bluuebunny/crossref_metadata_2025 huggingface dataset:
Model study of the sound propagation behind barriers of finite length
Authors: Pirinchieva R.
Published: October 2005

A scale model study is presented of the sound diffraction behind thin barriers of finite length on hard, reflecting ground. The measurements have been carried out in an anechoic chamber using a point source for the frequency range 5–20 kHz and scaled barrier lengths from 5 to 120 cm, with a scale factor of 1:20. The line integral theory [see Embleton, J. Acoust. Soc. Am. 67, 43–45 (1980)] was used to calculate by numerical integration the sound field behind the barrier, taking into account the diffraction not only from the top edge of the screen, but also from both side edges. In each case the prediction scheme allowed for the reflection from the finite-impedance ground. Experimental curves of sound pressure level distribution behind barriers of different lengths and heights have shown very good agreement with theoretically predicted values.
Real-ear attenuation of earmuffs in normal-hearing and hearing-impaired individuals
Authors: Suter Alice H. , Lempert Barry L. , Franks John R.
Published: October 2005

Many of the 9 million workers exposed to average noise levels of 85 dB(A) and above are required to wear hearing protection devices, and many of these workers have already developed noise-induced hearing impairments. There is some evidence in the literature that hearing-impaired users may not receive as much attenuation from hearing protectors as normal-hearing users. This study assessed real-ear attenuation at threshold for ten normal-hearing and ten hearing-impaired subjects using a set of David Clark 10A earmuffs. Testing procedures followed the specifications of ANSI S12.6-1984. The results showed that the hearing-impaired subjects received slightly more attenuation than the normal-hearing subjects at all frequencies, but these differences were not statistically significant. These results provide additional support to the finding that hearing protection devices are capable of providing as much attenuation to hearing-impaired users as they do to normal-hearing individuals.
Directional dependence of interaural envelope delays
Authors: Middlebrooks John C. , Green David M.
Published: October 2005

Interaural envelope delays were measured in six human subjects as a function of the location of a movable sound source, bandpassed between 3 and 16 kHz. A total of 324 source locations were tested in horizontal and vertical increments of 10°. A method is described for estimating the complex directional transfer function of the external ear, independent of the position of the recording microphone in the ear canal. To compute interaural envelope delays, directional transfer functions from the left and right ears were convolved with a critical-band filter, the envelopes were computed, and the envelopes were cross correlated. Interaural envelope delays, as well as interaural group delays, varied somewhat with the center frequency of the critical-band filter and with the vertical location of the sound source. Nevertheless, to a first approximation, envelope delays measured in the ear canals increased monotonically with increasing angle of incidence relative to the median plane, as they would for two microphones on the surface of a rigid sphere. The results are discussed in relation to the possible contribution of interaural envelope delays to sound localization behavior.
Frequency discrimination of complex tones with overlapping and non-overlapping harmonics
Authors: Moore Brian C. J. , Glasberg Brian R.
Published: October 2005

These experiments address the following issues. (1) When two complex tones contain different harmonics, do the differences in timbre between them impair the ability to discriminate the pitches of the tones? (2) When two complex tones have only a single component in common, and that component is the most discriminable component in each tone, is the frequency discrimination of the component affected by differences in residue pitch between the two tones? (3) How good is the pitch discrimination of complex tones with no common components when each tone contains multiple harmonics, so as to avoid ambiguity of pitch? (4) Is the pitch discrimination of complex tones with common harmonics impaired by shifting the component frequencies to nonharmonic values? In all experiments, frequency difference limens (DLCs) were measured for multiple-component complex tones, using an adaptive two-interval, two-alternative, forced-choice task. Three highly trained subjects were used. The main conclusions are as follows. (1) When two tones have the first six harmonics in common, DLCs are larger when the upper harmonics are different than when the upper harmonics are in common or are absent. It appears that differences in timbre impair DLCs. (2) Discrimination of the frequency of a single common partial in two complex tones is worse when the two tones have different residue pitches than when they have the same residue pitch. (3) DLCs for complex tones with no common harmonics are generally larger than those for complex tones with common harmonics. For the former, large individual differences occur, probably because subjects are affected differently by differences in timbre. (4) DLCs for harmonic complex tones are smaller than DLCs for complex tones in which the components are mistuned from harmonic values. This can probably be attributed to the less distinct residue pitch of the inharmonic complexes, rather than to reduced discriminability of partials. Overall, the results support the idea that DLCs for complex tones with common harmonics depend on residue pitch comparisons, rather than on comparisons of the pitches of partials.
Two-dimensional sound localization by human listeners
Authors: Makous James C. , Middlebrooks John C.
Published: October 2005

This study measured the ability of subjects to localize broadband sound sources that varied in both horizontal and vertical location. Brief (150 ms) sounds were presented in a free field, and subjects reported the apparent stimulus location by turning to face the sound source; head orientation was measured electromagnetically. Localization of continuous sounds also was tested to estimate errors in the motor act of orienting with the head. Localization performance was excellent for brief sounds presented in front of the subject. The smallest errors, averaged across subjects, were about 2° and 3.5° in the horizontal and vertical dimensions, respectively. The sizes of errors increased, for more peripheral stimulus locations, to maxima of about 20°. Localization performance was better in the horizontal than in the vertical dimension for stimuli located on or near the frontal midline, but the opposite was true for most stimuli located further peripheral. Front/back confusions occurred in 6% of trials; the characteristics of those responses suggest that subjects derived horizontal localization information principally from interaural difference cues. The generally high level of performance obtained with the head orientation technique argues for its utility in continuing studies of sound localization.
Observables due to beam-to-mode conversion of a high-frequency Gaussian P-wave input in an aluminum plate in vacuum
Authors: Lu I. T. , Felsen L. B. , Klosner J. M.
Published: October 2005

Obliquely incident Gaussian P-wave inputs furnish important test signals for ultrasonic detection of flaws in layered elastic media. Unlike vertically impinging beams, multiply reflected oblique beams eventually are converted into guided modes. The horizontal ranges over which the field is beamlike or modelike must be well understood for selection of a diagnostic scheme with ‘‘good’’ features for detection and identification. An analytical study is conducted here to detail the beam-to-mode conversion and P–SV coupling processes in an aluminum plate. The rigorously based approach utilizes the complex source point (CSP) method for generation of the input beam, and offers ray, mode, and hybrid ray–mode alternatives for best characterization of the wave phenomena. Numerical reference results over a representative range of parameters, which furnish potentials and displacements in various cross sections and on the plate boundaries, are interpreted in terms of beam or mode observables and related to the spectral wavenumber content of the incident signal. These considerations lead to the formulation of certain rules that categorize the detected signal in a manner which may be helpful in the construction of inversion algorithms for flaw detection.
Acoustical measurement of fish abundance
Authors: MacLennan David N.
Published: October 2005

The theory of echo formation provides formulas relating echo energy to physical characteristics of the target. Single-target theory (applicable to counting isolated fish) is extended to the multiple-target case relevant to schooling fish. An echo-integrator equation relates fish density to echo energy integrated over a time gate corresponding to the depth channel of interest. Parameters include the equivalent beam angle, the expected backscattering cross section per fish, equipment sensitivity, and a time-varied-gain correction factor. Variation of environmental factors (sound speed and absorption) affects the parameter values. More important is the variation of biological factors (fish behavior and physiology) which affects backscattering cross section and target strength. Verification of the echo-integrator equation, depending upon the linearity principle concerning the addition of signals from randomly distributed multiple targets, is discussed in the fisheries context. The swimbladder is the dominant sound reflector in fish having one. Accordingly, fish targets may be classified as (a) bladder closed, (b) bladder open, or (c) no bladder. Within each category, fish of the same size have similar target strengths. Target strength variation with fish size, water depth, and time is discussed. Experimental target strengths are well scattered even for nominally similar fish. Nevertheless, useful information about fish stocks can be obtained through careful application of acoustical techniques.
A comparison of parametric and nonparametric detector performance levels in underwater noise
Authors: Nielsen P. A. , Thomas J. B.
Published: October 2005

The aim of the work is to determine and clarify the effects of some typical problems that arise when working with a real-noise source. To this end, stationarity, time correlation between data samples, and parametric and nonparametric detection for a large segment of Arctic under-ice noise are examined. In particular, the effect that correlation between samples has on the ability to detect a known signal in additive noise using several different parametric detectors is analyzed. Since most underwater environments are both nonstationary and non-Gaussian, nonparametric detectors are of interest also. Thus a comparison of the performance levels of several different nonparametric detectors is performed for four common underwater environments.
Characterizing auditory neurons using the Wigner and Rihacek distributions: A comparison
Authors: Eggermont Jos J. , Smith Geoff M.
Published: October 2005

Because of their dynamic properties, most sounds can best be characterized in the combined frequency-time (FT) domain. Powerful frequency-time characterizations are the Wigner distribution function (WDF) and the Rihacek energy density function (RDF). In the present paper several new concepts are introduced such as using the WDF to characterize the tuning of auditory neurons under wideband noise stimulation and a new method to quantify phase lock of auditory neurons to a wideband noise. No appreciable differences were found between the WDF and RDF in narrow-band signal representations. However, the differences between the WDF and RDF increase as the bandwidth of the signal increases. When signals are buried in uncorrelated background noise, the average FT function of these signals may be obtained through averaging the FT functions for each signal plus noise segment. The WDF takes at least a factor 2 more in time to compute than the RDF. The FT functions can be used to characterize (linear) filters by averaging FT functions of input-noise segments that precede threshold crossings of the filter’s output signal. Both the WDF and the RDF were used to characterize auditory neurons from the midbrain in anurans; the WDF always had a smaller bandwidth than the RDF. By comparing the spectrum of the reverse correlation function and the average spectrum of the noise segments preceding the spikes, a quantification of the amount of phase lock of the auditory neuron to the noise is obtained.
Efficient across-frequency integration in short-signal detection
Authors: van den Brink Willem A. C. , Houtgast Tammo
Published: October 2005

A series of experiments was performed on the influence of bandwidth on the masked threshold of brief deterministic signals in continuous broadband noise. The signal bandwidth is quantified by the number (n) of constituent 1/3-oct bands. For n increasing from 1 to typically 9, the masked-threshold level in the constituent 1/3-oct bands is found to decrease by 8 log(n). This integration rule is obtained when each of the 1/3-oct bands covered by the signal equally contributes to detection, i.e., that, for each of these 1/3-oct bands, the difference between signal level and the individual masked-threshold level is the same. It was found that this integration rule also applies to noncontiguous signal spectra and that it remains intact over a broad range of masker levels. Commonly, the masked threshold of compound signals (for instance, n frequency components with a spacing of typically 1/3 oct), relative to the masked threshold of single-component signals, has been described by a 5 log(n) integration rule. However, this rule was obtained for signal durations of typically 100 ms or more. For the present brief signals (typically 10 ms or less), the across-frequency integration is found to be more effective.
Sensitivity to envelope coherence
Authors: Green David M. , Richards Virginia M. , Onsan Zekiye A.
Published: October 2005

Measurements are reported on the ability of observers to discriminate whether the envelope of two amplitude-modulated sinusoids are in phase or out of phase. Spacing between the two carriers was either 2/3 or 4/3 octave, and the depth of modulation was varied to determine threshold. Discrimination performance improved as the level of the carriers increases up to about 60 dB SPL. The frequency locus of the two carriers (geometric mean of the two frequencies), which varied from 500 to 8000 Hz in different experiments, had little effect on discrimination accuracy. Discrimination performance was relatively constant for modulation rates below 100 Hz and deteriorates for higher modulation rates. These results are compared with data obtained from comodulation masking release experiments.
Infant speech-sound discrimination in noise
Authors: Nozza Robert J. , Rossman Reva N. F. , Bond Linda C. , Miller Sandra L.
Published: October 2005

The effects of noise on 7- to 11-month-old infants’ speech-sound discrimination (/ba/vs/ga/) were determined using a conditioned head-turn procedure. Variation in performance as a function of signal-to-noise ratio (S/N) was estimated by testing each infant at four S/N’s (−8, 0, 8, and 16 dB). Adults were tested for comparison at four S/N’s (−12, −8, −4, and 0 dB). The S/N’s were chosen based on pilot data. Performance varied monotonically with S/N for both age groups, but infants required greater S/N than adults to achieve comparable levels of performance. Both groups were also tested using an adaptive (1-up, 1-down) threshold procedure with a 3-dB step size. There was a group mean difference in threshold of 5.8-dB S/N favoring the adults. Weighted group psychometric functions, derived from the responses obtained in the adaptive runs, showed good correspondence with the data points at the four S/N’s. The slopes of these functions were the same (7.5%/dB) for infants and adults. The results suggest that infants are at a greater disadvantage than adults when processing speech in noise and that concern over the effects of a noisy environment on the acquisition of language is justified. In addition, the adaptive threshold procedure can be used as an efficient way to estimate the limits of discrimination ability as a function of S/N or intensity, both for individual subjects and for groups of subjects, in developmental research.
Analysis of backscattering data in the Tyrrhenian Sea
Authors: Preston John R. , Akal Tuncay , Berkson Jonathan
Published: October 2005

Research has been conducted in the Tyrrhenian Sea basin to measure low-frequency basin reverberation and relate the measurements to backscattered strength. The receivers were a towed horizontal array and a suspended vertical array. Standard Italian Navy depth (74 kg) and SUS (0.818 kg) charges were used as sources. The unique factor in this broadband basin reverberation experiment was that data were collected simultaneously from the horizontal and vertical array. Towed-array data were taken on multiple headings; combining different headings permitted removal of the ambiguous directionality information from the towed array. Frequency-domain beamformed data are presented and reveal that, in the area examined, basin features are the strongest backscatterers. Receiver beam level data are presented primarily for 300 Hz, but some data are given for 100, 450, and 730 Hz as well. Estimates of backscattering strengths of different features are provided and are in the range quoted in other publications. A comparison of the horizontal and vertical array scattering strengths estimated for some selected basin features is also included. Statistics for the dominant backscatterers are given, and the results from one basin feature are presented showing detailed backscatter structure.
High-frequency reflection and scattering by multicomponent rough surface distributions
Authors: Lucas R. J. , Twersky V.
Published: October 2005

Earlier forms for the coherent reflection and incoherent scattering by multicomponent mixtures of bosses on rigid or free base planes [V. Twersky, J. Acoust. Soc. Am. 29, 209–225 (1957)] are applied to recent high-frequency results for aligned hemiellipsoidal bosses [R. J. Lucas and V. Twersky, J. Acoust. Soc. Am. 83, 2005–2011 (1988)] to investigate continuous distributions in boss size. Approximations for the coherent reflected intensity and incoherent differential scattering cross sections are obtained in terms of integrals of simple functions and a general probability density. To provide illustrations, numerical computations and graphical results are based on truncating the two-parameter gamma probability density function P(t;m,v) with t as a dimensionless variable that scales one or more boss dimensions, m as the mean value of t, and v as the normalized variance (ranging from zero to unity). For v small, P is Gaussian and reduces to a delta function as v approaches zero (to reproduce one-component results). More generally, the curve of P is skewed, and as v approaches unity P reduces to the exponential for the simplest Poisson case. Graphs are shown for cases where one (e.g., keel depth), two (e.g., base axes), or all three dimensions of the protuberances are randomized. The essentials are indicated by plots versus angle of incidence, with v as the parameter. The coherent intensity and the associated forward and backscattered incoherent differential scattering cross sections per unit area are emphasized.
Analysis of echoes from a sphere which includes the directivity of a transmitter and a receiver
Authors: Ueda Mitsuhiro , Morimatsu Eishi
Published: October 2005

Analysis of an echo from a solid elastic sphere has been limited to the case of a plane incident wave so far. Plane-wave theory is modified to the case of spherical incident wave first, and then it is modified to the case of an arbitrary ultrasonic transmitter and receiver by considering them as a set of point sources and point receivers, respectively. Using this relation, the general properties of the echo scattered from the sphere are analyzed first, and then the analysis is carried out in full for important practical cases; that is, the sphere is placed on the focal point of a circular concave transducer and in the focal plane of the transducer. It is applied to study the material dependence of the echo from the sphere placed at the focal point of the transducer as well as that of the sidelobe structure of the echo. These results are compared with those experimentally obtained using metal and plastic spheres of 3 mm in diameter, which show very good agreement.
Scattering of sound by sound from two Gaussian beams
Authors: Darvennes Corinne M. , Hamilton Mark F.
Published: October 2005

The scattering of sound by sound from Gaussian beams that intersect at small angles is investigated theoretically with a quasilinear solution of the Khokhlov–Zabolotskaya nonlinear parabolic wave equation. The analytical solution, which is valid throughout the entire paraxial field, is a generalization of a result obtained previously for parametric receiving arrays [Hamilton et al., J. Acoust. Soc. Am. 82, 311–318 (1987)]. Significant levels of scattered sum and difference frequency sound are shown to exist outside the nonlinear interaction region. An asymptotic formula reveals that sound is scattered in the approximate directions of k1±k2, where kj is the wave vector associated with the axis of jth primary beam. Computed propagation curves and beam patterns demonstrate the dependence of the scattered radiation on interaction angle, source separation, ratio of the primary frequencies, and source radii. Comparisons are made with the farfield results presented by Berntsen et al. [J. Acoust. Soc. Am. 86, 1968–1983 (1989)], which are valid for arbitrary interaction angles, source separations, and amplitude distributions.
The influence of wind and temperature gradients on sound propagation, calculated with the two-way wave equation
Authors: Nijs L. , Wapenaar C. P. A.
Published: October 2005

A method is introduced to calculate the influence of wind and temperature gradients in stratified media on sound propagating above an absorbing ground surface. It is based on the ‘‘two-way wave equation’’ for the Fourier transforms of the sound pressure P and its derivative V. The vector containing P and V is stepwise extrapolated through the medium in the direction perpendicular to the ground surface, fulfilling the boundary conditions at the ground surface, at a top level, and at the source height. The propagation equations for P and V appear as simple plane-wave equations, and computer (CPU) time within each layer is very low. Therefore, many thin layers (in the order of centimeters if desired) can be applied, and any complicated gradient can be used. Calculations for a homogeneous atmosphere, with a computer program based on this model, show an excellent agreement with previous models. When a wind profile is present, results are mainly compared with measurements by Parkin and Scholes [P. H. Parkin and W. E. Scholes, J. Sound Vib. 1, 1–13 (1964); 2, 353–374 (1964)]. They show very good agreement in the no-wind and downwind cases. In the upwind situation, agreement is very good below 500 Hz. Above this value the model does not predict sound to penetrate into the shadow region, while Parkin and Scholes found (low) sound levels.
A numerical approach to rough-surface scattering by the parabolic equation method
Authors: Spivack Mark
Published: October 2005

The parabolic equation method is an effective approach when the acoustic wave field is incident at low grazing angles onto a rough surface. The method consists of an integral equation and an integral, the first of which yields the surface field derivative. The main part of this paper is concerned with an approximation to this equation, valid when wavenumber times surface height is up to order unity. The approximation has several advantages. First, it allows a decomposition of the equation into deterministic and stochastic components. The stochastic part depends only locally upon the surface in certain regimes, and this can give rise to a great reduction in computational expense. Some basic statistical moments of the stochastic component are also considered. These are nonstationary, but for the incident field a simple stationary transformation is found, which can therefore be compared with Monte Carlo simulations using far fewer realizations. These results are demonstrated computationally. The final part of the paper describes the numerical implementation of the full parabolic equation method. Both of the integrals contain singularities, and these are treated semianalytically.
The time-marched fast-field program (FFP) for modeling acoustic pulse propagation
Authors: Porter Michael B.
Published: October 2005

Fast-field programs (FFPs) have emerged as an important tool for predicting transmission loss in an ocean waveguide. Such models have been primarily used for time-harmonic sources; however, pulses or other broadband sources may be treated by Fourier synthesis. A new technique is developed that provides a direct solution by marching the solution forward in time. As an example of the method, a pulse incident on an interface between two homogeneous half-spaces is considered. Snapshots of the pulse in time illustrate graphically the effects on the reflected and transmitted waves. Second, an interesting hyperbolic cosine profile is considered that leads to repeated focusing as the pulse propagates out in range.
Underwater acoustic ambient noise levels on the eastern Canadian continental shelf
Authors: Zakarauskas Pierre , Chapman David M. F. , Staal Philip R.
Published: October 2005

This paper reports the analysis of shallow-water ambient noise levels collected by Defence Research Establishment Atlantic during 14 cruises over the period 1972 to 1985. A weighted average is formed to de-bias the samples, with the aim of answering the question: ‘‘If one were to pick a site randomly on the eastern Canadian continental shelf at a random time, and perform an ambient noise measurement, what would be the expected noise level?’’ The samples are also grouped according to whether they were taken on the Scotian Shelf, the Grand Banks, or the Flemish Cap, and according to season. The frequency range covered is 30 to 900 Hz. The weighted mean and standard deviation of the noise levels are presented, as well as the correlation coefficient between the noise levels and wind speed. The results show that the eastern Canadian continental shelf as a whole presents levels that are characteristic of areas with high shipping density and good acoustic propagation, with the Scotian Shelf showing generally higher ambient noise levels. Finally, a comparison with Piggott’s ambient noise measurements [C. L. Piggott, ‘‘Ambient sea noise at low frequency in shallow water of the Scotian Shelf,’’ J. Acoust. Soc. Am. 36, 2152–2163 (1964)] on the Scotian Shelf is made, and an explanation for the observed difference is suggested.
Effectiveness of mode filtering: A comparison of matched-field and matched-mode processing
Authors: Yang T. C.
Published: October 2005

In a previous paper [T. C. Yang, J. Acoust. Soc. Am. 82, 1736–1745 (1987)] an eigenvector (EIG) mode decomposition method was used as the basis for modal beamforming (matched-mode processing). In this article, it is pointed out that matched-field processing is equivalent to a modal beamforming method with a different mode decomposition algorithm (denoted as the MFP-mode decomposition method). The difference in performance between the two processing schemes can be understood in terms of the effectiveness of mode filtering of the mode decomposition method used. It is found that the difference between the MFP- and EIG-mode decomposition is small if the modes associated with the signal and noise are adequately sampled by a (filled) vertical array, and potentially large if they are not. The difference is analyzed analytically and illustrated numerically using vertical arrays in an Arctic and Pacific environment. Array gain and peak-to-sidelobe ratio are evaluated for the MFP and EIG processors under identical environmental conditions, and compared with the matched-mode processing assuming perfect mode decomposition. Examples are given to illustrate the use of mode filtering to suppress the sidelobes caused by sound-speed mismatch in the upper (<1000 m) water column in the Pacific. It is found that higher-order modes are essential in keeping the sidelobe level down, despite the fact that they are not as well resolved as the lower-order modes. The (lower) modes that are responsible for the sidelobes (due to the sound-speed mismatch) are identified. It is concluded that source localization is possible in a mismatch environment, if proper mode filtering is applied to the data.
Transmission loss of a multilayer panel with internal tuned Helmholtz resonators
Authors: Prydz R. A. , Wirt L. S. , Kuntz H. L. , Pope L. D.
Published: October 2005

Two different methods of analysis are presented for the transmission loss prediction of a multilayer panel with internal Helmholtz resonators. One of the methods, devised at the Lockheed Aeronautical Systems Company, is based on the pressure ratio methodology of Beranek and Work [J. Acoust. Soc. Am. 21, 419–428 (1949)], where the acoustic resonators are included as a side branch. The other method uses a transfer matrix approach where the resonators are embedded in and move with the trim panel. Comparisons are made between numerical results obtained from the two types of analyses. Except for a minor shift in the frequency of maximum transmission loss, the agreement is very close. The shift is a function of the ratio of the nozzle throat area to the resonator surface area. When this ratio goes to zero the results from the two approaches become identical. Experimental results are shown for a double-wall panel with resonators tuned to 240 Hz and are compared against predictions with good agreement. The results show that the use of tuned acoustic resonators in a double-wall structure can provide large increases in transmission loss at selected frequencies. In addition, both theory and experiment show that for singly tuned resonator arrays the large transmission loss increase at the resonance frequency is accompanied by a large decrease in the transmission loss over a limited frequency range above the resonance frequency.
Improved active sonar detection using autoregressive prewhiteners
Authors: Kay Steven , Salisbury John
Published: October 2005

An active sonar receiver is proposed that more closely approximates the optimal prewhitener and matched filter receiver than currently existing active sonars. The prewhitener is based on an autoregressive model for the reverberation that is shown through analysis of real data to be an accurate model. At the output of the prewhitener a generalized likelihood ratio test detector is implemented. Experimental range-Doppler maps of real data demonstrate that low-Doppler targets are more easily detected than with current FFT processors.
A population study of auditory-nerve fibers in unanesthetized decerebrate cats: Response to pure tones
Authors: Kim D. O. , Chang S. O. , Sirianni J. G.
Published: October 2005

Population responses of single auditory-nerve (AN) fibers to 1-kHz pure tones were investigated in unanesthetized decerebrate cats. Driven rate (spikes/s), Fourier component amplitude (spikes/s), and d′e =(driven rate)/(standard deviation) were examined. These results extend the presently available information. First, the AN population responses, previously observed only in anesthetized animals, are now observed in unanesthetized animals. The anesthetized and unanesthetized animal responses are qualitatively similar. Second, a detectability measure d′e was applied, à la signal detection theory, to populations of AN fibers. A previously unknown behavior of the low-spontaneous-rate (SR) (<15 spikes/s) fibers was observed: the d′e increased to large values at moderate and high stimulus levels, giving rise to a large, sharply tuned peak in the d′e -place profile, even when the driven rate was saturated. This behavior was absent in the high-SR (>15 spikes/s) fibers. This observation suggests that the d′e of the low-SR fibers is the best code for stimulus intensity among the different response measures examined, and that the d′e -place profile may be a more precise spatial code for stimulus frequency than the commonly considered rate-place code.
NoSo and NoSπ detection as a function of masker bandwidth in normal-hearing and cochlear-impaired listeners
Authors: Staffel Jon G. , Hall Joseph W. , Grose John H. , Pillsbury Harold C.
Published: October 2005

NoSo and NoSπ detection thresholds for a 500-Hz pure-tone signal were measured as a function of masking noise bandwidth in normal-hearing and cochlear hearing-impaired subjects. NoSo and NoSπ critical bands were derived from the bandlimited noise functions. A notched noise measure of the monaural critical band was also obtained for each ear. One hypothesis tested was that an asymmetrical monaural critical band would result in a relatively steep improvement of the NoSπ detection threshold as a function of decreasing masker bandwidth and would, therefore, be associated with a wider binaural critical band. This was hypothesized because the outputs of the left and right auditory filters would be more decorrelated the greater the interaural difference in the monaural critical band. However, as the noise bandwidth was narrowed, the decorrelation would lessen, resulting in a relatively steep improvement in NoSπ detection. Results indicated that the masking level difference (MLD) was smaller and that the monaural critical bands were generally wider in cochlear-impaired listeners. NoSo and NoSπ critical bands were somewhat larger in the cochlear hearing-impaired listeners having relatively wide monaural critical bands. There was a significant correlation between monaural critical band asymmetry and the NoSπ critical band; however, this correlation was insignificant when a control was employed for the critical band in the worse ear. Therefore, the present results did not support a strong association between monaural critical band asymmetry and the width of the NoSπ critical band.
On the reflection functions associated with discontinuities in conical bores
Authors: Gilbert J. , Kergomard J. , Polack J. D.
Published: October 2005

The decomposition of reflections by finite tubes into successive elementary reflections by the input, discontinuities, and output of the tubes is interesting for its physical reasoning and for calculations using multiconvolution. In the case of tubes with conical bores, Martinez and Agulló [J. Martinez and J. Agulló, J. Acoust. Soc. Am. 84, 1613–1619 (1988)] have stated that in certain cases, the elementary reflection functions can be growing (causal) exponentials. Although such functions do not have Fourier transforms, Agulló et al. have assumed that they are inverse Fourier transforms of functions of frequency. It is shown that this assumption is erroneous. but by using a Laplace transform (i.e., by introducing losses through a complex frequency), we have also shown that for a general class of physically feasible systems, the decomposition into growing exponentials is valid.
Ultrasonic characterization of AlbunexR, a new contrast agent
Authors: Bleeker H. J. , Shung K. K. , Barnhart J. L.
Published: October 2005

A new, promising ultrasound contrast agent, Albunex■ (Molecular Biosystems, Inc., San Diego, California), is increasingly being used in experimental myocardial contrast echocardiography. Albunex■ is a suspension of stable microencapsulated air bubbles prepared from 5% albumin. Contrast agents like Albunex■ have been used to estimate regional myocardial perfusion. Current methods that use contrast media to assess myocardial perfusion in vivo from ultrasonic images are neither reliable nor absolute. The primary reason is that the fundamental relationship between ultrasonic properties and physical properties of the contrast media is unknown. Ultrasonic backscatter, acoustic velocity, and attenuation of Albunex■ as a function of its concentration, size, and temperature at 7.5 and 5 MHz were measured. The backscatter coefficient, which is an absolute measure of ultrasonic backscatter, was found to be approximately linearly proportional to the number of microspheres in Albunex■ (r=0.86) at low concentrations. The attenuation coefficient was found to be linearly proportional to the number of microspheres (r=0.96). An understanding of the relationship between the physical and ultrasonic properties of Albunex■ is essential if it is to be used to quantify myocardial perfusion.
Active control of stationary random sound fields
Authors: Nelson P. A. , Hammond J. K. , Joseph P. , Elliott S. J.
Published: October 2005

Previous work on the active control of sound has mostly used frequency domain formulations in order to establish the physical limitations of active methods. While entirely adequate for the prediction of the performance of active control systems designed to deal with deterministic primary fields, these methods cannot necessarily be applied in cases where the primary excitation is stationary random in nature. The application of frequency domain techniques often yields results for the optimal control strategy that require the secondary sources to act noncausally with respect to the primary sources. The work described here illustrates classical time domain methods for determining the performance limits of active noise control systems that are constrained to act causally. The first example considered is the minimization of the mean-squared acoustic pressure at a position in the field of a point monopole primary source by the introduction of a point monopole secondary source. The primary source radiates a stationary random signal and the secondary source is constrained to act causally with respect to the primary source. The active control of low-frequency random sound in enclosures is then addressed and the classical Wiener theory extended in order to deal with problems involving the minimization of multiple errors. The active control of a one-dimensional enclosed sound field is presented as a simple example. This theory is also used in presenting a third example that consists of a primary/secondary source pair radiating in a free field. The minimum acoustic power output of the source combination is calculated when the primary source radiates random sound and the secondary source is again constrained to act causally with respect to the primary source.
Acoustic radiation force on a small compressible sphere in a focused beam
Authors: Wu Junru , Du Gonghuan
Published: October 2005

A general expression for the acoustic radiation force on a small compressible sphere has been derived and expressed in terms of the time-averaged densities Ti and Vi of kinetic and potential energies, respectively, in the incident sound field. The results have been applied to two focused beams: Gaussian beams and piston beams. For both cases, the analytical expressions of the radiation forces on a small compressible sphere placed on the axis of the beams are calculated. The pertinent applications in acoustic levitation and bubble dynamics are discussed.
Offnormal incidence reflection-coefficient determination for thick underwater acoustic panels using a generalized ONION method
Authors: Piquette Jean C.
Published: October 2005

The ONION method is a reflection-coefficient measurement technique designed for use on data acquired from thick underwater acoustic panels in the frequency range 1–10 kHz, but may be used to frequencies as high as 25 kHz. The method extrapolates transient reflected-wave data using least-squares fitting to a multiple-layer panel model. A description of the method, as it applies to data acquired using a normally incident interrogating wave, is provided in J. C. Piquette, ‘‘The ONION method: A reflection coefficient measurement technique for thick underwater acoustic panels,’’ J. Acoust. Soc. Am. 85, 1029–1040 (1989). The present article describes research directed toward generalizing the method to allow for an offnormal incidence interrogating wave. Successful applications of the generalized method to data acquired under offnormal incidence are described. The generalized method was applied to measurements that were made on two different sample panels. The measurements involved one test temperature, two hydrostatic test pressures, and three test frequencies.
Dynamic response of an inert gas to slow piston acceleration
Authors: Wang Meng , Kassoy D. R.
Published: October 2005

This study describes a theoretical investigation of weak acoustic wave fields generated in a confined inert gas during slow piston acceleration processes. The piston velocity increases continuously from a zero initial value to a characteristic value, on the time scale of the piston passage from bottom to top of a cylinder. In contrast, the initial piston acceleration is finite. The gasdynamic phenomena occurring in the system are modeled by the unsteady, planar Euler equations, which are solved asymptotically in terms of the small characteristic piston Mach number ε. The finite initial piston acceleration is shown to initiate O(ε2) acoustic disturbances in the gas. A combination of multiple-time-scale methods and the Fourier series techniques is used to develop simultaneous descriptions of the evolving acoustic field and the accumulated bulk response of the gas to piston motion. It is found that the wave field remains linear throughout the period of piston motion. The wave amplitude is proportional to the initial acceleration of the piston and increases with the mean density of the gas.
An analytical solution for benchmark problem 1: The ‘‘ideal’’ wedge
Authors: Buckingham Michael J. , Tolstoy Alexandra
Published: October 2005

Results from an exact analytical solution for the field in a wedge with pressure-release boundaries (benchmark problem 1) are presented in the form of transmission loss as a function of horizontal range from the source. These computed curves contain no significant error, and thus may be used as a primary benchmark for establishing the accuracy of range-dependent, numerical propagation models.
Solutions to range-dependent benchmark problems by the finite-difference method
Authors: Stephen Ralph A.
Published: October 2005

An explicit second-order finite-difference scheme has been used to solve the elastic-wave equation in the time domain. Solutions are presented for the perfect wedge, the lossless penetrable wedge, and the plane parallel waveguide that have been proposed as benchmarks by the Acoustical Society of America. Good agreement with reference solutions is obtained if the media is discretized at 20 gridpoints per wavelength. There is a major discrepancy (up to 20 dB) in reference-source level because the reference solutions are normalized to the source strength at 1 m in the model, but the finite-difference solutions are normalized to the source strength at 1 m in a homogeneous medium. The finite-difference method requires computational times between 10 and 20 h on a super minicomputer without an array processor. The method has the advantage of providing phase information and, when run for a pulse source, of providing insight into the evolution of the wave field and energy partitioning. More complex models, including velocity gradients and strong lateral heterogeneities, can be solved with no additional computational effort. The method has also been formulated to include shear wave effects.
Reflection of plane acoustic waves from a layer of varying density
Authors: Robins Alvin J.
Published: October 2005

Analytical solutions of the Helmholtz equation are derived for the reflection of plane acoustic waves from a layer of varying density between two homogeneous media. The solutions show that the reflection coefficient is dependent on dimensionless vertical wavenumber and on the shape of the density profile in the intermediate layer. The most general case considered is one in which a step change in density is followed by a continuously varying density profile in a transition layer of finite thickness. Results are presented illustrating the dependence of reflection coefficient on wavenumber and on the structure of the transition layer. It is demonstrated that, in the limit of very low wavenumber or grazing incidence, the reflection coefficient tends to the Rayleigh coefficient appropriate to the overall density change between upper and lower media, while, in the high wavenumber limit, the reflection is given by the Rayleigh coefficient for the step change only.
Model of a bilaminar actuator for active acoustic control systems
Authors: Bao X.-Q. , Varadan V. K. , Varadan V. V. , Howarth T. R.
Published: October 2005

To increase levels of sound attenuation, it is often necessary to utilize active acoustic control methods. A complete active acoustic control system typically consists of sensing devices, an electronic control circuit, and an actuating device. Each component is designed to cause destructive interference of the impressed acoustic pressure field. A piezocomposite actuator is presented for active underwater attenuation control of an impressed acoustic field. The actuator is composed of bilaminate plates for absorption of both the reflection and the transmission of normally incident acoustic waves independent of the backing impedance. An electromechanical model of the bilaminate piezocomposite design with an analytical discussion is presented. An illustrated numerical example and discussion of system implications are included.
Reply to ‘‘Comments on ‘Passive synthetic arrays’ ’’ [J. Acoust. Soc. Am. 8
5, 1372 (1989)]
Authors: Autrey Samuel W.
Published: October 2005

This is a reply to the Letter to the Editor by Carey [J. Acoust. Soc. Am. 85, 1372 (1989)] that takes issue with ‘‘Passive synthetic arrays’’ [S. W. Autrey, J. Acoust. Soc. Am. 84, 592–598 (1988)]. The assertions in that letter are shown to be incorrect.
Contraction and relaxation-induced oscillations of the left ventricle of the heart during the isovolumic phases
Authors: Lewkowicz M. , Chadwick R. S.
Published: October 2005

A theoretical analysis is presented for the transient dynamical response of the left ventricle of the heart during the isovolumic contraction and relaxation phases of the cardiac cycle. Small oscillations of the left ventricular cavity pressure and wall motion are excited by the initial rates of filling and emptying of the ventricle as well as the rate of change in muscle fiber activation. The analysis applies to the genesis of the first and second heart sounds. The ventricle is modeled as a finite, thick-walled incompressible cylinder having a continuum of imbedded axial and circumferential active muscle fibers, which interacts with a fixed volume of an incompressible, ideal fluid. The solution is obtained using a two-timing asymptotic expansion procedure. The theoretical calculations of left ventricular pressure waveforms compare favorably with published recorded pressure waveforms. The amplitude spectra of computed waveforms contain information concerning the active elastic modulus of the fibers, which is a measure of cardiac contractility.
Bubble oscillations in the vicinity of a nearly plane free surface
Authors: Oguz Hasan N. , Prosperetti Andrea
Published: October 2005

The linear oscillation frequency of a bubble in the vicinity of a distorted plane free surface is calculated by a perturbation method. The approximate expression found is compared with numerical results valid for surface deformations of arbitrary magnitude. It is found that the approximate analytical result is quite good, provided that the deformation is small compared with the depth of immersion of the bubble. It is also shown that, unless the deformation of the free surface extends to distances at least of the order of an acoustic wavelength, the ‘‘image’’ bubble has the same source strength of the real bubble so that a dipolar acoustic emission can be expected in spite of the deformation of the surface.