bluuebunny/crossref_metadata_2025 huggingface dataset:
An alternative wide-angle PE formalism and hybrid split-step/finite-difference solution algorithm
Authors: Yevick David , Thomson David J.
Published: October 2005

The split-step Fourier algorithm is widely used in underwater acoustics to solve parabolic equations (PEs) resulting from split-operator approximations to the one-way wave equation. Recently, Porter and Jensen [J. Acoust. Soc. Am. 94, 1510–1516 (1993)] reported on a seemingly benign propagation problem in which two wide-angle split-operator PEs performed poorly compared to the standard PE. For a source and a receiver located within a leaky surface duct, each of the wide-angle approximations gave rise to anomalously high values of transmission loss at ranges where the leakage energy is refracted back into the duct. Moreover, the two wide-angle PEs were more sensitive to the choice of reference wave number than the standard PE. This behavior is contrary to previous experience with such split-operator approximations in underwater acoustics applications. In this paper, we present a higher-order PE formulation that retains most of the efficiency of the split-step algorithm while improving the accuracy of the propagated solutions. In its simplest form, the new formulation modifies the standard split-step Fourier method by additionally solving a tridiagonal system of equations at each range step. The accuracy of our algorithm is illustrated with several numerical examples.
A complete energy-conservation correction for the elastic parabolic equation
Authors: Collins Michael D. , Siegmann William L.
Published: October 2005

The accuracy of a partial energy-conservation correction for the parabolic equation (PE) method is excellent for range-dependent ocean acoustics problems involving fluid sediments [M. D. Collins and E. K. Westwood, J. Acoust. Soc. Am. 89, 1068–1075 (1991)] but only fair for problems involving elastic sediments [M. D. Collins, J. Acoust. Soc. Am. 94, 975–982 (1993)]. A complete energy-conservation correction has been developed to improve accuracy for the elastic case. A range-dependent elastic waveguide is approximated by a sequence of range-independent regions. The complete energy-conservation correction involves matching energy flux across the vertical interfaces between the range-independent regions. The flux condition and the PE operator are used to obtain a nonlinear boundary value problem for the transmitted field that reduces to a linear boundary value problem for the fluid case. The flux integral is related to the inner product for the normal modes of the depth separated elastic wave equation. This fact is exploited to derive an efficient iteration formula for the elastic case.
A hybrid marching scheme for acoustic propagation in range-dependent ocean environments
Authors: Goh JooThiam , Schmidt Henrik
Published: October 2005

Previously a hybrid boundary element and full wave solution technique was presented for two-dimensional acoustic propagation in laterally inhomogeneous, vertically stratified medium [W. Seong and H. Schmidt, J. Acoust. Soc. Am. Suppl. 1 86, S54 (1989)]. The technique involved dividing the range-dependent ocean into range-independent sectors with the field within each sector expressed as a boundary integral over the vertical sector boundaries in terms of a set of unknown boundary displacements. The approach gave both the forward propagating field and the backscattered component and was global since it involved the solution of a matrix relation, the coefficients of which were obtained from matching boundary conditions along vertical cuts distributed throughout the entire computational domain. However, for ocean environments with continuously changing bathymetry, the construction of this matrix in a global manner presents a severe computational load, especially in the amount of core memory required. This deficiency motivated the search for a more efficient formulation of the hybrid scheme. By employing the single-scatter approximation, where the back-scattered component from the forward vertical boundary is neglected, a computationally efficient forward marching scheme can be derived. The algorithm is illustrated using various test cases involving range-dependent bathymetric features.
Caustic formation and timing changes in long-range propagation from ocean sound-speed relative curvature
Authors: Duda T. F. , Bowlin J. B.
Published: October 2005

A depth-dependent parameter of ocean sound channels has been found, using numerical simulation, to strongly influence long-range deep-ocean geometrical ray propagation. This parameter, representing sound-speed relative curvature, is the second vertical derivative of sound speed divided by the square of the first derivative. Ray and wave front timing and intensity can be influenced within realistic ocean sound channels by unpredictable wave front triplications and caustics associated with high values of the parameter at ray turning points. The relative curvature behaves as a random variable because of ocean finestructure, causing unpredictability. The parameter has a higher rms value near the sound-speed minimum for both an internal-wave model and actual data. This effect is a plausible explanation of poor multipath resolution and identifiability late in 250 Hz (center frequency) pulse trains transmitted 1000 km in the North Pacific.
Range variation in a numerical average underwater acoustic propagation method
Authors: Oba Roger M.
Published: October 2005

A computational method of numerically computing average complex pressure field has been developed using a normal mode analysis of the forward propagating solutions of the Helmholtz equation [R. M. Oba, Acoust. Lett. 16, 56–61 (1992)]. It computes the transmission loss and phase for the average solutions in the continuous wave case in piecewise range-independent cases as well as a case of rapid fluctuations in range. Suppose the uncertainty can be probabilistically modeled within range intervals as ensembles of sound speeds. If for each sound speed in the ensemble the variation within each interval is relatively small, and if the variations within the ensemble are uniform, then modification of the existing method allows computation of average volumetric forward and backward propagation. This technique will be useful in matching this computed average field at some final range to a deterministic solution starting from that range on. This technique can also be used in the limit of small intervals of uncertainty. [Work supported by Naval Research Laboratory and the Office of Naval Research.]
Downslope sound propagation measurements in a laboratory scale model of the Santa Lucia escarpment
Authors: Glegg Stewart A. L. , Coulson Robert , Le Vigne Anthony
Published: October 2005

A laboratory scale model of the Santa Lucia escarpment (34°–35 °N, 121°–122 °W) has been built at a scale of 1/10 000, and this paper will describe the design of the model and the downslope propagation measurements which have been made. The bathymetry of this region is typical of the continental shelf on the west coast of the United States, and the model includes a large area with a 2° slope in shallow water (400 m depth full scale) which extends into a region with a 14° slope at the deeper end where it joins the abyssal plain (4000 m depth full scale). The slope also includes three dimensional features such as a spur which extends out into the abyssal plain. The base of the model is made of concrete simulating a rock basement, and the purpose of the study is to investigate the bathymetric refraction of propagating sound waves over a region with a variable slope. The measurements of the sound propagation will be compared with calculations based on adiabatic mode theory. [Work supported by ONR.]
A normal mode model for multilayered acoustoelastic ocean environments based on an analytic reflection coefficient method
Authors: Westwood Evan K. , Tindle C. T. , Chapman N. R.
Published: October 2005

A normal mode model has been developed for underwater acoustic propagation in an ocean environment with multilayered elastic media below and/or above the water column. The compressional (p-) and shear (s-) wave sound speeds may either be constant or have a gradient (1/c2 linear with depth) in each layer. Mode eigenvalues are found by analytically computing the downward- and upward-looking plane-wave reflection coefficients R1 and R2 at a reference depth in the water and searching the complex k plane for points where R1R2=1. The complex k-plane search is greatly simplified by following the path along which ‖R1R2‖=1. The ‖R1R2‖=1 path connects all the modes, which are found as points on the path where the phase of R1R2 is a multiple of 2π. The direction of the path is found from the analytic derivative d(R1R2)/dk. Leaky modes are found, and multiple channels are handled. The eigenvalue finding algorithm appears to be robust and efficient. [Work supported by the ARL Independent Research and Development Program.] a)On leave from Appl. Res. Lab., The Univ. of Texas at Austin. b)On leave from Dept. of Phys., Private Bag 92019, Univ. of Auckland, Auckland, New Zealand.
Thresholds for detection of a change in the displacement, velocity, and acceleration of a synthesized sound-emitting source
Authors: Wang Wen , Lutfi Robert A.
Published: October 2005

An experiment was conducted to determine if listeners, when judging the velocity and acceleration of a sound-emitting source, merely infer these quantities from overall source displacement. The acoustic waveform of a high-velocity source was reconstructed over headphones using principles of sound radiation and propagation from the moving source, an interaural time difference method, and a diffraction model of a spherical head. An adaptive 2IFC procedure was used to measure thresholds for detection of a change in the displacement, velocity, and acceleration of the source passing directly in front of the listener in a linear trajectory from left to right. The source traveled at 50 m/s for 4 s, at a distance of 5 m from the center of the head, point of closest approach. Thresholds for a rightward displacement showed little variability across listeners, averaging about 7 m. Velocity and acceleration thresholds varied from one listener to the next, but were in some cases less than the minimum value that could be inferred given the 7-m threshold for displacement. The results are discussed in terms of cues for discrimination related to changes in intensity, interaural delay, and Doppler effect. [Research supported by NIDCD.]
Minimum audible movement angle thresholds at low velocities
Authors: Costantino Brian L. , Perrott David R. , Vincent Linda C.
Published: October 2005

Auditory resolution of a moving sound source was determined in a real motion paradigm. Minimum audible movement angle (MAMA) and minimum visible movement angle (MVMA) thresholds were obtained for five subjects in a two-alternative, forced-choice, three-up/one-down, adaptive procedure as a function of velocity. Velocities from 0.15°/s to 40°/s were utilized to give a wide range of motion thresholds. The low-ended velocities are of particular interest because in a previous study Saberi and Perrott showed an increase of approximately 4–5 deg for velocities below 1°/second indicating a U-shaped function in the detection of MAMA thresholds [Saberi and Perrott, J. Acoust. Soc. Am. 88, 2639–2644 (1990)]. The results of the current study suggest that MAMA thresholds flatten or decrease for velocities below 1°/s. However, it should be noted that in the current study, real motion was employed in contrast to that of simulated motion which utilized a 1.8-ms click train with 5.7-cm speakers in the Saberi and Perrott study. In addition, when MAMA and MVMA thresholds are compared at the lowest velocities, MAMA thresholds are at or below MVMA thresholds. Implications of results will be discussed. [Work supported by NSF (BNS-9025118).]
Neural network models of sound localization based on monaural information and based on binaural information
Authors: Gilkey Robert H. , Janko James A. , Anderson Timothy R.
Published: October 2005

Neural networks were trained with back propagation to localize ‘‘virtual’’ sounds that could originate from any of 24 azimuths (−165° to +180°) and 6 elevations (−36° to +54°). The sounds (clicks) were filtered with head related transfer functions and transformed into 22-point quarter-octave spectra. The networks were composed of 22 to 44 input nodes, followed by 50 hidden nodes and 30 output nodes (24 azimuth nodes and 6 elevation nodes). With the ‘‘binaural’’ configuration, the interaural delay and the interaural difference spectrum were provided as inputs to the network. With the ‘‘monaural’’ configuration, separate networks were trained with the left-ear and the right-ear spectra; a third, arbitrator, network learned to localize based on the output of these two monaural networks (i.e., the activation levels of their azimuth and elevation nodes). The monaural configuration did not allow for binaural interaction in the traditional sense. Both configurations achieved performance comparable to humans, suggesting that for these conditions either monaural cues or binaural cues are sufficient to explain human sound localization. [Work supported by NIH-DC-00786, AFOSR-91-0289, and AFOSR-Task 2313V3.]
Prospects for computational techniques in basic engineering flow noise evaluations
Authors: Blake William K.
Published: October 2005

Advances in computational fluid dynamics techniques for predicting viscous turbulent flows suggest the plausibility for making appropriate extensions to aero- and hydroacoustics. Furthermore, the developments in our understanding of mechanisms of flow-induced sound and vibration over the past 30 years have provided useful insights on how to make these extensions. Using some example problems, this paper will explore how large-scale computation might play useful roles in the modeling of flow sources. Emphasis will be on subsonic sources of flow and vibration. Examples of some numerical modeling requirements for computational solutions for these model cases will be derived from the fundamental theories for these sources and their applications. Suggestions will be made for certain measurement techniques which could provide both code validation and necessary coefficients that may be necessary for empirically based code elements.
Development of the acoustics laboratory at Digital Equipment Corporation
Authors: Lotz Robert
Published: October 2005

Development of the acoustics laboratory at Digital Equipment Corporation began in late 1970’s. The goal was laboratory capabilities to support noise control engineering as part of computer product engineering. Technical and business requirements were identified and validated. Labs in other industries were visited. The best space available in the plant to construct the lab was at ground level, requiring a tight-fit design, then excavation, blasting, and innovative materials handling during construction. Absorption adjustments in the 200 cubic meter reverberant room and draw-away tests in the hemianechoic room resulted in qualification to ISO 3742 and ISO 3745, respectively; and chamber background levels have been quite acceptable. Instrumentation, computer tools, and lab practices have been continually enhanced since then for component and product data acquisition, diagnostics, data basing, reporting, product design computation and consultation, and installation noise prediction. The lab continues to achieve its objectives today in this competitive and rapidly evolving industry.
Digital revolution in acoustic signal processing
Authors: Bellisario Afarin O.
Published: October 2005

Numerical modeling of the acoustic behavior of physical systems, using general purpose computers, has been used as a research tool since the 1970s. However, the availability of specially optimized processors for digital signal processing has enabled the replacement of many bulky electrical or mechanical components with compact IC circuits in various real-world and research applications. The programmability of these processors even allow for ‘‘shrink wrap’’ algorithms to be available for running on standard DSP platforms. This paper focuses on the general attributes of today’s generation of DSP’s, some of their real-world applications in replacing mechanical/electrical elements in noise/vibration control, music synthesis, the generation of music hall effects in homes, and echo cancellation in recording studios, and the influence of DSP’s on acoustics/vibration research in the future.
From gigabytes to understanding
Authors: Williams Earl G.
Published: October 2005

A near-field holography data set typically consists of about 0.5 Gbytes of acoustic pressure data. After a bit of routine processing, data sets totaling 2.0 Gbytes are soon in our computer’s hands. An example is given with a recent experiment of how a typical data set is handled, processed, and reprocessed to eventually provide insight into the physics of vibration, radiation, and/or scattering. The nuts and bolts of the processing and how they have become optimized through years of experience will be discussed, for example, how such large data sets are dealt with, what kind of computers and computer programs, what signal processing algorithms and libraries, how data is visualized, what kind of display devices, computer networks, etc. In the quest for physical understanding, of critical importance is time, that is, it is not only what can be done that counts, but how fast it can be done. This is the purpose of this paper.
From numbers to understanding; the visualization of structural acoustic phenomena
Authors: Schenck Harry A.
Published: October 2005

Improved computational models and experimental capabilities provide the opportunity for understanding structural acoustic phenomena in detail and scope never before realized. However, the raw products of both physical and computational experiments are very large data sets. Thus in this field as in many others, scientific visualization has a key role to play in turning numbers into easily understood results. This presentation will include concepts, principles, and practical suggestions for visualizing the information represented in large multidimensional data sets of structural acoustic field and surface variables. Examples of concepts that will be presented include the display of magnitude and phase simultaneously in three-dimensional fields, volumetric display of dispersion relations, and the use of animation to understand parametric sensitivity. Representation of phenomena in the time domain and through spatial transforms will be illustrated, along with traditional spectral methods. Principles to be discussed include the use and misuse of color, the appropriate use of display geometry, scaling, making comparisons, and the avoidance of artifacts. The visualization tools and workstation environment used in generating this presentation, together with lessons learned and practical considerations, will be discussed. [Work supported by the Office of Naval Research.]
Visualizations for studies of ultrasonic scattering and propagation in medical applications
Authors: Waag Robert C.
Published: October 2005

Visualizations are used to show calculations of ultrasonic measurement system weight in scattering experiments, focal plane time history and point spread functions produced by beamforming algorithms, and scattered pressure and internal stress in a kidney stone model during lithotripsy. Knowledge of measurement system characteristics, especially the resolution, allows the design of scattering experiments for the determination of intrinsic parameters such as average differential scattering cross section or the power spectrum of medium variations. Depictions of focal plane time history and point spread functions that result from focusing algorithms in which wave front distortion is compensated, permit evaluation of algorithm performance. Simultaneous display of scattered pressure and internal stress in a kidney stone model following illumination by a plane-wave pulse elucidates the temporal buildup and reverberant decay of stress in a kidney stone during lithotripsy and shows scattering characteristics that may provide information about kidney stone disintegration. In each of these applications, visualization provides insight not readily obtained in other ways from the basic data.
Visualization of waves on a PC
Authors: Ceperley Peter H.
Published: October 2005

The author’s experience in visualizing waves and wave related calculations using Mathcad, as well as several other general purpose mathematical packages, will be presented. [See, for example P. H. Ceperley, Comput. Phys. 7, 274 (1993)] Mathcad is particularly good as an aid in understanding physical phenomena, as opposed to producing pretty pictures to show to other people (which other packages are generally better at). Its strength is its very interactive human interface, which allows calculations, text, and graphs of various types to be displayed on the page at will. It allows one to quickly alter numbers, equations, and graphs to speed up detective processes. Also of interest for wave and field research is Mathcad’s ability to create three-dimensional color surface plots. Examples will be shown which illustrate its interactive use in the display and fitting of experimental data, as well as in the display of theoretical results at multiple stages in a calculation. More limited results using the more powerful, but less interactive tools of Maple and Mathematica will also be shown.
Phase-space footprints of truncated periodicity: Time domain wave-oriented data processing
Authors: Carin L. , Felsen L. B. , Hsu T.-T. , Kralj D.
Published: October 2005

A companion paper [L. B. Felsen etal., paper 3aSAb8] has dealt with phase space processing techniques that extract frequency domain (FD) Bragg-modulated phenomenologies from scattering data pertaining to truncated periodic arrays of filaments. Displayed in subdomains of the (x,z,kx,kz)ω phase space accessed by windowed and global transforms, the footprints of the dispersive truncated Bragg modes are of special interest. These processing techniques are now extended and applied to short-pulse time domain (TD) synthetic scattering data from the filament arrays. The phase space is enlarged to (x,z,t,kx,kz,ω), with a correspondingly enriched choice of subdomain processing options and visualizations. Again, special attention is given to the truncated TD Bragg modes whose behavior is governed by an instantaneous dispersion relation extracted by the processing. Specific plots include the subdomains (t,ω), (t,px), and (ω,kx), which complement the FD results (px≡kx/ω). [Work supported by AFOSR.]
Acoustic scattering from a cylindrical shell insonified with a focused beam
Authors: Ezzaïdi Mohamed , Moudden Ali , Décultot Dominique , Maze Gérard
Published: October 2005

The backscattering acoustic spectra of a cylinder show resonances due to the circumferential waves. Other resonance types are seen on the spectra, they are due to the thickness e of the shell. When this thickness is thin, the circumferential waves can be compared with the Lamb waves. The phase velocity of these waves depends on the frequency. The Lamb waves (Ai,Si, i≳1) have, in normal incidence, a cutoff frequency. In this case, only the longitudinal wave generates some Lamb modes. In oblique incidence, the transversal wave also generates Lamb modes. As the thickness is proportional to an integer of half-wavelengths, it is easy to measure the thickness of a flat plate: The thickness resonances are clearly isolated. On the contrary, for a cylindrical shell, the thickness resonances are mixed with the circumferential resonances. It is difficult to determine the exact thickness resonance frequencies. To solve this problem, focused beam is used. The focal spot is smaller than the cylinder diameter. This method has been applied to shells with small diameter (10, 5 mm,...) and with thin thickness (0.5, 0.38 mm,...). The results are obtained with a good precision. [Work supported by the French government.]
Scattering of incoherently processed FM signals from arrays of rigid spheres
Authors: Baus Theresa A. , Radlinski Ronald P.
Published: October 2005

The peak target strength for arrays of rigid spherical scatterers insonified by FM signals which are incoherently processed is investigated for frequency, bandwidth, and spatial dependence. Examples of backscattering are given with two scatterers and line arrays of scatterers. Multiple interactions are ignored. For a fixed pulse length, increasing the bandwidth of an incoherently processed FM signal results in greater peak target strength due to increased opportunities for constructive interference from the scatterers. In the Rayleigh region, the ka dependence of an individual scatterer further enhances peak target strength as the bandwidth increases. Increased bandwidth also smooths the aspect dependence of the scattering pattern. Decreasing the spacing between the spheres decreases the number of ripples in the scattering patterns.
Time-frequency analysis of model based pulse scattering from a rough fluid-elastic interface
Authors: Bishop Garner C. , Huynh Quyen
Published: October 2005

A null field T-matrix formalism for plane-wave scattering from a doubly infinite fluid–solid interface with doubly periodic surface roughness [Bishop and Smith, J. Acoust. Soc. Am. 94, 1560–1583 (1993)] is extended to include scattering of broadband acoustic pulses. The scattered pressure field is expressed in terms of a Fourier integral over the spectra of the incident pulse and the T matrix. Time series are calculated for the scatter of a Gaussian amplitude-modulated linear frequency-modulated pulse from a fluid–elastic interface with sinusoidal and triangular roughness. The time series are analyzed using short time Fourier, Wigner, wavelet packet, and local cosine T-F transforms. The results are displayed graphically in a T-F plot in which the T-F plane is covered by rectangular boxes of dimensions Δt×Δω. These results are used to evaluate the ability of each of the transforms to address inverse scattering and identification problems. It is shown that the T-F analysis provided by the wavelet packet may be used to distinguish between the scatter from surfaces of different roughness and is superior to the T-F analysis provided by both the short time Fourier and Wigner transforms.
High resolution images in acoustic microscopy
Authors: Golan G. , Kenyon A. J. , Pitt C. W. , Griffel G.
Published: October 2005

A novel scanning acoustic microscopy technique for achieving high resolution acoustic images by employing thermal effects and image subtraction has been studied and demonstrated. Experiments were performed on a perspex block patterned with a machined grid on the reverse surface, and on a buried channel in similar material. It was found that using the image subtraction technique, short periods of sample heating can lead to a stronger pattern selectivity, because of the strong temperature dependency of the elastic parameters of the polymer. In previous SAM techniques improvement in resolution has been achieved through the use of special liquids as acoustic coupling media between the acoustic lens and the sample. The reported technique retains water as the coupling medium and the acoustic impedance matching is performed by varying the elastic parameters of the sample itself through direct heating. The temperature increase in the sample decreases the velocity of propagation of acoustic waves in the solid, and brings the acoustic impedance close to that of water. A theoretical model, including expressions for the acoustic aberrations, depth dependence, and acoustic impedance matching, has been derived. Examples of the results obtained are presented.
Investigation of a narrow-band signal structure in the vortex ring noise
Authors: Kopiev V. F. , Zaitsev M. Yu.
Published: October 2005

Experimental investigations of the vortex ring noise carried out recently in an anechoic chamber showed the presence of a narrow peak in averaged spectra near the frequency f=1200 Hz the width of which was 200 Hz. Separate realizations recorded previously by tape recorder were used for averaging. The main questions are: What is the mechanism of narrow peak appearance? What is the reason for such a peak width? In the first part of this work a careful comparison between the experimental data and the simplest theoretical model is conducted. Vortex ring modes that could be responsible for the radiation are obtained. However, the investigation of separate time realization considered in the second part of this work showed that the process of sound radiation by vortex ring is of a more complex nature than it was initially expected. The registered radiation appeared to have the form of short time flashing oscillations (wave trains) not connected in phase with each other. Observed peculiarities of the acoustic signal are inherent to the narrow-band random process, but not the mixture of the harmonic component and broadband noise as it seemed earlier, i.e., the mechanism of vortex ring sound generation is probably connected with an interaction between vortex core modes and the turbulence near the core.
Diagnostics of noise broadband acoustical sources from near-field measurements
Authors: Sidorovskaia N. , Turchin V.
Published: October 2005

Algorithms of reconstruction of moving random source characteristics by processing the data obtained from near-field measurements are presented. The first algorithm, the so-called algorithm of high-frequency approximation (HFA algorithm) deduced from an asymptotic solution of an integral equation (analogy of Maue equation) allows one to construct a near-field measurement technique for transformation to far-field characteristics. The processing procedure does not depend on the source model and needs only main trajectory parameters (motion angle, initial point). The second algorithm is founded on statistic estimation of source characteristics on the basis of the maximum likelihood method [algorithm of maximum likelihood estimation with regularization (MLER algorithm)]. In contrast with the HFA algorithm, the MLER algorithm allows one to solve the inverse problem of reconstruction of the elementary source distribution on the radiator. Unlike earlier presented methods, the optimum procedure of the pseudoinversion of the matrix representing the mapping of the source into near-field samples is obtained. This procedure minimizes the reconstruction total error. Both algorithms include the compensation of external spatially correlated noise. Representative results of the extensive numerical simulations and some natural experiments for various radiators are included.
A maximum-likelihood method for measuring binaural lateralization
Authors: Saberi Kourosh , Green David M.
Published: October 2005

The maximum-likelihood method is an adaptive psychophysical method. Previous work has outlined this procedure for signal detection tasks using either forced-choice [Green, J. Acoust. Soc. Am. 87, 2662–2674 (1990)], or Yes–No procedures [Green, J. Acoust. Soc. Am. 93, 2096–2105 (1993)]. The current study extends the maximum-likelihood procedure to lateralizing a 500-Hz tone and a sinusoidally amplitude modulated (SAM) signal with a 50-Hz modulation rate and a 3.5-kHz carrier. It is shown that by the sixth trial of the experimental run, an unbiased estimate of threshold may be obtained. Simulations show this procedure to produce less variable and less biased threshold estimates than the Up–Down methods. Data from human observers and from simulations show a theoretically ideal reduction in variability of threshold estimates with increasing number of trials. [Work supported by NIH and AFOSR.]
Lateralization of a frequency-modulated 500-Hz tone
Authors: Hukin R. W. , Darwin C. J.
Published: October 2005

Previous experiments [C. Trahiotis and R. M. Stern, J. Acoust. Soc. Am. 86, 1285–1293 (1989)] have shown changes in the lateralization of a bandlimited noise centered at 500 Hz with an ITD of 1.5 ms as its bandwidth is increased. The phenomenon is thought to be due to across-frequency integration of ITD information. The present experiment extends this phenomenon to an FM 500-Hz tone. With no FM the 500-Hz tone is heard on the lagging ear, but tones with appropriate rates and depths of FM are heard on the leading ear. Subjects indicated which of two FM tones (ITD=±1.5 ms) was heard more on the left. At 40 Hz FM, subjective lateralization changed at an FM depth of about 30%. At slower rates of FM, greater FM depth was needed to change subjective lateralization, so that at 5 Hz, about 45% modulation was needed. These results may be due to listeners integrating ITD information across both frequency and time. [Work supported by MRC.]
Auditory localization in noise. I. The effects of signal-to-noise ratio
Authors: Good Michael D. , Gilkey Robert H.
Published: October 2005

While several studies have examined localization of a single sound in the quiet, there have been few studies of localization in the presence of noise. In the present investigation, localization of a broadband click-train signal, masked by a broadband noise (located directly in front of the subject), was measured in the free-field as a function of signal-to-noise ratio. Subjects judged the direction of the signal, which could emanate from any of 239 spatial directions (surrounding the subject in azimuth and ranging from −45° to 90° in elevation), by pointing at a 20-cm-diam spherical model of auditory space. The ability to determine the direction of the signal decreased nearly monotonically as the signal-to-noise ratio was reduced. When the data were described in a ‘‘three-pole’’ coordinate system, it was evident that the accuracy of localization judgments, relative to the frontal plane (the ‘‘front/back’’ dimension), was degraded at much higher signal-to-noise ratios than was the accuracy relative to either the median plane (the ‘‘left/right’’ dimension) or the horizontal plane (the ‘‘up/down’’ dimension). Differences in the salience of cues for localization will be considered. [Work supported by AFOSR-91-0289.]
Auditory guided search: Free-field and synthesized
Authors: Perrott David B. , Cisneros John , Ericson Mark , McKinley Rich
Published: October 2005

Target discrimination tasks effectively test the ability to locate and interpret a given target. Adding sound, spatially located with the target, was hypothesized to decrease time necessary to localize and discriminate that target. This spatially correlated sound (SCS) was compared to spatially uncorrelated sound (SUS) at 260 points surrounding the subjects. Average reaction times of a two-alternative forced-choice discrimination task were better by over 150 ms in the frontal field (FE, +/−40 deg azimuth × +/−40 deg elevation) for SCS compared to SUS. Performance declined as targets moved into the periphery for both conditions, at an accelerated rate for SUS. When in the rear hemifield (RHF), SCS performance fell to almost two times what it was in the FF (780 vs 1325 ms), while it was almost three and one half times for SUS (934 vs 3310 ms). Spatially correlated synthesized sound (SSS) presentation over headphones proved to be about as accurate as SCS. Performance decline of FF vs RHF was about 2.5 to 1. These values show, especially in the RHF, ALCS’s provide almost as much information about target location as SCS cues and provide a subject with considerably more resources to succeed in a target acquisition task compared to SUS. [Work supported by NSF and AFOSR.]
Binaural and monaural influences on ‘‘free-field’’ masking for real and virtual sound sources
Authors: Ball Jennifer M. , Gilkey Robert H. , Good Michael D.
Published: October 2005

Gilkey and Good [J. Acoust. Soc. Am. 92, 2334 (1992)] found that the detectability of a click-train signal, presented in the free-field, increased dramatically when the signal was spatially separated from a noise masker, in either azimuth or elevation. In separate conditions, stimuli were filtered into low-, mid-, and high-frequency regions. The pattern of results suggested that the increases in detectability might have been based on monaural, rather than binaural, processing. Here, this study was partially replicated using ‘‘virtual’’ sound sources. The stimuli were filtered through head-related transfer functions and presented through headphones. Binaural, left-ear monaural, and right-ear monaural stimulus presentations were compared. Thresholds in the binaural condition were in good agreement with the free-field data. Although there was evidence under some conditions of a small binaural advantage at low frequencies, in general, binaural performance was no better than monaural performance with the ‘‘better ear.’’ Overall, the results suggest that the subjects of Gilkey and Good could have performed as well as they did based on monaural processing alone. [Work supported by NIH-DC-00786 and AFOSR-91-0289.]
Response measures used in free-field localization
Authors: Ericson Mark A. , D’Angelo William , McKinley Richard
Published: October 2005

Two response measures, verbal spherical angles, and manual pointing, were compared in a free-field localization task. Verbal spherical angles refers to speaking the stimulus’ azimuth and elevation angles with respect to the center of one’s head. Manual pointing refers to pointing the stylus tip of an electromagnetic position digitizer onto a 12-in.-diam sphere and recording the azimuth and elevation of the tip with respect to the center of the sphere. Continuous visual and aural stimuli were presented from 272 directions around each of the five subjects, whose head motions were unrestrained. The accuracy and reaction times of the subjects were measured and compared. Pointing was more accurate than the verbal response, 10 and 22 deg, respectively. Pointing was about three times faster than the verbal response. The manual pointing technique is being used in other experiments to compare free-field and headphone localization performance.
The continuity of auditory apparent motion for amplitude-modulated broadband noise
Authors: Kinoshita Ikuichiro , Aoki Shigeaki
Published: October 2005

The thresholds of the switching period Tth’s that produced the continuous or broken auditory apparent motion were measured, when the stimuli were successively presented by using three loudspeakers arranged in the horizontal plane. In the current experiment, the Tth’s were investigated for amplitude-modulated broadband noise as a function of the modulation frequency fm (ranging between 4–64 Hz) and the modulation index m (ranging between 0.0–1.0). In agreement with the previous study that employed signals including bandpass noise, speech, etc. as stimuli [I. Kinoshita and S. Aoki, Tech. Rep. Hear. Acoust. Soc. Jpn. H93-2, 1–8 (1993)], the Tth’s were diversified and generally larger than the one for an unmodulated (m=0.0) broadband noise; a monotonical increase in the Tth was indicated with the modulation index (e.g., the Tth ranged between 72–290 ms at a modulation frequency of 16 Hz), while the Tth was reduced as the modulation frequency increased (e.g., the Tth decreased from 360 to 135 ms at a modulation index of 1.0). These results indicate that the modulation causes the longer Tth’s and will be discussed in terms of the temporal variation of stimuli.
Analysis, modification, and synthesis of dolphin signature whistles
Authors: Morgenbesser Hugh B. , Buck John , Tyack Peter
Published: October 2005

A digital system was developed for analysis, modification, and synthesis of dolphin signature whistles. These whistles are modeled as frequency modulated signals with harmonically related frequency components. This system provides control over specific acoustic features which may be used in playback experiments. The analysis subsystem breaks the signal up into short overlapping blocks of time and estimates the fundamental frequency for each block. The system outputs frequencies and energies for the first three harmonics. The modification subsystem facilitates the following user-specified alterations to these features: Exclusion of one or more harmonic, variation of the energy envelopes, shifts and scalings of the frequency contours, and time dilation. The synthesis subsystem reconstructs each harmonic separately and then sums them to create the synthetic signal. Three different approaches for making smooth transitions between consecutive time blocks are compared.
Excursion requirements for low-frequency transducers when operated in electrically equalized loudspeaker systems
Authors: Rocha Jeffery A.
Published: October 2005

Electrically equalized loudspeaker systems are found in many professional applications where extended low-frequency response is desired. The maximum amount of equalization which can be applied is usually limited by the driver’s linear excursion. The excursion required is governed by the acoustical loading of the loudspeaker cabinet. It is possible to define the excursion based upon the acoustical parameters of the transducer-box equivalent circuit, using SPICE, for any given set of low-frequency response requirements. It is then immediately seen by the transducer designer whether these linear excursions are possible. Experimental measurements verify the results.
Bass piano wire inharmonicity
Authors: Zamir Lee , Allen Kathleen J.
Published: October 2005

Due to certain nonideal characteristics, namely stiffness, the overtones produced by piano wire deviate from a perfect harmonic series. The mechanism by which this occurs is well understood for the plain wire that is used in the treble region of a piano. The problem becomes much more complex, however, when considering the wound wire that is used in the bass section. Previous methods of analysis have had varying levels of agreement with my experimental data but they all lose correlation when applied to the lowest five to ten notes. [H. Fletcher, J. Acoust. Soc. Am. 36, 203–209 (1964)] There are certain assumptions or simplifications in these methods that breakdown for these large wires. Through finite element analysis techniques and further experimental investigation, it is hoped that an understanding of how the physical properties of bass piano wire affect its inharmonicity. This research may lead to a new wire design that may resolve tuning and tonal quality problems present in pianos today.
Bits of perspective on the advent of digital music
Authors: Hawley Michael
Published: October 2005

There has always been an interplay between musical tools and musical forms—a succession of instruments, from serpent horns to saxophones, and of styles, from chant to grand opera. All of this tends to echo aspects of the ambient technology infrastructure, for example, the rise of large buildings coincides to some extent with the breakup of Gregorian chant, and the development of steam engines and ironworks made possible the modern concert grand (and the globetrotting piano virtuoso). In much the same vein, broadcast and recorded media are associated with the wane of live concerts. This talk will discuss history and trends of music and technology. In particular, our culture is in the throes of a singular and massive transition from largely analog to largely digital media. The mass-deployment of MIDI and other computer-based music forms, and the diffusion of increasingly intelligent instruments, a symptom of that trend, has begun a drastic change that will lead...where?
Musical instruments, models, and machines
Authors: Gershenfeld Neil
Published: October 2005

A musical instrument is an analog computer. It integrates equations of motion based on applied boundary conditions and generates acoustic and tactile outputs. Understanding why the equations of motion is liked is a very hard problem, fabricating traditional instruments that duplicate desired equations of motion is a hard problem, but using new technology to emulate and generalize them is an increasingly easy problem. The remarkable confluence of sensing, modeling, computing, and communications that is blurring the distinction between using wood and strings or silicon to solve the same equations will be discussed. This includes ideas such as the active remote interrogation of passive structures for sensing, state-space reconstruction for building observational nonlinear models, cellular automata methods for efficient first-principles solution of continuum equations, and tools for real-time computing and communications on general-purpose platforms.
Hearing levels of a Swiss nonindustrial noise-exposed population in comparison with reference values in ISO 1999
Authors: Matéfi Laslo , Hohmann Beat
Published: October 2005

In Switzerland, the program for the prevention of occupational hearing loss is carried out by the Swiss National Accident Insurance Organization (SUVA). About 50,000 people are examined each year by means of six mobile examination units (‘‘audiomobiles’’). Most of these people are exposed to potentially hazardous noise at their working places. In recent years, several groups of nonindustrial noise-exposed people (NINEP) have been included also with the aim to create a control group data base. In order to get representative NINEP data, different criterion have to be considered and will be discussed. Social, national, and economic differences are probably as important as differences in noise exposure at work. The hearing of workers in different occupations is compared—distinct by age and sex—to the NINEP group. NINEP subgroups screened in different ways are compared and the results are discussed. All the hearing loss data are compared to the reference values in Appendix B of ISO 1999-1990. Some general considerations concerning the problem of arriving at comparable NINEP data for the evaluation of the effectiveness of hearing conservation programs will also be presented.
New analyses of hearing threshold data for nonindustrial noise-exposed populations (NINEPs) by age, sex, and race
Authors: Royster Julia D. , Royster Larry H.
Published: October 2005

Percentile distributions of hearing levels by age have been developed for male and female subjects in the black and white NINEPs which were originally published as mean data [L. H. Royster and W. G. Thomas, Am. Ind. Hyg. Assoc. J. 40, 504–511 (1979); L. H. Royster etal., Am. Ind. Hyg. Assoc. J. 41, 113–119 (1980)]. These reference populations were screened to exclude persons with more than 2 weeks of occupational noise exposure, but they include all other age-effect influences: Nonoccupational noise exposure, pathology, and presbycusis. The results for white subjects are similar to the U.S. Public Health Service data in Annex B of international standard ISO 1999:1990 (E). However, black subjects exhibit much better hearing, similar to the screened presbycusis data in Annex A of ISO 1999:1990 (E). The difference in threshold distributions will be explored for the better ear versus the binaural average, and for subgroups with different characteristics.
Software system for quantitative, observer-based analyses of aircraft noise
Authors: Reddingius N. H. , Sneddon M. D. , Smyth J. S. , Fidell S.
Published: October 2005

The National Park Service’s Overflight Decision Support System (NODSS) now provides a functional capability for estimating airspace volumes above points and areas on the ground that should be avoided if aircraft are to remain unnoticed. Calculations of these volumes are sensitive to aircraft noise emissions, background noise levels at observers’ positions, terrain shielding, atmospheric absorption, and geometric spreading. NODSS provides a complete and consistent framework for performing acoustic detectability-based calculations as well as more conventional aircraft emission-based analyses, and produces results that can be displayed in easily understood graphic form. The initial use intended for NODSS is to perform engineering analyses of airspace restrictions associated with restoration of natural quiet in Grand Canyon National Park, pursuant to Public Law 100-91. [Effort supported in part by U.S. National Park Service.]