bluuebunny/crossref_metadata_2025 huggingface dataset:
The role of speech rate in social evaluation
Authors: Crown Cynthia L.
Published: October 2005

Prior studies concerned with the relation of speech rate to the personality variable of extraversion and to social evaluation (i.e., interpersonal perception) have primarily examined monologues. The study reported here was designed to (a) explore such dependencies in dialogues, and (b) investigate the possibility that speech rate may be influenced by differences in interpersonal attraction. The 38 female pairs who participated in the study were selected on the basis of a sociogram designed to assess their attraction to each other prior to their participation in the study. Their sociogram scores divided them into three groups: those who liked each other, those who disliked each other, and those who were unacquainted with each other. Each pair engaged in a 30-min conversation from which the speech rate of each participant was electronically determined. The result of multiple regression analyses indicated that those women who spoke more quickly perceived themselves more positively, whereas those who spoke more slowly viewed their partners more positively. However, variations in rate were related to neither interpersonal attraction nor extraversion.
Two paradigms for examining the role of phonological stress in sentence processing
Authors: Gow David W. , Gordon Peter C.
Published: October 2005

The role of phonological stress in sentence processing was studied using a syllable monitoring task, as well as a new short-term memory probe task. In both tasks, target syllables were embedded in bysyllabic noun/verb homophones with syntactic category-dependent stress patterns. The stress and position within a word of target syllables were manipulated. Syntactic context was also manipulated to examine the role of syntactic constraints on the anticipation of stressed syllables. The syllable monitoring task reproduced the familiar facilitation of stressed syllable detection, and showed an effect of syntactic constraint. Stress facilitation was not found when targets were word-final. These results were interpreted in terms of stress-facilitated lexical segmentation and access processes. The short-term memory probe task introduced in this research was shown to be sensitive to stress, but not to other sentential factors that were examined. The application of this paradigm to issues in metrical phonology and short-term memory is discussed. [Work supported by AFOSR.]
Perceptual independence of pulse rates used in speech coding schemes for a multichannel cochlear implant
Authors: Lai W. K. , Tong Y. C. , Millar J. B. , Clark G. M. , van Hoesel R.
Published: October 2005

Two of the most important electric signal parameters for presenting useful acoustic speech information to the University of Melbourne/Nucleus multichannel cochlear implant are electrode position and electric pulse rate. The possibility of using pulse rate to encode acoustic speech information was investigated in this study. A set of 16 (4×4) orthogonal stimuli comprising all possible combinations of four different pulse rates on each of two fixed bipolar electrode pairs was presented to three cochlear implant recipients for identification. Tests for perceptual independence in terms of information transmission indicate that the two pulse rates presented on each of the two electrode pairs were perceived as partially separable dimensions. It may therefore be possible to use individual pulse rates on different electrode pairs in a speech coding scheme to present further acoustic speech information to a multichannel cochlear implant. [Work supported by NIH Contract NOI-NS-9-2400 and NIH Grant NS21027.]
A gating task to measure the contribution of contextual information in speech perception by cochlear implant wearers
Authors: Zwolan Teresa A. , Carrell Thomas D.
Published: October 2005

Contextual cue utilization by two groups of cochlear implant wearers was measured with a gating task. Group 1 consisted of implant wearers who were able to perceive open-set speech (unpracticed, unfamiliar words and sentences), while group 2 consisted of implant wearers who were unable to perceive open-set speech. Gated stimuli consisted of high- and low-predictability sentences taken from the speech perception in noise (SPIN) test. Subjects' responses were evaluated to determine the amount of acoustic information required for phoneme and word recognition as well as the percentage of phonemes and words identified in each type of context. Results indicated that open-set listeners required less information than non-open-set listeners to recognize phonemes and words. More importantly, a predictability-by-listener interaction demonstrated that open-set listeners were better able to make use of contextual cues than were non-open-set listeners.
Simplified tones for profoundly deaf tone-language speakers
Authors: Ching Teresa
Published: October 2005

The larynx frequency pattern of speech contributes significantly to speech communication when it is dependent on lipreading, especially in a tone-language context. Tests using syllable-based pitch contrasts as well as connected speech in controlled interactive situations show that receptive skills can be considerably improved both in Mandarin and Cantonese (two Chinese dialects). On the basis of findings which show that a sinusoidal presentation of voice pitch changes is more beneficial to a profoundly deaf person than the complete acoustic signal [S. Rosen and A. J. Fourcin, Speech, Hearing and Language, U.C.L. No. 1 (1983)], the present work adopts this simplification approach in an attempt to enhance prosodic skills in profoundly deaf Cantonese children. Results indicate that productive and perceptual skills can be significantly enhanced by using a simplified auditory presentation. These have important implications for the design of signal processing aids and the development of speech skills in the profoundly deaf tone-language speaker.
Axisymmetric vibration of spheroidal shells for high-aspect ratio
Authors: Chen Pei-Tai , Ginsberg Jerry H.
Published: October 2005

Assumed mode expansions based on Hamilton's principle were employed in an earlier presentation [P. T. Chen and J. H. Ginsberg, J. Acoust. Soc. Am. Suppl. 1 85, S139 (1989)] to study fluid-structure interactions of a submerged prolate spheroidal shell of low-aspect ratio. In that approach, a straightforward analysis of the in uacuo shell vibration properties was found to be efficient for the lower and upper branches of axisymmetric deformation, provided that the aspect ratio of the spheroid is not large. The present paper addresses methods by which the first few modes of both branches may be determined for cases of high-aspect ratio. In order to overcome numerical ill-conditioning, an iterative variational method is employed to seep the natural frequencies and mode shapes with increasing mode number. In addition, in order to ensure convergence to the upper branch, which occurs in a much higher frequency band that that of the lower branch, an artificial potential energy affecting only high wavenumbers is added. The model data derived in this manner are used to analyze a case of forced vibration of a submerged shell. The results are compared with Yen and Dimaggio's finite difference solution [J. Acoust. Soc. Am. 41, 618–626 (1967)]. [Work supported by the Office of Naval Research, Code 1132-SM.]
Response of an infinitely long cylindrical shell with a convected internal fluid
Authors: Leyrat G. , Cuschieri J. M.
Published: October 2005

The response of an infinitely long cylindrical shell excited by a point force, with an internal heavy medium (water) moving with a constant flow speed is examined. The solution is obtained by linearizing the differential equations describing the behavior of the fluid and the structure by using a spatial Fourier transform in the axial direction and a modal decomposition in the circumferential direction. The inverse Fourier transform of the solution of the linear set of equations gives the coupled solution for the shell/fluid system. Using this solution, input and transfer mobility functions are determined for a copper pipe of radius 0.025 m and wall thickness 1.5 min. For the input mobility functions, the nearfield effects combine with the coriolis effects of the moving fluid to produce instabilities of the circumferential modes at velocities that depend upon the mode number. For the transfer mobility functions, broadband peaks appear caused by the phase lag between waves of different circumferential mode numbers. These peaks are shifted and modified as the flow speed changes.
Response of a fluid-filled spherical shell submerged in an infinite fluid medium to a transient acoustic wave
Authors: Geers Thomas L. , Zhang Peizhen
Published: October 2005

Transient response problems involving fluid-filled shell structures submerged in infinite fluid media arise in various fields, including medicine, defense, and materials engineering. A canonical problem of this class possesses a spherical geometry, for which no solutions apparently exist. This paper delineates a relatively simple formulation and method of solution for this canonical problem and presents solutions for excitation by an incident step wave. The formulation begins with the familiar equations of motion for a thin spherical shell and the wave equation for the internal and external fluid domains. The Laplace transform is invoked, and the usual separation of variables method yields modal expressions involving Legendre polynomials and modified spherical Bessel functions. The explicit expressions for the latter are then manipulated to yield, after transform inversion, delayed differential equations in time for each response mode of the shell-fluid system, which are integrated numerically in time. Complete response solutions then follow by modal superposition, with special techniques being employed to improve modal convergence. To validate the methodology, numerical results are first presented for an incident step wave exciting a shell with mass density and acoustic velocity identical to the corresponding properties characterizing the same internal and external fluid. Results are then shown for a step-wave-excited steel shell containing water and submerged in water. [Work supported by DNA.]
Trends in hearing aids: From a technical to an audiological device
Authors: Lyregaard Poul Erik
Published: October 2005

In essence, a hearing aid is an acoustical amplifier, and has, therefore, traditionally been described in technical terms, as defined in the appropriate standards (IEC, ANSI). The advent of prescriptive fitting methods, plus a better understanding of the in situ acoustics, has had a significant impact on hearing aid design, but also on the way in which the acoustical properties are presented. Hearing aid dispensers tend to have an audiological rather than acoustical background, and the data to be employed in individual hearing aid fitting are also audiologically based. Hence, there is a strong case for manufacturers to adopt an audiological characterization of their hearing aids, even encompassing that hearing aid adjustments be calibrated in hearing loss and UCL units. Conversions between technical and audiological data for a hearing aid will be discussed and illustrated. In part, such conversions are contingent upon hearing aid measurements performed on the KEMAR manikin, one of Hugh Knowles' many contributions in the field of hearing aids.
Addressing Hugh Knowles' fundamental question: What would “high fidelity” mean for a hearing aid wearer?
Authors: Killion Mead C.
Published: October 2005

First assigned by Hugh Knowles to update Hugh's earlier estimate of the difference between real-ear and 2-cm3 coupler frequency-response curves for hearing aids, the author later did his Ph.D. research on the topic of the design and evaluation of high-fidelity hearing aids. Throughout this research, Hugh's probing questions were both useful and sometimes embarrassing. He made it impossible to give superficial answers to difficult questions about (1) the occlusion-effect (“hollow voice”) problem with closed-mold fittings, (2) what gain and frequency response would be required for a perception of high-fidelity sound by someone with a substantial high-frequency hearing loss, and (3) what directional/frequency-response accuracy in the hearing aid would be required for a “high fidelity” sense of the auditory space. Partly, as a result of Hugh's questions and contributions, a good solution has now been obtained to the occlusion-effect problem (reinventing Zwislocki's 1953 solution that he thought probably impractical), there are now transducers and “acoustic plumbing” techniques permitting 40-Hz to 16-kHz reproduction in hearing aids with fidelity comparable to that of professional studio monitors, in-the-canal hearing aids with directional properties comparable to that of the unaided ear, and a new integrated-circuit hearing-aid amplifier chip with level-dependent high-frequency emphasis that is designed to provide nearly “transparent” amplification to the wearer. Recorded demonstrations will be played if time permits.
Electroacoustic test methods for evaluation of hearing-aid performance
Authors: Beck L. B. , Burnett E. D. , Causey G. D.
Published: October 2005

The purpose of this paper is to describe the development of electroacoustic test procedures for evaluation of hearing aids by the Veterans Administration (VA). These methods have evolved from the use of the 2-cm3 coupler with pure tones as the input stimuli to the use of the KEMAR manikin with various types of complex noise as input stimuli. Hugh Knowles served as a consultant to the VA and his efforts, both for the VA and the research and development community at large, formed the framework for the evaluation program in place today. His contribution and leadership have resulted in procedures that permit the evaluation of a hearing aid in a manner very similar to the way it will be used by the wearer. Current use of FFT analysis and strategies for evaluating noise reduction circuits and in-the-ear hearing aids will be highlighted.
An experimental investigation of bubble clouds as sources of ambient noise
Authors: Yoon S. W. , Crum L. A. , Prosperetti A.
Published: October 2005

Collective motions of bubble clouds are experimentally investigated as an ocean ambient noise source. The experiments, performed in a water tank, show that the frequency due to the bubble column resonance depends inversely on the logarithm of the void fraction of the water/bubble mixture. It is also observed that the frequency depends inversely on the volume of the bubble column. These experimental results indicate that the volume resonance of the bubble column is one of the more likely mechanisms for an ocean ambient noise source between 100 and 1000 Hz. [Work supported by ONR and KOSEF (SWY).]
Vertical directionality of low-frequency wind-induced noise
Authors: Sotirin B. J. , Hodgkiss W. S.
Published: October 2005

The fine-scale structure of the directional ambient noise field is revealed using a large aperture (900 m) vertical array. Frequency and spatial spectral estimates are calculated during the passage of a local storm, providing a detailed study of ambient noise levels at low frequencies (15–130 Hz) as wind speed increases from 2 to 12 m/s over a 21-h period. Spectral levels of beams directed horizontally show the time and spatial variability of distant sources. Spectral levels of beams directed toward the surface reflect local noise sources and display a threshold type behavior with level changes up to 5 dB, suggesting the abrupt onset of a source mechanism such as breaking waves. Subsequent thresholds may indicate a change in source mechanism such as the conversion from spilling breakers to plunging breakers. [Work supported by ONT.]
The wedge assemblage method applied to backscattering from 3-D wind-driven sea surfaces
Authors: Keiffer Richard S. , Novarini Jorge C.
Published: October 2005

Numerical simulations of acoustic backscatter from fully 3-D wind-driven sea surfaces are performed and compared against experimental data. For this work, the directional seas considered are generated from either the Pierson-Moskowitz or JONSWAP spectrum and the back-scattering is calculated via a wedge assemblage (WA) method that has been generalized to handle 3-D surfaces. The WA approach is a time-domain method that provides the acoustic impulse response in terms of diffractions and image contributions from the scattering surface. The statistics of the scattered field are calculated by considering an appropriate ensemble of surfaces realizations. Comparisons between simulations and experimental data are examined for low frequencies (<500 Hz) and for grazing angles 10–89 deg.
Simulation of underwater acoustic scattering from large bubble plumes
Authors: Chin-Bing Stanley A. , Murphy Joseph E.
Published: October 2005

The finite-element ocean acoustic computer models, FFRAME and PE-FFRAME, have been used to gain insight into the dominant mechanisms involved in underwater acoustic scattering by large entrained bubble plumes near the ocean surface. Various types of incident acoustic fields (normal mode, PE, etc.) are used to interact with the plume while using the finite-element models to calculate the full-wave solution for the total field. Initially, the plume is treated as an acoustically soft boundary and various stages in the plume's life cycle are simulated. The simulation studies include the interaction of different propagating modes with the plume, and the effects of frequency and angle of incidence on the backscattered field.
The normal mode theory with bottom elasticity and range dependence
Authors: Arvelo Juan I. , Überall Herbert
Published: October 2005

The theory of normal modes has been extended to accommodate the effects of shear waves in the elastic ocean floor. The effect of absorption has also been included as the imaginary component of the shear and compressional wavenumbers. A semi-infinite elastic bottom basement layer has been included and it was found that the absorption causes the wavenumber spectrum of the radiating modes to be inherently discrete, hence, the number of radiating modes is drastically reduced relative to the ones obtained with the deep false boundary. The range dependence of the acoustic properties and of the boundaries of the ocean has been included by a modified version of the adiabatic normal mode theory. Comparisons are made with experimental measurements of the transmission loss of underwater explosives and with results from Collins' HEPE mode. [Work supported by the Independent Research Board of the Naval Surface Warfare Center and by NORDA.]
Measurement of shallow-water sound transmission on the New Jersey continental shelf
Authors: Maiocco Lynne , Carey William , Parssinen Edwin , Doutt James
Published: October 2005

Calibrated acoustic transmission measurements were made on the New Jersey continental shelf in the vicinity of the AMCOR 6010 borehole, an area with known geophysical properties. The water depth was 70 m and the experiment was conducted under known oceanographic conditions as shown by measurements of salinity, temperature, and sound speed versus depth. The acoustic measurements were performed with a vertical array of 24 equally spaced hydrophones, one of which was on the bottom. A calibrated source was used to obtain transmission loss data to ranges of 26 km at discrete frequencies from 50 to 600 Hz. The data from three hydrophones were processed for a 4-km range using techniques developed by Frisk and Lynch [J. Acoust. Soc. Am. 76, 205–216 (1984)] to yield wavenumber spectra. The transmission results of three hydrophones at 50 Hz were found to have an interference pattern clearly resulting from two modes. Low-frequency data (20 to 50 Hz) were obtained at discrete locations from the buoy to yield values of normal boundary impedance. Transmission loss data were obtained at eight frequencies, 50 to 600 Hz, along a constant depth and gradually changing depth radial. The constant depth radial results were compared to calculations performed with IFD-PE and SAFARI using a geoacoustic model based on measured geophysical properties. Predicted and measured levels generally agreed, however, differences in the computed and the measured modal interference patterns were observed.
Enhanced propagation in a foamy medium
Authors: Cowley A. , Harrison C. H. , Baird A. , Gedrich T. V. , Ferrari C. A.
Published: October 2005

Entrained bubbles have well-known effects on the acoustic properties of water, namely the velocity of sound, absorption, and backscattering. These, in turn, have a strong influence on sound propagation and reverberation in the vicinity of a ship. A numerical model is discussed that uses as input data bubble population densities derived from hydrodynamic and bubble transport calculations, on a three-dimensional rectangular grid. Acoustic properties as a function of frequency are computed at each grid point using classical formulas, accounting for both resonant and non-resonant contributions. Relatively fast algorithms have been developed to calculate the three-dimensional ray paths (curved or straight rays), and transmission loss contours for arbitarily positioned source and receiver. Reverberation level as a function of time is determined assuming straight rays, but including simple multipaths and surface backscattering. Some interesting examples are presented including cases in which transmission may be enhanced by ducting or focusing at certain frequencies.
The sources of trans-spectral coherence in the radiation from rotating blades
Authors: Vaidya P. G. , Girard Jeffrey J. , Moulten Carey L.
Published: October 2005

Rotating blades (installed in fans, propellers, and helicopters) radiate both discrete and broadband sound. This radiation can be examined for the coherence of the trans-phase. Trans-phase is the phase difference between the components at two different frequencies when one of them has been sent through a nonlinear filter. If the trans-phase shows coherence, the signals are said to be trans-spectrally coherent. Various sources of the rotating machinery noises contribute to the coherence in different ways. Existence of such coherence can help in the diagnostics and identification of the sources and in the detection and classification of the machinery in presence of extraneous noise.
Accurate spectral estimation of multiple sinusoids using an FFT
Authors: Burgess John C.
Published: October 2005

Accurate estimates of amplitude, frequency, and phase of multiple sinusoids in a signal can be obtained using a method described earlier [J. C. Burgess, J. Acoust. Soc. Am. Suppl. 1 83, S92 (1988)]. FFT size required for specified spectral resolution, as well as some computational aspects, is discussed.
Low peak factor waveforms for efficient system analysis using a fast Fourier transform
Authors: Burgess John C.
Published: October 2005

Low peak factor waveforms [M. R. Schroeder, Number Theory in Science and Communication (Springer, New York, 1984)] can be designed so that multiple harmonics in the waveform appear to have the ideal spectrum of a true Fourier transform when the waveform is analyzed using a fast Fourier transform (FFT). If the harmonics in the waveform are restricted to those having an integral number of cycles in the FFT data record length, there is no leakage, and the magnitudes and phases of the Fourier coefficients are identical to those of the harmonics in the waveform. Low peak factor waveforms drastically reduce peak power requirements. For example, the peak power requirement for a waveform having 63 harmonics can be reduced by a factor of about 30. Waveforms can be designed having harmonics with constant, octave, and close approximations to one-third octave frequency differences. This approach can provide an efficient alternative to other methods in current use for measuring system transfer functions. [Work supported in part by the National Research Council of Canada and the Lanikai Foundation.]
Application of the fast Hartley transform to industrial noise analysis
Authors: Gan Woon S.
Published: October 2005

The application of the fast Hartley transform (FHT) to industrial noise analysis to replace the fast Fourier transform (FFT) is proposed. The Hartley transform of a real-valued function is also real-valued. Its evaluation does not involve complex functions and it has the same formula for forward and inverse transform. Thus it has a speed advantage over FFT. The type of FHT algorithm chosen is due to H. V. Sorensen et al. [IEEE Trans. Acoust. Speech Signal Process. ASSP-33, 1231–1238 (1985)]. As an illustration, the two-channel FFT signal processor is chosen. The computation of the following functions is repeated: autocorrelation function, cross-correlation function, power spectrum, and the acoustic intensity of real-numbered data by replacing the FFT with FHT. A detailed analysis of the time that can be saved in computation by using FHT is made. A key advantage of the Hartley transform technique for large autocorrelation is the use of the same formula for transforming and inverse transforming.
Finite element simulation of an active noise-canceling scheme
Authors: Mohammed Osama A. , Ramdin Kurt S. , Raman Vijay , Yen Kang
Published: October 2005

This paper details a finite-element computer simulation scheme of an axis-symmetric physical setup of an active noise canceling headset. The results of this simulation show the distribution of sound-pressure levels inside the ear canal. Two-dimensional simulations of an axis-symmetric configuration of the ear cup were carried out. A point source was used to cancel out the noise. From these simulations it can be concluded that the noise inside the ear canal can be reduced. The amount of reduction depends on the strength and phase of the canceling source. On the other hand, a point source is not a practical device. To overcome this problem, new structures for speakers have to be created and simulated to check if their radiation is in accordance with that of practical devices. An algorithm proposed earlier by these authors for the estimation of the SPL noise reduction that can be accomplished inside the ear canal has been added. [Work supported by the Florida High Technology and Industry Council.]
Sound-pressure distribution on a cross section of a duct for active noise cancellation
Authors: Yen Kang , Mohammed Osama A. , Raman Vijay , Ramdin Kurt S.
Published: October 2005

Active cancellation techniques for low-frequency noise have been applied to some industrial plants. Field tests show very useful results. In the past, research work has concentrated on the theory of minimization of the enclosed sound field along one axis. However, in practice the direction of the sound source is uncertain. Especially when the noise source is very close to the duct, the direction of the sound wave will be a dominant design factor in order to achieve the cancellation at the other end of the duct. In this paper, a set of directional and finite-size cancellation sources inside the duct is considered. The sound-pressure distribution across any cross section in the duct is the primary research interest. The results provide information on the design of noise cancellation devices such as noise cancellation earphones. [Work supported by the Florida High Technology and Industry Council.]
Active attenuation of pure tones in a lined duct using adaptive filtering
Authors: Laget Dominique , Hixson Elmer L.
Published: October 2005

Fan noise is characterized by strong discrete blade passing tones added to broadband aerodynamic noise. In air conditioner ducts, absorptive liners provide reduction of the noise at the higher frequencies but are ineffective at lower frequencies. An active adaptive controller is mostly effective for low-frequency narrow-band noises. In this study, noise control combining a duct liner and an adaptive attenuator has been implemented. The stability and convergence properties of the adaptive attenuator depend on the compensation of the transducers and acoustic delays. Compensations using an IIR filter, a delay, and an FIR filter designed with the LMS algorithm have been studied. After simulation, the controllers based on a filtered-x and a delayed-x LMS algorithm have been implemented on a digital signal processing board. Measurements inside and outside a lined air conditioner duct showed significant pure-tone attenuation, especially when a novel filtering-x method is used.
Numerical simulation of a sound field generated by flow around a circular cylinder
Authors: Zhu Z. C. , Zeng L. Z.
Published: October 2005

Flow around a circular cylinder is a classical problem, but it is difficult to simulate a Karman vortex street in the wake numerically. In previous numerical works artificial rotation had to be added to the cylinder to separate the vortex from it. Obviously it would influence the original solution. In addition, body conditions were usually not satisfied correctly in many numerical works. There are two major improvements in this paper. (1) In order to separate the vortex, it is proposed that only one asymmetrical grid-point condition is given at the farfield of the wake. (2) A recent treatment is used to guarantee that body conditions are satisfied precisely. Then numerical simulation of flow around a circular cylinder can be completed correctly and successfully, with the application of ADIBGE schemes to solve incompressible N-S equations in the form of stream function and vorticity at Reynolds numbers from 1 to 1000. Sound radiation is calculated by numerical integrating FW-H equations at an Re of 100. Then careful analysis and comparison are made for all parts of sound sources. Finally several useful conclusions are given.
On the transition between slip and rigid boundary conditions between two solid media
Authors: Wang Y. J. , Rokhlin S. I.
Published: October 2005

In order to describe the transition between slip and rigid boundary conditions, a model with a thin interphasial viscoelastic layer is introduced. By changing the relaxation time (ωτ), the transition from a viscous liquid to a solid layer is observed. When the interface layer is in the liquid state with very small shear nodulus the shear stress components on the boundary vanish, resulting in a slip boundary condition. On the other hand, after the layer is solidified, the two semispaces become welded with the transition of shear and longitudinal stresses to one another. It is shown that the ratio of a hydrodynamic boundary layer thickness to the interface layer thickness plays a major role in this transition. A simple approximate equation for the reflection coefficient from the interface is derived and compared to the exact solutions for different transitional stages of the boundary condition. Theoretical predictions are compared to experimental results.
Method for measurement of elastic constants of thin anisotropic membranes
Authors: Wang W. , Rokhlin S. I.
Published: October 2005

In this work the elastic properties of a thin layer of porous aluminum oxide obtained by phosphorus anodization have been studied by a novel ultrasonic technique. The thickness of the porous oxide samples is about 50 μm with through vertical pores of about 2000 Å diameter and porosity of about 70%. The ultrasonic technique is based on measurements of the angle of the minimum of the transmitted energy at oblique incidence on the immeresed sample. The effective elastic constants that have been theoretically calculated from the experimental data are in good agreement with that found from estimations of bulk properties of anisotropic porous media. The knowledge of the properties of porous anisotropic anodized Al2O3 layers has major importance for adhesive joint characterization since such layers play an important part in interphasial strength of the joint.
Surface motion reconstruction based on half-space field data
Authors: Bennink David D. , Pate Anna L.
Published: October 2005

A finite but otherwise arbitrary normal velocity profile at the boundary of a half-space is expanded in a set of orthonormal functions. The radiation problem is then formulated in terms of the moments of this profile. (Details of this formulation are given in the preceding abstract.) The radiation problem is discretized to obtain a matrix formulation and a singular value decomposition method is used to estimate the velocity profile from measurements of the radiated field. Additional constraints are imposed upon the derivatives of the profile in order to overcome the ill-posed nature of the inverse source problem. The accuracy of the method will be discussed in terms of both the positioning of the field locations (nearfield versus farfield) and the weighting of constraints. Examples of velocity profile reconstruction will be given for a few simple radiators. [Work supported by Center for NDE at Iowa State University.]
A prior knowledge based optimal Wiener filtering approach to ultrasonic scattering amplitude estimation
Authors: Neal Steven P. , Thompson Donald O.
Published: October 2005

In order to interpret measurements of the scattered wave field resulting from sonification of an object, the effects of the measurement system must be removed from the measured data. In ultrasonic nondestructive evaluation, estimation of the farfield scattering amplitude for a flaw in a material, e.g., a crack, void, or inclusion, involves removing the measurement system effects in the presence of electronic noise and noise due to the scattering of sound from inherent material features such as grain boundaries. The purpose of the work reported here was to evaluate an optimal Wiener filtering approach to scattering amplitude estimation. The filter was shown to determine an optimal estimate as the weighted average of the information derived from measurement of the scattered wave field and prior information about the flaw distribution. The optimal Wiener filter was evaluated and compared to a nonoptimal form of the Wiener filter. The optimal Wiener filter was shown to provide improved scattering amplitude estimates by filtering out the noise and by utilizing prior flaw information. [Work supported by the Director of Energy Research, Office of Basic Energy Sciences.]
The measurement and analysis of acoustic noise as a random variable in ultrasonic nondestructive evaluation
Authors: Neal Steven P. , Thompson Donald O.
Published: October 2005

Sound scattered from inherent material features such as grain boundaries becomes a source of acoustic noise when attempting to detect or characterize a flaw in a material, e.g., a crack, void, or inclusion. In formulating a stochastic model to describe a flaw detection or flaw characterization experiment, acoustic noise is often assumed to be an uncorrelated, Gaussian random variable with zero mean and known average power spectrum. The work reported here focuses on the measurement of acoustic noise and on the evaluation of these assumptions for the cases of scattering from grains in stainless steel and scattering from porosity in aluminum, respectively. An estimate of the average power spectrum is determined for each case. It is shown that in both the time domain and in the frequency domain the acoustic noise considered has zero mean and is reasonably uncorrelated and Gaussian. [Work supported by the Director of Energy Research, Office of Basic Energy Sciences.]
Psychometric functions for level discrimination in simulated and cochlear impairment
Authors: Buus Søren , Florentine Mary , Zwicker Tilmann
Published: October 2005

To determine whether the form of psychometric functions for 2I, 2AFC level discrimination changes with hearing impairment, measurements were obtained in listeners with cochlear impairments and normal listeners with impairments simulated by masking. Ten increment levels were presented in random order within blocks of 100 trials. Stimuli were tones at 0.25, 1, 8, or 14 kHz or white noises chosen to encompass a wide range of conditions and difference limens. Durations were either 10 or 500 ms and levels ranged from 50 to 90 dB SPL. For each condition, at least 20 blocks were presented in mixed order. Results show that the sensitivity d' is nearly proportional to ΔL(= 20 log [(p + Δp)/p], where p is pressure) over the entire range of difference limens for both simulated and real impairments as well as for normal listeners tested in the quiet. These results indicate that if the transformation from stimulus intensity to decision variable is affected by heating impairment or masking, the change affects both the mean and the standard deviation in the same manner. They also lend support to the notion that ΔL—plotted on a logarithmic scale—is an appropriate representation of level-discrimination performance. [Work supported by NIH.]
Noise reduction hearing aids: A tutorial
Authors: Bentler Ruth A. , Bergman Brenda B.
Published: October 2005

In the last few years, there has been a proliferation of (so-called) noise reduction hearing aids onto the market. While some of these designs are little more than multiband compression circuits, others employ adaptive filtering (dependent upon overall input level), adaptive compression (dependent upon duration of high-level input signals), and variations thereof. The purpose of this poster session is twofold: (1) using graphics and schematics an explanation of six available “noise reduction” circuits will be provided. This will serve as a tutorial for those audiologists attempting to keep up with design and marketing strategies for noise reduction; (2) preliminary results of a large clinical project begun late in 1988 aimed at determining how effective different circuits are for different configurations and/or degrees of hearing loss in new and experienced users will be presented. Success of effectiveness will eventually be measured over a 12-month period using (a) speech recognition (in quiet and noise), (b) portions of the Hearing Performance Inventory (Giolas and Owens, 1979), (c) expectations checklist (Seyfried and Anderson, 1989), (d) qualitative measure task (modified from Gabrielson and Lindstrom, 1985), as well as (e) ratings of “satisfaction” taken at 6 months and 12 months post fitting.
The role of burst duration in noise-induced hearing loss
Authors: Ward W. Dixon
Published: October 2005

Intermittent daily exposures to noise are less hazardous to hearing than steady exposures of the same energy. This reduction of hazard depends on the two fundamental parameters of intermittence: burst duration and pause duration. In order to examine their relative importance, groups of chinchillas were given daily exposure, Monday–Friday, for a period of 9 weeks, to a 700- to 2800-Hz noise at 110 dB SPL in the form of (a) a single 48-min burst; (b) forty 1.2-min bursts given every 12 min, hence with 10.8-min pauses; (c) ten 4.8-min bursts every 48 min (pauses of 43.2 min); and (d) ten 4.8-rain bursts every 12 min (pauses of 7.2 min). Exposure (a) produced an average permanent threshold shift (PTS) at 1, 2, 4, and 8 kHz of 18 dB, exposure (b) gave a PTS of 4 dB, and exposures (c) and (d) each generated a PTS of 13 dB. Burst duration is, therefore, the more important parameter, as increasing it by a factor of 4 more than doubled the damage, whereas changing the pause duration by a factor of 6 had no effect. This generalization was confirmed by results of a daily exposure to forty 0.12-min bursts at 120 dB every 12 min, which produced negligible PTS. [Research supported by NS12125.]
An efficient default hierarchy for pronouncing English
Authors: Mniszewski Susan M. , Hochberg Judith G. , Calleja Teri
Published: October 2005

A default hierarchy induction paradigm was trained to translate English words from standard orthography into their phonetic transcriptions. Associations between letters and phonemes were established by building a hierarchy of rules. The most common rules, at the lowest level of the hierarchy, are based on single letter-to-phoneme correspondences. Higher level rules, taking into account contexts of increasing numbers of surrounding letters, capture exceptions with respect to lower levels. The hierarchy of rules is then used to pronounce words in a test set by searching the rule structure from the highest to the lowest level. A previous version of this paradigm, called “HIERtalker” [An et al., Proceedings of the IEEE Int. Conf. on Neural Networks, Vol. II, 221–228 (1988)], used only rules in which the letter being pronounced was centered in its context of surrounding letters. Further work has shown that efficiency (in the sense of requiring fewer rules) is improved by the use of left- and right-oriented rules and search strategies allowing competition between rules representing different orientations. [Work supported by DOE Contract W-7405-ENG-36.]
A simple neural-model algorithm for pitch determination
Authors: Drumright David G.
Published: October 2005

A very simple algorithm has been developed, which gives spectral information similar to that provided by LPC algorithms. The method is based on a digital model of a reflecting tube with threshold-firing sensors along the tube. The sensor thresholds are adjusted to keep their firing rates moderate. An optional second level of feedback allows for training the model to give judgments of F0. Without the second level, the algorithm runs in real time on a PC AT type of computer. The algorithm is one feature of a program with general facilities for handling speech.
A Japanese text-to-speech conversion system using pitch-controlled residual wave excitation
Authors: Iwata Kazuhiko , Ozawa Kazunori , Mitome Yukio , Watanabe Takao
Published: October 2005

A Japanese text-to-speech conversion system was developed that could generate fluent speech by concatenating CV and VC units. A new synthesis method was proposed, where residual waves were used as excitation signals for an LPC synthesis filter in all portions of each unit. LPC filter coefficients were calculated by approximating spectral envelopes extracted by the improved cepstral analysis method, which was less affected by pitch frequency than the conventional LPC method. Therefore, the synthetic speech generated by the proposed method had extremely high quality, even when pitch frequencies of the residual waves were widely changed. Moreover, in order to realize natural rhythms, phoneme duration was determined by new rules. The rules were created by taking into account the various contexts (such as preceding and following phonemes and the position of a phoneme in a phrase), and by statistically analyzing a large speech database. A comprehension test using 1000 phonetic balanced words showed that the proposed system achieved 97.4% of high accuracy rate.
Interactive utterance modeling with Delta for research in speech synthesis and speech perception
Authors: Hertz Susan R.
Published: October 2005

Productive research in speech synthesis and perception requires the ability to readily test the effect of selected modifications to an utterance (e.g., shortening of voice onset time). With the Delta system software [J. Acoust. Soc. Am. Suppl. 1 85, S124 (1989)], “multistream” utterance representations (deltas) consisting of coordinated user-defined acoustic or articulatory units can be built and manipulated interactively, and the effects of modifications tested through synthesis. To facilitate experimentation in the acoustic domain, a set of Delta procedures have been developed for interactively generating an initial delta for a Klatt synthesizer in any natural language. This delta contains a phoneme stream, streams for synthesizer parameters, and “slots” for perceptually important acoustic events such as stop bursts and formant transitions. With such a delta, the experimenter can easily manipulate acoustic values and their timing. Such interactive utterance modeling is being used to develop synthesis rules for several English dialects and for Hindi, and soon the approach will be extended to other languages. The system will be demonstrated. [Work supported by U.S. Dept. of Education and NIH.]
A knowledge-based temporal alignment system involving stochastic methods
Authors: Kabré H. , Malet J. F. , Pécatte J. M. , Pérennou G. , Vigouroux N.
Published: October 2005

In this paper the principle behind an automatic signal alignment system, used for speech database labeling, is described. This system is based on both a phonetic knowledge subsystem and a stochastic training procedure. It operates in two passes. During the first one, for each frame, temporal cues are worked out, similar to those handled by the hand labeler. The frames are next recombined into phonetic events that are assigned broad labels—e.g., vocalic, fricative, liquid, sonorant, occlusive, etc. The second pass aligns phonetic events onto a phonetic transcription—provided by an expert. It resorts to a stochastic model entailing (i) a phonetic knowledge base and (ii) a channel modeling first-level decoding errors. The training of the phonetic knowledge base consists in fine tuning the assessment of probability weights. This is done on hand-labeled corpora. Results, on both French and English corpora, will be presented, attention being paid to the problem of system multilingual adaptability.
Phonological rule set complexity as a factor in the performance of a very large vocabulary automatic word recognition system
Authors: Seitz Philip F. , Gupta Vishwa N. , Lennig Matthew , Kenny Patrick , Deng Li , O'Shaughnessy Douglas , Mermelstein Paul
Published: October 2005

Generative phonological rules were used to derive surface forms of words for the 89 000-word lexicon of the INRS-Télécommunications speaker-dependent, isolated word recognizer. The representation of the surface forms is in terms of the 41 phonemelike subword recognition units that are trained and recognized using 25-component continuous mixture hidden Markov models. Three sets of surface forms were generated by three phonological rule sets that differ in the amount of phonetic and phonological variability they represent. The three dictionaries thus derived were of different sizes, having ratios of surface to base forms of 1.04, 1.23, and 2.40. Recognition experiments were run on the speech of one talker. Representing pronunciation variations explicitly in the dictionary affords a recognition performance advantage, but this advantage is partially offset by a higher ratio of surface to base forms, which leads to a larger and more densely populated search space and greater confusability of subword unit sequences. [Work supported by the Natural Sciences and Engineering Research Council of Canada.]
Resonant multiple scattering from bubble clusters
Authors: Gilbert Kenneth E. , Wang Lintao
Published: October 2005

Multiple scattering from clusters of bubbles has been computed for N = 2 to N = 200 bubbles. It was found that as multiple scattering increases, the resonant frequency of the bubble cluster decreases. For example, for a 1-cm-radius cluster composed of bubbles with single-bubble resonances at 3000 Hz, multiple scattering calculations predict collective resonances at approximately 2000, 1000, and 600 Hz for N = 10, 100, and 200, respectively. Preliminary measurements have been made for N = 2 to N = 7. The measured resonant frequencies are in good agreement with the theoretical predictions. Details of the theoretical and experimental results will be discussed along with approaches for applying the multiple scattering method to large clusters with irregular shapes.
Plane-wave scattering from a random fluid half-space
Authors: Tang Dajun , Frisk George V.
Published: October 2005

An integrodifferential equation method is applied to low-frequency plane-wave scattering by a random fluid half-space. The results of coherent reflection of a plane wave from a random fluid half-space have been previously reported [J. Acoust. Soc. Am. Suppl. 1 85, S70 (1989)]. In this paper, the random scattered field strength is calculated, assuming that the random sound-speed correlation function has independent correlation lengths along the vertical and horizontal directions. Numerical results obtained using this approach are compared to those obtained using the Born approximation. They show that the coherent energy loss due to scattering cannot be ignored at low frequencies, since the coherent wave can propagate to large depths in the random medium. This study has potential application to the understanding of ocean bottom reverberation problems.