ASW depends most on mid-frequency reflections within the first 80 ms
• Asymmetric waveforms require combining sine and cosine coefficients into one spectrum
• At a pressure-release boundary, sound pressure nulls while particle velocity doubles
• At a solid reflecting boundary, sound pressure doubles while velocity drops to zero.
• At least 16 non-expert listeners are needed for reliable speech-quality tests
• At low frequencies, loudspeaker signals add coherently so delay and amplitude trade off.
• Atmospheric physics creates striking asymmetries in how sound travels
• Attack and release phases lasting milliseconds largely determine a note’s timbre
• Attention shapes what a listener perceives as loud in complex soundscapes
• Attentive focus on a particular timbre changes which auditory stream the ear follows.
• Audible amplitude change detection varies with signal properties
• Audible changes mark when two sine tones enter the same critical band
• Audible frequencies span an enormous range of wavelength sizes.
• Audible pressure range spans over a million to one, forcing a logarithmic scale
• Audience humidity from breathing changes a venue’s high-frequency acoustics over time
• Audience moisture exhaled into a room dynamically shifts high-frequency reverberation
• Audio coding systems compress signals to fit limited channels and storage
• Audio compression maximizes information by encoding only unpredictable signal parts
• Audio interfaces can give blind users richer information than vision-based systems.
• Audio symbols with high probability carry less information and need fewer bits
• Audiometric hearing loss maps directly onto missed timbral and speech-sound differences
• Audiometric tones are deliberately aperiodic so subjects cannot anticipate them
• Audiometry begins at 1 kHz because it is the most universally audible frequency
• Auditory attention requires temporal variation in any sound signal
• Auditory detection and discrimination are distinct abilities that can diverge
• Auditory expectation of reverberation is calibrated against visual room size.
• Auditory filter bandwidth determines which harmonics of a note get resolved
• Auditory filter bandwidth scales nonlinearly with center frequency
• Auditory filters are analyzed one component at a time to build up complex effects
• Auditory filters respond asymmetrically, favoring frequencies below center
• Auditory models frame perception as filtering, with hearing loss as spectral attenuation
• Auditory streaming splits one interleaved vocal line into two perceived lines
• Auditory thresholds are measured by adaptive level bracketing with a 50 percent response criterion.
• Averaging hearing loss across frequencies can mask a severe dead spot.
• Averaging hearing loss across frequencies can mask serious impairment
• Avoiding commensurate room dimensions prevents overlapping modal frequencies that cause acoustic problems
• Bach’s organ pedal leaps create perceptual pitch ambiguity for listeners.
• Bach’s organ writing can produce Shepard-like pitch ambiguity in listeners
• Bar vibration modes are spaced by squares of odd integers, far apart
• Baroque mansions achieved balanced acoustics through wood paneling absorbing low frequencies
• Basic binaural cues cannot fully explain front-back and elevation judgments
• Basilar membrane displacement produces an asymmetric filter response across frequency.
• Basilar membrane registers frequency by place along its graded length.
• Basilar membrane vibration envelopes are asymmetric around the peak
• Bass diffraction is negligible while midrange diffraction matters in cabinet design
• Bass strings are wire-wrapped to avoid the fourth-power inharmonicity of thicker cores
• Bass wavelengths are long enough to avoid cabinet diffraction but midrange wavelengths are not.
• Beamwidth halves each time the aperture spans twice as many wavelengths.
• Beatboxers synthesize percussion through vocal folds that are not percussion instruments
• Beatboxing exploits auditory streaming so one voice sounds like two lines
• Beats, roughness, and separation emerge at different frequency differences
• Below 0.3 average absorption, Sabine’s simpler equation is accurate enough.
• Below 213 Hz a living room must be analyzed by modal decay, not reverberation
• Below a horn’s minimum frequency, coverage angle widens with each lower octave.
• Below roughly 12.5 Hz the difference tone coincides with the beat frequency.
• Bending wave velocity tops out at the material’s transverse shear wave speed
• Bending waves disperse because their wavenumber grows only with the square root of frequency
• Bending waves disperse with frequency, so their waveshape is not preserved
• Bernoulli’s airflow across curved lips initiates their vibration
• Binary amplitude sequences can represent reflection and absorption in diffuser design
• Binary responses fail when the nature of a sound difference matters
• Binaural direction cues alone cannot distinguish front from back or vertical source angle
• Binaural realism degrades when the listener’s head differs from the recording head.
• Binaural recordings must be headphone-based because cross-ear coupling destroys the illusion
• Block-based gain setting guarantees overload-free coding at the cost of side information
• BMLD-based prediction models operate per spectro-temporal unit, not per channel
• Body resonances aligned to open-string fundamentals give each string instrument its timbre
• Body size relative to open-string pitch shapes each string instrument’s distinctive timbre.
• Bone conduction isolates cochlear function from outer and middle ear disorders
• Bone versus air conduction plots distinguish outer-ear from inner-ear hearing loss
• Bone-conduction audiometry bypasses outer and middle ear to directly assess inner ear function.
• Bone-conduction audiometry delivers sound mechanically to the skull rather than acoustically.
• Bookcases and carpets give practical domestic acoustic treatment for reflection zones
• Boosting energy near 2 kHz and 8 kHz makes a sound brighter and harsher
• Boosting rear-row level makes rear-wall reflections degrade intelligibility
• Boundaries alter bass output by changing the loudspeaker’s acoustical radiation resistance
• Boundaries concentrate sound power but do not affect the inverse-square spreading law.
• Boundary placement boosts sound intensity 3 dB per boundary by increasing directivity
• Boundary-induced source power changes scale with distance relative to wavelength
• Bounded sound waves change impedance with tube area, causing reflections
• Bow hairs are laid half in each direction so both strokes grip equally
• Brain structures supporting sound perception become harder to observe near the cortex
• Brass instrument bells reflect low frequencies early and pass high ones.
• Brass instrument design transforms odd-only pipe modes into a near-complete harmonic series
• Brass instruments are lip-driven pressure valves that behave acoustically like stopped pipes.
• Brass instruments compensate for fixed valve lengths with player-adjustable tuning
• Brass instruments differ sharply in how many harmonics their tones carry
• Brass instruments have no usable fundamental and start from their second mode
• Brass instruments operate acoustically as stopped-end tubes driven by lip reeds
• Brass instruments vary pitch by mechanically changing the length of their tubing
• Bright timbres stem from strong energy in harmonics above the seventh
• Built-in microphone equalization is insufficient; fully compensating proximity effect demands a variable equalizer.
• Bumps spaced at half a wavelength re-radiate in phase and vanish acoustically.
• Cabinet edges and centered drivers create diffraction that smooth curves minimize
• Cabinet shape and driver placement materially govern diffraction distortion
• Canceling speaker HRTFs can synthesize virtual surround sound when head position is fixed
• Cancelling both ears’ HRTFs requires a fixed speaker layout and a motionless head.
• Cascading open tone holes act as a frequency-based filter shaping woodwind timbre
• CBT arrays achieve constant beamwidth using only level shifts, not complex frequency-dependent processing.
• CBT arrays need only frequency-independent level shifts, unlike conventional line arrays
• CBT arrays taper drive levels from center to edges for constant beamwidth
• CD quantization noise is masked by loud signals across most of the audio band
• Ceiling reflections cannot substitute for side-wall reflections.
• Ceiling reflections do not match side-wall reflections for good listening
• Ceiling-reflected diffuse sound cannot substitute for real side-wall reflections
• Centrally mounted drivers between two edges cause coincidence effects that degrade sound.
• Centrally mounted drivers between two enclosure edges worsen response via coincidence effects
• Cents are the standard unit for reporting pitch in psychoacoustic experiments
• Certain processes are far easier to analyze in the frequency domain than in time
• Changing organ reed stops can create an illusory continuously descending pitch line
• Changing room shape relocates resonances but never removes them
• Chiff-voiced open pipes briefly sound an octave high at onset
• Chiff-voiced organ pipes begin with the second harmonic leading by about 30 ms
• Child soloists show a singer’s formant shifted upward because their vocal tracts are smaller
• Choice of audio coding domain depends on the application
• Chord consonance is computed from individual harmonic proximity across the notes
• Chord consonance is maximized when singers’ fundamentals are integer-related harmonics
• Chord dissonance rises as harmonic number increases and interval shrinks
• Chords sound consonant when singers’ fundamental frequencies are integer-related.
• Clarinet even harmonics are attenuated, not absent, from the spectrum.
• Classical concerts expose audiences to brief noise peaks above workplace safety limits
• Classical musicians must hear themselves via hall reflections alone
• Clear musical pitch arises from regularly repeating acoustic waveforms
• Clear pitch perception depends on periodic repetition of the pressure wave
• Clinical audiograms invert the vertical axis to show hearing loss downward
• Clinical audiometry samples hearing in octave-spaced tones from 125 Hz to 8 kHz
• Coarse stepped approximations of continuous amplitude shading barely degrade directivity
• Cochlear coiling makes inner nerve damage cause low-frequency hearing loss
• Cochlear damage widens auditory filters and impairs sound interpretation.
• Cochlear implants stimulate each cochlear place with calibrated pulses matched to ERB filter bands.
• Cochlear implants translate frequency bands into place-coded stimulation along the cochlea
• Codec effectiveness trades off against bits spent on side information
• Coded audio carries maximum information when the signal is maximally surprising
• Coding block length trades side-information cost against noise masking.
• Coincident room modes create audible frequency peaks that extra resonances must balance
• Color analogies naturally describe timbre, whose acoustic basis is overtones
• Combination tones exist below both source tones and vary in audibility
• Combination tones fill in missing harmonics of a perceived fundamental
• Combination tones fill in missing harmonics of an implied fundamental.
• Combination tones sit below the lower tone and vary greatly between listeners.
• Comparison category-rating tests measure whether processing improved or worsened a signal
• Comparison category-rating tests measure whether processing improves or degrades audio.
• Comparison rating tests use two-sided scales for signal processing effects
• Complex numbers conveniently represent coherent sound addition
• Complex representation handles propagation-delay phase shifts on speaker outputs
• Composers can blur a musical event’s perceived timing by manipulating harmonic-series membership
• Compressing a signal’s amplitude damages the syllabic variation listeners use to decode words.
• Compressions and rarefactions define sound as a longitudinal wave through a medium.
• Concentrated spectral energy damages the ear more than broadband noise of equal level
• Concentrating sound into a smaller solid angle raises horn sensitivity
• Concert audiences can briefly experience noise above damaging thresholds
• Concert hall reverberation times range from one to two seconds
• Conflicting visual and audio size cues make hybrid reverberation sound wrong.
• Conical bores suppress the reflection at the narrow end, yielding all harmonic modes
• Conical-bore instruments need only 12 chromatic holes because they overblow an octave
• Consonance and dissonance judgments stem from critical bandwidth mechanics
• Consonance judgments rest on psychoacoustic critical bandwidth
• Consonant intelligibility depends on reverberation time in the 1-2 kHz bands
• Constant on-axis response forces total radiated acoustic power to fall as directivity Q rises.
• Constant-directivity horns are efficient but cannot cover frequencies below 500 Hz
• Constant-directivity horns trade low-frequency reach for high efficiency and sustained output.
• Continuous-spectrum noises like ‘ss’ and ‘sh’ still convey pitch sensations.
• Controlled experimentation is how psychoacoustics gets knowledge from subjective reports
• Controlled experiments convert listening hunches into identified causes by isolating variables.
• Controlled experiments trace sound-feature variables against listeners’ behavioral responses
• Controlled listening experiments use pilot tests and counterbalanced stimulus orders to tame independent variables
• Controlled-reflection technique maximizes initial time-delay gap by suppressing early reflections
• Converting stereo to binaural sound requires modeling cross-coupling between speakers
• Convolving aperture shapes multiplies their far-field polar patterns.
• Correlated and uncorrelated sound sources sum by different rules
• Correlated sound sources add pressures while uncorrelated ones add energy
• Correlated sound sources cancel at points where path difference equals half a wavelength
• Correlated sound sources combine by direct pressure addition
• Correlated sound sources create spatial interference patterns that uncorrelated sources do not.
• Correlated sound sources sum to between zero and twice their individual amplitudes
• Correlated source summation swings from zero to twice amplitude by frequency
• Correlated source summation varies strongly with frequency.
• Correlation removal can be implemented in either the time or the frequency domain
• Counterbalancing task order across listeners cancels learning effects, unlike fixed-order testing.
• Critical bands are the most important psychoacoustic principle for music perception
• Critical bandwidth corresponds to the masker frequency range that fills a single auditory filter
• Critical bandwidth is the psychoacoustic basis of consonance and dissonance.
• Critical bandwidth widening means only low harmonics are resolved individually
• Critical distance is computed from room constant and source directivity
• Critical distance is short for typical domestic listening rooms
• Critical distance is the point where direct and reverberant sound levels are equal.
• Critical distance is where reverberant sound overtakes the decaying direct sound
• Cross-fingering works because an uncovered hole’s pressure-node position is affected by holes farther along
• Current acoustic measurements are too coarse to explain how listeners distinguish violins by maker
• Current acoustic models cannot explain how experts distinguish fine timbre differences between instruments
• Curved CBT arrays need no frequency-dependent signal processing
• Curved enclosures with off-center drivers minimize acoustic diffraction.
• Curved small enclosures produce the smoothest acoustic response, ideally spherical
• Curved surfaces can focus reflected sound and enhance its intensity
• Curved surfaces concentrate sound locally while draining power elsewhere in the room
• Curved-surface airflow, not the Bernoulli effect, drives lip vibration.
• Cyclic sound paths create standing waves when path lengths match half wavelengths
• Cylindrical stopped pipes support only odd-numbered modes, making instruments sound an octave lower
• Cylindrical stopped-pipe bores support only odd-numbered modes, forcing extra finger holes
• DCR tests measure how much processing degrades an audio signal
• Dead rooms with only direct sound reveal low-level audio detail clearly.
• Decibel sound levels cannot be added directly because sound powers combine, not levels
• Decibel sound levels cannot be added directly because they are logarithmic
• Decibel values must be converted to ratios before they can be added
• Decibels cannot be summed directly because they are logarithmic power ratios
• Decor of Baroque halls absorbed low frequencies, balancing the room’s acoustics.
• Deep porous absorbers give flat wideband absorption at the cost of space
• Deeper rear cavities and heavier panels lower a resonant absorber’s tuning frequency
• Degradation category-rating tests evaluate processed signals against the unprocessed original.
• Degradation mean opinion scores quantify how much processing has harmed a signal
• Delay-induced spherical wavefronts plus Legendre weighting yield frequency-independent beamwidth.
• Delay, distance, and intensity loss can be plotted on one common graph
• Delayed correlated sources produce frequency-dependent comb filtering
• Delayed fill loudspeakers should lead the acoustic signal by fifteen milliseconds.
• Delayed identical sound reproduces a pitch whose frequency equals the inverse delay
• Delayed-noise comb filtering can generate memorable pitch patterns
• Dense wall fabrics with empty space behind them become unwanted absorbers
• Denser materials slow sound while stiffer bonds speed it up
• Designing filters that filter without altering perceived timbre is genuinely difficult.
• Designing for worst-case signal levels wastes capacity most of the time
• Designing quantizers for peak levels wastes bits that masking could reclaim.
• Detailed early-reflection structure matters as much as reverberation time for perceived acoustics
• Detecting sounds and discriminating between them are distinct hearing abilities.
• Detecting sounds does not guarantee discriminating them, which also needs intact critical-band processing
• Detection is prerequisite to but insufficient for sound discrimination
• Different noise types mask different specific speech sounds by matching their spectra
• Different voice qualities result from distinct vocal fold modes
• Differing tensions on paired drum heads deliberately spread nonharmonic modes to sound nonpitched
• Diffraction arises because air-pressure waves cannot abruptly end at an edge
• Diffraction occurs around solid objects as well as through openings.
• Diffuse rear reflections blend instruments while specular ones create hot spots
• Diffuse reflections create a larger, forgiving sweet spot in listening rooms
• Diffuse reflections help integrate and blend acoustic instrument sound
• Diffuse reverberant fields let power in the field be equated to sound pressure level
• Diffuse surfaces prevent sound focusing away from the optimum listening position
• Diffuse surfaces scatter sound hemispherically rather than mirror-like
• Diffuse wall reflections are temporally spread and lower in level than specular ones
• Diffuse-field assumptions hold at high frequencies but break down at low ones
• Diffuse-field equations fail in geometrically extreme rooms
• Diffuser depth must scale with the lowest frequency it can scatter.
• Diffusers must be sized between an eighth and half a wavelength to scatter sound.
• Diffusing early reflections lowers their level without absorbing the sound energy
• Diffusing sequences keep their scattering properties because scaling for wavelength preserves their Fourier transform.
• Diffusion reduces early reflection level without absorbing sound energy.
• Diffusion replaces sparse loud early reflections with a dense set of quiet ones
• Diffusive surfaces scatter sound only within a bounded frequency band.
• Digital noise can be shaped to hide where hearing is least sensitive
• Digital signal processing made constant-directivity array loudspeakers practical and popular
• Digital transforms replace continuous time with a sample index.
• Direct sound carries signal information uncontaminated and governs speech intelligibility
• Direct sound intensity falls with the square of distance from the source
• Direct sound preserves signal information because it has not yet hit any boundary.
• Direct sound weakens by a factor of four for every doubling of distance
• Directional horns suppress spatial aliasing when they control directivity below the aliasing frequency.
• Directive drivers spatially filter out array side lobes that would otherwise alias.
• Directive loudspeaker elements act as spatial filters reducing array side lobes
• Directivity and elevated placement produce even coverage across most of a deep audience.
• Directivity depends on wavelengths fitting across the radiating aperture
• Directivity of any uniformly driven aperture depends on wavelengths fitting its length
• Discrete transforms represent signals only at finite discrete frequencies, unlike infinite Fourier series
• Discretizing a signal into levels is inherently lossy and produces error
• Dispersion causes bending waves to lose their original waveshape
• Dispersive media split wave speed into phase and group velocity
• Dissimilar layers and reduced coupling raise real-world sound isolation.
• Distance from a source changes the balance between direct and reverberant sound
• Distance makes sound quieter and duller as air absorbs high frequencies
• Distinct high partials make tones sound cutting or rough and powerful
• Distributing absorption randomly scatters sound while concentrated absorption leaves modes intact
• Dividing mean free path by sound speed gives time between reflections
• Domestic hi-fi listening is dominated by reverberant rather than direct sound
• Double glazing can be worse than single glazing when panes match
• Double-slope decay makes a space sound drier than its T60 predicts
• Doubling acoustic pressure raises SPL by exactly six decibels
• Doubling loudness requires 10 dB, not 3 dB or 6 dB
• Doubling pressure yields 6 dB and a tenfold increase yields 20 dB.
• Doubling uncorrelated sound sources adds only 3 dB of loudness
• Driver spacing exceeding half a wavelength causes spatial aliasing.
• Drum families differ structurally in head number, bowl shape, and playing excitation
• Drum membranes always have one clamped-edge circular node
• Drum-kit drums are deliberately tuned non-pitched so they fit any musical key
• Drums produce indefinite or definite pitch depending on head tuning match and snares.
• Dynamic auditory cues can override clear visual evidence of source location
• Each added boundary surface boosts sound intensity by 3 dB up to three boundaries
• Each cochlea contains about 15,500 hair cells in two distinct types
• Each instrument traces a distinctive dynamic path on the tristimulus diagram
• Each instrument’s timbre is defined by its unique recipe of harmonic partials
• Each octave rise halves how many hearing-range harmonics a note carries
• Each person’s directional hearing cues are learned and specific to their own ears
• Each person’s unique ear shape requires individualized acoustic rendering for 3-D audio
• Each reflecting boundary doubles sound intensity at a point, adding 3 dB
• Each surface reflection removes a fixed fraction of sound energy.
• Ear and brain adaptation dulls timbre modifications, spurring ever more boost
• Ear localizes to first-arriving sound unless delayed signal exceeds 12 dB louder
• Ear sensitivity depends jointly on frequency and on sound level
• Ear sensitivity variation cannot be fixed with simple treble and bass boost
• Ear trumpets could be matched to specific hearing loss profiles.
• Early equalizers corrected recording technology flaws now largely eliminated
• Early place theory assumed pitch requires the fundamental frequency present in the sound wave.
• Early reflection intensity depends on both travel distance and the reflecting surface’s absorption.
• Early reflections must arrive within 20 ms or they are perceived as echoes.
• Early reflections must be suppressed about 15 dB to stay inaudible
• Early reflections must be suppressed by about 15 dB before they disturb stereo imaging.
• Early reflections must stay below roughly 15 dB to avoid blurring a stereo image
• Early wall reflections pin perceived sound space to room size
• Echoes arise only when reflected sound exceeds a 30-millisecond delay
• Effective EQ depends on predicting the room’s acoustics at a given mic position.
• Effective low-frequency diffusion requires physically deep wells, about 34 cm for 500 Hz
• Effective sonification must fit acoustic changes within listeners’ audible limits.
• Effective sound diffusers ideally need no physical depth at all.
• Effective sound diffusion demands surface depths proportional to the target wavelength
• Effective sound diffusion requires bumps sized between an eighth and half a wavelength.
• Effective sound isolation requires airtight construction, which conflicts with ventilation needs.
• Efficient audio coding discards predictable signal components and encodes only novelty
• Efficient transmission codes mirror each symbol’s individual information content.
• Elastic lips and vocal folds self-oscillate once energy input begins.
• Elastic lips sustain oscillation only when energy is continuously input
• Electrolaryngography measures vocal pitch directly from fold contact, bypassing microphone bleed.
• Electronic reverberation enhancement suits amplified venues better than pure acoustic ones.
• Element-based acoustic models scale poorly as wavelength shortens
• Ellis translated klangfarbe as quality of tone to avoid ambiguous English words
• Enclosure volume and driver diameter can be decoupled in non-spherical cabinets.
• Encoding surfaces as binary absorber-reflector strips produces sound diffusion
• Energy at 3–5 kHz lets trained singers cut through an orchestra
• Energy density and impact rate jointly explain small-room acoustic behavior
• Engineers must speak pitch, not frequency, to communicate with musicians
• Engineers who discuss sound in musicians’ pitch terms communicate faster and get better results.
• Entropy encoding maximizes information carried by each bit pattern
• Entropy is minimized when all symbols occur with equal probability.
• Equal distances from two loudspeakers eliminate all room-induced destructive interference
• Equal frequency ratios occupy equal perceptual distances regardless of starting pitch
• Equal room dimensions boost and distort low-frequency loudspeaker response
• Equal room dimensions make standing wave modes coincide and distort low frequency sound
• Equal signals from both speakers create zero interaural phase difference
• Equal temperament beats stay within the 5% critical bandwidth so consonance survives
• Equal temperament cannot be tuned by ear because it offers no physical beatless guide
• Equal temperament tuning made organ mutation stops sound beaty and unfashionable
• Equal temperament’s added beats made mutation stops unfashionable until early-music revivalism.
• Equal tempered tuning divides the octave into semitones of the twelfth root of two.
• Equal-amplitude sounds in phase double in amplitude while out of phase they cancel
• Equal-probability symbol sources are the most efficient, wasting no bits.
• Equal-tempered accompaniment and training actively suppress singers’ natural just intonation
• Equalization adjusts sound by perceived timbre, not measured accuracy
• Equalization boosts or cuts frequency ranges by adding or subtracting a filtered signal.
• Equalization masks a mode’s audibility but leaves the room itself untouched
• Equalization persists because acoustic environments and microphone placement still create faults.
• Equalization selectively boosts or cuts frequencies rather than removing them
• Equalizer adjustments at specific frequencies produce predictable perceptual shifts in timbre
• Equalizer types differ by how many filter parameters users can vary.
• Even harmonics appear in clarinet sound despite its closed cylindrical resonator
• Even harmonics in clarinet tones are suppressed but not absent from the spectrum
• Even identical instruments playing in unison produce uncorrelated waveforms.
• Even identical notes vary measurably across every repeat performance
• Even noise without periodic waveforms can carry a weak sense of pitch.
• Even nonperiodic noise sounds convey a weak pitch sensation based on spectral energy distribution
• Even steady musical notes constantly change in their sound
• Every acoustic instrument reduces to a source feeding a modifier that yields an output
• Every additional loudspeaker in a distributed system reduces critical distance and worsens intelligibility.
• Every brass instrument comprises mouthpiece, mouthpipe, main pipe, and flared bell
• Every listening test has extraneous variables that must be controlled
• Every musical sound results from a source plus a filtering modifier
• Every room has a frequency below which modal effects break the diffuse-field assumption
• Every room has a low-frequency limit below which modal effects dominate
• Every room has a low-frequency limit below which modes, not diffusion, govern sound behavior.
• Every room has a low-frequency limit below which reverberation time is meaningless
• Every stringed instrument is analyzable as a sound source plus modifiers
• Exaggerated pitch contours and spelling alphabets both raise speech intelligibility in noise.
• Exciting multiple room modes instead of one reduces frequency and spatial sound variation
• Extra channel signals come from matrixing existing feeds or dedicated channels.
• Extra surround channels mainly aim at consistent localization and height illusion
• Extreme room shapes break reverberation-field formulas because sound visits surfaces unevenly
• Fabric-covered diffusers absorb sound instead of scattering it
• Falling reverberant energy makes a speaker sound duller and less present, reducing mix clarity.
• Fan-shaped halls sacrifice good lateral reflections for seating capacity and sightlines.
• Far-field array behavior depends on assuming the observation distance is effectively infinite
• Far-field array response equals the point-source sum with an added incidence-angle phase shift
• Far-field polar response at a given frequency is governed by the wavenumber k
• Fast-changing acoustic signals necessarily occupy a wider spectral bandwidth.
• Faster airflow through a constriction reduces pressure against the walls
• Faster sound decay broadens spectral peaks in observed spectra
• Female speakers’ higher pitch is less masked by reverberation and noise
• Fill speakers should aim only at uncovered zones to avoid spraying stray energy.
• Filter transition regions can extend a full octave from the cut-off frequency.
• Filtering a signal distorts its time response as well as its spectrum
• Filtering a signal is multiplication of responses and instruments themselves filter sounds.
• Filtering alters timbre, but nearby spectral boosting can partially offset the loss
• Filtering always alters timbre despite the goal of subjective transparency
• Filtering and resonance are more naturally understood in the frequency domain.
• Filtering in the frequency domain is simply multiplicative on spectra
• Filtering is easier to describe in the frequency domain than in time.
• Filtering removes unwanted frequency bands using high-pass, low-pass, and notch strategies
• Filtering signals equals multiplying their spectra by the filter function.
• Filters and the signal’s time response are physically coupled tradeoffs
• Filters attenuate gradually over a frequency range rather than cutting sharply
• Filters used for spectral analysis inevitably alter a signal’s time response.
• Finite boundaries in any wave-reconstruction array introduce edge distortions absent from idealized infinite sources
• Fixed additive valve lengths cannot correctly tune all brass notes
• Flanking transmission paths, not the partition itself, limit sound isolation
• Flue languid height precisely governs pipe speech timing and stability
• Flue pipes sound through a self-oscillating vortex called the air reed.
• Flush-mounting a speaker in a wall reduces the low-frequency rise to 6 dB
• Flute and piccolo span B3-D7 and B4-D8 respectively
• Flute harmonics enter one by one about 80 ms after the fundamental begins.
• Flute notes have a characteristic breathy onset with harmonics entering gradually after the fundamental
• Formants are standing-wave resonances of the vocal tract itself
• Fourier derived integral equations that extract harmonic coefficients from any signal.
• Fourier’s heat-transfer theorem now underpins all periodic signal analysis
• Free field radiation requires no reflecting boundaries near the source.
• Free reeds produce fewer high harmonics because absent shallot closing never fully halts airflow.
• Free reeds produce fewer high harmonics because airflow never fully stops
• Free reeds vibrate without shallot contact, tuned by a movable wire.
• Free space is a stricter concept than an unbounded region
• Free space requires freedom in all directions, not just unboundedness.
• Free space requires unobstructed propagation in all directions
• Free space requires unobstructed propagation in every direction, not mere unboundedness.
• Free-field radiation requires the source to be far from reflecting surfaces
• Frequency and time resolution are mutually exclusive trade-offs in any analysis of sound.
• Frequency discrimination depends on basilar membrane displacement separation
• Frequency modulation within a resonance peak produces visible amplitude modulation
• Frequency sorting in the basilar membrane can be reproduced in physical models
• Frequency-dependent absorption makes reverberation time vary with frequency, altering timbre.
• Frequency-domain coding using masking avoids transmitting masked sound information
• Frequency-domain thinking makes filtering easier than time-domain
• Front and back sound localization requires other mechanisms than ITD
• Full-bandwidth array directivity is practically infeasible due to aliasing cost
• Fully masked spectral components vanish from perception and thus from a sound’s timbre.
• Gamelan avoids perfect octaves because bell partials are inharmonic
• Gas pressure results from molecules bouncing off surfaces
• Gestalt grouping principles explain how the ear streams alternating tones
• Glottal airflow cycles through closed, opening and closing phases
• Good frequency resolution and good time resolution can never coexist
• Good noise isolation reduces preferred playback levels and hearing damage
• Good sound absorbers make terrible sound isolators
• Good sound absorption does not imply good sound isolation
• Good sound isolation requires airtight structure that still lets occupants breathe
• Graphic equalizers use fixed filter banks while parametric ones use variable controls
• Groups of nerves firing in staggered cycles code frequencies above 300 Hz
• Hair-cell damage widens acoustic filter bandwidth, harming speech separation and heightening dissonance perception
• Halving a speaker’s radiating angle doubles its critical distance.
• Hammer impact speed and mass trade off to shape a piano’s tone
• Handedness predicts which ear listeners hear high tones in
• Hard acoustic boundaries support standing waves at odd numbers of quarter wavelengths
• Hard boundaries reflect sound because pressure energy cannot cross a zero-velocity point.
• Harder blowing grows higher harmonics and brightens or rasps wind-instrument tone.
• Harmonic bands crowd closer together as frequency rises because of the ERB scale.
• Harmonics above the seventh exceed auditory filter bandwidth and cease to be resolved
• Harmonics appear as horizontal lines at fixed vertical positions
• Harmonics are integer related to the fundamental, whereas partials need not be.
• Harmonics are unresolved when critical bandwidth exceeds harmonic spacing.
• Harmonics cease being resolved when their spacing falls below the ERB
• Harmonics resolve only while their spacing exceeds the ear’s ERB
• Harmonics stop resolving once filter bandwidths exceed the fundamental’s spacing.
• Harmonics up to the seventh drive perceived consonance of complex-tone chords
• Harmony is sustained through chords of two or more simultaneous notes
• Having all harmonics up to the fifth or seventh also shapes timbre
• Head intensity-shading fails below the frequency where head size reaches one-third wavelength
• Head movement resolves directional ambiguity and explains headphone sounds seeming internal
• Head shading creates interaural level differences used for sound localization
• Head shadowing and binaural hearing raise sensitivity by six decibels or more
• Head-related pressure reflections and binaural hearing boost sensitivity by about 6-12 dB
• Head-related transfer functions are directional filters used to place sound in space.
• Head-related transfer functions are measured with in-ear microphones in anechoic chambers
• Head-related transfer functions spatially pan monaural sources by modeling each ear’s directional filter response
• Head-tracking externalises sounds that otherwise seem stuck inside the head.
• Headphone stereo fails because head filtering is missing or double-applied.
• Hearing adapts to focus on the audible 20-30 dB of decay
• Hearing adapts to timbre modifications, driving escalating compensation until a rest break resets perception
• Hearing aid effectiveness depends on measured speech communication benefit, not loudness alone
• Hearing damage first appears at the ear canal’s resonance frequency near 4 kHz
• Hearing damage widens acoustic filter bandwidth, impairing speech separation and sensitivity
• Hearing discrimination is measured phonetically by scoring correct speech sounds.
• Hearing loss impairs both sound detection and sound discrimination, from four distinct causes
• Hearing resolves close tones through beats, rough fusion, then separation bands
• Hearing sensitivity peaks near 4 kHz, so noise must be frequency-shaped to be inaudible
• Hearing thresholds are clinically tested at octave-spaced frequencies covering the speech range
• Hearing thresholds are mapped in 5 dB steps to balance precision against test duration
• Hearing two tones as separate requires separate basilar membrane peaks
• Hearing’s frequency-dependent sensitivity makes noise-position control possible
• Heavier panel absorbers resonate at lower frequencies
• Helmholtz derived timbre’s spectral dependence from theory alone, without critical-band data
• Helmholtz resonance is the trapped-air cavity resonance exploited in perforated-panel absorbers
• Helmholtz’s 1877 timbre rules anticipated modern psychoacoustic understanding
• Helmholtz’s 1877 timbre rules remain valid despite predating critical bandwidth theory
• Helmholtz’s nineteenth-century timbre rules anticipate the modern critical bandwidth and tristimulus analyses
• Helmholtz’s timbre rules map onto the axes of modern timbre-space experiments
• High harmonics must be dropped in the treble because they become inaudible
• High larynx closed quotient makes singing acoustically more efficient
• High-frequency deterrents work amid ambient music because music lacks energy up there
• High-frequency deterrents work because music has little energy above 16 kHz
• High-pass, low-pass, and notch filters each remove a distinct band of unwanted sound
• High-pass, low-pass, and notch filters each remove specific unwanted signal components
• High-pitched voices and instruments conventionally sit to the left of lower ones.
• Higher absorption reduces the first-reflection energy that feeds the reverberant field
• Higher closed quotient increases vocal efficiency through reduced lung sound leakage
• Higher formants above the third shape vocal identity and projection
• Higher material density slows sound propagation speed
• Higher piano notes require progressively greater octave stretching
• Higher source directivity reduces the reverberant field, equivalent to tonal darkening.
• Higher-order ambisonics captures acceleration components to improve localization accuracy
• Higher-order ambisonics improve localization accuracy and ease surround-sound format conversion
• Higher-order combination tones can be misheard as actual played notes
• Higher-pitched voices resist masking by reverberation and noise in public spaces
• Higher-pitched voices resist reverberation and noise masking better than lower ones.
• Highly articulated music needs drier acoustics than slow harmonic music
• Horn coverage angle doubles for every octave below the minimum directivity frequency.
• Horn size physically constrains the frequency range of directional coverage
• Horn size sets the lowest frequency a coverage angle can hold.
• Horsehair’s directional grip enables sustained bowed-string notes
• How bumps reflect sound depends on bump height relative to wavelength.
• How fast a sound changes dictates how wide its spectral bandwidth must be.
• How much one tone masks another depends on their frequencies and amplitudes.
• HRTFs must be stored as large azimuth-elevation tables because interpolation fails.
• Huffman coding achieves within one bit of source entropy but scales poorly
• Huffman coding builds comma-free variable-length codes that need no separators between words.
• Huffman coding’s efficiency collapses for small symbol sets because bits are integral
• Human audible pressure range spans over a million to one amplitude ratio
• Human hearing focuses on the first 20 to 30 decibels of reverberant decay
• Human hearing operates near the physical noise floor of air itself
• Human hearing operates near the theoretical physical limits imposed by thermal noise
• Human hearing spans a 6.4-million-fold pressure range from faintest to painful
• Human hearing spans a ratio of millions between faintest and painful sound
• Human high-frequency hearing declines markedly already in the twenties
• Human upper hearing range collapses above 16 kHz during the twenties
• Human vocal folds evolved as lung-protecting valves, not for communication
• Humans are exposed to environmental sound from before birth.
• Humans can vocalize desired sounds, aiding synthesis design
• Humid air carries sound faster than dry air because water molecules are lighter
• Ideal public address speakers radiate a frequency-constant angle of sound energy
• Ideal reverberant fields are hard to achieve in practice, requiring compromise
• Ideal reverberation is a smooth exponential decay whose rate depends on context.
• Ideal reverberation time depends on situation but follows broad general rules
• Identical signals at both ears produce a centered phantom image.
• Impedance changes with bounding surface area and cause reflections in instruments.
• Impedance discontinuities split an arriving wave into reflected and transmitted parts
• Imperfect phase alignment in ANC contaminates the wanted signal
• Impulse response analysis replaces mode analysis once mode density gets high.
• In a normal living room, reverberant sound dominates the direct sound from loudspeakers.
• In domestic listening rooms the reverberant field dominates the direct sound.
• In homes, air leaks dominate sound isolation more than the glazing itself
• In normal home listening the reverberant field dominates direct sound from speakers
• In organ reed pipes the resonator does not control the reed’s frequency
• In very small rooms the air mass loads sources and reduces their radiated output.
• In-filling components between harmonics cancel wave repetition to represent aperiodic sounds
• In-phase sounds double while out-of-phase sounds cancel to silence
• Incompressible cochlear fluid requires the round window to counter-move with the stapes
• Increased horn directivity trades front-row coverage for back-row coverage.
• Increasing speaker directivity reduces the reverberant field like a tone control does
• Independent partitions sum dB attenuations but real ones underperform this
• Independent partitions sum their attenuation in decibels
• Individual ear shapes give each listener a unique head-related transfer function
• Individual harmonics relate to each other as musical intervals, not just numeric frequency ratios.
• Individual instrument note onsets are irreproducible even under identical conditions
• Individual pinnae shapes make spatial hearing uniquely personal and learned
• Inertia delays wave transmission and the medium returns to rest afterward
• Information content in a signal rises with its unpredictability to the receiver.
• Information content increases as a message becomes less probable
• Information content is measured by occurrence probability and tied to bit count
• Information content relates to the number of bits required to transmit it
• Information only counts when a signal is actually transmitted
• Information theory borrows the term entropy from physics because its equation is analogous.
• Initial time delay gaps of 8-20 ms are naturally achieved in most rooms
• Instrument filtering adds predictability that audio coders exploit as redundancy.
• Instrument identity depends largely on attack and release transients, not steady tone
• Instrument identity relies heavily on note onset and offset phases
• Instrument makers shape bore profiles to tune woodwind modes around tone holes
• Instrument makers tune pipes by flaring ends or adjusting bore shape
• Instrument note onsets vary hugely across takes, players, and environments
• Instrument onset timbre varies so much that any single recording is only an example
• Instrument sound depends on whether restoring force comes from material stiffness or from stretching
• Instrument sound equals input spectrum multiplied by body response
• Instrument spectra blend source and modifier effects, revealing pluck position.
• Instrument spectra vary with player, loudness, pitch, and room
• Instrument stiffness versus tension determines how widely its modes are spaced
• Instrument timbre comes from waveform shape, not pitch
• Instrument timbre develops over time as higher harmonics fade in after the fundamental.
• Instrument timbre identity persists across notes via a stable tristimulus position
• Instrumental notes begin with the fundamental before higher harmonics appear
• Instruments sharing pitch sound different because waveform shape drives basilar-membrane timbre
• Instruments with lower partials up to the sixth sound rich yet sweet
• Instruments with split keys granting separate enharmonic pitches never gained popularity among keyboard players
• Intelligent equalization requires understanding a space’s direct-to-reverberant balance
• Intelligibility is measured by proportion of words heard correctly, not by perceived quality
• Intelligibility testing works by having listeners distinguish acoustically close consonant pairs
• Intelligible speech design requires direct sound to exceed reverberant and noise.
• Inter-channel coherence ICC measures signal similarity normalized from 0 to 1
• Interaural time and intensity cues can cancel each other in direction perception.
• Interaural time delay calculations must account for sound traveling around the head
• Interaural time delays alone cannot fully determine sound source direction
• Interference becomes negligible when the interfering wave drops below an eighth of the other
• Interference matching speech’s fluctuation rate is more distracting than its level predicts.
• Interference pattern of correlated sources varies with both listener position and frequency
• Interfering noises resembling speech’s modulation fool attention and exceed their level
• Interfering sound waves pass through each other because their energy flow is preserved
• Interval consonance ranking derives from critical-bandwidth proximity between harmonics
• Inverse-square law governs direct sound intensity even inside enclosed spaces
• Inverted and delayed sound sources cancel at low frequencies but reinforce higher up
• Inverting a signal cancels sound at all frequencies when paths are equal
• Just diatonic scale derives non-tonic notes from the dominant chord
• Just-tuned chord progressions predict key drift of a full semitone
• Kettledrum tone quality depends on strike position relative to nodal rings
• Kettledrums retune by pedal over a five-semitone range
• Known node positions let makers predict pitch changes from bore reshaping
• Lacking high harmonics makes instruments sound dark or dull
• Languid height determines whether a flue pipe speaks promptly, slowly, or not at all.
• Languid height governs air direction and thus promptness of pipe speech
• Large path-length variation between listeners produces uneven sound pressure levels and feedback.
• Large path-length variation causes poor direct-to-reverberant ratio and feedback
• Large pipe organs include stops sounding one or two octaves below foundation pitch
• Larger reflection-free listening volumes require proportionally larger absorption patches
• Larger rooms exhibit lower critical frequencies because volume scales with size
• Larger speaker size relative to wavelength enables higher directivity
• Late early reflections can ruin performers’ ability to hear each other and maintain ensemble
• Lateral early reflections arriving within 20 ms create perceived spaciousness
• Legendre shading tapers array drive from maximum at center to zero at the edges.
• Legendre-tapered drive levels give spherical-cap arrays constant beamwidth with low side lobes
• Legendre-tapered spherical transducer caps achieve constant beamwidth above a cut-off frequency
• Leq integrates squared pressure over time to estimate noise exposure
• Leq integrates squared pressure over time to yield an equivalent constant sound level
• Leq measurement integrates squared pressure to yield an equivalent constant level
• Leq measurements assess cumulative noise exposure linked to hearing loss
• Leq reduces varying noise to an equivalent constant sound level
• Lift on curved surfaces comes from a pressure gradient, not the Bernoulli effect
• Line-array beam patterns become progressively more directive as frequency rises.
• Lip and vocal fold vibration is driven by Bernoulli pressure reduction from airflow
• Lip constriction lowers the frequencies of all vocal tract formants
• Lip oscillation sustains continuously while airflow and adduction persist
• Lips only begin oscillating as a reed once airflow passes a threshold
• Listener position matters because violin radiation direction varies with frequency
• Listeners base perceived loudness on sounds they attend to, ignoring others
• Listeners lose the ability to identify instruments once onset and offset phases are removed.
• Listeners stream interleaved tones by pitch rather than by ear of delivery
• Listeners vary widely in how long masking persists across time.
• Listening analyzes periodic sound into integer-multiple sine components
• Listening rooms impose their own acoustics on top of recorded sound
• Logarithms convert weighted geometric means into simple absorption sums
• Logarithms in any base can be computed using natural logarithms.
• Long bass wavelengths diffract less because they exceed cabinet dimensions
• Long bass wavelengths relative to cabinet size make bass diffraction negligible.
• Long wavelengths diffract around obstacles more than short wavelengths do
• Long-ringing reeds restrict vibration to a narrow frequency band
• Long-term averaging of spectra suppresses random background noise
• Long-term safe noise limit may lie below the 85 dB standard
• Longitudinal and shear waves preserve waveshape because phase velocity is frequency-independent
• Longitudinal wave speed in solids depends only on density and Young’s modulus
• Looped pitch stimuli can generate an endlessly ascending scale that never actually climbs
• Loss of peripheral compression alone impairs recognition of low-level syllables between louder ones
• Lossless audio coding works by predicting and stripping signal correlations
• Lossless audio compression halves data size with ratio varying by signal type
• Lossless coding halves data size while leaving the signal bit-identical
• Lossy audio codecs improve encoder-side while keeping decoder standards fixed
• Lossy audio encoding drops completely masked components from transmission
• Lossy coding exploits psychoacoustic models to make removed information inaudible
• Loud low-frequency tones mask high frequencies more than high tones mask low.
• Loud masking sounds asymmetrically mask higher frequencies more than lower ones
• Loud noise damage can make the cochlea spontaneously generate its own noise
• Loud speech can mask its own high-frequency components via upward spread of masking
• Loud tones can render simultaneous tones partially or wholly imperceptible
• Louder maskers increasingly mask high frequencies over low ones.
• Loudness grows when sound energy spills beyond a single critical band
• Loudness increases when sound energy spreads across critical bands to fire extra nerves
• Loudness is measured by listeners assigning numbers relative to a reference sound.
• Loudness is measured psycho-acoustically via magnitude estimation against a reference
• Loudness is measured subjectively against 1 kHz reference tones
• Loudness meters use frequency weightings as a compromise because ear response varies with level
• Loudness perception is essentially logarithmic
• Loudness perception is logarithmic, which is why decibel scales fit hearing so well.
• Loudness perception requires a tenfold intensity increase to double
• Loudspeaker array behavior is analyzed as point-source superposition
• Loudspeaker cabinet volume sets the lower frequency limit
• Loudspeaker enclosure shape and size alter frequency response via diffraction
• Loudspeaker enclosures should be curved near drivers and avoid central mounting.
• Loudspeaker enclosures should curve near drivers and offset them vertically
• Loudspeaker makers use engineered wood to avoid directional sound speed
• Loudspeaker position determines which room modes get excited, as with bowed strings
• Loudspeakers only beam sound once their radiating part exceeds the wavelength.
• Low absorber rooms raise overall sound level because reverberant energy persists longer
• Low absorption levels increase reverberant field level by sustaining sound energy
• Low absorption raises the real critical frequency above the mean-free-path estimate.
• Low absorption sustains strong modal variation in rooms at higher frequencies than predicted
• Low adjacent harmony voices need wide spacing to avoid a muddy sound
• Low frequencies cannot form diffuse fields because boundaries create standing waves
• Low frequencies radiate omnidirectionally while high frequencies become directional
• Low-frequency sound diffusion requires impractically deep diffuser wells
• Low-frequency transport noise masks speech formants and cripples intelligibility
• Low-frequency transport noise masks the vowels that carry speech meaning
• Low-level tones mask lower frequencies more than higher ones, reversing at high levels
• Lower frequencies diffract more around obstacles than higher frequencies do
• Lower temperature slows the speed of sound predictably via thermal molecular motion
• Lowering ambient noise lets wanted sound play quieter and distort less.
• m-sequences offer flat Fourier properties suited to diffusion design
• Making a code comma free removes the need for separator bits
• Mallet hardness and strike location shape which drum modes get excited
• Mallet hardness and strike position shape a drum’s brightness
• Many listeners prefer passive noise cancellation because it leaves the wanted sound unmodified
• Masked audio components below the masking threshold need no bits transmitted
• Masked sound components are perceptually ignored within complex sounds
• Masked spectral components need no bits because they are inaudible
• Masking asymmetry grows as masker amplitude increases, favoring upward masking
• Masking is quantified by the shift in a listener’s hearing-threshold curve
• Masking limits how much combination tones interfere with other instruments’ pitches
• Masking noises that mimic speech’s rhythm disrupt intelligibility more than loudness predicts
• Masking reduces how prominently audible combination-tone beating is
• Materials with movable boundaries support wave types beyond longitudinal waves.
• Maternal rubella can cause profound hearing loss in a fetus
• Mathematically sequenced well patterns were developed to produce well-defined diffusion in rooms.
• Matrixing encoded multichannel audio onto two-channel media at the cost of channel separation
• Maximal-information audio signals have flat spectra across all frequencies
• Measured fundamental frequency per cycle drifts with tuning, intonation, and vibrato
• Measured instrument spectra blend source harmonics with resonance modifiers
• Measuring noise exposure over time requires Leq, not standard meter settings
• Membrane absorbers deliver wide-band absorption in only 30 cm of depth.
• Microphone directivity in impulse response recording controls perceived reverberation amount
• Microphones become more directional at higher frequencies, so off-axis reverberant pickup tilts dark.
• Microphones grow more directional at higher frequencies, dulling captured reverberant sound.
• Middle-ear muscles automatically stiffen to block loud sounds
• Minimum speaker count maximizes intelligibility; directivity can substitute for fewer sources
• Mirror-image rooms let engineers locate exactly which wall patches to treat
• Mismatched drum-head tensions deliberately destroy definite pitch in bass and snare drums
• Missing-fundamental illusions depend on a musical context being established first
• Mixed sound absorptions create multi-slope reverberant decay curves
• Mixing intensity and delay stereo cues produces spurious cues that confuse the ear
• Mixing noise with its own delayed copy can create a sense of pitch
• Modal absorption is frequency dependent, stretching decay time at standing-wave frequencies
• Modal bandwidths break diffuse-field assumptions when sound paths exceed the mean free path
• Modal behavior intensifies as wavelength approaches the room’s mean free path
• Modal behavior makes diffuse-field analysis fail in a mid frequency range
• Modal decay analysis governs room acoustics below the critical frequency
• Modal decay differs from diffuse decay because reflection paths are fixed, not random.
• Modal density rising smoothly with frequency predicts perceptually smooth room response
• Modal strength depends on reflection count, not just mode type
• Modern acoustic diffusion uses mathematically structured well patterns
• Molecular inertia makes air particle velocity change smoothly and continuously
• Molecular inertia makes sound particle velocity trace a cosine, smoothly not abruptly
• Monotonic mode growth predicts perceptually smooth room response
• Most acoustics texts attribute trumpet brilliance to treble boost
• Most instruments occupy only a portion of the audible frequency range
• Mounting a driver off-center reduces cabinet diffraction effects.
• Moving away from a source makes reverberant sound progressively dominate while it stays constant
• Moving sources make reflected-sound comb-filtering audible as pitch rather than mere colouration
• MP3 compression reaches an 11-fold reduction unattainable by lossless means
• Multichannel speaker designations follow a full-range plus sub count convention
• Multidimensional scaling maps perceived timbre differences between instruments
• Multiple open holes jointly shift the effective pressure node
• Multiple subwoofers cancel the room modes caused by one subwoofer.
• Multiplying element and array patterns can eliminate spatial aliasing without a physical line source
• Musical illusions only work when the listener’s expectations have been pre-established
• Musical instrument notes have a gradual onset before steady state
• Musical intervals between adjacent harmonics shrink as harmonic number rises
• Musical intervals trace directly to harmonic-series frequency ratios
• Musical notes have distinct onset, steady-state, and offset acoustic phases.
• Musical taste and harmonic conventions evolve as innovations become tradition
• Musicians describe sounds using pitch, loudness, and timbre descriptors
• Musicians need acoustic reflections within 20 ms for comfortable ensemble playing
• Musicians need sound reflections within twenty milliseconds to play together.
• Musicians rely on room reverberation for perceived output lift and feel uncomfortable without it.
• Mutation stops reinforce harmonics not aligned with the octave series
• Naive raters in sufficient numbers yield more reliable subjective quality judgments than experts.
• Narrow frequency bands allow removal of correlation and silence flagging
• Narrow-bandwidth equalization causes audible ringing in both boost and cut modes.
• Narrow-bandwidth equalization causes filters to ring audibly in time
• Narrower filter bandwidths lengthen time responses and create audible artifacts
• Narrower frequency bands make audio signals less correlated and easier to quantize
• Narrower signal bandwidth necessarily spreads signals over longer time spans
• Narrowing a filter’s bandwidth inevitably spreads the signal in time, harming timbre
• Narrowing a filter’s bandwidth spreads the signal in time, worst near the cut-off.
• Narrowing a horn’s coverage angle concentrates energy and raises its on-axis sensitivity.
• Narrowing constrictions self-amplify until the lips slam shut and cut airflow
• Natural complex sounds make level changes easier to hear than pure tones do.
• Natural-sounding speech can still be poorly intelligible
• Nearby walls boost loudspeaker bass output, and designers counter this in many ways.
• Negative SPL readings indicate levels below the reference threshold.
• Nerve firing intervals are integer multiples of the stimulus period
• Neural signals encode auditory information through spike timing patterns
• Neurons firing irregularly across harmonics reveal the fundamental period
• New information in a signal equals whatever cannot be predicted from its past
• Nicking the pipe edges smooths onset and suppresses high-frequency components
• No filter can have both fast time response and narrow bandwidth simultaneously.
• No single frequency-weighting can accurately meter loudness at all levels
• Noise above 85-90 dBA causes permanent hearing damage
• Noise and reverberation both mask consonants, which are about 20 dB quieter than vowels
• Noise damage accumulates as an integrated dose where loudness trades against duration
• Noise exposure is measured as an integrated dose, trading level against duration
• Noise exposure permits trading louder sounds for shorter duration within a fixed dose
• Noise masks consonants because they are about 20 dB quieter than vowels, harming intelligibility.
• Noise shaping exploits hearing sensitivity to boost effective CD signal quality
• Noise shaping exploits our most sensitive frequency range to raise CD signal-to-noise ratios.
• Noise that varies at speech-like rates hijacks auditory attention
• Noise-based pitch perception arises from spectral energy center of gravity but is weak.
• Noise-based sounds like ss and sh can carry pitch despite nonperiodic waveforms.
• Noise-derived pitch sensation comes from spectral energy centroid location.
• Noise-induced hearing loss first appears near 4 kHz as an audiometric notch
• Noise-like sounds have non-repeating waveforms and continuous, inharmonic spectra
• Non-environment listening rooms isolate direct sound to reveal low-level audio detail
• Non-environment rooms kill the reverberant field while keeping early reflections for occupants
• Non-environment rooms suppress all but direct sound for clearer audio detail and imaging
• Non-periodic continuous-spectrum sounds like fricatives can still evoke pitch sensations
• Non-simultaneous masking lets sounds mask others at temporal offsets, not only together
• Non-uniform quantization reduces audible noise by spacing levels by probability.
• Non-uniform quantizer spacing exploits masking and signal statistics to cut noise
• Nonharmonic mode relationships make struck bars and membranes sound clangy.
• Nonharmonic natural modes give struck percussion instruments their clangy identity
• Nonperiodic, one-off signals require analysis tools beyond the Fourier series.
• Nonsense-word tests remove top-down language repair, giving an unbiased intelligibility measure
• Nonuniform quantizers reduce perceived noise by spacing levels of rare louder signals farther apart
• Normal bone conduction with poor air conduction localizes damage outside the inner ear
• Note onsets are composed of many separable acoustic cues
• Note onsets have staggered harmonic entry times that define each instrument’s timbre
• Note-onset timbre differs measurably across instruments and lasts tens to 160 milliseconds.
• Objective sound amplitude measures do not describe perceived loudness.
• Objects must span a radian of phase shift before meaningfully interacting with sound.
• Objects near two-thirds wavelength produce interference because reflection and diffraction combine
• Oblique room modes depend on all three dimensions
• Oboe notes begin on the second harmonic before the fundamental arrives
• Octave stretching on pianos emerges directly from string inharmonicity
• Octave-band averaging can blur perceptually noticeable absorption variations
• Octave-spaced pure tones can demonstrate individual high-frequency hearing limits
• Odd-numbered partials alone make a tone sound hollow or nasal
• Of four brass instruments, only the trumpet shows prominent high harmonics
• Ohm’s acoustical law held that a pitch requires physical power at that frequency
• One decibel approximates the smallest loudness change humans can perceive
• One tone can make another tone inaudible through masking
• Only a few organs worldwide have 64-foot sub-bass stops
• Only about 1,400 discriminable pitches exist across the entire audible range
• Only bores keeping fixed mode ratios as holes open are musically useful
• Only exact integer-fraction plucking positions completely suppress harmonic modes
• Only harmonics below the fifth to seventh are individually resolved in timbre perception
• Only harmonics separated by more than the ear’s critical bandwidth are resolved
• Only some combination tones are readily perceived, and none by all listeners.
• Only the first five to seven harmonics are resolved regardless of fundamental
• Only two classical theories, place and temporal, explain pitch tracking.
• Onset attack mechanisms are vital acoustic cues to instrument identity
• Open and closed pipe ends reflect pressure pulses with inverted phase
• Open and stopped organ pipes audibly start on different overblown harmonics
• Open and stopped organ pipes correspond to open-open and open-closed acoustic tubes.
• Open pipe ends invert compression pulses while closed ends preserve them
• Open pipe ends invert pulse polarity while closed ends preserve it
• Open-open pipes resonate at the same half-wavelength frequencies as closed pipes.
• Open-open tubes support standing waves with the same half-wavelength rule
• Operatic vibrato modulates pitch at 5.5-7.5 Hz
• Operating a loudspeaker near three room boundaries imposes a 9 dB low-frequency rise
• Operating with three nearby boundaries gives a fixed 9 dB low-frequency rise
• Optimal reverberation and strength targets differ sharply by music type
• Optimal reverberation time depends on the music being performed.
• Orchestral accompaniment masks the voice’s fine spectral detail in recordings
• Orchestral seating aligns with a pitch-to-side perceptual bias in listeners.
• Orchestral seating rearrangements change the stereo image listeners actually hear.
• Organ builder breaks one mixture stop’s ranks across successive octave ranges
• Organ builders exploit a pitch illusion to extend sub-bass without actual low pipes
• Organ builders exploit missing-fundamental perception to create bass notes no pipe actually sounds
• Organ flue pipes divide into open-open and open-closed acoustic types
• Organ pipe resonator shape follows construction convenience more than acoustics
• Organ pipes begin notes one brief harmonic above their sustained pitch
• Organ pipes exemplify both reed and flue wind-instrument sound sources
• Organ reeds self-oscillate through repeated contact-with-shallot cycling.
• Organ resultant stops create the sensation of pitches below any note actually sounding
• Organ stop footage inversely relates to the harmonic number it reinforces.
• Organists shape timbre by selectively reinforcing harmonics of one series
• Overblowing a pipe jumps its pitch to the next resonator-supported harmonic
• Overblowing to higher modes lets players reuse the same fingerings to climb registers
• Overlapping three adjacent mode bandwidths at one frequency point dramatically reduces sound variation
• Overly loud speech masks its own higher frequencies and gets less intelligible
• Overtone numbering is offset one step behind harmonic numbering
• Overtone singers employ very high harmonics, around the twentieth to twenty-fifth.
• Overtone singing needs an extremely narrow, movable vocal-tract resonance over a single harmonic.
• Overtones count from the first frequency above f0, offset by one from harmonic numbering.
• Panel absorbers convert sound into internal friction and respond to pressure
• Panel stiffness grows with thickness, so simple mass-law absorption models fail on thick panels.
• Panning between two speakers recreates a phantom source positioned by interaural phase shift
• Parallel smooth surfaces generate rapid flutter echoes as energy bounces between them
• Partials become harmonics only when integer-related to the fundamental
• Partition mass provides isolation at higher frequencies until coincident resonances degrade it
• Partition sound attenuation rises with frequency except at coincident resonances, which damping reduces
• Passive noise cancellation avoids contaminating the wanted signal that active cancellation risks
• Path-length differences between correlated sources create spatial interference zones with silent spots.
• Pedal-operated kettledrums produce glissandos when retuned mid-vibration.
• Perceived loudness does not map one-to-one onto acoustic amplitude
• Perceived loudness doubles with every 10 dB increase, not every 3 dB of intensity.
• Perceived loudness meters must weight frequencies as a compromise across sound levels.
• Perceived loudness of complex sounds depends on what the listener attends to.
• Perceived musical pitch arises from regularly repeating acoustic pressure waves.
• Perceived pitch can arise from harmonics alone even when no actual fundamental sounds
• Perceived pitch intervals match log-frequency ratios regardless of register
• Perceived pitch is governed mainly by fundamental frequency, not loudness or duration
• Perceived pitch is set primarily by fundamental frequency.
• Perceived reverberation depends on early decay, not total T60 measurement
• Perceived roughness to smoothness shifts exactly at the critical bandwidth boundary
• Perceived warmth has two independent routes and boost near 3 kHz projects a voice
• Perceiving time-varying data requires sound dimensions in which change is readily heard.
• Percentage of lost consonants maps directly to intelligibility quality thresholds
• Perception completes masked vocal phrases using semantic context to make them continuous
• Perception of two tones progresses from beats to roughness to smoothness
• Perceptual adaptation to timbre dulls equalizer effects and drives endless compensatory boosts.
• Perceptual codecs spend bits only on audible components, discarding masked ones
• Perceptual coders exploit masking to cut audio storage and bandwidth
• Perceptual coding achieves compression by discarding only subjectively inaudible information
• Perceptual coding works by maximizing the surprise value of coded signals
• Perceptual frequency scales are quasi-logarithmic and straighten masking thresholds
• Perceptual spectrographic modeling extends from instrument timbre into speech, forensic audio, and singing.
• Perceptual thresholds vary by listener but share common ordering
• Perceptually discarded information can never be recovered by decoding
• Percussion excitation is a short impulse rather than a sustained tone
• Percussion instruments divide into bar, membrane, and plate classes
• Perfect pattern control across the audio range is physically infeasible due to aliasing cost
• Performance-space reverberant fields distort recordings with uneven frequency response
• Performers need foldback to hear themselves while performing.
• Periodic sounds give line spectra while aperiodic sounds give continuous spectra
• Periodic spectra show discrete partials spaced inversely to period
• Periodicity in a sound’s waveform is what makes it perceptible as having pitch.
• Periodicity in the acoustic waveform determines whether a sound is perceived as pitched.
• Phantom tones are heard despite being absent from the presented sound
• Phase locking fails above 5 kHz, so high-pitch perception relies on place theory
• Phase velocity and group velocity differ in dispersive media
• Phase velocity measures single-frequency speed while group velocity tracks the signal envelope
• Phase-based localization tops out at 743 Hz at 90 degrees
• Phase-based sound localization becomes ambiguous beyond 180 degrees phase shift
• Physical room constraints can make ideal reverberation targets unattainable
• Piano hammer bounce-back prevents it from damping string vibration
• Piano octave stretching grows with pitch as strings shorten
• Piano octaves are stretched because lower notes’ overtones are inharmonic
• Pilot tests and controlled constants eliminate confounding in listening experiments
• Pinna and concha resonance determine front-behind sound localization
• Pinna reflections form comb filters encoding sound direction in three dimensions
• Pinna shape creates learned, person-specific comb filters that aid sound localization
• Pipe bore constriction shifts mode frequencies depending on standing-wave node position
• Pipe constrictions shift mode frequencies up or down depending on node position
• Pipe length sets pitch while diameter and material shape harmonic strength
• Pipe organ stops can reinforce non-octave harmonics to reshape timbre
• Pipe organ stops reinforcing non-octave, non-unison harmonics are called mutation stops
• Pipe organs function as additive synthesizers by combining harmonics through stops.
• Piriform sinuses add spectral zeroes that boost the singer’s formant
• Pitch can be perceived at fundamental frequencies entirely below the hearing range.
• Pitch change depends only on fold tension and vibrating mass adjustments.
• Pitch discriminability is judged by binomial-significant forced-choice correctness
• Pitch extraction methods still derive a correct f0 from incomplete harmonic series
• Pitch is perceived from firing intervals even when nerves skip cycles
• Pitch measurement inherently requires subjective listener judgments
• Pitch perception combines place-based and timing-based neural analysis of complex tones
• Pitch perception depends on harmonics being integer multiples of the fundamental
• Pitch perception derives from the highest common factor of sound components
• Pitch perception is encoded by the timing of neural firing in the cochlea
• Pitch perception is explained by two competing physiological models of the inner ear
• Pitch perception rests on universal psychoacoustic mechanisms across all cultures
• Pitch, though subjective, is measurable in Hz by matching against a sine wave
• Place and timing theories fail to explain some complex pitch perception phenomena
• Place theory cannot explain pitch for sounds composed only of components below 50 Hz
• Place theory cannot fully explain how pitch is extracted from components
• Place theory claims pitch is encoded by basilar membrane position
• Plucked-string spectra reveal the plucking point through nulled harmonics
• Plucking a string at a node suppresses every Nth harmonic
• Plucking position on a string determines which harmonic modes are suppressed
• Plucking position on a string suppresses modes with nodes there
• Poisson ratio links a material’s shear and Young’s moduli via lateral strain
• Poorly coupled added volume produces a double-slope reverberation fault.
• Porous absorbers damp sound through velocity-induced frictional losses
• Porous absorbers only work where sound wave velocity is nonzero
• Porous and resonant absorbers combine to achieve broad frequency coverage.
• Porous materials absorb sound because their large surface area creates frictional losses.
• Porous sound absorbers work best when positioned at a quarter wavelength from a wall
• Positive feedback drives the reed to slam shut
• Practical audio devices rest on underlying acoustics principles
• Practical filters attenuate over a finite frequency range rather than cutting off instantly.
• Practical wavefield arrays need shaped layouts to reduce boundary artifacts
• Pre-echo artifacts from digital filtering are easily unmasked and subjectively disturbing.
• Predictable pure tones bore the ear while spectrally rich, unpredictable signals hold interest.
• Predictable signal components carry no information and can be recomputed rather than stored
• Predictable sounds carry less information than unpredictable broadband signals
• Presbycusis differs between men and women as hearing ages
• Presbycusis lets devices weaponize high frequencies against only young ears
• Presbycusis raises the audible high-frequency floor with age, so tones vanish for adults
• Presentation method, room acoustics, and listener state are independent variables in listening experiments
• Presented together, tones reveal far finer pitch differences via beats than sequential comparison
• Pressure dynamics make vocal folds close faster than they open
• Pressure feedback drives the reed into full closure across each cycle
• Pressure is scalar while velocity gives a sound wave its direction
• Prevailing nineteenth-century views treated the ear as receiving motion souvenirs from sound sources
• Prevailing wind direction should guide where one lives relative to noise sources
• Primitive root sequences provide an alternative periodic well-depth pattern for diffusers.
• Projecting a spherical wavefront onto a flat horn mouth produces a cosine roll-off.
• Psychoacoustic data are collected listener opinions, whereas acoustic data are physical measurements.
• Psychoacoustic experiments on listeners require controlled sound levels to prevent hearing damage.
• Psychoacoustic findings are experimentally established yet often lack anatomical or physical explanation.
• Psychoacoustic findings often lack known anatomical or physical mechanisms
• Psychoacoustic knowledge rests entirely on subjective listening tests, not direct measurement
• Psychoacoustic models evolve because encoders are unconstrained by decoders
• Psychoacoustic models govern which audio information can be discarded unnoticed
• Psychoacoustic noise shaping achieves an effective 120 dB SNR on CD audio
• Psychoacoustic noise shaping boosts perceived CD signal-to-noise beyond actual bit depth
• Psychoacoustic quantization is the only stage in audio coding that removes signal information
• Psychoacoustic testing measures subjective listener opinions under controlled independent variables
• Psychoacoustic theories are tested by constructing synthetic sounds that confound current explanations
• Psychoacousticians test theories by deliberately creating sounds that current theories cannot explain.
• Psychoacoustics developed in parallel, not from, physiology and was shaped by communication engineering
• Pure ratio intervals on three chords generate the entire just diatonic scale
• Pure tones alone cannot sustain musical interest
• Pure tones are the standard audiometric stimulus because they are frequency-specific.
• Pure tones sound soft and pleasant but lack power
• Pure-tone audiometry measures detection thresholds, not the ability to discriminate sounds
• Pure-tone instruments readily produce audible combination tones
• Pure-toned high instruments like piccolo make combination tones easy to perceive.
• Pythagorean scales are constructed by stacking perfect fifths
• Quadratic residue number sequences generate periodic well-depth patterns for acoustic diffusers.
• Quadratic residue sequences generate diffusor well depths using modular arithmetic
• Quantization always injects noise that lowers audio signal-to-noise ratio
• Quantization converts continuous signals to discrete levels and is lossy
• Quantization error manifests as added noise that reduces the signal-to-noise ratio
• Quantization error manifests as audible noise, which dither is used to randomize
• Quantization is inherently lossy because it discards information between input and output
• Quantizer levels need not be powers of two when entropy coding is used
• Raising a directive horn reduces front-to-back path variation but loses low frequencies.
• Raising a sound’s loudness shifts its perceived pitch except near 1-2 kHz.
• Raising airflow or lowering cut-up makes flute pitch jump discretely
• Raising airflow or lowering the cut-up makes flue pitch jump discontinuously
• Raising frequency narrows an array’s main lobe because the visible region expands
• Random noise and unpredictable spikes carry maximum information due to flat spectra.
• Rapid airflow cutoffs generate harmonic-rich tones in reed instruments
• Rapid changes in a sound’s amplitude envelope force broader spectral bandwidth
• Real audible sound pressures span a range of one million to one in amplitude
• Real audio signals cluster toward small amplitudes, not the uniform statistics quantizers assume.
• Real audio signals leave many coding levels unused and waste bits.
• Real audio signals spend more time at low amplitudes than high amplitudes
• Real musical f0 measurements drift cycle to cycle and need averaging
• Real musical instruments produce measurable distortion tones that fall near other scale notes
• Real piano strings force a stretched tuning to avoid audible beats
• Real string stiffness makes higher partials travel faster than the ideal thin-string model predicts
• Real string stiffness slightly raises wave velocity with frequency, affecting timbre.
• Real strings produce inharmonic partials whose deviation grows steeply with thickness
• Real walls operate in the mass-controlled region where performance rises 3 dB per octave
• Recording methods and cues for stereo are chosen independently of the listening arrangement.
• Recovery from a large noise dose takes days to weeks
• Rectangular loudspeaker cabinets produce significant diffraction even with off-center drivers
• Rectangular shoebox halls naturally produce abundant lateral reflections prized for sound quality.
• Redirecting reflections creates uneven hot areas, so absorption is often chosen.
• Redirecting reflections creates uneven hot spots while absorption shortens reverberation
• Reducing audio data rate is the precondition for distributing audio over limited channels
• Reducing background noise improves both safety and perceived sound quality
• Redundancy removal effectiveness trades off against the resources needed to encode side information
• Reed excitation injects all harmonics even when resonator rejects even modes
• Reed instruments can use resonators shorter than full pipe length
• Reed instruments need no resonator reinforcement of their fundamental frequency
• Reed instruments show fundamental frequency rising during note onset
• Reed material and support determine whether tone is focused or flexible.
• Reed oscillation is a damped spring cycle renewed by airflow
• Reed oscillation is self-sustaining through a positive feedback loop of airflow and pressure
• Reed shutoff produces sharp acoustic pressure variations that drive the resonator
• Reed vibration is self-sustaining spring oscillation against a shallot
• Reed-driven woodwinds act as stopped pipes while flute-type reeds act as open pipes.
• Reflected sound intensity is reduced in proportion to the absorption coefficient
• Reflected sound obeys an inverse square law in addition to surface absorption losses
• Reflecting boundaries create image sources that alter loudspeaker load impedance.
• Reflection from a bumpy surface depends on bump height relative to wavelength.
• Reflection from rough surfaces smooths detail once wavelengths exceed roughness
• Reflection intensity bounds can be graphed from delay and distance alone
• Reflection path length can be computed geometrically using the image-source method.
• Reflection phase at a boundary depends on whether acoustic impedance rises or falls
• Reflections arriving later than about 20 ms are heard as a distinct echo
• Reflections arriving near the 50 ms echo threshold can severely disrupt performers
• Reflections delayed beyond about 30 milliseconds are perceived as distinct echoes.
• Register holes exploit mode geometry to suppress one mode while sparing others.
• Register holes force a jump to a higher mode by removing harmonic support.
• Register holes suppress unwanted modes by exploiting pressure nodes and maxima
• Regularly spaced surface bumps diffuse sound only at specific frequencies
• Relative loudspeaker amplitudes shift the perceived direction of a phantom image.
• Relative phase and path distance determine whether correlated waves cancel or reinforce in space.
• Removing a sound’s fundamental leaves its perceived pitch unchanged
• Removing conflicting reflections improves perception of low-level audio detail
• Removing inter-sample correlation itself discards no information and stays lossless
• Removing redundancy in a signal raises its information density
• Removing signal redundancy trades effectiveness against available encoding resources.
• Removing the fundamental frequency does not change perceived pitch.
• Replacing average absorption with summed open-window area handles mixed materials.
• Reported sound levels almost always refer to pressure, matching human hearing
• Reshaping a room’s walls only obscures resonances rather than removing them
• Resonance placement differences explain each string instrument’s characteristic tone
• Resonant panel absorber formulas lose accuracy for thick panels because stiffness is ignored
• Resonator geometry shapes which harmonics appear in an instrument’s sound
• Resonator shape determines which harmonics a wind instrument emphasizes
• Reverberant field dominates home hi-fi listening from just 2 meters away
• Reverberant fields help performers produce louder output, so musicians favor live rooms
• Reverberant fields sound bass-heavy and require context-specific equalization
• Reverberant fields supply most perceived loudness in normal rooms
• Reverberant fields typically color perceived sound with excess bass that equalization must fix
• Reverberant sound is assumed to strike all surfaces with equal probability in modeling.
• Reverberant sound level is position-independent while direct sound varies with position
• Reverberant sound level rises as room size shrinks while decay time shortens
• Reverberant sound level stays constant with listener position because reflections arrive from all directions
• Reverberant sound level stays constant with listener position while direct sound fades with distance
• Reverberant sound stays constant across a room while direct sound falls with distance
• Reverberant-field level follows directly from the source’s frequency-dependent radiated power.
• Reverberation calculations break down for tunnel-like room shapes.
• Reverberation can be tuned by budgeting equivalent open window area.
• Reverberation decay should be a smooth exponential of constant dB per unit time
• Reverberation equations fail in extreme geometries where surfaces are visited unequally
• Reverberation evenness across frequencies is controlled by adding or removing absorbing materials
• Reverberation gives performers a perceived output lift, so they prefer live rooms
• Reverberation time depends on both surface absorption and time between reflections
• Reverberation time depends on room volume and surface area, not absorption alone.
• Reverberation time equations must incorporate frequency-dependent absorption coefficients
• Reverberation time formulas rest on statistical assumptions about sound fields
• Reverberation time spans from 0.2 seconds to 10 seconds across rooms
• Reverberation times range from fractions of a second to ten seconds
• Reversing stereo channels or turning around can replicate hearing music from behind the orchestra
• Rhyme tests pinpoint exactly which consonant distinctions a speech system fails.
• Rhyming word lists isolate consonant confusion as the measure of speech intelligibility
• Rich, sweet timbre requires present-but-modest partials up to about the sixth
• Right-ear treble preference may explain traditional orchestral seating arrangements.
• Room absorption and source directivity shift the critical distance only by their square roots
• Room absorption determines the steady-state level of reverberant sound
• Room acoustic design must begin with speaker placement decisions
• Room acoustics act as a second system cascaded after the instrument
• Room acoustics act as another system modifying the instrument’s output
• Room acoustics are tuned by budgeting equivalent open window absorption area
• Room acoustics, background noise, fatigue, and stimulus order all bias listening results.
• Room axial mode frequencies are set by the distance between opposing surfaces
• Room boundaries alter a sound source’s radiated power, not just its perceived sound
• Room boundaries alter loudspeaker output through reflected image sources
• Room materials’ absorption coefficient varies with frequency
• Room mode density rises with frequency squared, so uneven mode counts create audible peaks
• Room modes are determined by the linear dimensions of opposing surfaces
• Room modes produce multiple decay times instead of a single exponential reverberation
• Room resonances occur at frequencies where whole numbers of wavelengths fit between surfaces
• Room reverberant fields typically rise in bass, necessitating bandwidth-matched equalizer cuts for distant miking
• Room reverberation equilibrium mirrors a leaky bucket’s steady-state water level
• Room reverberation time can be tuned by budgeting open-window-equivalent absorption
• Room reverberation time varies with frequency, altering perceived timbre
• Room size and absorption both govern how long reverberant sound takes to build up.
• Room size and source directivity alone determine where direct sound gives way to reverberation.
• Room size in acoustics is defined by critical frequency relative to generated sound
• Room sound energy decays exponentially through repeated surface absorptions
• Room standing waves cause sound pressure to vary with listening position
• Sabine defined absorption in units of equivalent open window area.
• Sabine defined total room absorption as equivalent open-window area.
• Sabine measured absorption as equivalent open-window area, verified experimentally
• Sabine measured sound absorption by comparing a material to a fully open window.
• Sabine’s 1898 equation launched scientific room-acoustics design.
• Sabine’s 1898 formula let acousticians predict a room’s reverberation time, reshaping auditorium design.
• Sabine’s 1898 formula made reverberation time the dominant auditorium design parameter
• Sabine’s formula launched the science of architectural acoustics
• Sabine’s reverberation formula founded architectural acoustics but fails for high absorption
• Scale-factor precision trades quantizer utilization against side-information bit cost
• Scholes held that overtones alone condition timbre
• Scientific pitch notation numbers change between B and C, not across octaves symmetrically.
• Sealing the ear canal blocks background noise before it reaches the ear
• Selectively cutting maskers and boosting masked content unmask a sound within a mix.
• Sequences whose autocovariance approximates a delta function make effective sound diffusers
• Setting electronic delay about 15 ms beyond the acoustic delay yields two acoustic advantages.
• Severe audio degradation flips the measured variable from annoyance to word accuracy
• Severe distortion shifts the perceptual measure from annoyance to word intelligibility
• Shape-dependent spring constants make vibration speeds complex in solid objects
• Sharper waveform corners require more high-frequency components
• Shear forces make one part of a material slip past another
• Shear waves resist deformation via stiffness rather than tension
• Shear waves stay non-dispersive while bending waves disperse with frequency
• Shearing applies force at right angles making material parts slip
• Shepard demonstrated a pitch illusion likened to Escher’s endless staircase
• Shepard tones shift harmonics within a fixed envelope to create circular pitch
• Shepard-tone ambiguity arises in real music such as Bach’s organ works
• Shifting a source spatially adds a progressive phase change to its polar pattern
• Shifting resonance ratios as holes open would wreck woodwind tuning
• Shoebox-shaped Victorian halls naturally produced excellent acoustics
• Signal delays can make flat arrays behave like curved ones.
• Signal delays can substitute for curved physical geometry in shaping wavefronts
• SIL and SPL always change in proportion even where their absolute values differ
• Simple acoustic rules only work when applied with context-specific analysis and design.
• Simplified head approximations localise sound better than detailed individual head models.
• Simulation refractory periods do not capture real auditory phase-locking constraints
• Simultaneously presented tones sharpen pitch discrimination down to beat disappearance
• Sine wave inputs achieve 1.76 dB better quantization SNR than uniform signals
• Sine waves are fully described by amplitude, frequency, and phase
• Sine waves are the simplest periodic excitation, and all complex waveforms decompose into them
• Sine waves of varying frequency, amplitude, and phase can construct any waveform
• Singers change vocal pitch only via fold stiffness and vibrating mass
• Singers unconsciously resist tuning drift, with sopranos and altos anchoring pitch flatward.
• Singing in just intonation causes pitch drift when modulating through keys
• Singing is humanity’s most versatile instrument and a real research field
• Singing is modeled as vocal-fold excitation filtered by a time-varying vocal tract
• Singing just-tuned chords predicts rapid pitch drift flat
• Singing the target sound aids electronic synthesis approximation
• Singing voice acoustics is now a legitimate research field
• Single-speaker directivity cannot produce even coverage across an audience
• Sitting near walls creates repetition-pitch coloration in perceived sound
• Sixty degrees total between speakers is the recommended angle for ideal stereophonic listening.
• Slightly detuned strings produce a richer, slower-decaying sound while large detuning sounds bad
• Slightly mistuned piano strings produce richer, longer-lasting sound
• Slow and fast meter responses trade averaging for capturing rapid level changes
• Small horn mouths fail to provide advertised directivity at speech frequencies
• Small loudspeakers radiate omnidirectionally because they are small relative to wavelength
• Small rooms alter source impedance and reduce sound radiation at low frequencies
• Small singing bowls can train singers’ voice projection via formant components
• Smaller listening areas require less acoustic absorption treatment
• Smaller rooms produce louder reverberant fields but shorter reverberation times
• Smooth curved cabinet surfaces optimize frequency response over faceted ones.
• Snare wires add their rattle by vibrating against the lower head when struck
• Snell’s law governs how sound bends across temperature boundaries.
• Solid obstacles act as anti-openings, diffracting waves around themselves the same way openings do.
• Solo string lines split into perceived parts via frequency proximity grouping
• Some digital filters create audible artifacts that precede the signal
• Sonification design must match data resolution to minimum audible differences
• Sonification design must match data variation to the minimum audible change.
• Sonification either speeds data into hearing range or maps data to sound features.
• Sonification maps measured data onto audible changes exploiting auditory perception
• Sonification maps sensor data onto sound synthesis to convey information audibly.
• Sonification renders time-varying data through pitch and timbre changes
• Sound absorption and sound isolation are confounded but behave inversely in materials
• Sound absorption coefficients vary dramatically by surface material and frequency
• Sound amplitude is measured as pressure in pascals because ears detect pressure
• Sound amplitude is measured as root mean square pressure because ears sense pressure
• Sound becomes mechanical movement via the eardrum and ossicle chain
• Sound bends around objects small versus wavelength and reflects off large ones
• Sound bends around obstacles because pressure changes cannot end abruptly at shadow boundaries
• Sound bends around obstacles through a process called diffraction
• Sound bends like light because temperature changes its speed.
• Sound decays exponentially on reflection, but cyclic paths create standing waves
• Sound diffraction arises from gradual pressure decay, not abrupt wave cutoff
• Sound diffraction strength depends on wavelength relative to opening size
• Sound diffracts strongly only when wavelengths exceed opening size
• Sound fields are symmetric between source and receiver when the medium is still
• Sound intensity and power can each be expressed in decibels
• Sound intensity and pressure levels match except where reflections alter arriving waves
• Sound intensity changes perceived pitch of pure tones except near 1–2 kHz
• Sound intensity falls off as distance squares only in free field, boundary-free conditions
• Sound intensity falls with distance due to energy spreading, not absorption
• Sound intensity is energy flow per unit area, spanning a trillion-to-one range
• Sound intensity level and sound pressure level are roughly equivalent
• Sound intensity measures energy transfer rate per unit area
• Sound intensity measures power density as watts per unit area
• Sound intensity spans a trillion-fold range, forcing a logarithmic scale
• Sound is localized to the first arriving acoustic source despite delayed reinforcement
• Sound isolation depends on flanking paths more than the barrier itself
• Sound isolation doubles only when partitions are acoustically independent of each other
• Sound isolation fails mainly through flanking paths like pipes and air leaks.
• Sound isolation is defeated mainly by flanking paths like air leaks and structure.
• Sound level convention defaults to pressure level because it is measurable and matches hearing.
• Sound level is commonly taken to mean sound pressure level
• Sound level meters need time averaging to estimate varying pressure levels
• Sound level multiplications become decibel additions in reflections
• Sound moves too fast for heat transfer, so air behaves adiabatically
• Sound must vary over time to hold listener attention
• Sound not only weakens but dulls with distance due to air absorption
• Sound peaks above 130 dB can damage hearing instantly
• Sound power and intensity levels are unaffected by source correlation, unlike pressure level
• Sound power level measures a source independent of surrounding acoustic context
• Sound power level measures total radiated acoustic power referenced to 1 picowatt
• Sound power, unlike intensity, is direction-independent and context-free
• Sound pressure and particle velocity components are always in phase
• Sound pressure is in phase with molecular velocity, not with displacement
• Sound pressure level and intensity level coincide only without reflecting surfaces
• Sound pressure level is the standard because it matches hearing
• Sound propagates through a medium via masses connected by intermolecular forces.
• Sound propagation can be modeled as coupled masses and springs
• Sound propagation depends on interactions with objects and the medium
• Sound pulses pass through each other unchanged via superposition
• Sound reaching the far ear must travel around the head
• Sound refracts because temperature changes its speed in air.
• Sound refracts between air masses of different temperatures just as light does between materials.
• Sound requires a physical medium and is fundamentally a mechanical disturbance
• Sound requires a physical medium to propagate through
• Sound scatters or diffracts around loudspeakers based on wavelength relative to size
• Sound spectrum analysis depends on banks of filters to decompose signals
• Sound speed depends on the medium’s density and springiness
• Sound speed in solids depends only on density and elastic modulus
• Sound speed varies nearly linearly with temperature across normal living conditions
• Sound spreads three-dimensionally rather than traveling on a fixed beam
• Sound travels at a finite speed because medium mass introduces time delays between molecules.
• Sound waves are longitudinal waves of alternating compression and rarefaction
• Sound waves diffracting at speaker cabinet edges distort the frequency response
• Sound waves form standing patterns only when wavelength relates to reflector spacing
• Sound waves reflecting between two boundaries can form standing waves
• Sound waves superimpose linearly because their energy flow is preserved.
• Sounds shorter than 200 ms are perceived quieter, scaled to burst length
• Sounds under 200 milliseconds need more amplitude to be perceived equally loud
• Sounds with prime-numbered partials lack a common fundamental pitch
• Spatial audio reproduction splits into cue-based systems and sound-field reproduction systems
• Spatial audio systems create illusions of direction and space rather than reproducing them exactly
• Spatial audio uses psychoacoustic illusions called phantom images to mimic real sound sources
• Spatially proportional room dimensions create coincident resonant modes
• Spatially static resonant modes cause sound level to vary across a room
• Speaker and instrument directivity at high frequencies alters perceived timbre
• Speaker directivity specifications can be misleadingly marketed without effect on speech intelligibility.
• Speaker drivers’ directivity at high frequencies causes spectral dips in the reverberant field
• Speaker energy aimed at rear walls degrades intelligibility via reflection and interference
• Speaker sensitivity gives SPL at one meter per watt of power input.
• Speaker size dictates the lowest frequency sustaining a given coverage angle
• Speaker size sets the minimum frequency it can beam at a given angle.
• Specially designed rooms can feel anechoic while retaining reverberation and stable stereo imaging.
• Spectra of only odd harmonics produce a clarinet-like timbre
• Spectrogram gray-scale plots show how a sound’s spectrum changes over time
• Spectrograms plot frequency against time to show both at once
• Spectrum components are partials and only integer-related ones are harmonics
• Spectrum magnitude, not phase, drives the perceived timbre of a sound
• Specular early reflection intensity falls off with the square of distance
• Specular reflections produce audible comb filtering and false source images
• Speech intelligibility and naturalness are distinct perceptual dimensions that do not track each other
• Speech intelligibility depends chiefly on direct sound, not the reverberant field.
• Speech intelligibility depends on high direct-to-reverberant sound ratios
• Speech intelligibility indexes average signal-to-noise ratio across critical bands
• Speech needs at least 7.5 dB signal-to-noise ratio for adequate intelligibility
• Speech perception requires both detection and discrimination of sound components
• Speech processors partition sound using ERB bandwidths and calibrated pulse amplitude per electrode.
• Speech quality tests need at least 16 naive potential users for statistical power
• Spherical loudspeaker enclosures require a diameter far exceeding the drive unit size
• Split-key instruments giving separate enharmonics never gained popularity
• Splitting audio into narrow frequency bands removes redundancy and enables efficient perceptual quantization
• Spreading a sound’s energy across more critical bands increases perceived loudness without raising intensity
• Standardized listening rooms match domestic listening enough to validate psychoacoustic tests
• Standardized reference listening rooms let psychoacoustic test results generalize to ordinary domestic settings.
• Standing wave frequencies of a system are called its modes, ranked by order
• Standing wave node positions let one predict bore-change effects on mode frequencies
• Standing waves are a subclass of the more general phenomenon called resonant modes
• Standing waves can form without any physical boundaries via interfering sources.
• Standing waves can form without two bounding reflectors
• Standing waves from room boundaries prevent diffuse sound fields at low frequencies.
• Standing waves in pipes occur only at odd multiples of the lowest frequency
• Standing waves in tubes need specific node-antinode patterns at boundaries
• Standing waves occur at all integer multiples of a fundamental frequency
• Standing waves qualify as resonant modes and need not require two boundaries
• Standing-wave modes produce several decay times, leaving excess energy at modal frequencies
• Standing-wave modes require one boundary per spatial dimension.
• Statistical room-acoustics models fail for elongated geometries where surfaces aren’t uniformly visited
• Steady-state reverberant level is set by the balance between input power and absorption loss
• Stereo directionality was long mistaken for a pure intensity effect
• Stereo loudspeakers localize sound by summing signals to create interaural phase differences
• Stereo reproduction injects room and speaker location cues unrelated to the recording
• Stereo speakers must form a 60-degree equilateral triangle around the listener
• Stiffness-based instruments space their modes wider apart than tension-based ones.
• Stopped flue pipes sound their harmonics in a characteristic staggered order producing a chiff
• Stopped organ pipes sound the same pitch at half the length
• Stopped organ pipes trade space savings for weaker harmonic spectra
• Stopped pipes save space but sacrifice even-numbered harmonics
• Strike placement on a drum excites modes by their amplitude at that point
• Striking a cymbal with different beaters changes its output spectrum.
• Striking a membrane at a nodal line silences that mode entirely
• Striking a timpano then moving its pedal produces a glide effect
• String and pipe resonance differ in transverse versus longitudinal wave motion
• String pitch depends on mass per unit length, length, and tension
• String stiffness raises the frequency of a piano’s higher modes relative to the fundamental.
• String tension directly controls transverse wave speed, enabling instrument tuning
• Stringed instruments are classified by how energy enters their strings
• Stringed instruments radiate low frequencies omnidirectionally but high frequencies directionally
• Strong diffuse reflections can make a room sound anechoic while still reverberant.
• Strong energy above the fifth harmonic makes instruments sound bright
• Struck bars vibrate in non-harmonically related transverse modes
• Study results confirm that attack onset dominates timbre perception
• Sub-harmonics are lower frequencies whose harmonic series also contain all given components.
• Sub-octave organ ranks extend the sounding range far below the written foundation pitch.
• Subjective listening tests pose comparative questions rather than recording physical quantities.
• Subjective perception of sound cannot be predicted simply from its acoustic components
• Subjective speech ratings need at least sixteen listeners for statistical power
• Subjective tests have no right answers because each listener’s perception is unique
• Summed nerve-firing intervals across the basilar membrane reveal the fundamental period
• Superposition lets complex waveforms be solved as sums of sine wave solutions
• Supported standing-wave frequencies in bounded systems occur at odd multiples of the lowest
• Suppressing early reflections in one listening zone widens the perceived time-delay gap
• Suppressing early reflections through absorption or angled walls maximizes the initial time-delay gap.
• Surface absorption can override the usual ranking of room mode strengths
• Surface absorption reduces total sound power, so reflections fall below inverse-square predictions.
• Surface absorption weakens early reflections below inverse-square-law predictions
• Swear-word logatoms must be excluded because taboo words change the talker’s intonation
• Symmetric two-sided scales are needed when processing may improve a signal
• Tangential modes depend on two room dimensions rather than one
• Tangential modes require half-wavelength fits in two dimensions
• Targeted absorber addition flattens reverberation time variation across frequencies
• Temperature layers in the ocean duct sound and enable very long range propagation.
• Temperature layers in water can hide submarines from sonar.
• Temperature refraction makes distant concertgoers hear more noise than nearby ones
• Tempo can be made to loop infinitely via layered sped-up copies with faded amplitudes
• Temporal and place theories explain pitch by different neural and mechanical mechanisms.
• Ten times as many identical sound sources are needed to double perceived loudness
• Tenors avoid formant tuning because their range leaves a higher f0-to-formant ratio.
• Tension directly governs the velocity of transverse string vibrations.
• Testing one ear requires masking the other to isolate it, especially the poorer ear
• The 1970s loudness button wrongly assumed a flat frequency response
• The 60 dB reverberation metric is impractical in real listening conditions
• The A-weighting filter is now standard for all sound levels for consistency
• The acoustic reflex causes a dangerous spiral of turning volume up as hearing dulls
• The acoustic reflex creates a feedback spiral of ever-increasing volume
• The acoustic reflex is too slow to protect against sudden loud sounds
• The acoustic reflex is too slow to shield the ear from gunshots
• The air reed’s fundamental frequency is set solely by resonator length and end condition
• The audible harmonic count of a note depends on hearing’s 16 kHz upper limit
• The audible pressure range spans six orders of magnitude from threshold to pain.
• The auditory system can discard uninformative temporal information during detection
• The auditory system prefers component sets that are adjacent harmonics
• The B♭1 fundamental arises implicitly from F3 and B♭2 harmonics
• The basilar membrane’s mechanics enable the inner ear’s frequency analysis
• The basilar membrane’s tapering shape enables it to sort sounds by frequency.
• The bell acts as a frequency-dependent reflector that shifts brass mode frequencies.
• The bell reflects low frequencies deep inside and high ones near its rim
• The Bernoulli effect couples airflow velocity to reduced pressure at constrictions
• The Bernoulli effect is a wrong explanation for vocal fold vibration
• The bowl’s trapped air shifts kettledrum mode frequencies from ideal values.
• The brain processes sounds inside a critical band as a single lumped item.
• The clarinet lacks even harmonics at low tones but approaches odd-even parity higher up
• The cochlea coils 2.75 turns into three fluid-filled chambers
• The cochlea is a 2.75-turn spiral tube with a leak at its apex
• The cochlea is a three-chambered fluid-filled spiral tube
• The cochlea transduces vibrations into nerve signals at the oval window
• The cochlea’s logarithmic place analysis explains why musical intervals sound the same across registers.
• The cochlear filter response is an exact mirror of the basilar membrane displacement curve
• The concha, acting as a resonant cavity, enables front-behind localization
• The critical band concept is the key to understanding perceived pitch, loudness, and timbre
• The decibel multiplier calibrates integer changes to the ear’s just-noticeable differences
• The decibel scale derives from the smallest perceivable change in loudness.
• The decibel unit is calibrated to the smallest perceivable loudness change
• The diffuse field assumption underpins room acoustics predictions but fails at low frequencies
• The direct sound carries speech clarity, unlike reflected sound.
• The direct-to-reflection gap must stay under 30ms or become audible echoes.
• The ear analyzes complex sounds by superposition of individual frequency responses
• The ear and brain adapt to sounds, dulling perceived changes over time.
• The ear can perceive ILDs even where they never occur naturally.
• The ear decomposes complex sounds into summed place responses
• The ear functions as a bank of frequency-selective filters tuned to basilar membrane places
• The ear groups alternating notes into separate streams based on timbre differences.
• The ear groups sound into critical frequency bands that shape loudness perception
• The ear is far less sensitive to phase than to amplitude of frequency components
• The ear resolves low-frequency direction by measuring phase shift
• The ear uses spectral shape to judge timbre, so nearby frequencies can compensate cuts.
• The ear’s amplitude sensitivity varies with loudness and tonal complexity
• The ear’s frequency response is level-dependent, favoring mid frequencies at low volume
• The ear’s frequency response means quiet playback dulls bass and treble
• The ear’s full dynamic range exists only near 4 kHz
• The ear’s just-noticeable amplitude difference depends on signal type, level, and frequency
• The eardrum converts air pressure changes into mechanical middle-ear vibrations
• The English term timbre entered music through poor translation alternatives
• The English word timbre derives from a contested translation of German klangfarbe.
• The English word timbre was once judged a foreign, mispronounced borrowing not worth preserving.
• The ERB equation can be algebraically rearranged to find a filter’s center frequency
• The ERB equation can be inverted to solve for filter center frequency.
• The ERB operationalizes critical bandwidth into a calculable filter-centered quantity
• The factor of 20 in SPL makes one decibel equal a just-noticeable loudness step
• The flute lacks the seventh harmonic that groups it with oboe and clarinet
• The flute’s breathy onset is broadband high-frequency noise lasting about 70 milliseconds
• The Fourier transform extracts a frequency component by untwisting it into DC.
• The Fourier transform pair converts signals between time and frequency domains
• The Fourier transform unwinds periodic signals into flat DC
• The fundamental frequency of a pitched sound is measurable from its waveform period
• The fundamental period emerges from pooled nerve firing intervals across all places
• The gap between direct sound and reverberation encodes the perceived size of a space
• The hearing system analyzes complex sounds by linear superposition of component responses
• The hearing system performs far richer acoustic analysis than a spectrogram captures.
• The human hearing system compromises between frequency and time resolution, unlike ideal filters
• The human vocal tract behaves acoustically as a stopped pipe about 17.5 cm long.
• The just-noticeable difference quantifies the smallest audible change in a sound
• The kettledrum bowl lowers its head modes toward harmonic ratios
• The kettledrum outputs a missing-harmonic series yet still conveys a clear pitch
• The middle ear amplifies sound pressure through three mechanical mechanisms
• The middle ear both transmits sound and protects against loud noise
• The middle ear functions as an impedance transformer matching air to fluid
• The Millington-Sette equation reduces to the simpler Sabine equation when absorption is below 0.3
• The mosquito tone’s canonical frequency is 16.8 kHz with octave test steps
• The mosquito’s exclusionary sound was repurposed as private teen ringtones
• The Norris–Eyring reverberation formula replaces Sabine’s for absorptive rooms.
• The number of harmonics audible halves with each octave rise in pitch
• The ossicles convert air vibrations into mechanical movements that drive cochlear fluid
• The ossicles mechanically transform impedance to overcome fluid resistance
• The outer ear both locates sound sources and boosts certain frequencies
• The outer ear’s resonant shapes enable front-back sound localization
• The phon scale defines loudness relative to equal-loudness matches at 1 kHz
• The pipe organ is history’s first additive synthesizer for timbre
• The player’s lips form a variable double reed making brass a stopped-pipe acoustics
• The precedence effect makes the listening room’s first reflections mask the recorded hall’s acoustics
• The precedence effect was quantified by an experimenter named Haas
• The Pythagorean comma requires redistribution across fifths for multi-key tuning
• The resultant bass illusion works best when the higher harmonic is voiced softer.
• The reverberant field is an integrated sum of intensities arriving from all directions
• The room constant R is the reciprocal of the absorption-to-surface term
• The room containing a played instrument is itself a second modifying system
• The room is itself an acoustic system in the model chain
• The round window serves as a pressure-release valve for incompressible cochlear fluid.
• The Sabine equation works well when absorption coefficients stay under 0.3
• The same high-frequency targeting lets teens create adult-invisible ringtones
• The same standing wave mathematics applies to strings and acoustic systems
• The second brass resonance mode is the lowest usable musical pitch
• The seventh harmonic gives an organ mixture a reed-like timbre
• The shoebox hall’s shape and Victorian ornament together created superior acoustics
• The singing voice is the most versatile of all musical instruments
• The singing voice reduces to a source plus a shapeable vocal-tract filter
• The sound arriving first dominates perceived direction beyond 673 microseconds of delay
• The source excites all harmonics even when a resonator only supports odd ones.
• The spectrum of partials forms a unique recognizable signature for each instrument
• The speech transmission index measures intelligibility via a modulation transfer function
• The speed of sound rises about 0.59 meters per second per degree Celsius
• The standard reverberation time T60 is defined by a 60 dB sound energy decay.
• The superior olivary complex sharpens phase locking to enable left-right sound localization
• The term overtone numbers components above the fundamental, offset by one from harmonics
• The tympanic membrane is 13 times larger than the stapes footplate in humans.
• The unresolved seventh harmonic imparts reedy timbre to instruments like the clarinet
• The vocal folds vibrate via pressure changes from curved airflow, not the Bernoulli effect
• The vocal tract behaves acoustically like a stopped tube of constant diameter
• The voice operates between meaning and sound, resisting reduction to either
• The wave equation fλ equals c applies to any wave in any medium
• The word timbre derives from klangfarbe meaning color of tone
• The word timbre originally meant kettle drum and postage stamp
• Thirteen chords of just-tuned harmony can accumulate a semitone of flatness.
• Timbral differences between instruments can be plotted dynamically through critical bands
• Timbral terminology is ambiguous across different users
• Timbral words mean different things to different speakers
• Timbral words shift meaning between different users
• Timbre analysis requires a method that preserves when acoustic changes occur
• Timbre can be graphed as relative energy across fundamental, resolved, and unresolved harmonics.
• Timbre comprises spectral content and how frequencies and amplitudes shift over a note
• Timbre depends on staggered harmonic entry times and spectral extent
• Timbre has no objective right answer and must be measured by averaged listener ratings
• Timbre is conditioned entirely by the strength and presence of overtones
• Timbre is defined as the residual difference once loudness, pitch, and duration are matched
• Timbre is defined negatively, as whatever differs between equal-pitch, equal-loudness sounds
• Timbre is described through a rich vocabulary of quality adjectives
• Timbre is judged subjectively with no right answer, unlike pitch or loudness
• Timbre is the cue by which listeners identify which instrument produced a note
• Timbre lacks a correct answer, so it must be measured via adjective-pair ratings.
• Timbre lacks a one-dimensional subjective scale unlike pitch and loudness
• Timbre perception arises from basilar membrane vibration patterns, not pitch alone
• Timbre perception research remains too coarse to explain fine discriminations like distinguishing violin makers.
• Timbre reduces to a weighted ratio of fundamental and resolved versus unresolved harmonics
• Timpani are orchestral drums retuneable to follow key modulations.
• Timpani kept usable across key changes because they retune quickly
• Tinnitus after exposure signals damage even below the 85 dB threshold
• Tinnitus arises when the cochlea spontaneously generates its own noise
• Tinnitus persists audibly even amid loud scanner noise.
• Tone hole lattice layout shapes woodwind timbre as bright or dark
• Tone-hole fingering requires modes to keep constant frequency ratios as holes open
• Tone-hole opening must preserve modal frequency ratios for correct tuning
• Tones with lower partials up to the sixth sound rich yet sweet
• Tongue and lip constriction locations systematically shift formant frequencies from neutral values
• Total sound absorption depends on material area, not just absorptivity.
• Total surface area can outweigh absorption efficiency in controlling room reverberation
• Total surface area determines a material’s acoustic dominance more than its absorption coefficient alone
• Trained singers keep vocal folds closed longer each cycle than untrained singers
• Trained singers show higher glottal closed quotient than untrained singers
• Translating frequency-domain talk into note pitches improves engineer-musician collaboration
• Transverse shear waves travel at about 60 percent of longitudinal wave speed.
• Transverse waves move material sideways rather than compressing it along the travel direction
• Transverse waves move particles side to side rather than push-and-pull.
• Trapped air in a cavity oscillates like a spring-and-mass system, producing Helmholtz resonance
• Trombone and tuba differ dramatically in harmonic richness on the same note
• Trombone slides overcome tuning compromises that valves require
• Trumpets need a movable tuning slide because fixed third-valve length is inaccurate
• Truncated ellipsoidal speaker enclosures match spherical performance with more design freedom
• Trying to hear audio over loud ambient noise can cause noise-induced hearing loss.
• Tuning systems determine whether auditory combination tones align with harmonic series
• Turbulent airflow produces the nonpitched noise sounds of speech like sh and ss.
• Turning around on a live recording reverses perceived orchestral space
• Turning down volume suppresses perceived bass and treble relative to midrange
• Twelve stacked fifths slightly overshoot seven octaves by the Pythagorean comma
• Twelve stacked perfect fifths fail to close the octave exactly
• Two close sine tones produce amplitude beats at their frequency difference
• Two close tones progress through beats, roughness, then smooth separation as their gap widens
• Two competing theories explain how the ear tracks pitch
• Two correlated sources combine constructively or destructively depending on frequency
• Two equal uncorrelated sound sources add only 3 decibels.
• Two equal uncorrelated sources raise level less than two correlated ones.
• Two pure tones fuse into a single rough tone below the critical bandwidth.
• Two simultaneous pure tones generate an audible tone at their difference frequency
• Two simultaneous pure tones may fuse rather than be heard separately.
• Two spaced loudspeakers suffice to create phantom sound sources in front of a listener
• Two tones are perceived as separate when no single filter captures both
• Two tones fuse into one unless their separation exceeds critical bandwidth
• Two tones sound rough until no single ear filter captures both
• Two vibrations with an irrational frequency ratio never exactly repeat their combined pattern
• Tympanic membrane buckling doubles the force delivered to the malleus
• Tympanic membrane buckling doubles the force delivered to the middle ear bones.
• Unambiguous pitch perception requires a periodically repeating acoustic waveform
• Uncontaminated direct sound is required for clear speech intelligibility in rooms
• Uncorrelated ambient signals create an impression of directionless surround sound.
• Uncorrelated sound sources increase in level less than correlated ones do
• Uncorrelated sound waves sum by power, not by amplitude.
• Uncorrelated source addition ignores phase and depends only on signal power.
• Uncorrelated source addition is phase-independent unlike correlated addition
• Uncorrelated source totals ignore the relative phases of signals.
• Undersampled signals alias because high-frequency components mirror below the Nyquist frequency
• Understanding vocal acoustics helps musicians approximate synthesized sounds by ear.
• Unequal symbol probabilities always reduce a source’s entropy below its bit capacity
• Unequal wave amplitudes quickly reduce interference effects below 1 dB peak variation
• Uneven decay rates make a room ring at and boost specific frequencies.
• Unevenly absorbed or coupled spaces produce multi-sloped decay curves instead of one smooth exponential.
• Uniform Fourier-magnitude diffusion structures scatter sound evenly at any incidence angle.
• Uniform quantization noise becomes audible below roughly 60 dB signal-to-noise, or ten bits.
• Uniform quantization noise requires about 60 dB signal-to-noise to stay masked
• Unnicked organ pipes produce a sharper, clearer consonantal attack tone called chiff.
• Unpredictable sound presentation prevents subjects from faking hearing-loss test results
• Untrained listeners judge sine-wave consonance continuously, with no special role for conventional intervals.
• Untrained listeners judge two-tone consonance continuously without musical interval prominence
• Untuned organ mutation stops generate audible beats when chords are played
• Upper harmonics swing more in frequency than lower ones during vibrato
• Upward note bending fails because airflow cannot reinforce both resonances at once
• Variable-length codes can be made comma-free without separator bits
• Verbal timbre descriptors are ambiguous across speakers but map onto specific spectral regions.
• Very dead rooms lack a true reverberant field entirely
• Very low bit rates require working in the frequency domain to skip masked regions
• Very low-bit-rate coding requires frequency-domain processing because masking is a spectral phenomenon.
• Vibrato modulates frequency while tremolo modulates amplitude
• Vibrato swings upper harmonics more widely than lower ones on a linear frequency scale.
• Violin body tone is dominated by just two low-frequency resonances
• Visual cues of room size override what the ears actually hear.
• Visual cues override directional sound cues in film dialog perception
• Visual size cues override matching audio reverberation judgments
• Vocal fold pulses produce harmonics falling 12 dB per octave
• Vocal fold vibration can be measured through external neck electrodes via electrical impedance.
• Vocal folds evolved to protect the lungs, not to produce speech sound.
• Vocal pitch is controlled through fold tension and vibrating mass
• Vocal quartets empirically drift flat as just intonation predicts
• Vocal tract resonance multiplies each vocal fold harmonic to shape the sung vowel spectrum
• Voice training measurably raises the larynx closed quotient across singers.
• Volume changes alter perceived frequency balance, not just loudness
• Volume compression worsens speech intelligibility by flattening syllabic amplitude cues
• Vowel formant frequencies are predictable from the location of vocal tract constrictions
• Vowel production shifts formants by perturbing vocal tract geometry like a woodwind instrument
• Wall boundaries alter loudspeaker output by modifying acoustic impedance seen by the driver
• Wave propagation in 3D objects grows extremely complex beyond thin strings
• Wave propagation speeds depend on shape beyond simple strings
• Wave scattering behavior is governed by object size relative to wavelength
• Wavefield synthesis fails unless the room is completely anechoic
• Wavefield synthesis requires a unique signal per loudspeaker
• Wavefield synthesis yields realistic fields but stays niche due to speaker costs.
• Wavenumber encapsulates dispersion and velocity changes useful for plate propagation calculations
• Waves transitioning from bounded to unbounded space create interfering reflections
• We mostly listen to recorded sound in small rooms, not performance halls
• Weak interfering waves under one eighth intensity produce negligible comb filtering
• Weak musical pitch perception can persist without peripheral phase locking.
• Weighted geometric means of absorption coefficients let reverberation equations handle mixed materials
• Western harmony gradually adopted higher-numbered harmonic intervals over history.
• What a listener hears differs from an instrument’s raw output
• What the ear perceives can diverge from sound’s physical components.
• Where a gong is struck changes its sound from harmonic chime to noisy shimmer
• Where you strike a percussion instrument governs which modes sound
• Whether combination tones are musical depends on the tuning system
• Whole tone and leading notes derive from intervals above the dominant in just intonation
• Wide organ pipes suppress upper harmonics in their output
• Wide pipes are weak in upper harmonics
• Wide-band membrane absorbers achieve dead-room acoustics in just 30 cm depth.
• Wide-band membrane absorbers shrink non-environment rooms to fifteen square meters
• Widely spaced harmonics smear or hide formant peaks in high notes
• Widened critical bands make it harder to separate components of complex sounds like speech
• Widening a sound’s bandwidth beyond one critical band raises perceived loudness at unchanged intensity.
• Wind bends sound waves because air velocity changes with height
• Wind gradients bend sound waves and explain downwind audibility
• Wind instruments divide into reed and non-reed sound-source types
• Wood grain direction drastically changes the speed of sound through the material
• Wood paneling and tapestries absorbed sound well enough to balance baroque music rooms
• Wood paneling naturally absorbs low frequencies and balanced mansion hall acoustics
• Woodwind flue instruments act acoustically as pipes open at both ends
• Woodwind instruments enter their harmonics in staggered sequences on attack
• Woodwind perturbation theory predicts how articulator constrictions shift formant frequencies
• Woodwind reeds act as pressure-controlled valves while flue instruments act as flow-controlled valves
• Woodwind tone holes must preserve mode frequency ratios to stay in tune