bluuebunny/crossref_metadata_2025 huggingface dataset:
Matched-mode processing schemes of a moving point source
Authors: Chen Han-Yang , Lu I-Tai
Published: October 2005

Matched-mode processing schemes are presented to estimate the frequency, velocity, depth, and range information of a time-harmonic point source moving uniformly in a stratified oceanic waveguides. When the velocity of a moving source is much smaller than the sound velocity of the waveguide, the acoustic field generated by this moving source can be represented in terms of normal modes with different Doppler-shifted frequencies and, hence, different changes of modal eigenvalues. The shifted frequencies of each mode can be derived by the conventional approach for the Doppler effect in an unbounded medium with the corresponding modal phase velocity as the equivalent sound velocity of the medium. Then, the shifted modal eigenvalue is obtained by solving the transverse resonance equation of the waveguide. The frequency information alone is good enough to estimate the source velocity and frequency (unshifted), but the eigenvalue information has to be included to estimate the source depth and range. Processing schemes over the time domain (using a single receiver) and that over both space and time domains (using a vertical receiving array) are discussed in details with emphasis on deriving practical criteria of choosing processing parameters. These criteria are good for both shallow and deep oceans with arbitrarily stratified sound profiles.
A fast simulated-annealing algorithm for time-domain beamforming
Authors: Kuperman W. A. , Collins Michael D. , Perkins John S. , Davis N. R.
Published: October 2005

Most signal processing algorithms are based on the frequency-domain cross-spectral density matrix, which is constructed from a large time window of data from an array of hydrophones. Forming this matrix involves averaging the data, a process that wastes a great deal of potentially useful information. By working with an optimal time-domain beamformer, which uses all of the information in the received time series, it is possible to determine the bearings and signatures of plane-wave sources with a small time window of data and relatively few hydrophones. Due to the recent development of efficient methods such as simulated annealing for solving optimization problems involving many unknown parameters, optimal time-domain beamforming has the potential to become a very useful approach. Simulated annealing is analogous to slowly cooling a pure liquid to obtain a perfect crystal. For the optimal beamformer, the unknown bearings and time series appear to be analogous to a mixture of liquids of different freezing points, and it is necessary to scale out this difference to prevent the bearings from crystallizing separately. Examples will be presented to illustrate the effects of source level and bandwidth, noise level, array size, and number of hydrophones and to demonstrate that a priori information can be utilized to enhance the optimal beamformer.
Measurement accuracy of MVDR adaptive beamformers
Authors: Andrew Rex , Ho Khanh , Schmid Charles
Published: October 2005

Minimum variance distortionless response (MVDR) adaptive beamformers minimize noise under the constraint of maintaining a known signal amplitude response in the steered direction of the beam. These criteria make the MVDR beamformer attractive for measuring underwater signals from a known location in a noise background consisting of a number of point source interferences. This paper investigates the accuracy of measuring signals in such an environment by using a frequency domain implementation of a MVDR adaptive beamformer to process simulated and at-sea data. Examples will be shown of measurements using a 32-hydrophone planar array. Of special concern was the effect on measurement accuracy due to beam pointing error, adaptation time, and adaptive filter errors introduced by MVDR calculations. Also of interest was the trade-off of adaptation before or during the presence of the signal. Results are shown for three types of data scenarios: (1) a simulated sinusoidal signal and interference point noise sources in spatially uncorrelated noise, (2) a projected underwater sinusoidal signal in ambient noise, recorded off the eastern coast of Vancouver Island, Canada, and (3) an injected, simulated sinusoidal signal combined with recorded at-sea noise. Results showed that the MVDR beamformer significantly increased signal-to-noise ratios by canceling interfering noises, but did so at the expense of the measurement accuracy.
Waveforms with minimum power fluctuation
Authors: Hartmann William Morris , Pumplin Jon
Published: October 2005

Pumplin's algorithm [J. Pumplin, J. Acoust. SOC. Am. 78, 100-104 (1985)] is used to find waveforms that have the smallest possible power fluctuation for a given power spectrum. Such optimized waveforms also tend to have very low crest factors. For a wideband case, the algorithm must be run anew for each particular power spectrum. It has been applied to spectra where the first N harmonics have equal amplitude (N = 1,2,…,31). Alternatively, spectra of simple waveforms can be used as input; thus the algorithm generates, for example, the pseudo pulse wave, the pseudo triangle, or the pseudo half-wave-rectified sine. For narrow-band cases, some generality is possible because the operation of the algorithm is unchanged by a rigid translation of the spectrum along the frequency axis. The resulting signals are useful as optimally smooth maskers or stimuli evoking periodicity pitch. In general, it is expected that minimum-fluctuation waveforms can play an important role in the search for monaural phase perception. Psychoacousticians and physiologists are invited to recommend their favorite power spectra to our digital waveform factory. [Work supported by the National Institutes of Health.]
Auditory temporal acuity in hearing-impaired listeners for frequency-modulated signals
Authors: Madden John P. , Feth Lawrence L.
Published: October 2005

Recent studies of auditory temporal acuity in the hearing impaired have tended to focus on the resolution of amplitude modulation. This study compares the temporal resolution of normal and hearing-impaired subjects in an experimental paradigm using frequency-modulated stimuli. Three hearing-impaired and three normally hearing subjects had to distinguish two sinusoidal signals. The first, the glide signal, moved from its initial frequency over a linear path to its final frequency. The second, the step signal, was the same except that its trajectory followed a series of discrete steps in frequency. As the number of steps in the step signal increased, the duration of the individual steps decreased, and the signal more closely approximated the glide. The center frequencies of the signals were 500, 1000, and 4000 Hz. The subjects were asked to identify the interval containing the step signal in a 2-Q, 2AFC task. When the signals were presented at equal sensation levels, the impaired subjects exhibited significantly poorer discrimination than the normally hearing subjects, indicating a reduced ability to resolve the step modulation. Implications for formant processing by the hearing impaired will be discussed. [Work supported by a grant from AFOSR.]
The roles of three possible cues to dynamic vowel identity
Authors: Dutton Dawn L.
Published: October 2005

Recent research [Nearey and Assmann (1986);Strange (1989)] suggests that the changes in vowels over time are perceptually relevant to their identification. Possible “cues” for dynamic vowels include the frequency locus of spectral peaks over time, duration, and extent, rate, and direction of peak movement over time. The possible roles of duration and extent and rate of peak movement over time in cuing the [ʌ]-[a] distinction were examined using synthetic [ʌ]-[a] series. For each of three continua, two of the cues were varied while the third was held constant. Pilot testing showed that the continua were ambiguous on the basis of frequency alone and direction of spectral peak movement over time was nondistinctive. The ability of subjects to use the two vowel categories consistently was the basis on which the efficacy of the cues was judged. Duration was found to reliably cue vowel identity while no evidence of an effect of either rate of movement of spectral peak change or extent of spectral peak movement was found. Results will be discussed with respect to potential acoustic “cues” for dynamic vowels. [Work supported by NIDCD Grant DC 00219 to SUNY at Buffalo.]
On the perceptual vowel space of Modern Greek
Authors: Fourakis Marios S. , Hawks John W.
Published: October 2005

In this paper, 7703 vowel tokens were synthesized using the Klatt synthesis program. The values of F0, F1, F2, and F3 of each token were such that the token represented a unique point in a plane of the auditory perceptual space proposed by Miller [J. Acoust. Soc. Am. 85, 2114–2134 (1989)]. Each token was 400 ms long and had a rise-fall pitch contour. The tokens were identified by one native speaker of Modern Greek (MSF) as one of the five Greek vowels [i, e, a, o, u], or not as a vowel of Greek. In the latter case, the subject was then asked to indicate the closest Greek vowel to which the token could be assigned. Additionally, all tokens were rated for clarity on a scale from 1 (poor) to 5 (excellent). The results suggest that the vowels of Greek may be represented as nonoverlapping subspaces in the auditory perceptual space. The subspaces corresponding to the three point vowels of Greek ([i, a, u]) exhibit extensive overlap with the subspaces occupied by the same vowels in English, as determined from a similar experiment with a native speaker of American English. The subspaces corresponding to the Greek midvowels ([e, o]), on the other hand, show partial overlap with the spaces for American English [ɛ, ɔ]. The subspaces corresponding to the American English lax vowels [I, æ ʌ, u] are not used in Modern Greek. These results will be compared to subspaces obtained from production data for American English [Miller (1989)] and for Greek [Jongman et al., Language and Speech (in press)]. [Work supported by NIDCD.]
Can principal components be identified as distinctive features?
Authors: Nguyen-Trong N. , Santi S. , Cave C.
Published: October 2005

It is often considered that principal components analysis (PCA) makes it possible to define distinctive features of vowels. In this case, they are represented as points in an acoustic space. Applied to the vocalic system of French, a PCA gives three first factors that are usually associated with the three main distinctive features. The psychological reality of such factors has been less investigated, despite the importance of the question. The PCA is based on the notion of variance, stating that the descriptive value of a factor is proportional to the initial variance that it “explains.” Can the perceived information be represented as linear combinations of the original acoustic variables, that maximize the explained variance? An identification test from 60 realizations of the French oral vowels was carried out. Each vowel was synthesized and transformed through five different orthogonal projections. The corresponding confusion matrices confirm the phonetic interpretation of the axes. However, a divergence between explained variance and correct identification percentage is noticed, which is explained through a hypothesis that contradicts the acoustic dispersion theory.
Scattering of plane waves from objects having surface impedance discontinuities
Authors: Ferri Aldo A. , Rogers Peter H.
Published: October 2005

The scattering of acoustic plane waves from partially coated submerged objects is investigated. On the boundary between the uncoated and coated portions of the surface, an impedance discontinuity exists giving rise to sharply varying surface pressures and normal velocities. In the limiting case in which the uncoated surface has infinite impedance and the coated surface has zero impedance, it can be shown that a singularity in surface normal velocity exists. When finite and nonzero impedances are considered for the hard and soft boundary conditions, respectively, the singularity in normal velocity apparently disappears, but the rapid variation in surface pressures and normal velocities still presents numerical difficulties. Two specific systems are investigated: an infinite cylinder and a finite cylinder. Both systems are coated over two unconnected regions on their surfaces. The infinite cylinder is analyzed using a spectral expansion with the series coefficients evaluated using an overdetermined collocation technique. The finite cylinder is analyzed using a boundary element approach based on the standard Helmholtz integral equation. CHIEF points are used to avoid ill-conditioning at problem frequencies. [Work supported by ONR.]
Hybrid wavenumber integration—Boundary element approach to structural acoustics simulation
Authors: Schmidt Henrik , Ricks David , Getstoft Peter
Published: October 2005

Wavenumber integration (WI) techniques are well established for analysis of wave propagation in laminated or stratified media, inherently satisfying all boundary and radiation conditions. However, their area of applicability is restricted to separable geometries such as infinite laminated plates and cylindrical or spherical shells. In structural acoustics the boundary element method (BEM) is traditionally applied to determine the radiated and scattered field in the surrounding fluid, requiring a discretization of the entire surface of the structure under consideration, with the interior being treated by, e.g., finite elements. Here, a hybrid WIBEM approach is applied to study the wavenumber conversion induced by facets such as stiffeners and grooves in an otherwise perfectly laminated medium. By choosing the Green's function for the laminated medium, only the interface between the facet and the layered medium contributes to the surface integral in Green's theorem. Thus, only the boundary of the inhomogeneity needs to be discretized, leading to a highly efficient numerical implementation. The method is applicable to all problems where the major part of the structure is of separable geometry. Thus, in particular, it is intended for analysis of scattering and radiation by coated cylindrical shells with attached structures. In contrast to traditional BEM approaches, the scattered field is determined in the wavenumber domain, allowing direct physical interpretation in terms of modal coupling. Here the approach is illustrated by analyzing the wavenumber conversion in flat, coated plates with irregularities such as stiffeners and grooves. [Work supported by ONR.]
Studies on surface discretization in boundary element method applications
Authors: Pijaca J. , Prasad M. G.
Published: October 2005

The implementation of the boundary element method (BEM) to sound radiation problems consists of discretizing the vibrating surface into a number of elemental radiators and then applying the Helmholtz integral equation for predicting the acoustic field. As with other numerical techniques such as the finite element method, the accuracy of the model is dependent upon, among other factors, the discretization scheme. Creating an efficient mesh is important in order to insure accurate results with a reasonable amount of computational effort. Results of work conducted comparing the use of planar as well as linear and quadratic isoparametric elements are shown. The case studies involve modeling the acoustic field and radiation efficiency of a pulsating sphere at various wavenumbers and modeling the directivity pattern of a vibrating finite cylinder. Procedures for checking accuracy of solution are compared such as matrix condition number and velocity potential of interior points.
A banded matrix boundary element method
Authors: Main Geoffrey L. , Hahn Steven R.
Published: October 2005

Boundary element methods (BEMs) for the Helmholtz integral equation at frequency f result in matrices of order f4 elements requiring of f6 operations for the solution. Scattering or radiation problems for objects of more than a few ka in size result in prohibitively large computational problems. A method is developed for estimating the contribution of matrix elements corresponding to points more than 2–3 wavelengths apart in terms of matrix elements of points closer than the 2–3 wavelengths. The result is a banded matrix with a fixed bandwidth independent of f, of order f2 elements, and requiring of order f2 operations for the solution. The estimation for the points farther apart than 2–3 wavelengths is constructed in terms of the partial sums of closer points. For many objects of interest the bandwidth is approximately 1000, making the problems tractable. Combining the banded BEM with a finite element model of the object has the effect of increasing the bandwidths of the matrices by a relatively small amount. [Work supported by ONR.]
Elastic wave scattering from slender elongated inclusions
Authors: Hackman Roger H. , Lira Raymond
Published: October 2005

At a previous meeting of the Society [Lim and Hackman, J. Acoust. Soc. Am. Suppl. 1 85, S131 (1989)], an analysis of the elastic wave scattering from elastic spherical inclusions in a rigid, low-density, elastic host medium, impedance matched to water was presented. It was concluded that the compressional (P) wave coupling to the center-of-mass translation of the scatterer lead to a highly efficient, resonant conversion of P waves to shear (S) waves within the composite. Since S waves are effectively trapped in the coating by the dynamics of the water/syntactic foam interface, this mechanism could lead to a highly efficient, sound-absorbing material. Here, the scattering from elongated inclusion shapes is analyzed to address the effects on the PS conversion efficiency and address the potential for resonant PS conversion by coupling to the rotational degrees of freedom of such an inclusion. The calculational tool for this study is an improved version of the spheroidal-coordinate-based T matrix. The formal details of this approach are also presented.
Effects of a fluid boundary layer on the interaction of sound waves with a plate
Authors: Ingard K. Uno , Akay Adnan
Published: October 2005

The influence of a flow boundary layer on sound radiation from, and sound transmission through, a plate is discussed. The angular dependence of the transmission loss has been calculated. Difficulties encountered in the numerical analysis for the determination of the angler-averaged value of the transmission loss in a diffuse field are discussed. An asymptotic closed-form expression obtained for the averaged transmission loss shows that, at high frequencies, the transmission coefficient for a damped plate is proportional to ω−3. The conditions under which the flow increases or decreases the transmission loss or the radiated sound intensity are investigated. [Work supported by NSF and ONR.]
Dynamic finite element for anisotropic thermoviscoelastic structures
Authors: Chern J.-T. , Salamon N. J.
Published: October 2005

A numerical analysis is developed that solves a linear anisotropic viscoelastic solid subjected to nonhomogeneous temperature distribution and dynamic loading. The constitutive law is of hereditary integral type. The relaxation modulus is in the form of a Prony series. To conserve computer memory and solution time, it is formulated so that one does not have to integrate back to the time origin. Several examples are given to evaluate the accuracy of the solution technique. [Work supported by The Applied Research Laboratory at The Pennsylvania State University.]
Dynamics of elastic ship laminae
Authors: Dömeci M. Cengiz
Published: October 2005

This paper is addressed to the macromechanical analysis of dynamics of an elastic ship laminae within the effective stiffness concept of composites. The laminae that may possess arbitrary number of layers is immersed within an ideal fluid of infinite extent. In the first part, a variational principle [N. Sarigül and M. C. Dökmeci, AIAA J. 20, 1173–1175 (1984)] is extended by Friedrichs's transformation [M. C. Dökmeci, IEEE Trans. UFFC-35, 775–787 (1988)] so as to formulate a unified variational principle of fluid-solid interaction. In the second part, a set of two-dimensional governing equations of laminae is systematically derived by means of a variational principle together with a series representation in the thickness coordinate for the fields of displacement and hydrodynamic pressure that are chosen as a basis of derivation. The significant effects of laminated composites are considered, including those of the dynamic interactions between, and the hydrodynamic pressure on, the layers. The governing equations accommodate all the types of motions of laminae. In the third part, emphasis is placed upon a theorem of uniqueness in solutions of the governing equations and special cases.
Reconstruction of surface velocity on axisymmetric structures having nonseparable geometry
Authors: Sarkissian Angie
Published: October 2005

An algorithm is developed for reconstruction of the normal velocity on the surface of an arbitrary axisymmetric structure from measurements of pressure in the nearfield. This high-resolutinn reconstruction procedure is better suited for broadband data than the singular value decomposition method previously implemented [W. A. Veronesi and J. D. Maynard, J. Acoust. Soc. Am. 85, 588-598 (1989); Borgiotti et al., J. Acoust. Soc. Am. Suppl. 1 84, S173 (1988) ], enabling reduced computation time and providing a new expansion for the surface field with a set of functions that can be expressed as real functions, independent of frequency and independent of ones choice of a measurement surface. The boundary element method is applied to compute the orthonormal set of functions and a projection matrix relating nearfield pressure values to surface normal velocity.
Structural acoustics motivated sensor, actuator, and control research at MIT
Authors: von Flotow A. H.
Published: October 2005

Research into dynamics and control of large flexible spacecraft has a history of several decades. More recently, application of these techniques to naval structural acoustics has received attention. The dynamic similarity of these two applications is uncanny. This presentation will offer a broad overview of several years of work performed in the Dept. of Aeronautics and Astronautics at MIT in the area of sensor and actuator development for active control of structural dynamics and structural acoustics. Early focus was on piezoceramic actuation of thin metal and composite plates. Current projects include piezoceramic actuation of machinery mounts; piezoceramic actuation of aerodynamic surfaces; piezoceramic and electrostrictive actuation of optical components; piezoceramic actuation of specific traveling wave components in bending; piezeoceramic actuation and sensing of truss structures; customized spatially distributed strain sensing with PVDF film; low stiffness spacecraft mounting systems with PVDF film actuation; and basic comparison and classification of a range of solid-state actuation possibilities.
Characterization of distributed piezoelectric actuators for structural excitation
Authors: Clark R. L. , Fuller Chris , Wicks Al
Published: October 2005

Past studies have been devoted to the application of surface mounted or imbedded piezoelectric elements for structural excitation and development of theoretical predictions for the response of the structure [E. F. Crawley and I. deLuis, AIAA J. 25 (10), 1373–1385 (1987)]. The thrust of the present work is to determine the range of linearity of structural response due to excitation by pieozoelectric actuators as well as to compare experimentally measured responses to theoretical predictions for the case of a simply supported beam. Emphasis was placed on illustrating distributed excitation, both experimentally and analytically, with multiple independently driven actuators. Results indicate that the theoretical model provides a viable means of determining appropriate locations of piezoelectric actuators for exciting desired modal distributions in the structure. [Work supported by ONR.]
Piezocomposite actuator designs for an active underwater acoustic control system coating
Authors: Howarth Thomas R. , Bao Xiaoqi , Varadan Vijay K. , Varadan Vasundara V. , Benjamin Kim C.
Published: October 2005

The design and development of piezocomposites for use as the actuating device in an underwater acoustic control system coating will be discussed. Both flexible piezoceramic and 1–3 piezocomposite designs are explored as actuator candidates in the thin-walled polymer matrix encapsulant coating. The actuators will be presented in single layer form for the absorption of normally reflected acoustic pressure waves. A bilaminate design will be shown for the absorption of both the normally reflected and transmitted acoustic waves. Experimental measurements will be used for performance comparison of different composite and lamination designs.
Supervised and unsupervised neural net learning algorithms for modeling music perception
Authors: Bharucha Jamshed J.
Published: October 2005

Neural net learning algorithms permit the development of physiologically plausible computational models of learning in the auditory domain. Supervised learning algorithms are applicable when the acoustic environment provides target vectors that can guide learning. Musical sequences can be learned in this manner if each successive element in a sequence trains the expectancies generated by an accumulated representation of the sequence thus far [P. M. Todd, Comput. Mus. J. 13, 27–43 (1989); J. J. Bharucha and P. M. Todd. Comput. Mus.] J. 13, 44–53 (1989)]. Even when target vectors are absent, learning of auditory patterns can be accomplished via unsupervised learning algorithms. Both forms of learning seem to be necessary in order to account for a range of phenomena in music perception, including hierarchical and sequential representation of both specific musical patterns as well as abstract patterns that typify cultural regularities. [Work supported by NSF.]
Hierarchical sequential networks for music composition
Authors: Todd Peter M.
Published: October 2005

Sequential neural networks using time-delay feedback connections can be taught to reproduce musical sequences. By altering the network's controlling parameters, new sequences can be composed that are based on features of the previously learned examples [P. M. Todd, Comput. Mus. J. 13, 27–43 (1989)]. The current work expands this approach by adding an additional sequential network, running at a slower “speed,” which produces the parameters to control a faster-running network hierarchically positioned above it. This allows the slower network to control high-level structural aspects of the final musical output produced by the faster network, resulting in longer, more structured sequences. Issues remain in how best to divide up the sequence learning appropriately between the two hierarchical networks.
Musical acculturation in the first year of life
Authors: Lynch Michael P. , Eilers Rebecca E. , Oller D. Kimbrough
Published: October 2005

Questions concerning the developmental timetable for the acquisition of musical knowledge have been studied very little. This study of infants suggests a precocious onset of musical scale knowledge, an important aspect of music cognition, as findings indicate that infants better perceived mistunings in native than in nonnative scales. Six- and 12-month-olds were tested in noticing increases in the frequency of a randomly selected note in a melody based on the Western major scale (most common Western scale based on the semitone), the Western augmented scale (infrequently used Western scale but also based on the semitone), and the Javanese pelog scale (nonnative scale not based on the semitone). An operant head-turn procedure was used in which the well-tuned version of each melody was played repeatedly through a loudspeaker, and the infants learned to turn their heads toward the speaker in anticipation of a visual reinforcer when they noticed a frequency change in one of the melody notes. The 12-month-olds' performance was better on the major scale than on either the augmented or pelog scales. In contrast, the 6-month-olds' performance was better on both the major and augmented scales than on the pelog scale, and their performance was similar on the major and augmented scales. These findings suggest that substantial musical acculturation occurs as early as 6 months of age and that infants may proceed from acquisition of knowledge of the semitone as a basic structure of Western music to acquisition of more specific knowledge of the major scale by 12 months of age.
Cochlear mechanics
Authors: Dallos Peter
Published: October 2005

In this tutorial review new concepts and experimental results about the excitation of cochlear hair cells are described. First, an overview of the operation of the auditory system is provided. Traditional concepts of hydromechanical frequency analysis by traveling waves are considered next. Finally, provocative new results, suggesting local mechanical amplification processes in the cochlea, are presented and the mechanoelectrical transducer action of hair cells is discussed.
Acoustic echo cancellation with a transform domain adaptive filter
Authors: Kuo Sen M. , Zhao Huan
Published: October 2005

The adaptive filtering technique is one of the major approaches to control the acoustic echoes in teleconferencing systems or hands-free telephony systems. A brief overview of the performance of different adaptive algorithms for acoustic echo cancellation [cf, S. M. Kuo and H. Zhao, J. Acoust. Soc. Am. Suppl. 1 86, S56 (1989)] will be presented. A transform domain adaptive filter (TDAF) algorithm is proposed to construct an acoustic echo canceler that has a convergence rate faster than that of the conventional LMS adaptive transversal filter with colored signals. The proposed approach does not introduce extra delay during the process so that it can be used in the application where delay is not allowed. Different orthogonal transforms such as the discrete Fourier transform (DFT), discrete cosine transform (DCT), and discrete Hartley transform (DHT) are used for signals with different eigenvalue spread. Real-time implementations of this class of canceler with digital signal processors (DSP) will also be discussed.
Nonlinear pulse propagation in isotropic elastomers
Authors: Carman G. P. , Cramer M. S.
Published: October 2005

The nonlinear Burgers' equation for shear wave pulses in isotropic hyperelastic materials is derived for arbitrary initial strain states. Conditions under which negative and/or mixed nonlinearity is observed will be presented. Specific nonlinearity coefficients will be evaluated for the well-known elastomer model developed by Blatz and Ko [Trans. Soc. Theol. 6, 199–210, 287–292 (1962)]. Results are expected to be valid for a special class of isotropic hyperelastic materials including foamed and solid polyurethane rubbers.
Interaction of weak shocks with a simple fluid-fluid interface: Numerical simulation
Authors: McDonald B. Edward , Kuperman W. A.
Published: October 2005

The nonlinear progressive wave equation (NPE) model [J. Acoust. Soc. Am. 81, 1406 (1987)] has been used to simulate low grazing angle, steady-state weak shock interaction with a planar interface defined by a discontinuity in sound speed across a fluid-fluid interface. Model simplification results from taking the density to be continuous. This condition may be relaxed when early results have been diagnosed. The theory includes a term due to distortion of the interface by the wave, although it is estimated to be insignificant in the cases considered here. For a shock incident on a fluid of higher sound speed, it is possible to give a simple theoretical estimate for a critical grazing angle below which finite amplitude wave energy does not penetrate to great distances. When the grazing angle is above this value, numerical results indicate a finite amplitude head wave, with some reradiation of energy back across the interface, and a deep penetration shock offset in space from the incident shock. [Work supported by NRL and Sandia National Lab.]
Nonlinear wave propagation in a fluid layer
Authors: Hamilton M. F. , Zabolotskaya E. A.
Published: October 2005

The propagation of finite amplitude sound in a homogeneous fluid layer that is bounded below by a rigid surface and above by a free surface is investigated theoretically. Solutions are derived for the second-harmonic pressure generated by a tone that propagates in a single mode. It is shown that dispersion causes the modal structure of the second-harmonic component to vary with range. A nonlinear Schrödinger equation [Zabolotskaya and Shvartsburg, Sov.-Phys. Acoust. 33, 221–222 (1987)] is used to investigate the propagation of narrow-band pulses, and analytical results are obtained. Pulse propagation is found to be stable in a liquid layer, as is the case for narrow-band pulses in nonplanar modes of rigid circular ducts filled with air [A. H. Nayfeh, J. Acoust. Soc. Am. 57, 803–809 (1975)]. The case of a gaseous layer is also considered. Near cutoff, pulse propagation in a gaseous layer is found to be unstable, and pulses eventually separate into envelope solitons. [Work supported by the Texas Advanced Research Program (MFH) and the National Science Foundation (EAZ).]
Effects of acoustically driven oscillating flow on hydrodynamics of the flow around small spherical particles
Authors: Yavuzkurt Savash , Ha Man Y.
Published: October 2005

This theoretical study explores the effects of acoustically driven oscillating flows on the flow behavior around small spherical particles. Its application is in the area of enhancement of combustion of pulverized coal particles and coal water slurry fuel droplets via augmentation of heat and mass transfer to and from these particles that are entrained in the steady mean flow. The particle is 100 μm in diameter. The sound-pressure levels are between 140–160 dB, resulting in peak acoustic velocities of 2.5–50 m/s. The acoustic Reynolds numbers based on peak acoustic velocity and particle diameter are 16–150. The phenomenon is examined at 50 and 2000 Hz corresponding to Strouhal number 0.0002 and 0.08 in order to investigate the effects of low and high frequencies. The velocity field around the particle, the separation angle, and wall shear stress are calculated by solving the two-dimensional, unsteady laminar Navier-Stokes equations with a numerical technique. From the results, it can be seen that at 50 Hz the flow field around the particle is governed by the particle curvature and magnitude of the free stream velocity whereas, at 2000 Hz, the acceleration of the flow with time (due to oscillation) is also an important factor. This results in different heat transfer behavior due to different separated regions at both frequencies under the same free stream velocity. For example, at a Reynolds number 16, steady flow does not separate; neither does the flow at 50 Hz. However, at 2000 Hz, a separated region is observed due to the free stream flow acceleration. These results are discussed in detail.
Discrimination of frequency-glide direction
Authors: Neill Mary E. , Feth Lawrence L.
Published: October 2005

It is often assumed that the relevant information in formant transitions includes the direction as well as the extent of frequency excursions. A study to determine the ability of listeners with normal hearing to determine the direction of linear frequency modulation of sinusoidal tones has been devised. To more closely approximate the listener's task in processing speech, music, or other environmentally important sounds, the initial frequency of each transition was selected at random from within a predefined range. Thus, for each interval of a 2Q, 2AFC listening task, the listener hears a frequency glide of the same duration and extent. In one of the two middle intervals, the direction of the transition is reversed to produce the “signal.” Since each of the four glides begins and ends on frequencies selected at random, the listener cannot rely on simple pitch differences to determine which interval contained the reversed glide. Our preliminary results will be discussed, with emphasis on the effect of the width of the “roving frequency” range on listener performance. [Work supported by a grant from AFOSR.]
Auditory enhancement and suppression in normal-hearing persons with reduced speech recognition in noise
Authors: Thibodeau Linda M.
Published: October 2005

Auditory enhancement has been suggested as a possible mechanism underlying one's ability to understand speech in noise. Using a forward-masking paradigm, enhancement effects were evaluated in two groups of normal-hearing persons who differed in their ability to recognize high-frequency words in noise. Masked thresholds for a 2000-Hz probe were compared between a standard condition in which a four-component harmonic complex including 2000 Hz was the masker and an enhancing condition in which a three-component adapter not including 2000 Hz preceded the masker. One explanation for the enhancement effect involves an adaptation of suppression that results in an increase in the masker effectiveness in the 2000-Hz region. Therefore, adaptation of suppression was also evaluated by measuring forward-masked thresholds for a 2000-Hz probe as a function of the duration of a suppressor tone added to a 2000-Hz masker. Although all subjects showed some degree of enhancement, there were differences in the reduction in suppression that occurred with increased suppressor duration.
Peripheral filter shape and central decision rule in frequency discrimination: A neural network approach
Authors: Zeng Fan-Gang , Turner Christopher W. , Smith Robert L.
Published: October 2005

An artificial three-layer neural network is constructed to study possible mechanisms in auditory frequency discrimination. The 110 input units, resembling the peripheral neurons, have two sets of filter shapes: idealized eighth nerve isointensity functions and Patterson-style filters [Moore and Glasberg, Hear. Res. 28, 209–255 (1987)]. Twenty-four intermediate units, similar to critical bands, process information from the input units. The output unit makes a frequency discrimination decision based upon an excitation difference in the intermediate units. Single-band and multiband decision-making rules are tested. When using the idealized eighth nerves as the peripheral filter shape, neither the single-band nor a linear-combination, multiband model satisfactorily predicts experimental data. However, by adjusting the weights from the intermediate units to the decision-making unit by the Delta Rule [Rumelhart and McClelland, Parallel Distributed Processing 1 (MIT, Cambridge, MA, 1988), pp. 444–459], it is found that a weighted-band model can best predict the observed data. The weighted-band model assigns different weights to each band according to its excitation difference: The larger the difference is, the greater the weight. When using Patterson-style filters as the input units, none of the above three models provides a satisfactory prediction. The results suggest that (a) the peripheral filter shape may play a more important role than the central decision rule in frequency discrimination, and (b) the weighted-band model may be a more appropriate central decision rule.
Categorical perception revisited, yet again: Frequency and frequency ratio resolution by “possessors” and other musicians
Authors: Burns Edward M. , Campbell Shari L. , Arehart Kathryn H.
Published: October 2005

Macmillan et al. [J. Acoust. Soc. Am. 84, 1262–1280 (1988)] recently demonstrated the advantages gained by applying experimental techniques and analyses derived from a model of intensity resolution to the phenomenon of categorical perception of speech. In this paper, the application of these techniques to both the perception of musical intervals (frequency ratios) by musicians with relative pitch, and the perception of frequency by musicians with absolute pitch is discussed. The results suggest that the perception of frequency ratios by musicians differs from perception along most unidimensional psychophysical continua (characterized by the “7 ± 2” principle) in two ways. First, identification resolution for frequency ratios is much better than for stimuli along 7 ± 2 continua. Second, musical interval perception seems to be a case of ideal categorical perception (as originally defined) in that musicians apparently can discriminate between two musical intervals only to the extent that they can differentially identify them. Although the identification resolution for frequency by absolute pitch possessors is much better than that of nonpossessors (whose identification resolution is consistent with the 7 ± 2 limitation), their discrimination resolution for frequency remains much better than their identification resolution. [Work supported by NIDCD.]
The perception of complex, naturally occurring, nonspeech auditory stimuli: Walking perception
Authors: Li Xiao-Feng , Logan Robert J. , Pastore Richard E.
Published: October 2005

The act of walking across a hard surface by humans imposes a nonarbitrary structure on an acoustic waveform. This research investigated the sensitivity of listeners to this structure and evaluated the ability of listeners to extract certain information about the source of the sound. Three experiments were conducted. In experiment 1, stimuli produced by eight walkers (four male and four female) were presented to listeners in an AX (same/different) walker discrimination task. Experiment 2 evaluated listeners' ability to label the walkers by gender. Finally, experiment 3 assessed the ability to identify the individual walkers. The results suggested that the listeners could discriminate between the walkers and that listeners can recover some of the physical characteristics of the source based on the auditory signal.
Pattern recognition within spectrally isolated regions of broadband complex sounds
Authors: Bashford James A. , Warren Richard M.
Published: October 2005

The iteration of recycled noise segments (RNs) can be heard readily at infratonal repetition frequencies (below 20 Hz,) with “whooshing” (which becomes resolved into components such as “thumps” and “clanks”) heard from 1–4 Hz and “motorboating” (involving a featureless timbre) heard from 4–19 Hz [Guttman and Julesz, J. Acoust. Soc. Am. 35, 610 (1963)]. As previously reported, repetition is heard at all center frequencies when a 1/3-oct filter is swept through broadband motor-boating or whooshing RNs. The present study examined the recognition of individual RNs presented first with limited spectral bandwidths, and then embedded in broadband “parents.” Accurate performance required bandwidths greater than a critical band and center frequencies less than 4.8 kHz. These results, and those from a second experiment involving identification of narrow-band derivatives of previously heard broadband RNs, will be discussed in terms of (1) the relative contribution of different spectral regions to recognition of long-period waveforms and (2) the ability to perceptually segregate broadband patterns into discrete spectral regions. [Work supported by AFOSR and NIH.]
Auditory memory for long-period random waveforms
Authors: Brubaker Brad S. , Warren Richard M.
Published: October 2005

Recycled noise segments (RNs) are heard as a repetitive “whooshing” from the lower limit of effortless iterance detection (1 Hz) through 4 Hz [Guttman and Julesz, J. Acoust. Soc. Am. 35, 610 (1963)]. The present study found that after listening to a whoosing range RN, the frozen noise segment could be identified at a later time (well beyond the limit of echoic storage) when it was presented as part of a longer “parent” frozen noise segment. For RNs with frequencies above the whooshing range, recognition of the frozen noise segment was difficult or impossible when presented as a portion of a longer parent. Implications concerning the nature of repetition detection at different iterance rates will be discussed.
Design and evaluation of FIR filters for digital hearing aids with arbitrary amplitude and phase response
Authors: Nielsen Lars B. , Wu Ed , Hoffman Michael W. , Buckley Kevin , Soli Sigfrid D.
Published: October 2005

An automated design algorithm has been developed for design of arbitrary amplitude and phase response FIR filters for heating aids. This method uses a weighted least-squares frequency sampling approximation with these features: (a) Frequency samples specifying amplitude and phase can be arbitrary and closely spaced in critical regions: (b) each frequency sample can be weighted; and (c) interpolation can be done between samples using units appropriate to hearing. A method for objective evaluation that provides a perceptually based measure of the closeness of fit has also been developed. Filter responses and their objective evaluations are presented for a number of typical hearing losses that include compensation for the transducer's amplitude and phase response.
Laboratory and field evaluation of hearing aid performance in noise
Authors: Sullivan Jean A. , Eisenberg Laurie , Allsmart Cathy , Soil Sigfrid D.
Published: October 2005

A within-subjects design was used to evaluate the performance of three digitally programmable hearing aids in 15 adults with mild to moderate sensorineural hearing loss. The aids were fitted and programmed according to the three manufacturers' specified procedures. There were two laboratory measures of performance: (a) quality ratings while listening to continuous speech in quiet and in two common noise environments (cafeteria noise at 7 dB S/N and machine noise at 12 dB S/N), and (b) intelligibility of CUNY topic-related sentences in cafeteria noise at signal levels of 65 and 80 dB SPL. Also, field evaluations of each aid were conducted in 10 of the subjects for 6 days. Each day, speech intelligibility and sound quality in quiet, low-noise, and high-noise conditions were rated. Performance of the aids will be compared and the validity of laboratory and field methods for assessing hearing aid performance in noise will be discussed.
A microcomputer workstation for auditory neurophysiology
Authors: Vrieslander Julian D. , Capranica Robert R.
Published: October 2005

An inexpensive microcomputer-based system has been developed for fast digital synthesis of complex sound stimuli, recording of neural activity, and interactive on-line histogramming and plotting of response properties. The hardware requirements for stand-alone synthesis and spike recording capability are an Apple Macintosh II, a GW Instruments data acquisition card, and a modest amount of interfacing and signal conditioning electronics. Off-line analysis of acquired data files can be performed on any current Macintosh. Stimuli can be computed synthetic sounds or digitized natural sounds; both types are presented at sampling rates of up to 100k samples/s with concurrent timing of spikes at 10-μs resolution. The system is especially suited for experiments in which sounds must be rapidly resynthesized as a stimulus parameter is changed under manual or automatic control. A modular software design allows installation of new synthesis and analysis routines, or new drivers for alternate hardware interfaces. Synthesis parameters and raw spiketime data are saved to disk files in a format that allows exact recreation of stimuli and precise correlation of spikes with temporal features in those stimuli. [Work supported by NINCDS.]