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Oct 1977

Volume 62, Issue 4, pp. 813-1059

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Effect of arbitrary temperature and flow profiles on the speed of sound in a pipe

Baldwin Robertson

J. Acoust. Soc. Am. Volume 62, Issue 4, pp. 813-818 (1977); (6 pages) | Cited 1 time

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An expression for the wave number of the fundamental mode in a uniform duct of arbitrary cross section is obtained correct to first order in V/c and in c1/c0, where V is the velocity of the fluid in the pipe and c=c0+c1 is the local sound speed with c0=constant. Correction terms are also obtained. The calculation applies to a fluid with the equation of state of an ideal gas or to a liquid and is valid for frequencies well below cutoff. The speed c and velocity V can be arbitrary functions of position across the cross section given only ∇⋅V=0. For the gas, the wave number, which is a constant, can be expressed in terms of the average density and the total mass flow rate.
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43.20.Hq Velocity and attenuation of acoustic waves
43.20.Mv Waveguides, wave propagation in tubes and ducts

Coaxial circular spherical array for ultrasonic imaging

Shuji Shibata, Toshio Koda, Shuntetsu Matsumoto, and Joji Yamaga

J. Acoust. Soc. Am. Volume 62, Issue 4, pp. 819-824 (1977); (6 pages) | Cited 1 time

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A new type of receiving array has been designed and constructed for the purpose of visualizing an object by using the ultrasonic wave at low frequency. The array consists of 24 receiving elements around three coaxial circles on a spherical surface. The multiplication of three additive outputs, each of which is the sum of the eight element outputs on a circle, is performed for the signal processing of the array. After theoretical and experimental studies, it has been proven that the array proposed is appropriate for the high‐directional receiver in the ultrasonic visualization system. A preliminary experiment on imaging the object in the air has been carried out.
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43.20.Rz Steady-state radiation from sources, impedance, radiation patterns, boundary element methods
43.35.Sx Acoustooptical effects, optoacoustics, acoustical visualization, acoustical microscopy, and acoustical holography
43.38.Ar Transducing principles, materials, and structures: general
43.60.Gk Space-time signal processing, other than matched field processing

Propagation of noise along a finite impedance boundary

C. I. Chessell

J. Acoust. Soc. Am. Volume 62, Issue 4, pp. 825-834 (1977); (10 pages) | Cited 8 times

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The purpose of this paper is to show that the essential properties of the propagation of noise over real soil surfaces can be modeled using the properties of fibrous absorbent materials and the appropriate theory of sound propagation along a finite impedance boundary. This work is an expansion of the original proposals of Delany and Bazley [J. Sound Vib. 16, 315–322 (1971)]. The model predictions are shown to be in good agreement with a variety of experimental ground absorption measurements. The application of the model to the measurement of aircraft reverse thrust noise is described.
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43.28.Fp Outdoor sound propagation through a stationary atmosphere, meteorological factors
43.50.Vt Topographical and meteorological factors in noise propagation

Reflection and radiation due to a quadrupole near a fluid interface

R. Dash

J. Acoust. Soc. Am. Volume 62, Issue 4, pp. 835-846 (1977); (12 pages)

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The reflexion of sound at an interface between two fluid half‐space in relative motion (one in motion and the other at rest), but with different density and different sound speed, is considered. The reflexion coefficient is found in an explicitly closed form. A brief discussion of the effect of mean velocity, i.e., corresponding Mach number on different noise generating quadrupole directivities concludes the presentation of the paper. The graphs provided illustrate this aspect in a rather more convincing way.
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43.28.Ra Generation of sound by fluid flow, aerodynamic sound and turbulence
43.20.Fn Scattering of acoustic waves
43.20.Bi Mathematical theory of wave propagation

Project TASMAN TWO: Low‐frequency propagation measurements in the South Tasman Sea

R. W. Bannister, R. N. Denham, K. M. Guthrie, and D. G. Browning

J. Acoust. Soc. Am. Volume 62, Issue 4, pp. 847-859 (1977); (13 pages)

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In February 1975, a low‐frequency acoustic propagation experiment, Project TASMAN TWO, was conducted in the South Tasman Sea to the west of the South Island of New Zealand. The aim was to study low‐frequency acoustic propagation in a region having considerable variations in oceanic environment. Sound signals, set to detonate at the SOFAR axis, were air dropped out to a range of 3000 km along paths chosen to sample three distinct water masses (Subtropical Water, Australasian Subantarctic Water, and Circumpolar Subantarctic Water) and to traverse a large bathymetric feature (the South Tasman Rise). In addition, convergence zone (RSR) propagation in subtropical water was investigated with the sound signals set to detonate at 18 m. The results show that the three individual water masses have significantly different attenuations at frequencies below 500 Hz. It was established that Circumpolar Subantarctic Water has an attenuation nearly five times greater than Subtropical Water. For SOFAR propagation conditions, major acoustic transmission discontinuities were not always observed at the boundaries between the water masses, but strong shadowing effects with level changes of as much as 15 dB were clearly associated with the bathymetric feature. The experimental data for both SOFAR and RSR propagation are compared with predictions made using the parabolic equation model. Although the predictions match most general trends observed in the data, poor‐detailed agreement was achieved where rapid horizontal gradients in either sound speed or water depth were experienced.
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43.30.Bp Normal mode propagation of sound in water

General analysis of ocean eddy effects for sound transmission applications

R. F. Henrick, W. L. Siegmann, and M. J. Jacobson

J. Acoust. Soc. Am. Volume 62, Issue 4, pp. 860-870 (1977); (11 pages)

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An analytical approach is used to obtain an approximate solution for deep‐ocean mesoscale eddies, including depth‐dependent effects. The solution is used in the development of an environmental‐acoustics model which relates acoustically relevant quantities, such as sound‐speed and current distributions, to eddy parameters. Parameters of the model are depth of influence, radius, rotational direction and maximum speed, and translational velocity. An application to a particular Gulf Stream ring is made, and the resulting current and sound‐speed structures are shown to be in qualitative agreement with observations. Then, general results are presented for rotational current structure, maximum horizontal sound‐speed change, and maximum SOFAR‐axis elevation as functions of eddy radius and peak current speed. It is shown explicitly how these quantities change significantly with eddy size and strength. This model provides a basis for subsequent analytical studies of sound transmission through an arbitrary eddy or eddy field.
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43.30.Bp Normal mode propagation of sound in water
43.28.Py Interaction of fluid motion and sound, Doppler effect, and sound in flow ducts

Excess sound propagation loss in a stochastic environment

Hans G. Schneider

J. Acoust. Soc. Am. Volume 62, Issue 4, pp. 871-877 (1977); (7 pages)

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A general scheme is outlined to explain excess propagation loss in sound channels in which the stochastic variability of the environment is combined with the usual deterministic effects on propagation loss. As a result, the excess loss in a shallow‐water sound channel is a function of both the bottom loss and the stochastic variability of the environment. A method, previously reported by the author [Acustica 35, 18 (1976)], to ’’handle’’ stochastic sound‐speed variations in ray‐tracing routines is applied to model the excess propagation loss for the Hudson Bay data. The model results are in excellent agreement with the experimental data without fitting the parameters and exhibit the dependence predicted above. Additional results show that difficulties usually inherent in deterministic ray tracing as, for example, the focusing of energy at the source depth and sharp caustics may easily be overcome by using a stochastic ray‐tracing approach.
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43.30.Bp Normal mode propagation of sound in water
43.30.Ft Volume scattering
43.20.Dk Ray acoustics
43.60.Cg Statistical properties of signals and noise

Models for the amplitude fluctuations of narrow‐band signals and noise in the sea

R. J. Urick

J. Acoust. Soc. Am. Volume 62, Issue 4, pp. 878-887 (1977); (10 pages) | Cited 3 times

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The sound from a steady distant sinusoidal source in the sea fluctuates in amplitude because of multipath interferences, surface motion, internal waves, microstructure, and other propagation effects. Such fluctuations are found to follow a Rician or modified‐Rayleigh distribution having as a parameter the fraction of random power in the received signal. At the output of conventional sonar processor—namely, a narrow‐band filter, a squarer, and an integrator—the fluctuation statistics are determined by the propagation processes occurring in the sea between source and receiver. On the other hand, ambient sea noise is found, from analyses of field recordings, to have fluctuation statistics determined by the processor itself; ambient noise samples at the processor output obey a chi‐square distribution having a number of degrees of freedom equal to twice the bandwidth‐time product of the processor, as would be expected from a Gaussian input. The two distributions—Rician power for signals and chi‐square for noise—while formally different, have remarkable similarities in the limits. In short, the output fluctuation statistics of narrow‐band signals and Gaussian ambient noise can apparently be predicted from estimates of the degree of randomness introduced by the prevailing propagation conditions, and from a knowledge of the processor, respectively, provided these statistics remain stationary during the analysis period.
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43.30.Bp Normal mode propagation of sound in water
43.30.Nb Noise in water; generation mechanisms and characteristics of the field

Coherent ray propagation through a Gulf Stream ring

N. L. Weinberg and X. Zabalgogeazcoa

J. Acoust. Soc. Am. Volume 62, Issue 4, pp. 888-894 (1977); (7 pages)

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A model study using coherent ray‐tracing methods was based on a cold‐water eddy measured in the Altantic. The eddy, whose diameter is about 320 km was passed through the propagation path at a rate of 4 km/day between a fixed source and fixed receiver separated 445 km. A brief description of the range‐dependent model required for this study is presented. The major effects of the eddy were in changing the ray type (number of cycles, RR–RSR, etc.) and in the travel time which had a major effect on the multipath resultant. Time series of amplitude and phase are presented with a qualitative comparison to previously measured data.
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43.30.Bp Normal mode propagation of sound in water
43.20.Dk Ray acoustics
43.28.Py Interaction of fluid motion and sound, Doppler effect, and sound in flow ducts

Reciprocal acoustic transmission in a midocean environment

Peter F. Worcester

J. Acoust. Soc. Am. Volume 62, Issue 4, pp. 895-905 (1977); (11 pages) | Cited 6 times

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The simultaneous transmissions of acoustic pulses in opposite directions between two point in midocean has been used to separate current effects on acoustic propagation from the effects of sound‐speed structure. Two 2250‐Hz sources were suspended at about 1‐km depth from ships 25 km apart. Differences in travel times of the oppositely traveling pulses are interpreted in terms of ray‐averaged currents. The measured values contain an unknown contribution from ship drift, but differences between upper and lower ray paths present in this experiment are insensitive to drift. Further, the nonreciprocity in pulse shape and amplitude of opposing transmissions is found to be of the same order as the fluctuations in time of one‐way transmissions. This demonstrates that currents contribute to acoustic scattering.
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43.30.Bp Normal mode propagation of sound in water
43.20.Dk Ray acoustics
43.28.Py Interaction of fluid motion and sound, Doppler effect, and sound in flow ducts

Band‐limited power flow into enclosures

L. D. Pope and J. F. Wilby

J. Acoust. Soc. Am. Volume 62, Issue 4, pp. 906-911 (1977); (6 pages) | Cited 5 times

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Acoustic power flow into enclosures is an area of continued research activity. Recent work [Pope and Wilby, BBN Report No. 3286 (1976)] resulted in band‐limited power‐flow formulations which account for coupling of structure and volume modes in the low‐frequency regime and allow for a general description of the exciting external random pressure fields.
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43.55.Ka Computer simulation of acoustics in enclosures, modeling
43.55.Cs Stationary response of rooms to noise; spatial statistics of room response; random testing
43.20.Tb Interaction of vibrating structures with surrounding medium

Multimicrophone signal‐processing technique to remove room reverberation from speech signals

J. B. Allen, D. A. Berkley, and J. Blauert

J. Acoust. Soc. Am. Volume 62, Issue 4, pp. 912-915 (1977); (4 pages) | Cited 9 times

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It is well known that room reverberation can significantly impair one’s perception of sounds recorded by a microphone in that room. Acoustic recordings produced in untreated rooms are characterized by a hollow echolike quality resulting from not locating the microphone close to the source. In this paper we discuss a multimicrophone digital processing scheme for removing much of the degrading distortion. To accomplish this the individual microphone signals are divided into frequency bands whose corresponding outputs are cophased (delay differences are compensated) and added. Then the gain of each resulting band is set based on the cross correlation between corresponding microphone signals in that band. The reconstructed broadband speech is perceived with considerably reduced reverberation.
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43.55.Br Room acoustics: theory and experiment; reverberation, normal modes, diffusion, transient and steady-state response
43.55.Jz Sound-reinforcement systems for rooms and enclosures
43.72.-p Speech processing and communication systems
43.60.Gk Space-time signal processing, other than matched field processing

Spectrum of the lagged product in cross correlation

Henry M. Beisner

J. Acoust. Soc. Am. Volume 62, Issue 4, pp. 916-921 (1977); (6 pages)

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Cross correlation has been used for detection and parameter estimation of moving targets. Various correlation techniques consist, essentially, of methods to estimate the spectrum of the correlation product. The power spectrum of the lagged product is derived. The time‐lag dependence of the spectrum is found to be proportional to the squared magnitude of the signal autocorrelation function. It is shown that, if the bandwidth of the modulation noise is small compared to the signal bandwidth, considerable gain can be obtained by correlation.
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43.60.Cg Statistical properties of signals and noise
43.20.Bi Mathematical theory of wave propagation

Variance bounds for passively locating an acoustic source with a symmetric line array

G. Clifford Carter

J. Acoust. Soc. Am. Volume 62, Issue 4, pp. 922-926 (1977); (5 pages)

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The maximum‐likelihood (ML) processor is presented for passively estimating range and bearing to an acoustic source. The source signal is observed for a finite‐time duration at several sensors in the presence of uncorrelated noise. When the speed of sound in an isovelocity medium and the sensor positions are known, the ML estimator for position constrains the source to sensor delays to be focused into a point corresponding to a hypothesized source location. The variances of the range error and bearing error are presented for the optimum processor. It is shown that for bearing and range estimation, different sensor configurations are desirable. However, if the area of uncertainty is to be minimized, then the sensors should be divided into equal groups with one‐third of the sensors in each group.
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43.60.Gk Space-time signal processing, other than matched field processing
43.30.Vh Active sonar systems

Operator formulation for maximum‐likelihood estimation of acoustic emitter delay and Doppler

B. Fisher and L. R. Weill

J. Acoust. Soc. Am. Volume 62, Issue 4, pp. 927-929 (1977); (3 pages)

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A general, compact, and quickly derived solution to maximum‐likelihood estimation of acoustic emitter delay and delay rate can be achieved by representing the signal propagation to each of M sensors as a matrix operator containing the unknown parameters. A general solution is first derived in which the emitted signal is unknown but nonrandom. This solution can incorporate the combined effects of propagation parameters such as wide‐band Doppler, frequency‐dependent attenuation, multipath time delays, and signal phase inversion due to some types of reflection. The result is simplified if only wide‐band Doppler, single‐path delays, and frequency‐independent attenuation are present and, in the case of two sensors, it reduces to simple cross correlation. The matrix approach is also applicable to casses where the emitted signal is modeled as a random process.
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43.60.Gk Space-time signal processing, other than matched field processing
43.30.Vh Active sonar systems
43.30.Cq Ray propagation of sound in water

Cochlear micromechanics—a mechanism for transforming mechanical to neural tuning within the cochlea

Jont B. Allen

J. Acoust. Soc. Am. Volume 62, Issue 4, pp. 930-939 (1977); (10 pages) | Cited 3 times

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A linear mathematical model is proposed which will account for the differences observed between mechanically measured data of Rhode (1971) for basilar membrane motion, and the responses of neural tuning curves (Kiang et al., 1974). We show that theoretical tuning curves may be derived from mechanical responses by forming the difference between the pressure across the basilar membrane and its displacement. Some ramifications of this proposal are discussed. We then propose a hypothetical physical model which could perform such a function.
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43.64.Kc Cochlear mechanics
43.64.Bt Models and theories of the auditory system
43.64.Pg Electrophysiology of the auditory nerve

Two‐tone auditory spectral resolution

Lawrence L. Feth and Honor O’Malley

J. Acoust. Soc. Am. Volume 62, Issue 4, pp. 940-947 (1977); (8 pages) | Cited 7 times

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Two component complex tones were synthesized so that each member of a complementary pair exhibited identical amplitude modulations; frequency modulations, however, were in opposite directions. Previous work has shown that the discriminability of the members of a complementary pair is related to the envelope‐weighted differences in their instantaneous frequency functions. Listeners report that they base their discriminations upon differences in pitch between complementary two component signals. We assumed that the high discriminability found for moderate separations between the two component frequencies would not obtain if the components were resolved into separate critical bands. Thus we could define the just discriminable (75%) frequency separation as a direct estimate of spectral resolving power without recourse to assumptions about masking effectiveness, loudness summation, or other subjective changes thought to depend upon auditory spectral resolving power. Our findings indicate that resolution bandwidths resulting from the discriminability of complex tone pairs approximate the width of the traveling‐wave envelope observed by von Békèsy. In light of our results, the implications for other critical bandwidth estimates are discussed.
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43.66.Fe Discrimination: intensity and frequency
43.66.Ba Models and theories of auditory processes

Detection and recognition of pure tones in noise

David M. Green, Daniel L. Weber, and Joseph E. Duncan

J. Acoust. Soc. Am. Volume 62, Issue 4, pp. 948-954 (1977); (7 pages)

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We examine the predictions of a new theorem relating signal identification (specifying a signal as a particular member of a set of potential signals) to signal detection (discriminating the presence of a signal). The theorem, derived in the context of signal‐detection theory, requires that the signals be equally detectable and orthogonal. Our sinusoidal signals are partially masked by noise and their intensities adjusted to produce equal‐signal detectability; we do not examine this assumption of the theorem. The theorem generally provides a reasonably accurate description of recognition performance for two‐signal and four‐signal conditions and is equally accurate for both the Yes–No and category‐rating procedures. In a preliminary investigation of the orthogonality assumption, we varied the frequency separation between two signals. When the frequency separation between two signals is small (20 Hz near 1 kHz), the theorem fails to provide a good description of performance.
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43.66.Ba Models and theories of auditory processes
43.66.Dc Masking
43.66.Fe Discrimination: intensity and frequency
43.66.Lj Perceptual effects of sound

Monaural and between‐ear temporal gap detection: I. Single gaps

Irwin Pollack

J. Acoust. Soc. Am. Volume 62, Issue 4, pp. 955-960 (1977); (6 pages)

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Temporal gap thresholds were obtained for four signal combinations: monopolar (M) and random‐polarity (R) pulse trains; and, presentation of signals before and after a temporal gap to the same (S) ear or to different (D) ears. When a monaural spectral analysis of the signal across the gap is possible (conditon M‐S, gap sensitivity improves with longer and longer signals on each side of the gap, with extremely fine temperal gap sensitivity (∠1 μsec). Sensitivity decreases markedly for the other three conditions (∠103‐fold for condition R‐S; ∠104‐fold for condition M‐D or R‐D). There, gap sensitivity decreases with longer signals about the temporal gap. The poorer sensitivity with longer random‐polarity signals for monaural gap detection may be related to greater masking by the longer signals. The poorer sensitivity with longer signals for binaural gap detection may be related to binaural ’’precedence’’ effects. Strong asymmetrical effects upon gap detection were also noted.
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43.66.Mk Temporal and sequential aspects of hearing; auditory grouping in relation to music
43.66.Pn Binaural hearing
43.66.Rq Dichotic listening

Noncategorical perception of stop consonants differing in VOT

Arlene Earley Carney, Gregory P. Widin, and Neal F. Viemeister

J. Acoust. Soc. Am. Volume 62, Issue 4, pp. 961-970 (1977); (10 pages) | Cited 7 times

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The discriminability of bilabial stop consonants differing in VOT (the Abramson–Lisker bilabial series) was measured in a same–different task, an oddity task, and a dual response, discrimination–identification task. Subjects showed excellent within‐category discrimination in all three tasks after a moderate amount of training in a same–different task with a fixed standard and with feedback. In addition, discrimination performance continuously improved with increasing stimulus difference for both intra‐ and intercategory comparisons. Also, subjects were able to alter their identification responses so that well‐defined category boundaries fell at arbitrary values determined by the experimenters. These results are not compatible with a strict interpretation of the categorical perception of stop consonants.
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43.70.Dn Disordered speech
43.70.Fq Acoustical correlates of phonetic segments and suprasegmental properties: stress, timing, and intonation

Perceptual switching in bilinguals

Jeffrey L. Elman, Randy L. Diehl, and Susan E. Buchwald

J. Acoust. Soc. Am. Volume 62, Issue 4, pp. 971-974 (1977); (4 pages) | Cited 4 times

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Three groups of subjects, monolingual English speakers, monolingual Spanish speakers, and English–Spanish bilinguals, identified a series of naturally produced syllables which varied in voice onset time (from /ba/ to /pa/). The two monolingual groups differed substantially in their identification performance, with English speakers tending to label most of them as /ba/ and Spanish speakers tending to label most of them as /pa/. The bilingual subjects heard the test stimuli in both an English and a Spanish context, each designed to induce a particular language ’’set.’’ These subjects perceived a reliably greater number of the test items as /ba/ in the English context than in the Spanish. The magnitude of this perceptual switching effect depends on the listener’s degree of bilingualism.
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43.70.Dn Disordered speech
43.70.Fq Acoustical correlates of phonetic segments and suprasegmental properties: stress, timing, and intonation

Speaker identification by long‐term spectra under normal and distorted speech conditions

Harry Hollien and Wojciech Majewski

J. Acoust. Soc. Am. Volume 62, Issue 4, pp. 975-980 (1977); (6 pages)

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Two experiments were carried out in which long‐term spectra were extracted from controlled speech samples in order to study the effectiveness of that technique as a cue for speaker identification. In the first study, power spectra were computed separately for groups of 50 American and 50 Polish male speakers under fullband and passband conditions; an n‐dimensional Euclidean distance technique was used to permit identifications. The procedure resulted in high levels of speaker identification for these large groups—especially under the fullband conditions. In a second experiment, the same approach was employed in order to discover if it was resistant to the effects of variation in speech production—at least under laboratory conditions. Talkers were 25 adult American males; three different speaker conditons were studied: (a) normal speech, (b) speech during stress, and (c) disguised speech. The results demonstrated high levels of correct speaker identification for normal speech, slightly reduced scores for speech during stress and markedly reduced correct identifications for disguised speech. It would appear that long‐term speech spectra can be utilized to identify individuals from their speech—even in relatively large groups—when they are speaking normally or under stress (of the type studied); LTS does not appear to be an effective technique when voice disguise is employed. While this approach was utilized only in controlled laboratory experiments, it is suggested that it may have some merit for use in applied situations or as one of the features in a multiple‐vector approach.
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43.72.Fx Talker identification and adaptation algorithms
43.70.Dn Disordered speech

Study of variations in the male and female glottal wave

Randall B. Monsen and A. Maynard Engebretson

J. Acoust. Soc. Am. Volume 62, Issue 4, pp. 981-993 (1977); (13 pages) | Cited 7 times

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A reflectionless metal tube which can act as a pseudoinfinite termination of the vocal tract was used to collect glottal volume‐velocity waveforms produced by 10 male and female adult subjects. From each subject, glottal volume‐velocity samples were collected of normal, loud, and soft voice; falsetto and creaky voice; monosyllables with rising and falling intonation; and three‐syllable utterances containing primary lexical stress on one of the three syllables. Analysis of the data indicates a wide variation of the glottal waveform shape, its rms intensity and fundamental frequency, phase spectrum, and intensity spectrum. It is observed that as the fundamental frequency changes over time, the glottal source varies in one of two different ways. In one type of change, the harmonic relations in the glottal spectrum become steeper as fundamental frequency rises. In a different type of glottal‐wave change, relations between harmonics tend to remain the same despite a change in the fundamental frequency; the source spectrum in this case is simply shifted along the frequency and amplitude axes as a function of fundamental frequency. To account for these variations in the glottal source, at least three factors must be known: the sex of the speaker, the voice register in which he phonates, and the linguistic context in which the phonation occurs.
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43.70.Bk Models and theories of speech production
43.72.Ar Speech analysis and analysis techniques; parametric representation of speech

Effect of final‐syllable position on vowel duration in infant babbling

D. K. Oller and Bruce L. Smith

J. Acoust. Soc. Am. Volume 62, Issue 4, pp. 994-997 (1977); (4 pages)

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Previous research has suggested possible speech‐production and/or speech‐perception‐oriented causes for the temporal phenomenon referred to as final‐syllable vowel lengthening, yet little conclusive evidence has been adduced for either possibility. The present study represents an attempt to provide additional information concerning the nature of this phenomenon by approaching the issue from a developmental perspective. A much smaller amount of final‐syllable vowel lengthening was observed in the premeaningful vocalizations of a number of very young infants than in phonetically comparable utterances produced by adult speakers of English. It is concluded on the basis of this preliminary evidence that extensive final‐syllable lengthening observed in the productions of adult speakers of English seemingly constitutes a learned behavior.
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43.72.Ar Speech analysis and analysis techniques; parametric representation of speech
43.70.Bk Models and theories of speech production

Control of vocal‐tract length in speech

Carol J. Riordan

J. Acoust. Soc. Am. Volume 62, Issue 4, pp. 998-1002 (1977); (5 pages) | Cited 2 times

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Essential for the correct production of vowels is the accurate control of vocal‐tract length. Perkell [Psychology of Speech Production (MIT, Cambridge, MA, 1969)] has suggested that two important determinants of vocal‐tract length are vertical larynx position and lip spreading/protrusion, often acting together. The present study was designed to determine whether constraining lip spreading/protrusion induces compensatory vertical larynx displacements, particularly on rounded vowels. Upper lip and larynx movement were monitored photoelectrically while French and Mandarin native speakers produced the vowels /i,y,u/ first under normal‐speech conditions and then with lip activity constrained. Significant differences were found in upper‐lip protrusion and larynx position depending on the vowel uttered. Moreover, the generally low‐larynx position of rounded vowels became even lower when lip protrusion was constrained. These results imply that compensatory articulations contribute to a contrast‐preserving strategy in speech production.
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43.70.Bk Models and theories of speech production
43.72.Ar Speech analysis and analysis techniques; parametric representation of speech
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