bluuebunny/crossref_metadata_2025 huggingface dataset:
Wave Mechanics of Boundary-Layer Turbulence and Noise
Authors: Landahl Marten T.
Published: October 2005

Recent advances in the understanding of the structure of boundary-layer turbulence form the basis of a new conceptual model tot the fluctuating velocity field. In this, an “active” mode of small-scale motion is coupled to a large-scale “passive” mode by aid of wave-mechanical considerations. The small-scale motion originates during the intermittent “bursting” in the wall layer leading to excitation of large-scale, lightly damped, traveling shear waves. It is argued that the passive wave-like mode gives the major contribution to the pseudosound, whereas the active bursting mode serves as the predominant source for the radiative noise. It is shown that the bursting mode gives rise to compact noise sources of both dipole and quadrupole type whose strengths are related to the turbulent shear stress production.
Nonlinear Theory of Infrasonic Fluctuation Spectra
Authors: Moo C. A. , Pierce A. D.
Published: October 2005

In a previous paper [AGARD Conf. Proc. No. 115 (1972)], the authors have suggested that the 2–5 min period oscillations observed in the ionosphere during periods of thunderstorm activity in the troposphere may be explained in terms of the interaction of the velocity fluctuation fields in thunderstorm regions. This analysis is hem extended to the investigation of the nonlinear interactions of a system of propagating and evanescent internal acoustic-gravity waves. The development rests on Hamilton's principle for nonlinear acoustic-gravity waves with the Lagrangian approximated to third order. The coherent nonlinear interaction problem is discussed with the artifice of two primary waves which interact in a restricted region of space. Although two interacting gravity waves cannot produce a third propagating gravity wave of second order they may produce an acoustic wave in the atmosphere. The general problem of interacting random wave fields is formulated and the evolution of an initially monotonic gravity wave spectrum due to nonlinear interactions is discussed. In addition to affording a possible explanation of the anomalous ionospheric oscillations, the nonlinear interaction theory may explain the well known spectral gap which is found in the kinetic energy spectrum of atmospheric turbulence during periods of convective activity.
Nearfield Calibration of Navigational Sound Signals
Authors: Cook Bill D. , Barnes Frank N.
Published: October 2005

As an alternative to open field calibration of navigational sound signals (fog horns), these tests are being conducted in anechoic chambers. Modern sound signals are usually linear arrays which may be greater than 10 ft in length. These lengths combined with the dimensions of anechoic chambers require that calibration measurements be made in the nearfield of the array. Present techniques involve modeling the sound signal as a number of point sources and comparing the predicted and measured SPL as a function of distance from the source. Here we present an alternative procedure where the sound pressure and its phase are measured on a spherical surface enclosing the array. The sound pressure at all field points can then be calculated by using the solution of the wave equation in spherical coordinates. Some preliminary experimental results are shown.
Human Temporary Threshold Shifts (TTS) and Recovery from 24-Hour Acoustic Exposures
Authors: Krantz D. W. , Nixon C. W. , Johnson D. L.
Published: October 2005

Twelve volunteers experienced 24-h continuous exposures to a 13-oct band of noise centered at 1000 Hz. Exposure effects at levels of 75, 80, and 85 dBA were investigated in separate sessions occurring several days apart. Hearing thresholds were measured at twelve frequencies between 125 and 12 000 Hz prior to, during, and following exposure. TTS acquisition and recovery are discussed relative to (1) asymptote, (2) exposure interruption for hearing threshold measurement, and (3) other studies of long duration exposure.
The Ototoxicity of Methyl Mercury
Authors: Falk S. A. , Klein R. , Sanders G. M. , Lim D. J.
Published: October 2005

To provide understanding of deafness associated with methylmercury poisoning, we studied the pathological effects of methyl mercury hydroxide (MMH) on the nervous tissue and sensory epithelium of the inner ear. Young Hartley guinea pigs (250–500 g) were treated with daily subcutaneous doses of 2 mg Hg/kg body weight MMH, and the cochleas were examined by phase-contrast and electron microscopy. By the use of cochleographic reconstructions of surface preparations of the organ of Corti, the outer hair cells (OHC) were significantly damaged (P<0.05) at the level of 212 turns from the cochlear base. Pathology also occurred at 312, 112, and 12 turns but was not significant. Cellular pathology included derangement of orderly pattern of stereocilia, cytoplasmic vesiculation, and collapse of cell membranes. Study of the stria vascularis and auditory nerve and its spiral ganglion did not reveal any pathology. Thus, MMH exerts an ototoxic effect directly on the OHC and is a unique ototoxic agent, since aminoglycosidic antibiotics and ethacrynic acid cause initial pathology in the basal turns. The sensorineural hearing loss occurring in methyl mercury poisoning, which was presumed secondary to diffuse CNS pathology, now can be considered secondary to a primary cochlear lesion from organic mercurials.
Temporal Summation Function in Presbycusis
Authors: Corso John F. , Wright Herbert N. , Valerio Michael
Published: October 2005

The threshold-duration function was measured by brief-tone Békésy audiometry at 250, 1000, and 4000 Hz. At each frequency, threshold was tracked continuously for 1 min; thresholds were then tracked successively for interrupted tones of 500-, 200-, 100-, 50-, 20-, and 10-msec duration. All interrupted tones were presented for 1 min at a repetition rate of 1/sec with a rise-fall time of 10 msec; the attenuation rate for all tests was 2.5 dB/sec, on a Grason-Stadler E-800 audiometer. Subjects (nine male, seven female) were 51 to 57 years of age with no otological abnormalities; threshold-duration functions were obtained on both right and left ears. The results indicate the presence of temporal summation; but, as frequency is increased, the threshold-duration function is depressed compared to that of a young, normative group [H. N. Wright, J. Acoust. Soc. Amer. 53, 113(A) (1972)]. Five subjects from the older group with no previous history of noise exposure showed a pronounced frequency effect, with a shortened time constant at the higher frequencies. It appears, therefore, that temporal auditory summation is affected by the physiological disturbances of presbycusis, independent of noise exposure.
Field Coil Pick-Up Monitoring Hearing-Aid Outputs at an Artificial Ear and External Auditory Meatus
Authors: Mendelson G. L.
Published: October 2005

An investigation was done to evaluate the differences between measured hearing-aid loading by an artificial ear and by the real ear. Responses of the field coil from four aids and 11 receivers were measured both on the artificial ear and on the real ear canal. Input signals to the aid were at 15 discrete frequencies and presented at the level of 60 dB re 0.0002 dyn/cm2. Results indicated that the artificial ear (2-cm3 coupler) is reliable for average measurement of a given hearing-aid output. However, individual differences did occur in the form of minor intensity changes in the response obtained at the real ear canal as compared to the response obtained at the artificial ear. In a clinical setting the field-coil pick-up offers the potential of easily, quickly, and reliably monitoring the differences in hearing-aid frequency responses on different ears.
Monaural and Binaural Free-Field Speech Reception through Ear-Level Hearing Aids under Noise and Reverberation
Authors: Nábělek Anna K. , Pickett J. M.
Published: October 2005

Speech reception was studied in a room at reverberation times T = 0.3 and 0.6 sec, to compare binaural and monaural reception through ear-level hearing aids in the presence of a babble of eight voices. Some subjects (Ss) had normal hearing and some had severe sensorineural hearing impairment with bilaterally symmetrical audiograms. The speech and noise sources subtended an arc of 60° on a circle 11 ft in radius centered on the S's head. Binaural gain, the difference in speech to noise ratios for 50% correct speech reception comparing binuaral and monaural results, was 3 dB for normal Ss at both reverberant conditions. The binaural gain for impaired Ss varied from 3 dB to 0; those with previous binaural experience tended to show greater binaural gain. The prolongation of reverberation caused a reduction in reception. In presence of the masking noise, the reduction was about 15% correct response for both normal and impaired Ss. In quiet, normal Ss' reception was not influenced by the prolonged reverberation but the impaired Ss showed about 10% reduction in reception. [Supported by Gallaudet College].
Linguistic Units and Speech Production Theory
Authors: MacNeilage Peter F.
Published: October 2005

Today, one seldom hears of the unit of speech production. Speech error data, in particular, have made it clear that every linguistic unit from “phonemic phrase” through “word,” “morpheme,” “syllable,” and “phoneme” to “distinctive feature” is useful to the speech production theorist in that each allows him to conceptualize processes whereby the speaker manipulates unitary variables in his output. As some of these units must be “activated” prior to others in the organization of output, and as some units actually overlap structurally with others, we require a hierarchical model specifying the sequence of events leading to output and resolving problems of overlap. Some clues to the form of this model are available. Very little is known about the higher order units—phonemic phrase, word, morpheme—in this regard. For the lower-order units much conceptual difficulty remains, including questions asked to the physical manifestation of the syllable, and the inventory of phonemes and features. Most work in linguistic science proper has been of limited assistance here because of its virtually sole concern with contrastive properties of speech, and its failure to conceptually distinguish production from perception. Problems of sampling and interpretation may also limit further profit from speech error data. The most basic problem of speech production theory has always been that although speech must be conceptualized in terms of units, a speaker's output is a continuous strain in which movements of various articulators overlap in space and time. Studies of the dynamics of these movement sequences show promise of providing the best empirical base for theory of speech production in which linquistic units and observable physical processes are reconciled. Consideration of timing mechanisms and rhythmic organization will be a prerequisite to such a theory, and sets of rules relating citation form speech to spontaneous speech must be included. The invariants in this theory will be found in the goals being attempted rather than in the means being used to achieve them.
A Unified Theory Relating Speech Production and Perception
Authors: Lieberman Philip
Published: October 2005

The explanatory powers of the motor theory of speech perception (in the form elaborated at Haskins Laboratories) are enhanced when acoustic factors like those discussed by Stevens in his quantal theory are incorporated into a unified theory. The categorical discrimination of stop consonants may follow from neural feature detectors that have evolved to match the quantal effects of the human supralaryngeal vocal tract. Recent experimental data indicate that the primate auditory system is plastic, i.e., neural “feature detectors” can be set up for auditory events that are important to the animal. The decoding of speech in man appears to be the result of arrays of feature detectors matched to the quantal constraints of the supralaryngeal vocal tract. The effects of hemispheric lateralization indicate that active modeling of the vocal tract by an “analysis-by-synthesis” model does not appear to be the correct neural implementation. The unified theory is consistent with a phylogenetic model for the evolution of human speech production and perception. The evolution of the human supralaryngeal vocal tract probably preceded the evolution of the neural mechanisms necessary for decoding human speech.
Noise Mechanisms in the Inflation of the Automotive Safety Air Bag
Authors: Hickling Robert , Posey Joe W.
Published: October 2005

Free-field noise from a compressed-gas air bag inflator was recorded and spectrally analyzed. Tests were run with different types of silencing devices inserted in the manifolding, and with and without the bag attached. Comparisons between the noise spectra obtained in these tests permitted the identification of different mechanisms of noise generation. The largest peak in the spectrum was generally below 20 Hz and is due to a monopole-type of source associated with the overall discharge of gas from the inflator system. Other noise sources are associated with the motion of the surface of the bag, “organ-pipe” sources within the manifolding, and broadband jet noise associated with the flow of gas through the diffuser slots into the air bag.
Prediction of Transmission Loss by the Finite-Element Method
Authors: Young Cheng-I James , Crocker Malcolm J.
Published: October 2005

A numerical technique based on the finite-element method has been developed for analyzing the performance of systems of acoustic elements including expansion chambers. The theories developed are, first, the variational formulation of the acoustic field existing in the system and, second, the finite-element approximate solutions of the variational problems. The predictions of the transmission loss then are made by forming the equivalent acoustic four-terminal transmission network in which the acoustic four-pole constants are calculated from the finite-element method. As used, the finite-element approach is perfectly general and may be applied to any system component with arbitrary boundary geometry, provided this may be realized by an assembly of rectangular elements. The method is applied to simple expansion chamber models because the results are tractable and theoretical results from acoustic filter theory are available for comparison purposes. The results from the various models bring out an important fact: the accuracy of the prediction of transmission loss implies that the variational formulation and finite-element approximations are adequately applicable to a number of practical applications.
Mufflers for Small Engines: Some Phenomena Revealed by Analysis of Waveforms
Authors: Brammer A. J. , Piercy J. E.
Published: October 2005

The physical phenomena present in simple cavity mufflers attached to a small one-cylinder, two-cycle engine have been studied to help understand why devices used in practice often bear little resemblance to systems predicted by linear theory. The acoustic waveforms both inside the muffler and radiated by the tailpipe were recorded, together with the engine performance and static back pressure, as the resonance frequency and the geometrical configuration were varied. The pulse at the exhaust port approximated to a half sinusoid of 12-msec duration and 6-psi amplitude, which is consistent with the cylinder and port forming a nonlinear resonator. The cavity mufflers superimposed an oscillation at the engine firing frequency, this being the effective back pressure experienced by the motor. This alternating pressure also corresponds to the driven linear response of the system. Evidence was found for shock formation during propagation in air or within long tailpipes. There was often some ringing excited by the transient nature of the pulse observable in the radiated waveform.
The Measurement of Noise from Moving Vehicles
Authors: Hixson E. L. , Warnix J. L.
Published: October 2005

A new method for measuring vehicle noise is proposed that dynamically corrects the sound pressure for the source distance as the vehicle passes. The result is a sound pressure at unit distance from the vehicle. When logarithmic units are used, the quantity is defined as a noise source level (NSL). A system using a sound level meter, traffic radar, and means for a polar plot of NSL was built and evaluated. For an omnidirectional noise source on a vehicle, spherical spreading was verified and a polar plot, uniform with track angle, was produced. The directivity of a directive source on the vehicle was measured during a pass-by and under stationary conditions. Comparable results were obtained. The vehicle noise measurement method provides rapid data acquisition of source levels and directivity for noise research, a basic measure for noise certification of vehicles, and an improved means for noise ordinance enforcement.
Fourier Analysis of Spherical Aberrations
Authors: Clark J. A. , Afkar A.
Published: October 2005

Aberration in imaging systems due to the presence of spherical surfaces is usually treated by the expansion of the aberration function in terms of a complete set of polynomials that are orthogonal over the interior of a circle of unit radius. This method involves a set of complicated programs, and the interpretation of the components in the expansion in terms of physical quantities is not possible. To avoid such complications, Fourier transform techniques are used. This method is applied to determine the diffraction pattern of a circular acoustical reflector. It is shown that the amplitude distribution function using this technique and the language of J. W. Goodman's Introduction to Fourier Optics is consistent with experimental results obtained in the laboratory. In particular, the effects of varying the radius of the spherical surface, index of refraction, and wavelength in the reduction of spherical aberration are obvious in the resulting equation. [This work supported by ARPA and monitored by ONR.]
Exact Response of Piezoelectric and Isotropic Acoustic Imaging Faceplates
Authors: Ahmed M. , Whitman R. L. , Korpel A.
Published: October 2005

The amplitude and phase responses of resonant and nonresonant acoustic imaging faceplates of arbitrary thickness are analyzed. The faceplates are important components in several acoustic visualization systems such as the Sokolov tube and the laser-scanned acoustic camera. Knowledge of the response is important because it governs the resolution of the image and the aberrations which are present, even if other components of the system are perfect. Computer plots of the responses of some typical piezoelectric and isotropic faceplates are presented. These plots are derived from an analysis which is exact and takes into account the effects of evanescent longitudinal and shear waves when they are present, as well as losses due to internal friction in the plate material. The resolution of different systems which use a faceplate and the contribution of the faceplate to image aberration will be compared and discussed.
Capacitance Microphone Static Membrane Deflections—Comments and Further Results
Authors: Warren J. E.
Published: October 2005

The static deflections of the membrane diaphragm of capacitance microphones due to electrical biasing are governed by a nonlinear differential equation. Since the governing equation is nonlinear, deflection and stability results are usually obtained using numerical methods. The static deflections and their stability values determine the operating range of certain microphone parameters and influence the sensitivity and frequency response. The results of two past investigations using different forms for the nondimensionalized nonlinear differential equations are reviewed and compared. [J. E. Warren, A. M. Brzezinski, and J. F. Hamilton, J. Acoust. Soc. Amer. 52, 711–719 (1972); F. W. Fraim et al., J. Acoust. Soc. Amer. 53 (to be published)]. Results are presented for circular, annular, and strip membranes using a third and perhaps more appropriate form for the nonlinear differential equation. Stability results for the three types of membrane systems for flat stationary electrodes with openings are also presented. Since the third nondimensionalized form is similar to that used by Teer, his results for an optimum stationary electrode curvature for circular membranes may now be applied to situations with two dielectric layers of different thicknesses and permittivities [K. Teer, Acustica 15, 256–263 (1955)].
Transmission of Sound through a Rectangular Membrane
Authors: Leehey Patrick , Sledjeski Leo A.
Published: October 2005

A rectangular thin tensioned membrane was set in a large essentially rigid baffle and excited by nearly plane sinusoidal sound waves over the frequency range from 200 to 2300 Hz. Directivity patterns and pressure levels of transmitted sound were measured for both normally incident and off-axis excitations. Membrane displacement response was measured for normal excitation over the frequency range from 300 to 700 Hz in order to determine response patterns near two strong antiresonances. The membrane wave speed was 30% of the sound speed and the membrane was heavily air loaded. Measurements were compared with a low-frequency Galerkin-process solution for the membrane response and transmitted sound field.
Outdoor Tests of Ceramic Microphones
Authors: Djuric S. V.
Published: October 2005

Sixteen General Radio 1- and 12-in. ceramic microphones have spent over one year outdoors, located on the roof of a building in Concord, Massachusetts. Performance of the microphones was checked daily by a special setup. Precise measurements of microphone parameters were made periodically in the laboratory. The average change of sensitivity at 500 Hz throughout the testing period was ±0.2 dB. The average changes in the frequency-response curves at different frequencies were of the same order of magnitude. The measuring setup is described and the measurement data presented.
Matrix Quadraphony with Logic Enhancement
Authors: Bauer Benjamin B. , Gravereaux Daniel W. , Budelman Gerald A.
Published: October 2005

Matrix quadraphony is a technique for “matrixing” or combining several original program channels having presssigned angular positions into two audio-frequency channels (e.g., of a stereophonic disk record) for reproduction over four loudspeakers which surround the listener. Practical utility and realistic spatial reproduction of the original channels requires the use of a matrix code and a position-responsive time-sharing mechanism capable of (a) full compatibility with existing stereophonic and monophonic sound reproduction media and devices, (b) providing adequate positional cues, codable into the matrixed channels and recoverable therefrom, and (c) complementary cooperation with the psychoacoustics of human hearing. A description of a quadraphonic disk system (The SQ™ Disc Record System) responding to the above requirements is given.
Some Psychoacoustic Phenomena Related to Quadraphonic Reproduction
Authors: Bauer Benjamin B. , Milner Paul
Published: October 2005

Heretofore unexplored psychoacoustic phenomena having special bearing on matrix quadraphony have been studied using centrally located observers facing a small end of a simulated rectangular living room with loudspeakers in the corners. It was observed that (1) monotonic all-pass phase-shift networks (“psi-networks”) do not perceptibly affect sound quality; (2) signals applied to the two front loudspeakers through differential quadrature filters appear to spread out and to shift towards the loudspeaker carrying the leading signal (“quadrature shift”); (3) directional cues attributable to signals from the front loudspeakers are relatively uninfluenced by lesser-magnitude similar signals applied to back loudspeakers (“front source dominance”); and (4) the perceived angular coverage of the field produced by the back loudspeakers is substantially smaller than that produced by the front loudspeakers (“back image contraction”).
Dynamic Range Compression for Tape Recording of Noise Data
Authors: Blanchard Byron E.
Published: October 2005

Conventional approaches such as the Dolby or DBX dynamic range compressors for tape recording are unsuited for noise data recording because the expansion process has an inherent time lag relative to the compression. The time lag occurs because the expander gain is controlled by rectified and averaged compressed signal. A system has been developed to permit exact tracking of the compression and expansion gains even in the transient case. An FM signal representing compression gain command is recorded on a separate tape channel. On playback, the output of an FM discriminator controls the expander gain. The lag of the low-pass filter following the discriminator is matched by an identical filter in the compression circuit between the compression gain command and the gain block. Input signals over a 90-dB range are accommodated and the compression and expansion track within 1 msec.
Measurements of Acoustic Scattering from Spheres
Authors: Hunter J. L. , Dossa D. D. , Fisch M. R.
Published: October 2005

Although acoustic scattering from spheres has received thorough theoretical treatment in ideal cases, relatively few basic laboratory measurements have been performed. We are conducting a systematic experimental study of the received strength of signal as a function of the relative sizes of the scattering sphere, the acoustic wavelength, and the position of the scattering sphere with respect to the axis of the transducer. The radiator is a 1-in.-diam quartz disk, the receiver a PZT cylinder of diameter 2 mm; the scattering spheres range in size from 4 mm to several centimeters in diameter. The sound field is scanned through 360° with the scattering sphere as center. This work originated in past attempts to compare observed signals from scatterers with idealized theoretical predictions of scattering from spheres, and the feeling that departures from idealized conditions in typical experimental apparatus may cause significant departures from results predicted theoretically. This paper will compare experimental results with theoretical predictions. [This work supported by Acoustics Branch, Office of Naval Research.]
Wave Propagation in a Region Having a Randomly Rough Boundary
Authors: Wenzel Alan R.
Published: October 2005

The problem of scalar wave propagation in a uniform medium bounded by a randomly rough surface is considered. A detailed treatment is given of the case in which the surface is a small perturbation of an infinite plane. It is shown that, if the surface roughness is statistically homogeneous, the coherent. (or mean) wave satisfies a generalized impedance boundary condition on the mean surface. From this boundary condition, the effective plane-wave reflection coefficient Cs is obtained. Generally, Cs depends on the direction of propagation as well as on the frequency and angle of incidence. However, if the surface is statistically isotropic, Cs is independent of the propagation direction. A relatively simple expression for Cs is obtained by assuming that the wavelength is much greater than the correlation length of the surface roughness. This expression shows that |Cs| <1, so that the amplitude of the coherent wave is diminished upon reflection from the surface. This is the result of scattering of energy out of the coherent wave by the surface roughness. It is also shown that this type of surface can act as a wave guide.
A Semi-Empirical, Nonlinear Wave Equation for Standing Waves in Cavities
Authors: Coppens Alan B. , Lane Christopher
Published: October 2005

A one-dimensional nonlinear wave equation with loss term based on the Rayleigh-Kirchhoff predictions for wall losses and speed-of-sound dispersion was found to be only partially correct in describing standing waves in tubes because the observed dispersion was not in agreement with the theoretical model [A. B. Coppens, J. V. Sanders, and J. Winn, J. Acoust. Soc. Amer. 51, 154(A) (1972)]. As a result, we have modified the nonlinear model so that the infinitesimal amplitude resonance frequencies and Q's of the cavity observed in practice can be inserted into the nonlinear wave equation. This means that the nonlinear distortion of the standing wave can be predicted on the basis of the measured infinitesimal amplitude behavior of the cavity. Comparison with experiment is good on the whole, although some discrepancies remain. This semi-empirical model has been extended to three-dimensional standing waves. The model predicts that nonlinear effects will be reduced appreciably if the normal modes of the cavity are not integrally related, and also predicts that in a cavity with pressure release walls the nonlinear distortion vanishes to second order.
A Perceptual-Anchor Model for Context Coding in Intensity Perception
Authors: Berliner J. E. , Durlach N. I.
Published: October 2005

Previous experiments on intensity perception show that resolution varies over the range of intensities employed in the experiment. Resolution is better near the edges of the range than in the middle, and, when a fixed reference stimulus is presented on each trial, resolution is improved in the vicinity of this stimulus. A revision of our initial theory [N. I. Durlach and L. D. Braida, J. Acoust. Soc. Amer. 46, 372–383 (1969)] is proposed to account for these phenomena. Specifically, a new context-coding model is developed in which the observer is assumed to use the edges of the range and the reference stimulus (when present) as perceptual anchors, and to estimate the intensity of the stimulus to be judged by comparing it with these anchors.
Effect of High-Frequency Tail on the Loudness of a 250-Hz Tone
Authors: Hellman Rhona P.
Published: October 2005

The effect of the peripheral spread of excitation to high frequencies on the loudness of a 250-Hz tone was measured by loudness matching and by magnitude estimation and magnitude production. All listening was binaural through earphones. Most of the data were obtained with a 250-Hz tone in the presence of a noise band 390–2400 Hz wide. The noise increased in intensity with the tone so that the signal-to-noise ratio remained constant throughout a given listening session. Loudness measurements were obtained using three signal-to-noise ratios: +10, 0, −10 dB. The data suggest that at low sound intensities the loudness functions for the 250-Hz tone in quiet and in the presence of the noise band are parallel. At sound-pressure levels above 60 dB the effect of the noise is to decrease somewhat the slope (exponent) of the loudness function. It appears that the contribution of high frequencies at high intensities does not appreciably affect the loudness of a 250-Hz tone.
Relative Loudness and Wavetrain Signals
Authors: Victoreen John A.
Published: October 2005

Relative loudness determinations are reported for three adults using damped wavetrain signals delivered by means of circumaural earphones. Measurements were made by observing the amount of sound-pressure change required to produce subjective loudness changes of 0.5 to 1 and 2 to 1 at various sound-pressure levels from about 0.002 μbar to about 100 μbar. Using signals having a decrement of 0.90 loudness was found to vary in direct proportion to sound pressure at 1100 Hz. An empirical expression is given which describes the variation of loudness sensation with sound pressure at other frequencies. This expression enables the calculation of equal loudness data which agree with experimentally measured values to within about ±1 dB over the usual frequency and sound-pressure range of the nondeficient ear.
The Effect of Internal Noise on Signal Detectability
Authors: Soderquist D. R.
Published: October 2005

Recent data related to internal noise [D. R. Soderquist and J. W. Lindsey, J. Acoust. Soc. Amer. 52, 1216–1220 (1972)] have shown that the normal heart function produces a physiological noise capable of masking low-frequency signals. In addition, other investigators have shown that noise within the ear canal (aural noise) may also be a low-frequency masker. Furthermore, it has been suggested that the intensity of the aural noise varies as a function of the volume enclosed by the earphones. This investigation was subsequently designed to reveal the relative effect of each type of internal noise (physiological or aural) on the detectability of a 100-Hz tone. Three well-trained listeners participated in a single-interval signal detection paradigm. Detectability (d′) was calculated for each subject under each of the three variables of interest: type of earphone (insert, supra-aural, circum-aural), listening condition (binaural, monaural—contralateral ear occluded, monaural—contralateral ear unoccluded), and temporal location of signal relative to the subject's EKG R-wave (0.0-, 150-, 300-, 450-msec delay). Results show an increase in the effect of aural noise as the volume under the earphones increase. In contrast, the effect of physiological noise does not diminish across the different conditions. That is, a decrease in sensitivity occurs at the 0.0 and 300-msec delays across all conditions regardless of earphone or listening condition.
Organizing a System for Continuous Speech Understanding
Authors: Woods W. A. , Makhoul J. , Wolf J. , Rovner P.
Published: October 2005

In spectrogram reading experiments [Klatt and Stevens, Boston Speech Conference (1972)], the performance obtained by human experts for phonetic segmentation and labeling was approximately 75% partially or completely labeled but correctly specified, 15% mislabeled, and 10% segments missed. When syntactic, semantic, and vocabulary constraints were applied, the success rate in identifying words was 96%. How does one design a computer system to match this performance at word identification? This paper describes the BBN speech understanding system, which is being designed to extract the meaning from continuous speech inputs in a limited semantic environment by using syntactic, semantic, and vocabulary information to overcome vagueness and errors in phonetic segmentation and labeling. The system contains components for acoustic feature extraction, phonetic segmentation and labeling, word retrieval from partial phonetic information, evaluation of word-match quality, and syntactic and semantic evaluation of partial utterances and proposal of words from context. A control component embodies the strategy for applying different components at appropriate times and places depending on conditions which will vary from point to point in an analysis.
Syntactic and Semantic Support for a Speech Understanding System
Authors: Woods W. A. , Nash-Webber B. , Bates M.
Published: October 2005

The syntactic and semantic components of the BBN speech understanding system serve a dual role. Their primary function is to select syntactically and semantically well-formed sequences of words from a lattice of possible word matches determined by the acoustical processing and lexical retrieval components of the system. They are also responsible for predicting words which may have been missed by the lexical retrieval routines but which are syntactically or semantically motivated by words that have already been found. Each predicted word is then verified against the acoustic waveform by a word matcher, which may in turn ask the acoustic component to reexamine the relevant section of the signal with that word in mind. This paper will describe the structures of the syntactic and semantic components and some aspects of their operation and interaction with each other and with the other components of the system. The paper will be illustrated with examples from simulations of the system.
Use of a Two-Pole Spectral Model in Speech Recognition
Authors: Makhoul John , Wolf Jared
Published: October 2005

In speech recognition applications, it is often desirable to make a gross characterization of the shape of the spectrum of a particular sound. The autocorrelation method of linear prediction analysis leads to an all-pole least-squares approximation to the signal spectrum. Hence an LPC analysis using two poles produces one possible gross characterization. The two poles are computed as the roots of a quadratic equation whose coefficients are the linear prediction parameters, which are simple functions of the autocorrelation coefficients R0, R1, and R2. The poles are either both real or form a conjugate pair in the z plane. This fact, together with the exact positions of the poles, is particularly useful in describing certain gross characteristics of the spectrum.
Speaker Identification Using Nasal Coarticulation
Authors: Su Lo-Soun , Li K. P. , Fu K. S.
Published: October 2005

A new approach which used the statistical properties of the nasal spectra was used to quantitatively study the coarticulation of nasal consonants with the vowels in the isolated /h ə′ CVd/ utterances. The Euclidean distance between two nasal spectra means followed by front vowels and by back vowels along each frequency dimension was used as the acoustic measure of the coarticulation of [m,n] and the following vowel context. The coarticulation of [m] was found to to be about 50% larger than that of [n]. Moreover, the strong speaker idiosyncratic characteristics were found for the coarticulation process, which is not likely to be modified in natural speech. Speaker identification was performed using the [m] coarticulation measure and a correlation decision criterion. The results indicated that the coarticulation cues are much better than nasal spectrum alone, which has already been demonstrated to be one of the best acoustic cues for speaker identification. [This work was supported by the AFOSR Grant 69-1776.]
A Two-Pass Procedure for Synthesis by Rule
Authors: Kuhn G. M.
Published: October 2005

A revised Holmes-Mattingly algorithm for speech synthesis from phonetic input makes it possible to produce speech when control parameter values have been generated only for the first syllable. The algorithm is fast enough to produce speeded speech for a reading machine. With its supporting program, it is being used as a teaching tool for the synthesis of varied languages in an undergraduate speech synthesis course. Pass 1 labels the syllable structure, the allophonic contexts, and the pitch contour of each breath group in the utterance. Pass 2 calculates the control parameter values, including F0, for each syllable in turn. Major effects of voiceless intervals on the F0 contour are provided for. Place of articulation is referenced proportionally in order to facilitate the changing of apparent head size. The algorithm is being used for computer control of the Haskins and Glace-Holmes parallel resonance synthesizers, and of the OVE III serial synthesizer.
AMDF Pitch Extractor
Authors: Shaffer H. , Ross M. , Cohen A.
Published: October 2005

A method is described for estimating the fundamental frequency component of voiced speech using the Average Magnitude Difference Function (AMDF). The technique, a variation of autocorrelation analysis, uses a difference signal formed by delaying the input speech various amounts and subtracting the delayed waveform from the original. The difference signal is always zero at delay = ø, and exhibits deep nulls at delays corresponding to the pitch period for a voiced sound having a quasi-periodic structure. Unlike the generation of the autocorrelation function, the AMDF calculations require no multiplications, a desirable property for real-time speech processing. Also, since the computations involve numerous repetitive operations, they are very suitable for fabrication in special-purpose hardware. The implementation of the AMDF pitch extractor will be described and experimental results presented to illustrate the basic measurement properties.
Implementation of a Linear Predictive Coder
Authors: Goldsberg A. J. , Ross M. J.
Published: October 2005

This paper will describe the design and implementation approach of a predictive coding system for speech compression based, in part, on the autocorrelation method [F. Itakura, IEEE Trans. Comm. COM-20, 4 (1972)], a computationally more efficient technique than the autocovariant method [B. S. Atal, J. Acoust. Soc. Amer. 50, 2 (1971), but which produces speech of similar quality. The design requires a 6400-Hz sampling rate on an input bandwidth of 3 kHz, a 10th-order predictive filter, an absolute magnitude difference function (AMDF) pitch extractor, and a pitch-synchronous interpolation of the synthesizer parameters to produce transmission rates of 2.4 and 4.8 k bps. The computational procedures to implement the algorithms with fixed-point arithmetic on a processor having a word size of 16 bits will be discussed with special reference given toward (a) calculating the first 11 autocorrelation functions, (b) inverting the 10×10 Toeplitz matrix and solving for the 10 filter coefficients, (c) computing gain and pitch and interpolating synthesizer parameter pitch synchronously, and (d) synthesizing speech with a recursive lattice filter. System performance will be demonstrated.
Impedance of Soft Ground and Its Effect on Practical Measurements
Authors: Piercy J. E. , Embleton T. F. W. , Olson N.
Published: October 2005

This paper is a continuation of paper AA5, 84th Meeting of The Acoustical Society, Miami, 1972 [J. Acoust. Soc. Amer. 53, 340 (A)(1973)]. The propagation characteristics of the ground wave are influenced by the ground impedance. The amplitude and phase of the ground-reflected sky wave (and hence the ground impedance ratio at oblique angles of incidence) have been measured at short ranges up to about 20 ft over grass by interference with the direct wave, and used (a) as an independent check of the value from the ground wave obtained in other geometric configurations and (b) to predict the sound field at greater distances for which direct measurement may be difficult. Such predictions and also measurements at ranges up to 1000 ft indicate shadow zones due to ground impedance of up to 40 Db. In a configuration of interest to measurement standards and community noise, via., source height 1–2 ft, receiver height 4–5 ft at a range of 50 ft over grass, there is a transmission dip of 10–15 dB near 1000 Hz: this dip would be more marked and at lower frequencies (due to “Lloyd's mirror” interference between the two sky waves) were the low-frequency energy not transmitted via the ground wave.
Model Studies of Effects on Motor Vehicle Noise of Shapes and Arrangements of Buildings along Urban Streets. I. Preliminary Survey and Theory
Authors: Knudsen Vern O. , Plane Vern C.
Published: October 2005

Literature searches revealed a paucity of relevant information on the “canyon effect” of motor vehicle noise along streets faced with buildings. Published accounts of the reviewed investigations furnish useful but incomplete data on various aspects of the overall problem. Our preliminary investigation, both experimental and theoretical, has led to significant results which we shall discuss. Reverberation between both reflective and absorptive pairs of walls has been investigated in models of various sizes and shapes located in the large UCLA anechoic chamber (30 ft long, 23 ft wide, and 21 ft high). The model scales ranged from 1:5 to 1:10. Noise sources employed in the tests were white noise, impulsive noise (pistol shots), and the noise of a one-cylinder variable-speed gasoline engine. Steady-state and decay measurements agree approximately with predictions provided by a one-dimensional reverberation theory, which is a simplified version of room reverberation theory. Possible architectural measures for the reduction of “canyon effect” noise levels are presented and discussed in the following paper.
Portable Wide-Dynamic-Range Amplitude Statistical Analyzer for Community Noise Measurements
Authors: Blanchard Byron E.
Published: October 2005

An instrument has been developed to perform amplitude statistical analysis of noise level in dBA over an 80-dB range. The instrument is portable, battery powered, and automatic; it records the results of the analysis for each hour, along with the time and date, on digital cassette tape for later computer processing. The 80-dB range is analyzed into 64 bits of 1.25-dB width at a sampling rate of 8/sec. Each sample results in incrementing one of 64 binary counters of 15 bits each. Once an hour, the contents of the counters are dumped onto digital tape and the counters reset to zero. The digital cassette is read into a time-shared computer for further processing into, for example, L10, L50, etc., Leq, NPL, HNL, or CNEL. The instrument is powered by a 12-V car battery for more than a week. The digital cassette lasts for more than three days without attention.
A Parametric Source for Deep Scattering Layer Investigation
Authors: Konrad W. L. , Carlton L. F.
Published: October 2005

A driver-transducer system capable of producing difference frequency source levels of 225 dB//μPam at 14 kHz is described. The primary mean frequency is 65 kHz with source levels up to 245 dB//μPam per frequency. Performance of the array agrees well with the Mellen-Moffett design prediction. The results of a sea test using the transducer as both projector and hydrophone against the deep scattering layer are presented. [Work supported by the Naval Ship Systems Command.]
Sounds from the Implosions of Steel Cylinders under Water
Authors: Price Robert S. , Shuler Verna K.
Published: October 2005

The Naval Research Laboratory recently conducted an experimental study of the collapse of steel cylindrical tanks lowered or dropped into the ocean. Twenty-four compartmentalized tanks were tested varying in length from 34 to 171.5 in. and in diameter from 8 to 30 in. The Naval Ordnance Laboratory made measurements of the acoustic signals emitted into the water as the tanks collapsed. The 13 tanks that were lowered to depth were recovered for inspection so that damage mode could be correlated with the acoustic measurements. These steel tanks generated acoustic signatures that in most cases differed significantly from those produced by explosion bubbles or by most imploding glass bottles. The observed negative onset of the implosion signatures from the steel chambers was apparently caused by the formation of an inward buckle. The subsequent pressure history could be shown to depend on the number and relative size of the buckles that form. The final stage of the collapse was indicated by a positive pulse.
Response of End-Fire Array to a Transient Excitation
Authors: Shooter J. A.
Published: October 2005

The impulse response of a thin end-fire array is presented. Beam patterns were computed by detecting the root-mean-square value of the received waveform. The patterns show a smoothing and graduate disappearing of the nulls with decreasing pulse length. The explanation is that the waveforms show a sharp spike on the leading and trailing edge of the pulse at angles in the beam pattern that are normally nulls. Because of the detection technique, no null is apparent. The spike is not evident at the lobes of the pattern. [This work was sponsored by the Office of Naval Research.]
The Directional Characteristics of Cylindrical Receiving Arrays with Non-Uniform Hydrophone Response and Optimum Weighting
Authors: Queen W. C.
Published: October 2005

In a recent paper [“The Directivity of Sonar Receiving Arrays,” W. C. Queen, J. Acoust. Soc. Amer. 47 (1970)], the problem of determining the directional characteristics of arbitrary configurations of point-source hydrophones was considered and results given for the special case of a cylindrical array. It was assumed that the array was transparent and consisted of omnidirectional, uniformly weighted elements. In this paper we extend these earlier studies to include both the effects of nonuniform weights as well as nonomnidirectional hydrophone response. The question of optimal weighting is considered and closed-form solutions obtained. These analyses are applied to the special case of an array formed on a cylindrical surface in a ring-stave configuration and results given in the form of beam patterns and directivity curves. Included are the omnidirectional case as well the special case where the hydrophones exhibit cardioid response functions.