bluuebunny/crossref_metadata_2025 huggingface dataset:
Dichotic Forward and Backward Masking of Speech and Nonspeech Sounds
Authors: Darwin C. J.
Published: October 2005

When a CV syllable is played to one ear and noise of similar over-all spectrum and duration to the other with a delay of ±60 msec, recall of the syllable is more efficient when the syllable leads the noise. This contrasts with previous results in which two CV syllables are played one to each ear and the advantage is for the lagging syllable. A second experiment using brief pitch modulated signals carried on a steady vowel /ɛ/, under dichotic presentation with onset asynchronies of ±25 msec, showed an advantage in recall for the lagging signal. Superiority of backward over forward masking under dichotic presentation thus appears not to be confined to stimuli differentially categorized as speech sounds, but does depend on the relationship between the potential masker and maskee. [Research supported in part by grant from NICHD.]
The Effect of Temporal Overlap on the Perception of Dichotically and Monotically Presented CV Syllables
Authors: Porter Robert J.
Published: October 2005

When stop consonant-vowel syllables, 390 msec in duration, are presented dichotically, with an onset asynchrony of 50–70 msec, subjects identify the temporally lagging stop more accurately than the leading. When the same pairs are presented monotically, the leading stop has the advantage. This difference has been interpreted in terms of central versus peripheral masking [M. Studdert-Kennedy, D. Shankweiler, and S. Schulman, J. Acoust. Soc. Amer. 48, 599–602 (1970)]. Dichotically, the signals travel separate peripheral pathways, and central masking effects are revealed. Monotically, the lagging signal is masked peripherally by the overlapping leading signal. This experiment was designed to determine the effect that the elimination of the temporal overlap would have on the dichotic and monotic masking functions. Syllables 75 msec long were presented at six asynchronies between 0 and 100 msec. The dichotic masking function was unaffected, whereas the monotic lead advantages were substantially reduced at the nonoverlapping asynchronies. Implications of these results will be discussed. [Work supported by NIH.]
Size of the Dichotic Right-Ear Effect as a Function of Alignment Criteria
Authors: Hannah J. E. , Thompson C. L. , Cullen J. K. , Hughes L. F. , Berlin C. I.
Published: October 2005

Synthetic CVs with V-O-Ts ranging from −30 to +90 msec, were aligned for dichotic presentation according to either onset, transition, V-O-T, “boundary” and/or combinations thereof. Twenty-four female right-handed subjects, who listened to messages aligned as “simultaneous” according to those landmarks, generated the following superiority of right ear over left ear: (1) onset (16.3%); (2) onset +V-O-T (12.8%); (3) onset +transition (12.5%); (4) transition (10.7%); (5) “boundary” +V-O-T (10.6%); (6) all factors aligned (9.0%); (7) V-O-T (9.0%); (8) “boundary” (7.6%); (9) “boundary” +transition (5.0%). Over-all intelligibility and laterality effects were not correlated. Unvoiced CVs dominated V-U pairings unless “boundary” and/or V-O-Ts were aligned; then voiced consonants were more intelligible. “Lag effect,” where the trailing member of a dichotic pair is more intelligible, was redefined; lag effects were largest when “boundary” was taken as the reference for simultaneity, and one transition lagged behind another.
Large Laterality Effects May Help in a “Bio-Assay” for the Elements of Speech Recognition
Authors: Berlin C. I. , Lowe S. S. , Thompson C. L. , Berlin H. L. , Schumacher M. T.
Published: October 2005

Patients with circumscribed temporal lobe lesions show large contralateral ear decrements in simultaneous dichotic speech tasks. However, when noise is in the ipsilateral ear, there is little decrement in the contralateral ear scores. As intensity of speech in the ipsilateral ear gradually increases above threshold, scores of the contralateral ear decrease markedly until, with even 30-dB intensity difference in its favor, the contralateral ear still shows a decrement from its monaural level. The nervous system seems to signal its recognition of “speech” in the ipsilateral ear by suppressing the performance of the contralateral ear.
Communication between Divers in Deep Submergence Atmospheres
Authors: Morrow C. T.
Published: October 2005

There are two obvious peculiarities of divers' speech in deep submergence conditions. At any depth, a diving mask, when worn, has an acoustic reaction on the vocal tract, deteriorating the intelligibility. For depths below 400 ft, the helium atmospheres used, either in a mask or in a habitat, shift the frequency band associated with speech intelligibility upward by as much as a factor of 3. The effects of both can be overcome if a microphone is suitably designed. The upward shift of “Mickey Mouse Effect” is generally credited with destroying intelligibility to such an extent that a downward shifting compensator becomes necessary. By itself, the upward shift does not destroy intelligibility, but there are other effects associated with diving and deep submergence that may make a compensator necessary even for face-to-face conversation in a habitat. At the present time, the performance of compensators is limited by the microphones available.
Translation of Divers' Speech Using Digital Frequency Warping
Authors: Zue Victor , Johnson Don , Oppenheim Alan
Published: October 2005

The composition and pressure of breathing mixtures used by divers translate frequency components of their speech nonlinearly. Recently, a method of forming a new sampled time function from a given finite-duration sampled time function in which the spectra of the two are related through a nonlinear warping in frequency has been discussed [A. Oppenheim, D. Johnson, and K. Steiglitz, “Computation of Spectra with Unequal Resolution Using the Fast Fourier Transform,” Proc. IEEE 59, 299 (1971)]. The technique is based on the expansion of the original time function in terms of a set of functions related to Laguerre functions. The coefficients in the expansion, when interpreted as samples of a new time function, have a Fourier transform that is equal to that of the original function with a nonlinear transformation in frequency. These coefficients are obtained by processing the original signal with a second-order network in cascade with a chain of first-order all-pass networks. To apply the technique to translation of divers' speech, individual pitch periods are processed by the above network. These modified pitch periods are than reconvolved with a train of pitch pulses to synthesize the translated speech.
The Detection and Spectral Analysis of Atmospheric Whistlers
Authors: Nitsche J. R. , Levine M. S. , Waters J. F.
Published: October 2005

Whistlers are low-frequency (100 Hz –100 kHz) electromagnetic waves that are found to occur in the Earth's magnetosphere. Whistlers may present various profiles in the time-frequency domain, but in general, they may be characterized as gliding tones in the audiofrequency range. Some forms of whistlers appear as relatively pure tones that decrease in frequency with increasing time. Others appear as bands of noise with progressively decreasing mean frequency. Although this work was done with particular regard to the study of atmospherics, it is pertinent to the study of any Doppler-influenced acoustic signature. This experiment had two phases; the first phase consisted of the design and implementation of a computer-controlled editor, which allowed the rapid detection and classification of whistlers on magnetic tape; the second phase was the analysis of the detected whistlers in the time-frequency-amplitude domain using a contour spectrogram. Based on the known time-frequency characteristics of the whistlers of interest, it was decided to base the design of the whistler editor on the concept of using a high-speed general-purpose computer to monitor the outputs of a series of contiguous 13-oct-band filters. Because of several characteristic features of the whistlers and because of their favorable signal-to-noise power ratio, the editor was designed to perform detection and classification operations. This paper describes the design and implementation of the whistler editor. Examples of the editor output and typical results of the contour spectral analysis are presented. Over 100 whistlers were detected, classified, and analyzed from five magnetic tapes obtained from the NASA OVI-10 satellite experiment. The probability of detection was estimated to be 75% with the detection threshold used. [Work supported by Electronic Fields Branch (NASA).]
Frequency Spreading of Scattered Sound through Impulse-Response Measurements
Authors: Spindel R. C. , Schultheiss P. N.
Published: October 2005

The frequency spreading of sound scattered from the windblown surface of a model tank was investigated. It was assumed that the surface-reflection channel can be approximated by a linear random time-varying filter. Thus characterization of the channel reduces to a statistical description of its stochastic system functions. The channel impulse response, h(τ,t), was taken as fundamental because of the relative ease with which it can be measured. The channel bifrequency function, H(ω,ν), was obtained by computer fast Fourier transformation of h(τ,t) with respect to both time axes. For narrow-band channel excitation, the ensemble average bifrequency function measures the average amount of excitation energy at frequency ω converted to scattered energy at frequency ν. Comparison of measured ensemble-average bifrequency functions with data obtained by single-frequency measurements subjected to conventional spectral analysis shows that the impulse response technique provides frequency spreading information for any given input frequency over the bandwidth of the acoustic pulse.
Digital Simulation and Analytical Prediction of Level Crossing Statistics for Linear Filtering of Non-Gaussian (Impulsive) Noise
Authors: Hearn J. J. , Smits T. I.
Published: October 2005

In many physical situations, it is important to determine the level crossing statistics resulting from impulsive noise excitation of linear narrow-band systems. The statistics thus developed can be used to obtain engineering approximations useful in describing system performance. A digital simulation approach, applicable to a variety of impulse strength distributions and with time statistics based on a Poisson process, was developed to generate the desired level crossing statistics. This approach circumvented the difficulties inherent in analytical derivations of the level crossing statistics. The simulation model was used to generate level crossing statistics associated with filtered impulsive noise whose strengths were exponentially and Gaussianly distributed (EDIS and GDIS). These statistics were obtained for intermediate values (1⩽γ⩽10) of the impulse-noise-density parameter, γ, which is defined as the product of the noise impulse occurrence rate and the system response time. A three-parameter distribution (generalized gamma) was found to fit the EDIS level crossing statistics quite well and analytical expressions were then derived relating γ to the distribution shape factor p, and p to the distribution scale factor a, for the intermediate γ values.
A 50-dB Increase in Sustained Ambient Noise from Fish
Authors: Fish J. F. , Cummings W. C.
Published: October 2005

The Salton Sea, with a surface area of about 350 square miles, is the largest inland body of water in California. Ambient noise was measured once a month in this salt-water sea from January to August 1970 with a calibrated recording system. There was no seasonal change in daytime levels. However, the night-time spectrum level at 1000 Hz increased by about 50 dB from January to May. This great increase in level resulted entirely from a chorus of sounds produced by the orangemouth corvina (a croaker), Cynoscion xanthulus. The seasonal peak of sound production coincided with the breeding season (April–June) of the 1 to 2 million corvina. During the winter, the Sea was exceptionally quiet at all hours with a spectrum level over the entire recording band approaching the empirical lower limit for ocean noise. The 50-dB increase in sustained ambient noise level is the greatest ever noted from underwater natural causes.
The Acoustic Damping of a Rectangular Panel
Authors: Wallace C. E.
Published: October 2005

The acoustic radiation damping of a rectangular panel vibrating at one of its natural frequencies is theoretically determined from the ratio of energy radiated per cycle to the vibratory energy of the panel. The panel is supported in an infinite baffle with either simple supports on all four sides, or with only hinges or clamped supports on two opposite sides. For simple supports on four sides, the damping of thin panels depends on the aspect ratio, with the square panel developing the smallest value. For thick panels, the damping is nearly independent of the aspect ratio. For panels supported on only two sides, thin panels exhibit nearly twice as much damping for clamped supports as they do for simple supports. The damping of the fundamental mode of a thick panel with calmped supports is less than one-half that of the same panel with hinged supports. However, the damping of the high numbered modes of this thick panel are unaffected by the support conditions. [Supported by the National Research Council and the NASA Langley Research Center.]
Control of Plate Vibrations through the Application of Constrained Viscoelastic Layers
Authors: Strickland Daniel Z. , Torvik Peter J.
Published: October 2005

Two configurational additions employing multiple anchored viscoelastic layers and one employing multiple unanchored layers were designed and applied to simply supported plates. The energy dissipated in both free and forced vibrations was determined experimentally for all configurations. The damping in forced vibrations was determined from the response curve at the first resonance. The damping in free vibration was determined by driving the plate at resonance, disconnecting the driving mechanism, and measuring the decay. For all configurations, the logarithmic decrement was essentially independent of the amplitude, and three to five times the decrement for an untreated plate. The unanchored damping addition is a two-dimensional version of the configuration described, analyzed, and tested by Plunkett and Lee [J. Acoust. Sec. Amer. 48, 150–161 (1970)]. The experimental results for an optimum structure were within 10% of the predictions of an analysis related to that performed by Plunkett and Lee. Limited data at the resonant frequency of a higher mode indicates that this addition retains its effectiveness.
Application of an Approximate z-Transform Method to the Analysis of Distributed Parameter Systems
Authors: Stepanishen P. R.
Published: October 2005

A transform approach is presented to compute at discrete time intervals the time responses of lumped or distributed parameter systems that exhibit wave propagation effects. The approach was previously developed by Boxer and Thaler [R. Boxer and S. Thaler, Proc. IRE 44, 89–101 (1956)] as a numerical method of solving linear and nonlinear equations and is an extension of the more familiar Laplace transform technique. Rather than utilize the inverse Laplace transformation to obtain the time function, a mapping of the Laplace transform of the function from the z domain to the z domain is performed via the transformation z = esT, where T is the desired sampling interval of the time function. Unlike the inverse Laplace transformation which requires the pole-zero structure of the transform, the sampled time function is readily obtained from its z-domain representation without using the theory of residues. Numerical solutions to several problems are presented to illustrate the advantages of the approximate Z-transform approach. Included within the numerical results are the time responses of distributed parameter transducer models.
Session WHH. Noise Workshop: Sound Power Measurement at Discrete Frequencies
Authors: Tichy Jim
Published: October 2005

The method of measurement of the sound power at single frequencies using reverberation chamber is still under development and a lot of both theoretical and experimental research is being done. The workshop will give the opportunity to discuss the most recent results of the research and the practical aspects of the measurements. The program of the workshop will cover: Introduction and Summary (J. Tichy); The Effect of the Sampling on the Variance of the Sound Pressure Squared (D. Lubman); The Effect of the Nearfield Reflectors (D. Ingalls); Dependence of the Radiated Power on the Sound Source Position (G. Maling); The Application of Rotating Vanes (J. Tichy); Qualification Procedure of Reverberation Chamber (C. Ebbing); R. Waterhouse and W. Lang will take part in the discussion.
Session WII. Musical Acoustics Workshop: Concert/Demonstration of New Bowed String Instruments Developed by Catgut Acoustical Society, Inc.
Authors: Neustadt Herbert M.
Published: October 2005

A classic problem of acoustics is: How can one build a violin having a tone quality that matches the tone of a Stradivarius? This problem has been under enthusiastic attack for many years by a group of researchers led originally by the late Professor Frederick A. Saunders of Harvard, and more recently by Carleen M. Hutchins. The group calls itself “The Catgut Acoustical Society” and has four hundred members in fourteen nations. Many of the members have achieved professional distinction as mathematicians, physicists, and musicians.

Forty years of work in a multidisciplinary situation involving technical researchers, violin makers, performing musicians, conductors, and composers as well as listeners have yielded a vast store of information on the conventional violin, viola, cello, and string bass. Many of these findings have been incorporated into experimental instruments built and tested by Carleen Hutchins in her basement laboratory and workshop resulting not only in improvements in conventional violins, violas, cellos, and basses but also in design and construction of a whole new family of eight violin-type instruments which give a uniform tone quality across the entire range of musical frequencies.

These eight new instruments are to be played by musicians of the National Gallery Orchestra and Mrs. Hutchins will lecture briefly.
Review of Integral Equation Methods in Sound Radiation and Scattering
Authors: Chertock George
Published: October 2005

Integral-equation methods are described for calculating the entire sound-pressure field when either the velocity distribution or, alternatively, the sound-pressure distribution is specified on an arbitrary closed surface. The theory is based on determining equivalent surface layers with either monopoles and/or dipoles. An indefinite number of appropriate integral equations are derived for the surface monopole and/or dipole density for each case and each boundary condition. Simple techniques are described for approximating each of the integral equations by a linear matrix equation with finite elements and for the numerical solution of the matrix equation. For every surface shape, and any specified ka, there exists an infinite sequence of complex characteristic numbers γi and associated sets of surface functions (sound pressure pi, normal gradient qi, dipole surface density μi, and monopole surface density σi) which are linear homogeneous functions of each other, and which provide a complete orthonormal basis for the solation to arbitrary Dirichlet or Neumann problems. In particular, for every closed surface, there exist two infinite sequences of characteristic wavenumbers such that, at one or the other of these wavenumbers, and for particular boundary conditions which are stated, each of the more common integral equations has an infinite or indeterminant solution. Special methods are described to eliminate the indeterminacy in such cases.
A Simple Method for Calculating the Pressure Field Generated by an Impulse on an Arbitrary Surface
Authors: Schomer P. D.
Published: October 2005

It is shown that the pressure at any point P in space generated by an impulse applied over a line or surface can be thought of as the summation of pressure responses due to each element of line or surface, where the “directivity pattern” of each element on this source surface is secθ, θ being the angle between the line from the source element to point P and the line or plane tangent to the curve or surface at this element. Examples of sources include an infinite plane, a line, and a disk. Further, it is shown that, for the disk, the on-axis response can be written without calculation, and in addition the Fourier transform can be performed, again without calculation.
The High-Frequency Directivity Pattern of a Velocity Source on a Rigid Cylinder
Authors: Feit David
Published: October 2005

The usual way of determining the farfield directivity pattern of a prescribed velocity source on a cylindrical surface is to use the method first introduced into the acoustics literature by Laird and Cohen [J. Acoust. Soc. Amer. 24, 46–49 (1952)]. The method consists of first expanding the source distribution as a Fourier series in the circumferential coordinate and a Fourier transform over the axial direction. From the latter information, a similar expression for the pressure can be evolved, and from this an asymptotic expression for the farfield pressure can be obtained. This results in an expression which is given in terms of a wave harmonic series. In the high-frequency range, this series converges slowly. The present paper directs itself towards a different representation of the radiated field, which is most useful in the high-frequency range where the wave harmonic series is most inefficient. A comparison of the results obtained by the two different representations is presented and the agreement is excellent. [This investigation was supported by the acoustics Program Branch of the Office of Naval Research.]
The Radiation of Love Waves from an Oscillating Cylinder
Authors: Hudimac A. A.
Published: October 2005

We consider the radiation of Love waves from a rigid circular cylinder imbedded part way into the upper solid layer overlying a semi-infinite solid and oscillating sinusoidally about its vertical axis. Only the radial component of the vector potential exists, and this is transformed into a scalar potential satisfying Helmholtz's equation; the boundary conditions are unusual for propagation in a fluid. Because the boundary values on the cylinder are mixed and the cylinder is finite in length, the mathematical formulation is converted into a pair of integral equations by means of a Green's function. The latter is obtained in two ways: using Hankel transforms, and using Weyl expansion for the source. In either case, the final integration is obtained by closing a contour. The contributions from the poles have the characteristics of Love waves. The contributions from the deformation of the contour about branch cuts have the property of lateral waves. The evaluation of the Green's functions on the surface of the cylinder enables us to replace the dual integral equations approximately by two sets of simultaneous algebraic equations for the unknown wavefield or its normal derivative. Lastly, the Green's integral is integrated numerically.
Beam Behavior within the Nearfield of a Vibrating Piston
Authors: Zemanek J.
Published: October 2005

A numerical investigation of the nearfield region of a vibrating piston was conducted. While existing literature is not exactly clear about how close to a transducer a regular beam pattern is formed, the results of this investigation clearly show the limits of operation one can expect. From one interpretation of literature, a −3-dB beam with a minimum spot size equal to the transducer diameter would be expected to form beyond an axial distance z = 3.89a2/λ. This investigation shows that such a beam forms at z = 0.75 a2/λ and had a minimum spot size equal to one-fourth the transducer radius. These results are verified for transducers with a/λ's ranging from 1 to 20 and can be extended with confidence to higher a/λ's. Similar results are also obtained for −1.5-, −4.5-, −6.0-, −7.5-, −9.0-, −10.5- and −12-dB beams.
Transmission Coefficients of Surface Cracks in the Path of Rayleigh Waves
Authors: Sinclair Rex
Published: October 2005

An approximate theory is presented for the transmission coefficients of surface cracks for incident Rayleigh waves. Predictions are given for a solid of Poisson's ratio 0:29 and are compared with the experimental results of Dalvi and Finch on surface flaws ou railroad wheels.
Research on Fan Noise Generation
Authors: Benzakein M. J.
Published: October 2005

The paper presents in some detail the results of analytical and experimental investigations on fan noise generation. It discusses the aeroacoustic characteristics of blade-passing-frequency tones (and their higher harmonics), combination tones, and broad-band frequency noise. The paper presents theoretical procedures for the prediction of fan noise generation as well as extensive comparisons between analytical and experimental data. Results of aeroacoustic investigations on broad-band noise conducted on two-dimensional cascades and on rotating rigs are also reported.
The Effect of Masking on SPL and Intraoral Air Pressure during Vocal and Whispered Speech
Authors: Brown W. S. , Brandt John F.
Published: October 2005

Last year at this convention the present investigators reported findings indicating that speakers increased their spl and intraoral air pressure with increases in a binaural masking noise even though they had been instructed to speak in a constant manner regardless of the presence or absence of noise. The present investigation was a further consideration of the effects of masking (i.e., auditory feedback) on articulatory performance. During the speakers' repetitions of sentences, a masking noise was presented through earphones at several SPLs (up to 107 dB). The speakers were instructed to ignore the masking noise and to repeat the sentences in a constant manner. Whispered sentences were included in order to reduce auditory feedback further, presumably bone conduction. Measures of average peak intraoral air pressure and SPL were made to determine the stability of the speakers' utterances. Again, SPL and intraoral air pressure were found to increase. However, these increases were slight, indicating that the speakers may have used sensory-motor information other than auditory to control their articulatory behavior.
Comparison of Subglottal Air Pressures Associated with /p/ and /b/
Authors: McGlone Robert E. , Shipp Thomas
Published: October 2005

Intraoral air-pressure differences between voiced-voiceless cognates have been attributed by most investigators to a transglottic pressure drop associated with voicing. This assumes pressures below the glottis of essentially the same magnitude during the production of these sounds. The present study tests this assumption. Airflow and subglottal air pressure were recorded as 10 adult males spoke syllables containing /p/ and /b/ in the intervocalic position. Each uttered the syllables stressing the initial vowel and again stressing the final vowel. The airflow trace was used to provide reference points for air-pressure measurement. First, pressure values were measured at a point of no air flow prior to consonant release lip opening and second, the pressure point that corresponded with maximum airflow after release. Analysis of these values showed no significant difference between the pressures at the two points of measurement nor between measures obtained for /p/ and /b/. Differences were found related only to stress. Pressures were greater when the final vowel was stressed.
Aerodynamic and Acoustic Events at the Release of Stop and Fricative Consonants
Authors: Stevens Kenneth N.
Published: October 2005

The production of stop and fricative consonants is characterized by (1) rapid mechanical movements at the supraglottal constriction and at the larynx, (2) changes in intraoral pressure and airflow, and (3) a sequence of acoustic events involving turbulence noise generation and glottal exictation of the vocal tract. A theoretical model that accounts for the observed events at the mechanical, aerodynamic, and acoustical levels is developed. This model consists of an acoustic tube with two time-varying constrictions—one at the glottis and one at the point of articulation of the consonant. The airflow and the turbulence noise generated at the constriction are determined from equations of aeroacoustics, and the conditions under which vocal-cord vibration is initiated are established on the basis of theories of vocal-cord operation. The model has been used to make calculations of airflow, pressure, and acoustic output for several classes of voiced, aspirated, and glottalized stop and fricative consonants. The calculations suggest that these consonants fall into natural classes with well-defined and stable acoustic correlates, based on whether the glottis is in a normal, spread, or constricted configuration and whether the vocal cords are stiff or slack. [Supported by NIH and by the U. S. Air Force.]
An Interpretation of Continuous Articulator Acceleration as a Muscle Forcing Function
Authors: Abbs J. H. , Netsell R.
Published: October 2005

Recordings of jaw acceleration (Jx/t2) were obtained in an attempt to provide some preliminary information on the nature of the masticatory muscle forcing function during speech. The impetus for this approximation was provided by Newton's second law of motion. Speech audio, intraoral air pressure (Po), jaw displacement (Jx), and jaw velocity (Jx/t) were simultaneously recorded with Jx/t2 to facilitate inferences to net muscle force. The temporal and magnitude characteristics of Jx/t2 appear to have implications for (1) the pattern of net muscle activity in jaw opening and closing, (2) the influence of gravitational acceleration, and (3) general articulatory mechanics.
Anatomical Constraints on Speech in Chimpanzee, Newborn Homo sapiens and Neanderthal man
Authors: Lieberman Phillip , Klatt Dennis H. , Crelin Edmund S.
Published: October 2005

The cross-sectional area function of the supralaryngeal vocal tract of chimpanzee and the muscular and anatomical constraints on possible variation, were determined by sectioning the cadaver of a young adult on the midsaggital plane. Vocal tract area functions that were directed towards making best possible approximations to the human vowels /a/, /i/, and /u/, as well as certain consonantal configurations, were modeled by means of a computer program. The chimpanzee vocal tract could not produce these extreme vowels since it lacked a supralaryngeal pharyngeal region like that of adult Homo sapiens. The vocal tracts of newborn Homo sapiens and chimpanzee are similar and resemble the reconstructed vocal tract of the fossil La Chappelle-aux-Saints Neanderthal man (Lieberman and Crelin, 1971). The lack of the vowels /a/, /i/, and /u/ in the phonetic repertories of these creatures may be concomitant with the absence of consonantal encoding and decoding and consequently a linguistic ability inferior to that of modern man.
Factor Analyses of Tongue Shapes
Authors: Ladefoged Peter , DeClerk Joseph L. , Harshman Richard
Published: October 2005

Tracings from x rays were made of the shape of the tongue in nine vowels as spoken by five speakers of General American. The midline of the tongue on each tracing was specified in terms of twelve numbers (two orthogonal measurements for each of six defined points). The resulting 9×5×12 matrix was analyzed by the PARAFAC procedure (Harshman, this meeting). The data indicate the extent to which the midline of the tongue can be described in terms of a small number of factors. The paper will discuss the relation between these factors and the following physiological parameters: (1) jaw opening; (2) forward pull of the posterior genioglossus muscle on the tongue root (and hence raising of the body of the tongue) as opposed to backward pull of the glossopharyngeus muscle; (3) upward and backward pull of the styloglossus muscle; (4) downward pull of the hyoglossus muscle; and (5) actions of the intrinsic muslces that control the bunching of the tongue.
Vocalization Rate of Deaf and Normal-Hearing Children
Authors: Asp Carl W. , Wood W. Scott , Keller James
Published: October 2005

The vocalizations of 65 deaf and 48 normal-hearing children, at age levels of 5, 7, 10, 13, and 17 years, were picked up with a throat microphone, transmitted by an FM system, and recorded on an electromechanical counter. The criterion measure was the number of vocalizations per minute. Deaf children vocalized at a significantly different rate as a function of age level, whereas normal-hearing children did not. Normal-hearing children vocalized significantly more than deaf children. [Work Supported by U. S. Office of Education and a NIH Biomedical Sciences Support Grant.]
Oral Sensory Perception in the Deaf
Authors: Bishop M. E. , Ringel R. L. , House A. S.
Published: October 2005

In recent years significant advances in the study of the relationships between oral sensory acuity and form discrimination in the mouth have been made through the use of a testing procedure that successfully discriminates between persons with normal speech patterns and persons with deficient articulatory patterns [R. L. Ringel et al., J. Speech and Hearing Disorders 35, 3–11 (1970)]. Little is known, however, about the oral sensory acuity and perception of the deaf, in spite of the fact that oral sensory feedback may be the only means of mediating speech commands and motor patterns in the absence of auditory function. Experiments were conducted that tested the oral sensory acuity and oral form discrimination of congenitally deaf high-school students. When compared with a similar hearing group, the deaf subjects demonstrated essentially normal sensory acuity—that is, normal two-point discrimination and difference limens for object size—but performed poorly on tasks calling for discrimination of shape differences. These results are discussed in terms of their relevance to theories of language and speech function. [This research was supported in part by the National Institute of Neurological Diseases and Stroke, National Institute of Dental Research, and by the Air Force Cambridge Research Laboratories.]
Mathematical Model for the Class IV Flextensional Underwater Acoustic Transducer
Authors: Rutledge James R. , Royster L. H.
Published: October 2005

A mathematical model for axisymmetric free vibrations of the Class IV flextensional underwater acoustic transducer shell is developed. The shell is approximated by an arc with varying radius of curvature and thickness. The eigenvalues and eigenvectors obtained from the approximate model for the simplified case of a 180° curved beam with horizontally guided boundary condition agree with those obtained utilizing a previous developed curved-beam model. The model of the shell is coupled to a discrete set of equations governing the piezoelectric stack. The results are then set up in matrix form and the free vibrations of the Class IV flextensional underwater acoustic transducer in a vacuum are obtained. Results for various design parameters are presented. [Work was partially sponsored by the Office of Naval Research, Acoustics Programs, and by the Naval Underwater Sound Laboratory.]
Computer Modeling of Oval Flextensional Underwater Transducers
Authors: Brigham Gerald A.
Published: October 2005

The computer modeling of radiating piezoelectric transducers involves, generally, the solution of combined electrical-mechanical-acoustical equations of state. For an oval flextensional type, we consider a longitudinally polarized sandwiched ceramic cylinder with lateral and bonding effects; a flexing and stretching metal oval shell of variable curvature and wall thickness; a confined inner acoustic stress field with (or without) compliant tubes; and a radiating external pressure field. The assumptions necessary to reduce the total combined equations to a single program that would fit the IBM 360 (i.e., 480 000 bits or approximately three million words) are discussed in some detail and justified wherever possible. Extensive comparison (both air and water operation) with an experimental transducer operating between 2500 and 5000 Hz shows the model to be in good agreement. The final program which used nearly full computer capacity, required approximately 5 min per frequency to calculate the transducer's input electric impedance.
The Transient Response of Arrays of Transducers
Authors: Stepanishen P. R.
Published: October 2005

A general approach is developed to compute the time-dependent velocities of transducers within an array resulting from a set of prescribed electrical inputs. It is noted the effect of the dynamic response of a transducer of any complexity and the effect of the time-dependent mutual interactions are included within the approach which is based on a Green's function solution to an initial boundary value problem. The solution of the initial boundary value problem is reduced to a set of time domain coupled convolution integral equations which can be easily solved on a digital computer. In addition to presenting a clear physical understanding of the transient behavior of arrays, the approach leads to a more basic understanding of array operation and limitations caused by its transient response. Specific application of the approach to the analysis of planar arrays mounted in rigid baffles is discussed and related to earlier work. Numerical results are also presented.
Radiation Characteristics of Baffled Pistons Transmitting through Domes
Authors: Ross John M. , Hill Tom W. , Gruber George J.
Published: October 2005

The principal mechanisms responsible for the reduction in radiated power and the severe beam pattern distortions induced by domes do not appear to have been fully studied with model underwater sound arrays. Performance results for bare model arrays were reported at the Atlantic City meeting. Network and field analyses accounted properly for the head velocity and farfield pressure data, respectively. In this paper, we present the results of a not-yet-complete experimental investigation using the same model array in the environment of glass-plate domes. To determine the effects of domes on radiated power and directivity pattern, similar types of measurements were made as with bare arrays. Geometrically linear arrays with up to five elements were studied. The mathematical analyses of the no-dome case were extended to provide a model incorporating the complexity produced by the addition of domes. In particular, it was sought to determine if the interelement and element/dome coupling effects could be superimposed to yield the head velocity of an array element transmitting through domes. The experimental and theoretical results and their comparison will be discussed.
Experimental Determination of Thresholds of Masked “Words” Comprising Tone Bursts
Authors: Berger Louis S.
Published: October 2005

Acoustic “words” comprising short tone bursts separated by silence were presented binaurally against a low-level (40 dB SPL over-all re standard reference) masking background of white noise. Eight listeners, using the two-alternative temporal forced-choice procedure, provided experimental data for determining the behavioral thresholds. Two hundred different signal formats of various tone burst frequencies (75–2500 Hz), burst lengths (10–200 msec), durations of interburst silence (10–200 msec), and number of bursts per signal word (7–32) were investigated. The literature [C. S. Watson and R. W. Gengel, J. Acoust. Soc. Amer. 46, 989–997 (1969)] suggests that one may expect thresholds to be strongly influenced by burst length and interburst spacing. However, results consistently were dominated by the tone burst frequency effects and the thresholds followed the usual threshold-versus-frequency curves for pure tones. Effects of variations in the other signal parameters were minor and ambiguous. The results can be explained by an aural “de-integration” time. [Study funded by International Digisonics Corporation, Chicago.]
Comparison of Masking by Combination Bands and Bands of Noise of External Origin
Authors: Greenwood D. D.
Published: October 2005

Combination bands generated by a pure tone and a narrow band of noise produce masking patterns in the frequency region where they occur. These patterns take the form of elevations that merge with, or stand in isolation from, the low-frequency side of the masked audiogram produced by the lower primary, depending on the separation of the bands from the primary. Masked audiograms produced by bands of external origin were also mapped. An “external” band was presented under two conditions: alone and then together with a more intense higher-frequency tone. Comparisons among conditions lead to these conclusions: (1) Addition of a higher-frequency more intense tone to a band of noise of external origin does not increase the masking maximum produced by the noise. This masking remains visible as an elevation on the low-frequency side of the masked audiogram produced by the tone—unless masking by the tone itself in the lower-frequency region of the noise exceeds the masking effect of the noise. (1) Consequently, it is inferred that masking produced by combination bands is also not significantly increased by the presence of the more intense higher-frequency primary stimuli. (3) Hence, by using signal-to-masker ratios obtained when narrow external bands are presented, one can estimate the intensity level of a combination band of comparable width from measurements of masked threshold in the region it occupies.
Loudness Level Summation for Octave Bands of Noise
Authors: Bauer B. B.
Published: October 2005

During a previous investigation [“Researches in Loudness Measurement” IEEE Trans. Audio Electroacoust. AU-14, 141–151 (1966)] a new law of loudness level summation was established. This law may be formulated as follows: Loudness level L of a random noise, measured in octave bands, is L = Lm+20 log Σn (pn/pnm), where Lm is the loudness level of the loudest band, pn is the sound pressure of the nth band of the noise, and pnm is the sound pressure of the nth band of the Lmth equal loudness contour. It is shown that this law, which can readily be implemented in a loudness level meter, is in reasonably close agreement with loudness levels recomputed from Stevens' equation for loudness calculation [S. S. Stevens, “Procedure for Calculating Loudness,” J. Acoust. Soc. Amer. 33, 1577–1585 (1961)].
Loudness Summation between Tones and Noise
Authors: Fishken David
Published: October 2005

The principles by which disparate sounds summate in loudness were investigated by measuring the loudness levels of white noise, with and without an added tone. The tone was 500, 1000, or 4000 Hz, added at 5 dB below, 5 dB above, or 15 dB above the SPL of the noise. The method of adjustment was used, and stimuli were presented at SPLs from 30 to 90 dB. A noise and tone combination was higher in loudness level than the noise alone; this difference was taken as the measure of loudness summation. As the level of the tone increased, relative to the noise, the amount of summation increased. The increase, however, was greatest when the SPL of the noise was 35 dB or lower. At all noise levels, summation was greatest when the tone was 4000 Results were compared to calculations obtained by four procedures, including Kryter's, Stevens', and Zwicker's. Zwicker's predicted the amount of summation best, differing by 1.5 phons, on the average. The success of Zwicker's procedure, which is based on masking patterns, supports the notion that tones and noise summate according to principles of masking.
Loudness Adaptation: Use of the Double Staircase Method and Heterophonic Balances
Authors: Stokinger T. E. , Cooper W. A. , Jones K. O.
Published: October 2005

Twelve normal-hearing listeners were adapted monaurally with a 4-kHz pure tone of 60 dB SPL for 5 min following 3 min of preadapting loudness balances. The control ear was simultaneously stimulated every 3 sec with a 0.25-, 1-, or 4-kHz comparison tone of 1- or 2-sec duration. The listener was instructed to compare the loudness of each comparison signal with that of the adapting signal. Comparison signal intensities were controlled by the listener through two recording attenuators with the comparison signal randomly switched to one or the other attenuator. The magnitude of adaptation was calculated at 20-sec intervals. Average adaptation was found to be significantly different with the three comparison signal frequencies used. The greatest amount of measured adaptation occurred with the 4-kHz comparison signal and the least amount with the 0.25 kHz signal. Comparison signal duty cycles of 13 and 23 yielded no difference in the magnitude of measured adaptation.
Acoustic Reflex Attenuation of Suprathreshold Tone Pips
Authors: McRobert H.
Published: October 2005

Subjects carried out equal loudness balancing of 20-msec tone pips in one ear, while a signal was presented to the contralateral ear (duration 500 msec, spacing 2.2 sec). Tone SPLs of 80 and 100 dB and frequencies of 250, 500, and 750 Hz were employed. The signal was either an octave band of noise centered at 2 kHz and having an over-all SPL of 115 dB, or a 2-kHz tone at the same SPL. A delay time between the onset of the signal and that of the first tone was varied in 25-msec steps (0–200 msec). The results indicate that the maximum effect occurs when employing 500-Hz tones at 100 dB and the noise signal (mean attenuation 4 dB). No effect was found when the frequency was 250 Hz, the SPL 80 dB, or when the 2-kHz tone was in use. A further study involved balancing tones in the quiet, presented alternately to both ears, their separation being varied in 25-msec steps (25–300 msec). No apparent acoustic reflex (AR) effect was found. This absence of AR effects in both studies will be discussed in the light of previous evidence found employing tones at threshold.
Ordering Procedure for Auditory Stimuli
Authors: Nakatani Lloyd H.
Published: October 2005

Sorting is an efficient procedure for rank ordering stimuli along a subjective continuum; moreover, this efficiency leads to enhanced subject motivation and greater reliability. A sorting procedure for auditory stimuli was implemented on a laboratory computer with mope display, magnetic disk for random access to digitally recorded stimuli, and digital-to-analog playback. The procedure initially presents a subject with a “list” of two stimuli which he must order. Subsequent stimuli are added one at a time to the bottom of the already-ordered list. The subject must then insert the new stimulus into its proper order in the list. Ordering is done by rearranging a displayed list of labels that correspond to the auditory stimuli. Any stimulus can be heard at any time by pointing to its label in the display. The procedure was used to obtain preference, comprehensibility, and naturalness judgments of synthetic speech with independent pitch, formant, and timing manipulation.
Effect of Pitch and Formant Manipulation on Speech Quality
Authors: Nakatani Lloyd H. , McDermott Barbara J.
Published: October 2005

Naturally uttered sentences were analyzed and then resynthesized with the range of pitch and formant excursions varied systematically; these sentences were ranked for preference, clarity, and naturalness in separate sessions. The ranges of the synthesis parameters f0, f1, f2, f3 were compressed or expanded about their respective geometric means in the log frequency domain by a factor k; pitch (k=0–1.60) and formant (k=0.55–1.20, all formants transformed simultaneously) variations were factorially combined to generate the sentence ensemble. The data were analyzed using multidimensional scaling procedures that accounted for differences among the listeners. The analysis showed that preference was related to both pitch and formants; clarity was generally related to formants; and naturalness was idiosyncratic. Formant compression or expansion generally lowered both preference and clarity rankings. The influence of pitch compression or expansion on preference varied with the range of the formant excursion.