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Journal of the Acoustical Society of America

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Apr 1984

Volume 75, Issue 4, pp. 1041-1321

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Quantitative analysis of the pseudo‐Rayleigh phenomenon

J. H. M. T. van der Hijden

J. Acoust. Soc. Am. Volume 75, Issue 4, pp. 1041-1047 (1984); (7 pages) | Cited 4 times

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Acoustic pseudo‐waves (leaky waves) along a plane interface are studied in the space‐time domain. Analysis of the Green’s function as derived by the Cagniard–de Hoop method allows a quantitative description of the propagation of a pseudo‐Rayleigh pulse along a fluid/solid interface. This includes a condition which limits the range of existence and an equation for the travel speed. A pseudo‐Rayleigh factor which determines the relative strength of the phenomenon is defined and the decay of pulsed wave motion along the interface is studied.
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43.20.Bi Mathematical theory of wave propagation
43.35.Pt Surface waves in solids and liquids
68.35.Gy Mechanical properties; surface strains
68.35.Iv Acoustical properties

Fresnel diffraction: Some extensions of the theory

Takahi Hasegawa, Naoki Inoue, and Kiichiro Matsuzawa

J. Acoust. Soc. Am. Volume 75, Issue 4, pp. 1048-1051 (1984); (4 pages) | Cited 2 times

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The rigorous expansion developed by the present authors for the velocity potential of a circular piston source [J. Acoust. Soc. Am. 74, 1044–1047 (1983)] is extended to include the space average pressure in the nearfield and the Fresnel diffraction of incident plane waves by an infinitely thin rigid disk. The present theoretical framework has the remarkable advantage that it includes neither approximations nor numerical integrations.
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43.20.Rz Steady-state radiation from sources, impedance, radiation patterns, boundary element methods
43.20.Fn Scattering of acoustic waves
43.20.Bi Mathematical theory of wave propagation

A finite difference solution for the propagation of sound in near sonic flows

S. I. Hariharan and Harold C. Lester

J. Acoust. Soc. Am. Volume 75, Issue 4, pp. 1052-1061 (1984); (10 pages)

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An explicit time/space finite difference procedure is used to model the propagation of sound in a quasi‐one‐dimensional duct containing high Mach number subsonic flow. Nonlinear acoustic equations are derived by perturbing the time‐dependent Euler equations about a steady, compressible mean flow. The governing difference relations are based on a fourth‐order, two‐step (predictor–corrector) MacCormack scheme. Difference equations for the source and termination boundary conditions are derived from the appropriate characteristic relationships. The solution algorithm functions by switching on a time harmonic source and allowing the difference equations to iterate to a steady state. A significant advantage with this approach is that the nonlinear terms can be retained and evaluated with only modest additional computer cost above that required for a linear model calculation. The principal effect of the nonlinearities was to shift acoustical energy to higher harmonics. With increased source strengths, wave steepening was observed. This phenomenon suggests that the acoustical response may approach a shock behavior at a higher sound pressure level as the throat Mach number approaches unity. Where applicable, comparisons were made with the calculations from a linear finite element algorithm. On a peak level basis, good agreement between the nonlinear finite difference and linear finite element solutions was observed, even though a peak sound pressure level of about 150 dB occurred in the throat region. Nonlinear steady‐state waveform solutions are shown to be in excellent agreement with a nonlinear asymptotic theory.
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43.28.-g Aeroacoustics and atmospheric sound
43.25.Cb Macrosonic propagation, finite amplitude sound; shock waves

Time spread of acoustic signals reflecting from a fixed rough boundary

Michael H. Brill, Xavier Zabal, and Stanley L. Adams

J. Acoust. Soc. Am. Volume 75, Issue 4, pp. 1062-1070 (1984); (9 pages)

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The time spreading of a spherical‐wave impulse of acoustic power reflecting from a not‐too‐rough fixed boundary (such as an ocean bottom) is computed using a simple geometric‐acoustic model. In that model, a ray reflects specularly from each boundary facet, and arrives at the receiver if and only if the position and slope of the facet provide the requisite specular path. In this way, the probability of reception of a ray from a particular facet is tied to the slope distribution of the reflecting boundary (assuming the depth is constant). A general expression for time spread is derived, and computed time spreads are presented for particular source/receiver geometries. Appreciable time spreads (up to half a second) are shown for benign ocean‐boundary conditions (rms slope ≲tan 15°). Time spreads increase dramatically with rms slope, decrease with increasing range, and increase with increasing depth from source/receiver to the boundary. It is emphasized that, for broadband signals, models must be in three dimensions (as opposed to two) and depart from plane‐wave idealizations.
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43.30.Cq Ray propagation of sound in water
43.30.Gv Backscattering, echoes, and reverberation in water due to combinations of boundaries
43.20.Bi Mathematical theory of wave propagation
43.60.Gk Space-time signal processing, other than matched field processing

Calculation of the second term in the geometrical acoustic expansion

D. C. Stickler, J. Tavantzis, and E. Ammicht

J. Acoust. Soc. Am. Volume 75, Issue 4, pp. 1071-1076 (1984); (6 pages)

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One method to estimate the validity of the leading term in the geometrical acoustics expansion is to calculate the second term in the expansion. This second term plays an essential role in the construction of the second term in Ludwig’s uniform smooth caustic ansatz and in the construction of the progressing wave smooth caustic ansatz. Therefore, the determination of this term is important in the assessment of each of these expansions. Expressions for this second term are derived valid for arbitrary index profiles near the source point and near a smooth caustic. Numerical examples are presented for the n2‐linear profile. These results show that near the source the second term is not singular, but near the caustic it is more singular than the leading order term.
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43.30.Cq Ray propagation of sound in water
43.20.Bi Mathematical theory of wave propagation
43.20.Dk Ray acoustics

Acoustic scattering by fish—Acoustic models and a two‐parameter fit

C. S. Clay and Barry G. Heist

J. Acoust. Soc. Am. Volume 75, Issue 4, pp. 1077-1083 (1984); (7 pages) | Cited 4 times

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The Rician probability density function (PDF) accurately describes acoustic scattering by individual live fish. A fish scattering model is proposed which contains concentrated scattering components (representing the swim bladder) and distributed scattering components (representing the fish skeletal structure). We associate the gross directional backscattering cross section with the concentrated component. The ratio of these two components, γ, may serve as a means of identifying different fish species and fish behavior from the scattering PDFs. Monte Carlo simulations of a swimming fish using the acoustic fish model give Rician PDFs.
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43.80.Jz Use of acoustic energy (with or without other forms) in studies of structure and function of biological systems
43.30.Gv Backscattering, echoes, and reverberation in water due to combinations of boundaries
43.20.Fn Scattering of acoustic waves

An improvement in the range resolution of ultrasonic pulse echo systems by deconvolution

R. N. Carpenter and P. R. Stepanishen

J. Acoust. Soc. Am. Volume 75, Issue 4, pp. 1084-1091 (1984); (8 pages) | Cited 1 time

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Ultrasonic pulse echo systems are often limited in range resolution by the bandwidth of the piezoelectric transducer. Significant improvements in the range resolution of such systems can be obtained by minimizing the effects of the transducer’s dynamic response on the overall pulse echo process. An approach to minimize the effects of the transducer is developed from linear system and impulse response techniques. In essence, the pulse echo voltage of interest is deconvolved with a pulse echo reference voltage which is obtained from an air/water interface in the nearfield of the transducer. A computer study of the pulse echo process and the deconvolution process is presented to illustrate the nature of the improvement in range resolution for several cases of interest. Finally, experimental results are presented to illustrate the improvement using commercially available transducers.
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43.30.Yj Transducers and transducer arrays for underwater sound; transducer calibration
43.35.Yb Ultrasonic instrumentation and measurement techniques
43.60.Gk Space-time signal processing, other than matched field processing

Development and application of multiple input models for structural noise source identification of forge hammers. Part I: Development

Martin W. Trethewey and Harold A. Evensen

J. Acoust. Soc. Am. Volume 75, Issue 4, pp. 1092-1098 (1984); (7 pages)

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The application of multiple input models to analyze structurally generated noise from a forge hammer is discussed. Part I of this article is intended to present the rationale for application and analysis of multiple input models for noise source identification. The development of the empirical models is reviewed and investigated to show how the terms in the model can be interpreted to mathematically simulate the selective wrapping approach to source identification. The interaction of the structural excitation forces and radiated structural noise is examined for a four‐piece forge hammer and provides an indication of the characteristic measurements needed to develop the multiple input model that is representative of the hammer’s sound radiation. The transducer requirements for application to forge hammers are examined through experiments performed on a laboratory test structure and a forge hammer column. The results indicate that a single, well‐placed transducer may be sufficient to characterize the sound radiation from a monolithic element. Part II of this article analyzes the application of the techniques to a Chambersburg ♯8 die forger under production conditions. The experimental results indicate that the ram is the dominant source of sound energy, the columns are secondary sources, and the yoke and anvil are minor sources when detected through a microphone at the operator’s position.
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43.40.At Experimental and theoretical studies of vibrating systems
43.50.Jh Noise in buildings and general machinery noise
43.50.Ed Noise generation

Development and application of multiple‐input models for structural noise source identification of forge hammers. Part II: Application

Martin W. Trethewey and Harold A. Evensen

J. Acoust. Soc. Am. Volume 75, Issue 4, pp. 1099-1104 (1984); (6 pages)

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The application of multiple‐input models to analyze structurally generated noise from a forge hammer is discussed. Part I of this article presented the rationale for developing and interpreting multiple‐input models for structural noise source identification. An investigation of the transducer requirements for characterizing the sound radiation from a monolithic element showed that a single, well‐placed accelerometer may be sufficient for each element. Part II of this article analyzes the application of the multiple‐input modeling technique to the structural noise source identification of a Chambersburg ♯8 die forger. A comparison of three‐, five‐, and seven‐input models applied to the forge hammer under production conditions indicates that as few as five transducers would suffice to characterize the sound contributions of the five structural elements. Analysis of these models indicates that the ram is the dominant source of sound energy, the columns are secondary sources, and the yoke and anvil are minor sources when detected through a microphone at the operator’s position. The analysis also shows that the coupling between the hammer structural elements is sufficient to render conventional wrapping identification methods unreliable for analyzing hammer noise.
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43.40.At Experimental and theoretical studies of vibrating systems
43.50.Jh Noise in buildings and general machinery noise
43.50.Ed Noise generation

In‐plane accelerations and forces on frequency changes in doubly rotated quartz plates

P. C. Y. Lee and Kuang‐Ming Wu

J. Acoust. Soc. Am. Volume 75, Issue 4, pp. 1105-1117 (1984); (13 pages) | Cited 2 times

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Two‐dimensional equations of motion of doubly rotated quartz plates for the thickness‐shear, flexure, and extensional vibrations under in‐plane initial stresses are employed to predict changes in the fundamental thickness‐shear frequencies due to initial stresses. Two types of initial stresses are considered: (1) stresses due to a pair of diametral forces, and (2) stresses due to steady accelerations for a three‐point ‘‘T’’‐shaped mount and a four point ‘‘+’’‐shaped mount configurations. Force sensitivity and acceleration sensitivity coefficients are computed and compared with experimental data and existing computed results. For both ‘‘T’’‐ and ‘‘+’’‐shaped mount configurations, mount orientations corresponding to the maximum and minimum of acceleration sensitivity are predicted.
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43.40.Dx Vibrations of membranes and plates
43.40.At Experimental and theoretical studies of vibrating systems
62.30.+d Mechanical and elastic waves; vibrations

Free vibration of a point‐supported circular cylindrical shell

Toshihiro Irie, Gen Yamada, and Yasunori Kudoh

J. Acoust. Soc. Am. Volume 75, Issue 4, pp. 1118-1123 (1984); (6 pages)

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An analysis is presented for the free vibration of an elastically or rigidly point‐supported circular cylindrical shell. For this purpose, the deflection displacements of the shell are written in a series of the deflection functions of beam. The dynamical energies of the shell are evaluated, and the frequency equation is derived by the Ritz method. For a rigidly point‐supported shell, the Lagrangian multiplier method is conveniently employed. The method is applied to a cylindrical shell supported at symmetrically located four points; the natural frequencies and the mode shapes are calculated numerically, and the effects of the point supports on the vibration are studied.
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43.40.Ey Vibrations of shells
43.40.At Experimental and theoretical studies of vibrating systems

State prediction for environmental noise system by a generalized adaptive function model

Mitsuo Ohta, Eiji Uchino, and Kazutatsu Hatakeyama

J. Acoust. Soc. Am. Volume 75, Issue 4, pp. 1124-1132 (1984); (9 pages)

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This paper describes a new attempt at predicting the fluctuations of an environmental noise system by use of adaptive functions. Principally, some kinds of correlations (in some cases, in the form of simple inertia) can be actually found between the recorded past data and the future values. In this paper, the basic correlative relationship of this kind is expressed universally with the use of newly introduced adaptive functions of both the linear, and nonlinear types. In prediction algorithms, discrete or continuous adaptive functions of orthogonal and/or nonorthogonal types are employed to meet the actual observation mechanism and to get the effective data processing. Finally, the effectiveness of the proposed method has been confirmed by computer simulation and application to actually obtained road traffic noise data.
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43.50.Lj Transportation noise sources: air, road, rail, and marine vehicles
43.50.Qp Effects of noise on man and society
43.60.Gk Space-time signal processing, other than matched field processing

A fast and simple nonlinear technique for high‐resolution beamforming and spectral analysis

Ronald A. Wagstaff and Jean‐Louis Berrou

J. Acoust. Soc. Am. Volume 75, Issue 4, pp. 1133-1141 (1984); (9 pages)

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A simple and fast approximate technique is presented that can generate high‐resolution spectra from the output of a conventional processor (i.e., beamformer or spectrum analyzer). The speed and simplicity of the algorithm results from two factors: the input data of the system are not used, only the conventional output data; the significant mathematical operations are simple additions and subtractions of log‐transformed quantities, which make the technique nonlinear in power and avoids matrix inversions. As a result, the technique is extremely robust, being relatively insensitive to system and measurement errors. The algorithm is explained and results are presented for both modeled and measured data.
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43.60.Gk Space-time signal processing, other than matched field processing

Imaging through an inhomogeneous layer by least‐mean‐square error fitting

Makoto Hirama and Takuso Sato

J. Acoust. Soc. Am. Volume 75, Issue 4, pp. 1142-1147 (1984); (6 pages)

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A new active ultrasonic imaging method through an inhomogeneous layer is proposed. It has the special feature that its effectiveness does not depend on the class of the objects to be imaged. In this method, first, a set of data is acquired by repeating transmission and reception for all possible combinations of pairs of transducers on the array, then the spatial frequency components of the object and the structure of the inhomogeneous layer are estimated from these data by means of least‐mean‐square error fitting. Since the data have redundancy for the parameters to be estimated, this process gives an optimum and stable estimation algorithm even when measurement errors and noise are included. The image is reconstructed from the estimated spatial frequency components through inverse Fourier transform. The effectiveness of this method is ensured by several numerical analyses and experiments.
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43.60.Gk Space-time signal processing, other than matched field processing
43.35.Sx Acoustooptical effects, optoacoustics, acoustical visualization, acoustical microscopy, and acoustical holography

Interrelation of different oto‐acoustic emissions

Eberhard Zwicker and Eberhard Schloth

J. Acoust. Soc. Am. Volume 75, Issue 4, pp. 1148-1154 (1984); (7 pages) | Cited 32 times

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Amplitude and phase of the sound pressure measured in the closed ear canal during stimulation with pure tones have been monitored as a function of frequency for subjects with and without measurable spontaneous emissions. The frequency spacing between neighboring maxima of the evoked emissions is closely related to that found between neighboring spontaneous emissions. Similar data are found with delayed evoked emissions. All three catagories, spontaneous, delayed, and synchronous evoked emissions are closely related to each other and to the fine structure of threshold in quiet.
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43.64.Kc Cochlear mechanics
43.64.Yp Instruments and methods

Diversity of adaptation patterns in responses of eighth nerve fibers in the bullfrog, Rana catesbeiana

Andrea L. Megela

J. Acoust. Soc. Am. Volume 75, Issue 4, pp. 1155-1162 (1984); (8 pages) | Cited 1 time

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The firing patterns of eighth nerve fibers in the bullfrog, Rana catesbeiana, were analyzed for responses to long duration tone bursts at best excitatory frequency (BEF) and at frequencies along the upper and lower boundaries of the excitatory tuning curve of each fiber. These firing patterns were used as an index of the degree of short‐term adaptation of each fiber. Amphibian papilla fibers (with BEFs 100–1000 Hz) exhibited marked diversity in their firing patterns to BEF tones, ranging from very flat or tonic (sustained responses throughout the duration of the stimulus) to very peaked or phasic (responding primarily or exclusively to stimulus onset). Moreover, the degree of short‐term adaptation shown by an individual fiber varied with stimulating frequency. The firing patterns of amphibian papilla fibers tended to become more tonic as stimulus frequency was lowered below BEF; conversely, as stimulus frequency was increased above BEF, firing patterns either showed little change from that at BEF, or became more phasic. A similar frequency dependence of adaptation has not been reported in responses of mammalian eighth nerve fibers with comparable BEFs. The firing patterns of basilar papilla fibers (BEFs greater than 1000 Hz) remained similar in response to both BEF and non‐BEF tones. These data reveal that the firing patterns and degrees of short‐term adaptation of amphibian papilla fibers vary considerably across the tuning curve, whereas those of basilar papilla fibers remain relatively more constant with changes in stimulating frequency. These data also suggest that the amphibian papilla and the basilar papilla may play different roles in the use of physiological adaptation as a dimension for coding of information in sounds by the bullfrog.
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43.64.Pg Electrophysiology of the auditory nerve
43.64.Tk Physiology of sound generation and detection by animals

Profile analysis: Critical bands and duration

David M. Green, Christine R. Mason, and Gerald Kidd, Jr.

J. Acoust. Soc. Am. Volume 75, Issue 4, pp. 1163-1167 (1984); (5 pages) | Cited 12 times

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The detection of an increment in the intensity of the central component of a multi‐component complex was measured as a function of the frequency spacing of the components and the duration of the presentation. The overall intensity of the complex was randomly varied on each presentation of the stimulus. Curiously, the increment becomes easier to hear as the range and density of the surrounding complex is increased. This increase in range and density is also effective in improving the detectability of the increment when there is no random variation in intensity, i.e., a conventional Weber fraction experiment. This is unlike the results obtained in many other critical‐band experiments where energy remote from the signal frequency has little or no effect. Measurement of the effects of signal duration showed that when presentations were shorter than about 100 msec a greater increment in intensity was required than for longer durations. These results with duration are similar to those obtained in other intensity‐discrimination tasks.
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43.66.Fe Discrimination: intensity and frequency
43.66.Ba Models and theories of auditory processes
43.66.Dc Masking

Identification and discrimination of rise time: Is it categorical or noncategorical?

Diane Kewley‐Port and David B. Pisoni

J. Acoust. Soc. Am. Volume 75, Issue 4, pp. 1168-1176 (1984); (9 pages) | Cited 2 times

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Previous studies have reported that rise time of sawtooth waveforms may be discriminated in either a categorical‐like manner under some experimental conditions or according to Weber’s law under other conditions. In the present experiments, rise time discrimination was examined with two experimental procedures: the traditional labeling and ABX tasks used in speech perception studies and an adaptive tracking procedure used in psychophysical studies. Rise time varied from 0 to 80 ms in 10‐ms intervals for sawtooth signals of 1‐s duration. Discrimination functions for subjects who simply discriminated the signals on any basis whatsoever as well as functions for subjects who practiced labeling the endpoint stimuli as ‘‘pluck’’ and ‘‘bow’’ before ABX discrimination were not categorical in the ABX task. In the adaptive tracking procedure, the Weber fraction obtained from the jnds of rise time was found to be a constant above 20‐ms rise time. The results from the two discrimination paradigms were then compared by predicting a jnd for rise time from the ABX discrimination data by reference to the underlying psychometric function. Using this method of analysis, discrimination results from previous studies were shown to be quite similar to the discrimination results observed in this study. Taken together the results demonstrate clearly that rise time discrimination of sawtooth signals follows predictions derived from Weber’s law.
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43.66.Fe Discrimination: intensity and frequency
43.66.Lj Perceptual effects of sound
43.66.Mk Temporal and sequential aspects of hearing; auditory grouping in relation to music

Amplitude modulation thresholds in chinchillas with high‐frequency hearing loss

Donald Henderson, Richard Salvi, Gary Pavek, and Roger Hamernik

J. Acoust. Soc. Am. Volume 75, Issue 4, pp. 1177-1183 (1984); (7 pages) | Cited 1 time

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Estimates of auditory temporal resolution were obtained from normal chinchillas using sinusoidally amplitude modulated noise. Afterwards, the animals were exposed to noise whose bandwidth was progressively increased toward the low frequencies in octave steps. The first exposure was to an octave band of noise centered at 8 kHz. Three additional octave bands of noise were subsequently added to the original exposure in order to progressively increase the extent of the high‐frequency hearing loss. The first exposure produced a temporary hearing loss of 50 to 60 dB near 8 kHz and elevated the amplitude modulation thresholds primarily at intermediate (128 Hz) modulation frequencies. Successive noise exposures extended the temporary hearing loss toward lower frequencies, but there was little further deterioration in the amplitude modulation function until the last exposure when the hearing loss spread to 1 kHz. The degradation in the amplitude modulation function observed after the last exposure, however, was due to a reduction in the sensation level of the test signal rather than to a decrease in the hearing bandwidth. The results of this study suggest that the high‐frequency regions of the cochlea may be important for temporal resolution.
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43.66.Gf Detection and discrimination of sound by animals
43.66.Sr Deafness, audiometry, aging effects
43.66.Mk Temporal and sequential aspects of hearing; auditory grouping in relation to music

Detection of frequency and rate modulation by the chinchilla

Glenis R. Long and William W. Clark

J. Acoust. Soc. Am. Volume 75, Issue 4, pp. 1184-1190 (1984); (7 pages) | Cited 1 time

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Thresholds for detection of frequency modulation were obtained from three chinchillas at seven frequencies between 320 Hz and 12.75 kHz and at three sensation levels (20, 40, 60 dB SL). These were compared with human thresholds at 40 dB SL obtained under the same conditions. Our data show no significant impact of sensation level and confirm Nelson and Kiester’s [J. Acoust. Soc. Am. 64, 114–126 (1978)] report that the chinchilla has much poorer frequency discrimination than man. In an attempt to assess separately the chinchilla’s capacity for temporal coding of frequency, thresholds for detection of a change in the modulation rate of amplitude modulated noise were obtained for nine modulation frequencies between 10 and 320 Hz. The chinchillas needed a larger rate change for detection than most humans but the relative difference was less than that found for frequency modulation detection.
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43.66.Gf Detection and discrimination of sound by animals
43.66.Mk Temporal and sequential aspects of hearing; auditory grouping in relation to music
43.80.Lb Sound reception by animals: anatomy, physiology, auditory capacities, processing

Interaural intensity discrimination: Insensitivity at 1000 Hz

D. Wesley Grantham

J. Acoust. Soc. Am. Volume 75, Issue 4, pp. 1191-1194 (1984); (4 pages) | Cited 9 times

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Recent data from three laboratories have replicated Mills’ [J. Acoust. Soc. Am. 32, 132–134 (1960)] finding that interaural intensity discrimination is relatively poorer for tones of 1000 Hz than for tones of either higher or lower frequencies. To get a finer look at this frequency effect, interaural intensity difference thresholds were obtained from four subjects for tones of several frequencies around 1000 Hz. An adaptive two‐interval forced‐choice procedure was employed, in which the overall intensity of the signals was varied randomly in order to prevent subjects from listening to monaural loudness changes. Despite large intersubject differences in overall sensitivity to interaural intensity differences, all four subjects showed a local peak in their threshold functions at or near 1000 Hz. This curious ‘‘1000‐Hz effect’’ might be explained by imagining that an interaural intensity comparator operates more efficiently as frequency increases, but that a peripheral interaural intensity difference to interaural‐time difference conversion contributes to laterality judgments for low‐frequency tones, thus acting to lower thresholds again for frequencies below 1000 Hz.
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43.66.Pn Binaural hearing
43.66.Fe Discrimination: intensity and frequency

The influence of pinnae‐based spectral cues on sound localization

Alan D. Musicant and Robert A. Butler

J. Acoust. Soc. Am. Volume 75, Issue 4, pp. 1195-1200 (1984); (6 pages) | Cited 16 times

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The role of pinnae‐based spectral cues was investigated by requiring listeners to locate sound, binaurally, in the horizontal plane with and without partial occlusion of their external ears. The main finding was that the high frequencies were necessary for optimal performance. When the stimulus contained the higher audio frequencies, e.g., broadband and 4.0‐kHz high‐pass noise, localization accuracy was significantly superior to that recorded for stimuli consisting only of the lower frequencies (4.0‐ and 1.0‐kHz low‐pass noise). This finding was attributed to the influence of the spectral cues furnished by the pinnae, for when the stimulus composition included high frequencies, pinnae occlusion resulted in a marked decline in localization accuracy. Numerous front–rear reversals occurred. Moreover, the ability to distinguish among sounds originating within the same quadrant also suffered. Performance proficiency for the low‐pass stimuli was not further degraded under conditions of pinnae occlusion. In locating the 4.0‐kHz high‐pass noise when both, neither, or only one ear was occluded, the data demonstrated unequivocally that the pinna‐based cues of the ‘‘near’’ ear contributed powerfully toward localization accuracy.
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43.66.Qp Localization of sound sources
43.66.Ba Models and theories of auditory processes

Concurrent minimum audible angle: A re‐examination of the concept of auditory spatial acuity

David R. Perrott

J. Acoust. Soc. Am. Volume 75, Issue 4, pp. 1201-1206 (1984); (6 pages) | Cited 9 times

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Minimum audible angle was measured for simultaneous acoustic events. Localization of concurrent events was found to be a direct function of the spectral differences between the events, the angle between the sources, and the location of the sources within the field defined by the subject. In the latter case, the m.a.a. was smallest with sources placed symmetrically about the listener’s median plane and maximal at the extreme lateral portions. Post‐hoc tests were completed which indicate that the spectral limits for concurrent localization is dependent both upon the angular separation of the sources and the position within the field as defined by the locus of the subject. The functions obtained approach the values reported by Mills [J. Acoust. Soc. Am. 30, 237–246 (1958)] as the temporal overlap between the concurrent events decreased. The present results suggest that a single localization function may exist with the optimal performance observed with fully successive stimuli and poorest performance in the condition involving simultaneous events. The implications of these results are discussed.
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43.66.Qp Localization of sound sources
43.66.Pn Binaural hearing

Speech recognition in a special case of low‐frequency hearing loss

Dianne J. Van Tasell and Christopher W. Turner

J. Acoust. Soc. Am. Volume 75, Issue 4, pp. 1207-1212 (1984); (6 pages) | Cited 3 times

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Recognition of speech stimuli consisting of monosyllabic words, sentences, and nonsense syllables was tested in normal subjects and in a subject with a low‐frequency sensorineural hearing loss characterized by an absence of functioning sensory units in the apical region of the cochlea, as determined in a previous experiment [C. W. Turner, E. M. Burns, and D. A. Nelson, J. Acoust. Soc. Am. 73, 966–975 (1983)]. Performance of all subjects was close to 100% correct for all stimuli presented unfiltered at a moderate intensity level. When stimuli were low‐pass filtered, performance of the hearing‐impaired subject fell below that of the normals, but was still considerably above chance. A further diminution in the impaired subject’s recognition of nonsense syllables resulted from the addition of a high‐pass masking noise, indicating that his performance in the filtered quiet condition was attributable in large part to the contribution of sensory units in basal and midcochlear regions. Normals’ performance was also somewhat decreased by the masker, suggesting that they also may have been extracting some low‐frequency speech cues from responses of sensory units located in the base of the cochlea.
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43.66.Sr Deafness, audiometry, aging effects
43.71.Ky Speech perception by the hearing impaired

The discrimination of vowel duration by infants

Rebecca E. Eilers, Dale H. Bull, D. Kimbrough Oller, and Diana C. Lewis

J. Acoust. Soc. Am. Volume 75, Issue 4, pp. 1213-1218 (1984); (6 pages) | Cited 1 time

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Three groups of nine 5–11‐month‐old infants provided evidence of discrimination of speechlike stimuli differing only in vowel duration. Ease of discrimination was directly related to the magnitude of the ratio of the longer to shorter vowel. Group one infants discriminated three vowel duration contrasts (with ratios of 0.33, 0.67, and 1.0) embedded in a synthetic /mad/ syllable; group two discriminated these same duration contrasts within the bisyllable /samad/, and group three in the trisyllable /masamad/. In all cases, the contrasting durations were carried by the last vowel of the synthetic word. These same three infant groups failed to provide evidence of discrimination of a final position released stop consonant contrast (/mat/ versus /mad/) cued by voice excitation during closure of the /d/ and not the /t/. These results suggest that vowel duration may be a primary cue for infants’ perception of the voicing of final position stop consonants.
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43.71.Ft Development of speech perception
43.71.Es Vowel and consonant perception; perception of words, sentences, and fluent speech
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