bluuebunny/crossref_metadata_2025 huggingface dataset:
Analysis of aperiodicities in nearly periodic waveforms
Authors: Schumacher Robert T.
Published: October 2005

Waveforms from the oscillations of sustained tone acoustic musical instruments are never precisely periodic. The source of their aperiodicity is instrument dependent, and typically small. With the exception of some studies on the period fluctuations of bowed string instruments, musical instrument aperiodicities have never been quantitatively characterized. This paper introduces two techniques for the characterization of nearly periodic waveforms: the norm difference method and the Fourier transform of sonograms. The former is appropriate for the study of cycle to cycle fluctuations, and yields its best results for fluctuations that have a typical time scale of up to 16 periods of the nominally periodic oscillation. The second method is appropriate for a frequency range of 2 Hz to approximately 100 Hz. The ranges of applicability overlap for pitch frequencies in the treble region. The techniques are described in detail, and explored with various computer-constructed waveforms with appropriate properties designed to probe the strengths and weaknesses of the methods. As an illustration of the application to real musical sounds, a French horn note is analyzed by both methods. They both reveal fluctuations in the sound of a note of fundamental frequency of 347 Hz in the range of 33–35 Hz, the frequency of the lowest frequency input impedance peak of typical horns in F.
Method to measure acoustic impedance and reflection coefficient
Authors: Keefe Douglas H. , Ling Robert , Bulen Jay C.
Published: October 2005

A frequency-domain based system for measuring acoustic impedance and reflection coefficient is described. The calibration procedure uses a least-mean-squares approximation to the Thevenin parameters describing the source and receiver characteristics in which the data measured on closed, cylindrical tubes are matched to a viscothermal tube model. The system is intended for use in acoustical measurement in human ear canals, in which the cross-sectional area of the ear canal at the point of insertion is imprecisely known. This area is acoustically estimated from the impedance data, and the reflection coefficient is calculated in terms of this area and the impedance data. Measurements on a variety of closed tubes show the method is accurate over the frequency range investigated (<10.7 kHz). The time-domain reflection function is evaluated by transforming the reflection coefficient from the frequency domain, but the finite bandwidth of the measured data limits the accuracy of time-domain response measurements. The method is well suited for frequency-domain measurements in human ear canals.
A method for accurate detection of time of arrival: Analysis and design of an ultrasonic ranging system
Authors: Figueroa J. Fernando , Lamancusa John S.
Published: October 2005

Many industrial applications demand accurate ranging operations, for example, robotic end-effector positioning and measurement of vibrations in machine monitoring tasks. For instance, measurement of the position of a robotic end effector in high-accuracy applications demands the following stringent specifications from a ranging system: (1) absolute ranging accuracy of ±0.025 mm (±0.001 in.), (2) working range of approximately 2 m (6.56 ft), (3) update rate >100 Hz, and (4) high reliability. It is shown that an ultrasonic ranging system that meets the above specifications may be feasible. The system model is based on measuring the distance between a transmitter and a receiver. A novel method to detect the instant when the signal arrives at the receiver is described. The method combines envelope peak detection and phase measurement to determine the time of arrival, and it gets close to meeting the accuracy criteria. To verify the analytical predictions, distances of up to 0.9 m were measured using a prototype system. The absolute accuracy of such measurements was ±0.150 mm (±0.006 in.). The tests performed indicate that the errors are caused primarily by air turbulence and may be decreased by averaging more samples. In this paper ultrasonic ranging from a transmitter to a receiver in a three-dimensional space setting is addressed.
Comments on ‘‘The influence of wind and temperature gradients on sound propagation, calculated with the two way wave equations’’ [J. Acoust. Soc. Am. 8
7, 1987–1998 (1990)]
Authors: Raspet Richard , Yao Lixin , Franke Steven J. , White Michael J.
Published: October 2005

In a recent paper Nijs and Wapenaar [J. Acoust. Am. 87, 1987–1998 (1990)] have developed a method for solving for the effects of horizontal winds on sound propagation in the atmosphere. In this development, Nijs and Wapenaar neglect terms containing the density gradient of the atmosphere which in their notation are comparable to the effects of the adiabatic temperature lapse rate on the sound-speed gradient. Their method represents a significant improvement over the use of an effective sound-speed gradient to express the effects of the wind only for high propagation angles. For these situations, one would expect errors due to the change in ambient pressure and density to be greatest. In this letter, a numerically attractive method to incorporate horizontal winds and a hydrostatic density gradient into a two-dimensional transform solution for sound propagation in the atmosphere is developed. In addition, Nijs and Wapenaar approximate the two-dimensional Fourier transforms by a single one-dimensional transform with little justification. Conditions are discussed under which such an approximation may be valid and it is demonstrated that these conditions are unlikely to be met by the two-dimensional kernal of Nijs and Wapenaar.
Reply to: ‘‘Comments on the influence of wind and temperature gradients on sound propagation calculated with the two-way wave equation’’ [J. Acoust. Soc. Am. 9
1, 498–500 (1992)]
Authors: Nijs L. , Wapenaar C. P. A.
Published: October 2005

As shown by Pierce [J. Acoust. Soc. Am. 87, 2292–2299 (1990)], the influence of the static pressure on sound propagation through the atmosphere is only of the second order. For very accurate predictions, the improvements suggested by Raspet et al. [J. Acoust. Soc. Am 91, 498–500 (1992)] can be used, but then other second-order effects and cross products should be taken into account as well. The two-dimensional Fourier transform can be used for three-dimensional sound-pressure predictions considering the method of stationary phase. Errors are well within 2% for common temperature and wind gradients, provided that the wind vector is parallel to the vertical plane through the source and receiver. For situations with a cross-wind vector full three-dimensional calculations are recommended. These situations were avoided in the original article [Nijs and Wapenaar, J. Acoust. Soc. Am. 87, 1987–1998 (1990)].
Short-term poststimulatory response characteristics of the human acoustic stapedius reflex: Monotic and dichotic stimulation
Authors: Cacace Anthony T. , Margolis Robert H. , Relkin Evan M.
Published: October 2005

Two experiments were performed to study short-term poststimulatory response characteristics of the human acoustic stapedius reflex in the time and intensity domains. In experiment 1, monotic magnitude-intensity functions (MIFs) were obtained for a 20-ms test stimulus preceded by a conditioning stimulus varying in duration (20, 50, 100, 500 ms) and level (−10, 0, +10 dB re: stapedius-reflex threshold) and temporally separated from the test stimulus by various interstimulus intervals (ISIs) (0, 20, 50, 100, 500 ms). Experiment 2 was similar in design except that conditioner and test stimuli were presented dichotically and fewer ISIs were used. Both experiments demonstrated that a prior conditioning stimulus produced significant increases in test-stimulus response magnitude. These poststimulatory effects were characterized by complex interactions among stimulus variables (conditioner duration, conditioner level, and interstimulus interval) with similar interactions occurring for both monotic and dichotic stimuli. A simple superposition effect of the responses to the conditioner and test stimulus does not account for the effect of prior stimulation since responses often exceeded the sum of the responses to the conditioner and the test stimulus alone.
Marine sediment classification using the chirp sonar
Authors: LeBlanc Lester R. , Mayer Larry , Rufino Manuel , Schock Steven G. , King John
Published: October 2005

The chirp sonar is a calibrated wideband digital FM sonar that provides quantitative, high-resolution, low-noise subbottom data. In addition, it generates an acoustic pulse with special frequency domain weighting that provides nearly constant resolution with depth. The chirp sonar was developed with the objective of remote acoustic classification of seafloor sediments. In addition to producing high-resolution images, the calibrated digitally recorded data are processed to estimate surficial reflection coefficients as well as a complete sediment acoustic impulse profile. In this paper, surficial sediments in Narragansett Bay, RI are used to provide ground truth for an acoustic model. Quantitative acoustic returns from the chirp sonar are used to estimate surficial acoustic impedance and to predict sediment properties. A robust acoustic sediment classification model that uses core samples to account for the local depositional environment has been developed. The model uses an estimate of acoustic impedance to predict surficial density, porosity, compressibility, and rigidity. The comparisons show a high correlation between the core-determined sediment properties and the estimates obtained from acoustic measurements.
Sonar attenuation modeling for classification of marine sediments
Authors: LeBlanc Lester R. , Panda Satchidanan , Schock Steven G.
Published: October 2005

An attenuation-based model for classification of marine sediments is developed for the chirp sonar operating in the frequency range of 2–10 kHz. A relaxation-time model is proposed that combines the various dissipative energy loss mechanisms of sound in marine sediments into a single parameter. Historical data were analyzed by converting attenuation values reported in ‘‘dB/m@kHz’’ to a single relaxation time value. Analysis of these previous attenuation measurements supports the use of a relaxation-time model. Based on this large collection of data, an empirical equation is developed that relates relaxation time to grain size (in phi units). Using this model, very little phase dispersion is observed for a correlated chirp pulse traveling through 40 m of sand, silt, or clay. Yet, this is not so for a pulse in the ultrasonic frequency range (0.2–1.0 MHz) traveling through only 10 cm of clay. Here, significant dispersion is noted. Because of the unique Gaussian-like shape of the correlated chirp pulse power spectrum, pulse elongation due to attenuation is minimized. Using the center frequency shift in the pulse spectrum, a new ‘‘instantaneous frequency’’ method of attenuation estimation is proposed that overcomes the problems associated with interfering reflections. Based on the relaxation-time model, the correlated chirp pulse was synthetically attenuated to establish a relation between the relaxation time and the center frequency shift. In situ sediment-type predictions from chirp sonar data using the instantaneous frequency method and analyses of core samples taken in the Narragansett Bay, Rhode Island are in good agreement.
Multipole sources in boreholes penetrating anisotropic formations: Numerical and experimental results
Authors: Leslie H. D. , Randall C. J.
Published: October 2005

A 2.5-D velocity-stress finite-difference code is described that models acoustic propagation in a borehole penetrating a generally anisotropic formation. The excitation may be a dilatation (monopole), or a point force (dipole) in an arbitrary direction. The anisotropic formation is homogeneous along the axis of the borehole but may be inhomogeneous in the transverse plane. The borehole cross section and location of the source in the borehole are arbitrary. Synthetic time-domain waveforms are displayed for arrays of monopole and dipole receivers deployed along the borehole axis in both fast and slow anisotropic formations. The specific anisotropy model employed for the numerical results is a transversely isotropic (TI) formation with its axis of symmetry inclined with respect to the borehole axis by an arbitrary angle. Flexural/shear and Stoneley wave slowness and attenuation estimates are extracted from the synthetic waveforms using a variant of Prony’s method for a range of borehole inclinations relative to the formation axis of symmetry. In a fast formation, with borehole and formation symmetry axes perpendicular, flexural mode dispersion curves for quasi-SV and SH polarizations separate only at low frequency where moderate attenuation is also observed. In a slow formation, distinct dispersion curves are obtained for quasi-SV and SH polarizations over the entire frequency range. Moderate attenuation is again observed at low frequency. Scaled laboratory experiments confirm the numerical procedure. Experimental and numerical waveforms for monopole and several polarizations of dipole excitation in a transversely isotropic model formation overlay with excellent agreement.
Auditory filter shapes at low center frequencies in young and elderly hearing-impaired subjects
Authors: Peters Robert W. , Moore Brian C. J.
Published: October 2005

Auditory filter shapes were measured for two groups of hearing-impaired subjects, young and elderly, matched for audiometric loss, for center frequencies ( fc) of 100, 200, 400, and 800 Hz using a modified notched-noise method [B. R. Glasberg and B. C. J. Moore, Hear. Res. 47, 103–138 (1990)]. Two noise bands, each 0.4fc wide, were used; they were placed both symmetrically and asymmetrically about the signal frequency to allow the measurement of filter asymmetry. The overall noise level was either 77 or 87 dB SPL. Stimuli were delivered monaurally using Sennheiser HD424 earphones. Although auditory filters for the hearing-impaired subjects were generally broader than for normally hearing subjects [Moore et al., J. Acoust. Soc. Am. 87, 132–140 (1990)], some hearing-impaired subjects with mild losses had normal filters. The filters tended to broaden with increasing hearing loss. There were not any clear differences in filter characteristics between young and elderly hearing-impaired subjects. The signal-to-noise ratios at the outputs of the auditory filters required for threshold (K) tended to be lower than normal for the young hearing-impaired subjects, but were not significantly different from normal for the elderly hearing-impaired subjects. The lower K values for the young hearing-impaired subjects may occur because broadened auditory filters reduce the deleterious effects on signal detection of fluctuations in the noise.
Frequency dependence of binaural performance in listeners with impaired binaural hearing
Authors: Gabriel Kaigham J. , Koehnke Janet , Colburn H. Steven
Published: October 2005

Binaural performance was measured as a function of stimulus frequency for four impaired listeners, each with bilaterally symmetric audiograms. The subjects had various degrees and configurations of audiometric losses: two had high-frequency, sensorineural losses; one had a flat sensorineural loss; and one had multiple sclerosis with normal audiometric thresholds. Just noticeable differences (jnd’s) in interaural time, interaural intensity, and interaural correlation as well as detection thresholds for NoSo and NoSπ conditions were obtained for narrow-band noise stimuli at octave frequencies from 250–4000 Hz. Performance of the impaired listeners was generally poorer than that of normal-hearing listeners, although it was comparable to normal in a few instances. The patterns of binaural performance showed no apparent relation to the audiometric patterns; even the two subjects with similar degree and configuration of hearing loss have very different binaural performance, both in the level and frequency dependence of their performance. The frequency dependence of performance on individual tests is irregular enough that one cannot confidently interpolate between octaves. In addition, it appears that no subset of the measurements is adequate to characterize the performance in the rest of the measurements with the exception that, within limits, interaural correlation discrimination and NoSπ detection performance are related.
Frequency-domain implementation of Griffiths–Jim adaptive beamformer
Authors: Chen Yuan-Hwang , Fang Hwai-Der
Published: October 2005

It is well known that the drawback of the time-domain least-mean-square (LMS) algorithm proposed by Widrow [Proc. IEEE 63, 719–720 (1975)] in adaptive filter applications is that its convergence speed decreases as the eigenvalue spread of the input autocorrelation matrix increases. However, this problem can be overcome by employing the transform-domain LMS method [Narayan et al., IEEE Trans. ASSP 31, 609–615 (1983)] which first transforms input time-domain signals into another transform-domain signals through DFT or some orthogonal transforms and then uses the self-orthogonalizing algorithm [Gitlin and Magee, IEEE Trans. Commun. 25, 666–672 (1977)] to optimize the variable weights of an adaptive filter. This method has been shown to offer great improvement in convergence rate over the time-domain LMS method. The objective of this paper is to apply the frequency-domain LMS algorithm with the self-orthogonalizing technique, called FLMS, to the Griffiths–Jim adaptive beamformer to accelerate the convergence rate for real-time processing of adaptive array signals. It is shown that the two minimum mean-square errors of the beamformer implemented in the time and frequency domains are identical. Computer simulations show that the adaptive beamformer using the FLMS exhibits faster convergence behavior and better performance of nulling jammers than that using the LMS, especially for the larger eigenvalue spread.
Effects of stimulus duration on the response properties of Pacinian corpuscles: Implications for the neural code
Authors: Checkosky C. M. , Bolanowski S. J.
Published: October 2005

Based on physiological and psychophysical data, it has been suggested that the neural code for threshold detection in the P channel, mediated by Pacinian corpuscles (PCs), may be 2–4 neural impulses/stimulus (Bolanowski et al., 1988). To further test the efficacy of this code, responses from PCs were measured at different vibratory burst durations. Since it is known that the P channel has the capacity to summate vibratory stimuli temporally within the central nervous system, the effect should also be present in the physiological results. Temporal summation predicts a decrease in threshold at a rate of −3 dB/doubling of stimulus duration. Responses from single PCs were integrated by counting neural spikes over the entire burst duration. Furthermore, real-time responses were integrated with a low-pass filter, more accurately modeling the central process. For the spike-counting scheme, a criterion of 4 impulses/stimulus showed a decrease in stimulus amplitude for increases in duration similar to that obtained psychophysically. The amplitude-duration function obtained with the low-pass filter, however, resulted in a function which did not follow that obtained psychophysically, regardless of the number of impulses/stimulus. Since the P channel is known to have a central integrator, it has been concluded that activity of a single PC afferent probably is not sufficient to signal threshold in the P channel.
Differences in auditory performance between monaural and diotic conditions. I: Masked thresholds in frozen noise
Authors: Langhans Angela , Kohlrausch Armin
Published: October 2005

Thresholds of a 5-ms, 1-kHz signal were determined in the presence of a frozen-noise masker. The noise had a flat power spectrum between 20 Hz and 5 kHz and was presented with a duration of 300 ms. The following interaural conditions were tested with four listeners: Noise and signal monaural at the same ear (monaural condition, NmSm), noise and signal identical at both ears (diotic condition, NoSo), noise identical at both ears and signal monaural (dichotic condition, NoSm) and uncorrelated noise at the two ears and signal monaural (NuSm). The signal was presented at a fixed temporal position with respect to the frozen noise in all measurements and thresholds were determined for different starting phases of the carrier frequency of the signal. Variation of the carrier phase strongly influenced the detection in the diotic condition and the masked thresholds varied by more than 10 dB. The pattern of thresholds for the monaural condition was less variable and the thresholds were generally higher than for the diotic condition. The monaural–diotic difference for specific starting phases amounted to as much as 8 dB. Comparison measurements using running noise maskers revealed no such difference. This relation between monaural and diotic thresholds was further investigated with eight additional subjects. Again, monaural and diotic thresholds in running noise were identical, while in frozen noise, diotic thresholds were consistently lower than monaural thresholds, even when the ear with the lower NmSm threshold was compared. For the starting phase showing the largest monaural–diotic difference, the thresholds for NoSm lay between the monaural and the diotic values. At other starting phases, the NoSm threshold was clearly lower than both the NmSm and the NoSo threshold. One possible explanation of the observed monaural–diotic differences relates to contralateral efferent interaction between the right and the left hearing pathway. A prediction based on this explanation was verified in a final experiment, where frozen-noise performance for NmSm was improved by simultaneously presenting an uncorrelated running noise to the opposite ear.
Fabrication and characterization of continuously cast clad metallic buffer rods
Authors: Jen Cheng-Kuei , Neron Christian , Miri Amir , Soda Hiroshi , Ohno Atsumi , McLean Alex
Published: October 2005

The fabrication, characterization, and ultrasonic measurements of clad metallic ultrasonic buffer rods are presented. The core and the cladding consist of tin/lead alloy and pure tin, respectively, produced by a vertical continuous casting technique. The acoustic velocity profiles across the rod diameter are measured using a 225-MHz line focus beam scanning acoustic microscope. A schlieren visualization system confirms the concentration of acoustic energy in the core. Spherical acoustic lenses have been fabricated at the ends of the rods. Focused and unfocused ultrasonic measurements at frequencies between 2 to 10 MHz and signals with excellent signal to noise ratios have been obtained.
Wave propagation interaction with free and fluid-loaded piezoelectric substrates
Authors: Nayfeh Adnan H. , Chien Hual-Te
Published: October 2005

A unified analytical treatment is presented that is supported with numerical illustrations of the interactions of ultrasonic waves with piezoelectric anisotropic half-space substrates. The solids are allowed to possess up to monoclinic anisotropic symmetry and associated piezoelectric coupling. The solids are also assumed to be loaded with water and subjected to incident acoustic beams at arbitrary polar and azimuthal angles. Simple analytical expressions for the reflection and transmission coefficients are derived from which all propagation characteristics are identified. Such expressions contain, as a by-product, the secular equation for the propagation of free harmonic waves on the piezoelectric substrate. It is found that piezoelectric coupling, as well as water, influence both types of modes. Higher symmetry, such as orthotropic, transverse isotropic, and cubic, are contained implicitly in this analysis. This paper also demonstrates that the motion of the surface and SH modes uncouple for propagation along axis of symmetry. For such cases, however, piezoelectric coupling can influence one of these types of modes depending upon the type of piezoelectric model adopted.
Harmonic generation in finite amplitude sound beams from a rectangular aperture source
Authors: Kamakura Tomoo , Tani Meiko , Kumamoto Yoshiro , Ueda Koji
Published: October 2005

Theoretical analysis and some experiments are performed on nonlinearly generated harmonic components in bounded sound beams emitted from a rectangular aperture source. The Khokhlov–Zabolotskaya–Kuznetsov equation, which takes account of nonlinearity, dissipation, and diffraction effects in the beams, is numerically solved by means of the alternating direction implicit difference method. Using a planar source of size 24×44 cm, axial sound pressures and beam patterns of the first three harmonics are measured in air for initially sinusoidal ultrasounds of 25- and 30-kHz frequency, and are compared with the theory. They are in relatively good agreement. Deformation of the source face from circular to rectangular shape results in the unclear appearance of pressure peaks and dips with propagation. Within the framework of these studies, the harmonic pressure levels in the far field are almost the same as from a circular aperture source with equal face area and equal initial pressure, independent of the source levels.
Development of the active doublet method for measuring small velocity and attenuation changes in solids
Authors: Roberts Peter M. , Phillips W. Scott , Fehler Michael C.
Published: October 2005

The measurement of small changes in elastic wave velocity and attenuation is important to a broad range of problems, such as earthquake prediction and early detection of rock failure in mines. Previous authors proposed a method for estimating small temporal velocity changes in the earth’s crust by analyzing progressive relative phase delays between the scattered waves of two signals generated by nearly identical earthquake sources, called doublets, recorded at different times at the same receivers. Several improvements have been made to the original method and are presented here. The reliability of measured velocity changes has been increased by using active, repeatable sources instead of natural earthquakes. The robustness of the analysis technique has been improved by eliminating unnecessary intermediate phase regression steps and thus reducing the sensitivity to spurious data. Finally, the phase-delay algorithm has been extended to allow measurement of small attenuation changes from relative amplitude decay rates. Using ultrasonic source and receiver transducers embedded in Plexiglas test samples, velocity changes as small as 0.01%, caused by ambient temperature variations in the Plexiglas, have been measured. Changes in attenuation on the order of 10%, due to permanent damage induced in one of the samples, have also been measured.
Tongue-tip trills and vocal-tract wall compliance
Authors: McGowan Richard S.
Published: October 2005

The effect of vocal-tract wall compliance on tongue-tip trills is to create a favorable pressure-flow relation at the tongue tip for sustained vibration. The governing equations are derived for a model based on this mechanism, and data on unvoiced trills are used to help set parameters for a numerical simulation of the model.
Automated detection of the tongue surface in sequences of ultrasound images
Authors: Unser Michael , Stone Maureen
Published: October 2005

An image processing system has been developed for a Macintosh II personal computer. It is designed to process sequences of sagittal tongue sections that are digitized in real time and stored in standard tagged image file format (TIFF). The successive processing steps are: (a) a low-pass filter for noise reduction, (b) a resampling of the sector of interest in polar coordinates, (c) a matched filter (vertical differentiator) for the enhancement of the tissue/air interface in the surface region of the tongue, and (d) an extraction of border points by searching for an optimal radial path along the angular dimension. This latter task is achieved by dynamic programming, which has the following advantages. First, due to the use of a global criterion to guide the detection, it is very robust. Second, as a result of certain restrictions of the allowable transitions, the extracted contours are smooth. Finally, the method permits the specification of particular predefined contour points. This system was implemented in a program that can handle image sequences in a fully automatic mode. Results obtained using ultrasound data are presented.
Acoustic impedance measurement: Plane-wave mode and first helical mode contributions
Authors: Dalmont Jean-Pierre , Bruneau Anne-Marie
Published: October 2005

A measuring probe has been built that is comprised of an electrostatic transducer as a source and two receiving microphones. This probe allows measurements of acoustic impedance to be made over a wide range of values and over a larger frequency range than hitherto possible in waveguides by separating the contributions of the plane and first helical modes to the impedance.
Resolving the picosecond characteristics of synchronous sonoluminescence
Authors: Barber Bradley P. , Hiller Robert , Arisaka Katsushi , Fetterman Harold , Putterman Seth
Published: October 2005

The resolution with which the synchronous picosecond flashes of acoustically generated light can be measured has been improved. The flash widths are now found to be considerably less than 50 ps and the jitter in the time between flashes can also be substantially less than 50 ps. The flashes of sonoluminescence appear to turn off very sharply without ringing or after pulsing.
Comments on ‘‘Quadratic detection of signals with drifting phase’’ [V. V. Veeravalli and H. V. Poor, J. Acoust. Soc. Am. 8
9, 811–819 (1991)]
Authors: Morgan Dennis R. , Smith Thomas M.
Published: October 2005

A connection is pointed out between the time-domain detection problem in the subject paper and another recent work [D. R. Morgan and T. M. Smith, J. Acoust. Soc. Am. 87, 737–747 (1990)] dealing with detection in the spatial domain. In both cases, a deflection criterion is assumed within the framework of a quadratic detector and exponential coherence model. Likewise, suboptimal structures are considered in both treatments that incoherently combine the detection statistics of a number of coherently processed segments. It is hoped that the connection raised here will serve to mutually complement and enhance the usefulness of both works.
A robust, low precision, environmentally adapted receiver algorithm
Authors: Brannan James R.
Published: October 2005

For a model problem of source localization, it is shown that a receiver algorithm based on maximum likelihood theory and which also includes a priori knowledge of the environment has high precision but is extremely sensitive to deviations of environmental parameters from exact values. Consequently a computationally intensive search over the parameter space is required to estimate the source location. An alternative, physically motivated, low resolution estimation algorithm, the Lévy receiver, based on the Lévy metric for probability distributions is presented. The Lévy receiver also incorporates a priori knowledge of environmental parameters but appears to be much less sensitive to uncertainty in these parameters. Furthermore, the amplitude of the surface generated by the Lévy receiver over parameter space exhibits trends which permits restricting the search to a relatively coarse grid.
Broadband matched-field source localization
Authors: Westwood Evan K.
Published: October 2005

A straightforward algorithm for broadband matched-field source localizaton is developed and subsequently applied to experimental data. For the two-receiver case, the algorithm involves correlating modeled and measured cross spectra and summing coherently over frequency. The extension to the multiple-receiver case is to perform the two-receiver algorithm on each pair of hydrophones and sum the complex results coherently. The frequency band over which the summation is made may be chosen to maximize the signal-to-noise ratio. Using an acoustic propagation model based on ray theory to produce modeled cross spectra, the broadband localization scheme is applied to an experimental dataset in which a pseudorandom noise source was towed past a bottom-moored vertical array in a deep-ocean environment. Localization is successful out to the maximum range of 43 km. The effects on the source localization of varying such parameters as the number of phones, bandwidth, and receiver aperture are examined. It is found that matching the autospectra as well as the cross spectra significantly degrades the localization, and that coherent summation over both frequency and phone pairs is superior to incoherent summation.
Occluded-ear simulator with variable acoustic properties
Authors: Egolf David P. , Kennedy William A. , Larson Vernon D.
Published: October 2005

Ear simulators were designed to replicate acoustical characteristics of the average adult ear. Due to variability of ear-canal geometry and eardrum impedance among individuals, the possibility of any one person exhibiting such ‘‘average’’ characteristics—especially if that person is a child and/or has a conductive pathology—is remote. Thus, ear simulators have been of only peripheral value when prescribing a hearing aid (a high output impedance device) to fit the acoustical requirements of a particular patient. Reported herein is development of a programmable artificial ear (PAE) that can account for individual differences in ear-canal geometry and eardrum impedance. It consists of a 2.0-cc coupler, microphone, amplifier, computer, PAE code, and a computer card and/or software for digitization and Fourier transformation. Required input data includes ear-canal dimensions, eardrum impedance, and output impedance of the hearing aid being tested. Sound-pressure recordings produced in the 2.0-cc coupler by the hearing aid are adjusted by the computer to what they would have been had the recordings been made at the eardrum of a particular patient wearing the same hearing aid. Good agreement was observed between experiment and theory for one test case involving a totally occluding miniature earphone.
Auditory nerve representation of a complex communication sound in background noise
Authors: Simmons Andrea Megela , Schwartz Joshua J. , Ferragamo Michael
Published: October 2005

A population study of auditory nerve responses in the bullfrog, Rana catesbeiana, analyzed the relative contributions of spectral and temporal coding in representing a complex, species-specific communication signal at different stimulus intensities and in the presence of background noise. At stimulus levels of 70 and 80 dB SPL, levels which approximate that received during communication in the natural environment, average rate profiles plotted over fiber characteristic frequency do not reflect the detailed spectral fine structure of the synthetic call. Rate profiles do not change significantly in the presence of background noise. In ambient (no noise) and low noise conditions, both amphibian papilla and basilar papilla fibers phase lock strongly to the waveform periodicity (fundamental frequency) of the synthetic advertisement call. The higher harmonic spectral fine structure of the synthetic call is not accurately reflected in the timing of fiber firing, because firing is ‘‘captured’’ by the fundamental frequency. Only a small number of fibers synchronize preferentially to any harmonic in the call other than the first, and none synchronize to any higher than the third, even when fiber characteristic frequency is close to one of these higher harmonics. Background noise affects fiber temporal responses in two ways: It can reduce synchronization to the fundamental frequency, until fiber responses are masked; or it can shift synchronization from the fundamental to the second or third harmonic of the call. This second effect results in a preservation of temporal coding at high noise levels. These data suggest that bullfrog eighth nerve fibers extract the waveform periodicity of multiple-harmonic stimuli primarily by a temporal code.
An improved formalism for rough-surface scattering. II: Numerical trials in three dimensions
Authors: Milder D. Michael , Sharp H. Thomas
Published: October 2005

The recently described operator-expansion formalism can in theory accurately compute the wave scattering from arbitrary rough surfaces so long as the Fresnel number, the Rayleigh height times the surface slope, is small at all roughness scales. A limited number of tests have previously confirmed this for simple one-dimensional profiles. Because of its efficiency the method can be implemented readily on two-dimensional surfaces of substantial size as well. Trial computations on doubly periodic ‘‘eggcrate’’ gratings, for which exact solutions are available, yield accuracies as good as in one surface dimension; for example, cross-section errors are 1% or 2% for Fresnel numbers of 0.3. A simulated random ocean surface on a 256×256 square grid produces bistatic scattering amplitudes that can be seen to combine Bragg diffraction at short scales with geometric shadowing at long scales.
Sea ice elastic moduli: Determination of Biot parameters using in-field velocity measurements
Authors: Williams K. L. , Francois R. E.
Published: October 2005

Elastic moduli of sea ice have been determined from high-frequency velocity measurements on small samples extracted from the arctic pack ice. The experimental apparatus allowed quick in-field observations of the velocities. The experimental procedure included measurement of the ice samples’ salinity and temperature; the relations derived here can be used to predict elastic moduli from knowledge of these more easily obtainable physical parameters. The elastic moduli determined are those appropriate for a specialized Biot model in which the ice is viewed as an isotropic material with closed pores. For this model, seven parameters are needed to predict acoustic velocities, all of which are easily interpreted physically and are given here in terms of their salinity and temperature dependencies.
The role of remote rate information and relative acoustic resemblance on phonetic representations.
Authors: Newman R. S. , Sawusch J. R.
Published: October 2005

Miller and Liberman [Percept. Psychophys. 25, 457–465 (1979)] demonstrated that listeners use vowel duration information when making the /ba/–/wa/ distinction. They also found that adding a final consonant to the syllables shifted the /ba/–/wa/ boundary. However, they did not examine the effects of changing the final consonant duration. Furthermore, by using stimuli that were voiced throughout, the potential effects of acoustic and auditory similarity between the initial consonant distinction and the succeeding phonemes could not be examined. In this experiment CVC syllables were used to examine the effect of this later-occurring speaking rate information on an initial /sh/–/ch/ distinction. The syllables ended with /s/, and both vowel and final fricative duration were varied independently. This allowed one to examine the effect of acoustic similarity on the coding of rate information. The relevance of these results to the auditory and/or phonetic coding of speaking rate information will be discussed. [Work supported by NIDCD Grant No. DC00219 to SUNY at Buffalo.]
The effects of coarticulation on Japanese voiceless consonant discrimination.
Authors: Kojima Hideki , Kimura Shinta , Loken-Kim K. H.
Published: October 2005

In this paper, the results of a study of automatic discrimination of Japanese voiceless fricatives in vowel (/a,i,u,e,o,n/)–consonant (/F,s,h/) –vowel (/a,i,u,e,o/) syllables are presented, and the effects of coarticulation on this discrimination are discussed. A total of 4032 speech data uttered by 16 male subjects were used for the study. A higher recognition rate was obtained when the acoustic features were extracted from the consonant and succeeding vowel transition region than when they were extracted from the consonant region or the succeeding vowel transition region alone. Significantly, the extracted features from the consonant–vowel transition region were most effective in the recognition of consonants succeeded by /a/, but less effective for those succeeded by /i/, /e/, and /o/. Furthermore, when /i/ and /e/ were preceded by /F/ and /s/ the recognition rate was low. Similarly when /o/ was preceded by /F/ and /h/ the recognition rate was low. This phenomenon can be explained from the perspective of consonant–vowel coarticulation.
Stable localized symmetric integral equation method for acoustic scattering problems
Authors: Zeng Xiaogang , Kallivokas Loukas F. , Bielak Jacobo
Published: October 2005

An energy-based infinite boundary element integral equation method is developed for the solution of two- or three-dimensional time harmonic fluid scattering problems. This method is essentially based on a domain decomposition that insures the validity for all frequencies, and uses a hypersingular operator that can be integrated readily by standard procedures for single layers. It leads to a set of sparse, symmetric discretized equations. Numerical experiments for a rigid circular cylindrical scatterer subjected to a plane incident wave confirm the stability of the new procedure, and serve to assess its accuracy for wave numbers ranging from 0 to 30, both directly on the scatterer and in the far field.
Dynamic internal response of fluid-loaded multilayered anisotropic media
Authors: Taylor Timothy W. , Nayfeh Adnan H.
Published: October 2005

Theoretical investigations are carried out on the interaction of ultrasonic waves with multilayered plates separating a fluid from a vacuum, another fluid, or a solid half-space. It is assumed that each constituent of the plate and the solid half-space possesses no less than monoclinic symmetry. The plate is subjected to incident acoustic waves (propagated through a fluid) at arbitrary angles from the normal as well as at arbitrary azimuthal angles. It is further assumed that all solid interfaces are either rigidly or smoothly bonded. Solutions are obtained for the individual layers that relate the field variables at the upper and lower layer surfaces. The response of the total system proceeds by satisfying appropriate interfacial conditions across the constituents. The internal response, consisting of complete descriptions of the variations of displacements and stresses, of both the layered plate and the half-space is determined. Reflection and transmission coefficients are derived as by-products of the analysis. Representative numerical results are given in order to delineate the influence of the various bounding media and the smooth bond assumption on the propagation process and internal response.
Sound radiation from vibrating bodies in motion
Authors: Wu Xiao-Feng , Akay Adnan
Published: October 2005

This paper presents an integral formulation for prediction of sound radiation from a vibrating body in motion. The formulation consists of volume and surface integrals. The volume integral represents the effect of the turbulent stress field induced by the motion of the vibrating body, while the surface integral represents the combined effects of the gradient of the turbulent stress field and the unsteady force exerted on the fluid by the surface, as well as the surface displacement effect. It is shown explicitly that source cancellations take place within the surface integral when turbulence effects are taken into account. Moreover, dimensional analysis demonstrates that the volume integral is negligible compared with the surface integral when the source is in rectilinear motion at low Mach numbers. If turbulence effects are neglected at the outset of the derivation, then there is no interaction between the turbulent stress field and the vibrating surface in motion and the resulting integral formulation is different. The fundamental difference is whether or not the effect of the gradient of the turbulent stress field is included in the integral formulation. Numerical examples of sound radiation from the dilating and transversely oscillating spheres moving rectilinearly in space at constant subsonic speeds are demonstrated.
Nonlinear propagation of plane and circular Rayleigh waves in isotropic solids
Authors: Zabolotskaya E. A.
Published: October 2005

Nonlinear Rayleigh wave propagation in an isotropic solid is investigated theoretically. Hamiltonian formalism is used to derive a set of coupled equations for the harmonic amplitudes. Both plane and circular waves are considered. Numerical results are presented for an initially monochromatc wave that propagates in steel. It is shown that the horizontal component of the particle velocity wave forms a shock profile, while the vertical component forms a pulse. An evolution equation for the waveform is derived.
Errors in attenuation measurements due to nonlinear propagation effects
Authors: Zeqiri Bajram
Published: October 2005

An investigation into the influence of finite amplitude distortion on narrow-band ultrasonic attenuation measurements is described. Measurements have been made using a through-transmission substitution technique in the nonlinear field of a 5-MHz plane-piston transducer driven under tone-burst conditions. Various transducer excitation levels were used to generate a range of shock parameters σ at the position of the measuring hydrophone up to a maximum of 3. The influence of the resulting loss in amplitude at the fundamental frequency has been studied by measuring the transmission properties of reference attenuators consisting of Dow Corning-710 fluid-filled cells of various thickness. The presence of nonlinear distortion in the acoustic waveform produces overestimates of the measured transmission coefficients. The magnitude of the error has been shown to depend on the value of σ, the small signal transmission loss of the sample and its position in the acoustic field. In some situations, the error was as high as 100%. A theoretical model that describes the variation of the transmission coefficient with σ has been derived and good agreement demonstrated between theory and experiment. The implications for attenuation measurements, both narrow band and broadband, are discussed.
Imaged travel time surface of the transverse cusp caustic produced by reflection from a curved surface and the shape of reflected transients.
Authors: Frederickson Carl K. , Marston Philip L.
Published: October 2005

As a curved wavefront propagates in a uniform medium it may form caustics. Catastrophe theory can be used to classify the structurally stable caustics, where the initial local shape of the wave front determines the classification. These caustics have an associated travel time surface that describes the merging and disappearance of rays as the caustic is crossed. The travel time surface is defined by the travel time of signals propagating along rays from the wave front to the observer. Experiments were performed to image an acoustical transverse cusp diffraction catastrophe along with its associated travel time surface. The finite difference in arrival times of the acoustical signals allows the resolution of the travel time surface. Reflection of a spherical source signal from a curved surface in water was used to produce an initial wave front with the general local shape that produces a transverse cusp. Calculations of the expected travel time surface compared well to images cut through the experimental surface. Acoustical signals traveling along rays that touch the caustic are shown to have the Hilbert transform shape of the incident signal. Those signals that do not touch the caustic have the same general shape as the incident signal. [Work supported by ONR.]