bluuebunny/crossref_metadata_2025 huggingface dataset:
Wave equation solution by step-marching in space.
Authors: Pai D. M.
Published: October 2005

This paper presents a step-marching method for wave equation solution in inhomogeneous media. The solution is the transmission signal. The method expands the solution in terms of plane waves, which, during propagation, are mixed by the medium inhomogeneity. Like the parabolic equation method, the present method is step-marching in space and thus efficient. From comparisons to analytic solutions, the present method proves to be more accurate than the parabolic equation method.
An introduction to the biology, chemistry, and physics of woodwind reed material.
Authors: Casadonte Donald J.
Published: October 2005

In this paper two questions are addressed: (1) What are the material science properties of woodwind reed material (Arundodonax) in musical settings (i.e., finished reeds), and (2) what are the effects of constant use of single reeds on their material and acoustic properties? Using scanning electron microscopy (SEM), the detailed biological ultrastructure of the material has been isolated and investigated. A range of sizes for cell wall dimensions has been studied as an indicator of ‘‘unstability’’ (i.e., reed quality). The three-dimensional cutting pattern in the manufacture of the finished reed produces a characteristic profile for a given reed, which has also been imaged by optical densitometry. These densitometry measurements are utilized to reproduce accurate three-dimensional surface maps of the reed density distribution, which are also correlated with reed quality. These two measurement techniques allow clarinet reed modeling using a finite element program (ANSYS) for study of the calculated forced motion, mode shapes, and resonance frequency against measured values (as, for example, in a video that has been generated of stroboscopically slowed motion of the clarinet reed in the mouth of a player by use of special fiber optic imaging). Three breakdown pathways of the reed material are discussed: (1) biological (bacterial infestation), (2) chemical (hydrolysis or lysing of the reed material over time), and (3) crystallite transitions in the microfibril structure.
An anharmonic oscillator model for a Helmholtz resonator excited by an air jet.
Authors: Cottingham James P. , Wiersma Jennifer L.
Published: October 2005

An earlier study by Khosropour and Millet investigated the internal spectra of several air jet excited Helmholtz resonators. In particular, an exploration was made of the dependence of frequency and amplitude of the Helmholtz mode on jet speed and jet width [R. Khosropour and P. Millet, J. Acoust Soc. Am. 88, 1211–1221 (1990)]. Additional measurements made on a variety of resonators show general agreement with this earlier study, but further study has been made of the nonlinear dependence on jet speed and jet width of the amplitude of the Helmholtz mode. A theoretical model has been developed in which the jet-resonator system is modeled as a damped harmonic oscillator driven by a periodic driving force. Some of the observed changes in amplitude with increasing jet speed are interpreted in the model as transitions from periodic to chaotic motion.
Acoustical and electronic instruments in jazz: An educational perspective.
Authors: Mash David S.
Published: October 2005

The personal desktop music production workstation is now a reality. New hardware and software configurations for the personal computer have brought powerful tools for sound design, digital recording and editing, mixing and processing, MIDI sequencing, and real-time music performance. The ability to combine live acoustical performance into the MIDI sequencing environment with a single intuitive graphical interface brings new challenges for both the musician and music education. What are the principal skills the musician needs to create and produce music in the 1990s? How do we develop these skills, and how will technology affect the teaching and learning processes? This lecture/demonstration will present new applications for the desktop music production workstation and discuss their implications for both the professional musician and the music educator. New approaches to teaching about sound design, orchestration, production, and recording techniques will be explored using the latest developments in MIDI and audio hardware and software packages for the personal computer. This presentation will focus on programs initiated at Berklee over the past 5 years and those currently in the planning process.
Acoustical and electronic instruments in jazz: A keyboard performance perspective.
Authors: Thurmond Dennis
Published: October 2005

A problem that has faced keyboard players since the advent of the electronic keyboard in the Jazz–Fusion–Rock idiom is the transfer of skills from acoustic keyboards to electronic keyboards. The pianists who moved over to electronic instruments found themselves with many new problems to solve, such as no left or middle pedal, pitch bend wheels, mod wheels, and multilayered keyboards. These problems were easily solved with a little time spent orienting to the new keyboard, but there was another problem that was not as easily solved−chord voicing on piano versus chord voicing on synthesizers. What would sound good on the piano would not necessarily transfer with similar quality to the electronic keyboard. This paper addresses the issue of different keyboard voicing techniques, as solved by Chick Corea and Herbie Hancock as their performing careers went from acoustic to electronic. Examples will be shown illustrating comping solutions in both the acoustic and electronic domain.
The nature of localized elastic mode on quasi-2D structure.
Authors: Zhou Minyao , Sheng Ping
Published: October 2005

Recent experimental measurements by L. Ye and G. Cody on the elastic modes of a quasi-2D structure, consisting of a 1-in. steel plate with periodically placed lucite blocks, have yielded the surprising observation of localized elastic modes. In order to see the nature of these localized modes, the eigenmodes of the structure are calculated by using the finite element analysis. The results demonstrate that there is a remarkable quasi gap in the bending wave to the lucite shear modes. For the perfect periodic structure, the mode in the gap have extremely low density of states. As a result, small random perturbation causes these elastic modes to be localized.
Short pulse sound-wave scattering from an infinite periodic array of rigid flat strips in a fluid.
Authors: Felsen L. B. , Borderies P. , Carin L.
Published: October 2005

Many submerged structures of interest contain periodically spaced supports or other periodic structural features. When insonified with waves at various frequencies, such periodicities are known to give rise to dispersive phenomena with well-defined pass and stop bands. However, for short pulse (SP) sound waves with their very wide frequency spectra, the scattered sound field exhibits features that are well resolved in the time domain, therefore not well parametrized by superposition of the infinite harmonic wave trains in the frequency domain. To understand the evolution of the SP induced transient field, it is desirable to operate directly in the time domain (TD). For clarification of relevant issues, a systematic investigation has been undertaken for the simple model of an infinite periodic array of flat rigid strips. A reference solution for SP plane-wave scattering has been constructed by an (integral equation)–(method of moments) algorithm implemented numerically via Floquet modes in the frequency domain. The results furnished by this numerical experiment are then parametrized in terms of the TD observables in the data−well-resolved short pulses, interspersed with trailing oscillations, etc. This TD observable-based parametrization (OBP) permits the monitoring of the buildup of the global dispersive spectra from the initially well-resolved multiple scatterings between adjacent strip elements. Wave-oriented signal processing, in the configuration-wave number phase space, systematizes the OBP implementation and explanation of the results. [Work supported by ONR.]
Scattering from shells with internal structural loading. Part I. General theory.
Authors: Guo Yue-Ping
Published: October 2005

This paper starts a series of systematic analytical studies on sound scattering from cylindrical shells with various forms of internal structural loading. The objectives of these studies are to understand the physical phenomena involved in the scattering process and to identify dominant mechanisms that control the acoustical behavior of internally loaded shells in the mid-frequency region. It is hoped that these studies will also help to establish physical principles and modeling techniques which enable orderly treatment of complicated internal structures in sound scattering problems. In the present paper, a general formulation is given for the sound-shell-internals-coupled problem with exact solutions that can easily accommodate different forms of internal loading. Solutions are explicitly expressed in terms of two coupling matrices, respectively, representing the geometric and dynamic effects of internal loading. The first matrix is completely determined by the loci at which the internal structures are attached to the shell, independently of the details of the internal system. The dynamic coupling between the shell and its internals is entirely accounted for by the second matrix which relates the forces exerted on the shell by the internal structures to the shell displacements at the attachment points. This matrix can be found by considering the dynamics of the internal structures alone and involves only properties of the internal system. Examples are given for some forms of internal loading, which clearly illustrate the methodology employed and are of practical interest. Detailed analyses and discussions of the derived results will be given in a companion paper. (Y.-P. Guo, following abstract.) [Work supported by ONR.]
Hybrid ray-mode parametrization of acoustic scattering from submerged internally loaded thin cylindrical elastic shells: Direct construction from continuous wave spectra.
Authors: Guo Yue-Ping , Felsen Leopold B.
Published: October 2005

In a companion paper [L. B. Felsen and Y. P. Guo, preceding paper], an observable-based hybrid ray-mode parametrization (OBP) for a broad class of axially-invariant two-dimensional acoustic plane wave scattering from submerged internally loaded thin cylindrical elastic shells has been developed by reformulation of solutions constructed in terms of angular harmonic resonant modes. Now, the solution is developed directly in terms of the continuous wave-number spectra that lead naturally to the hybrid ray-mode OBP, thereby incorporating the wave phenomenon from the beginning. The previous resonant-mode and present traveling-wave formulations are compared at various stages. These comparisons are aided by the Poisson sum formula which establishes the analytic connection between the resonant-mode series and the wave-number continuum, showing the exact equivalence of the two approaches. The paper concludes with observations concerning the numerical implementation of these alternative algorithms, especially in regard to characterizing the induced forces on the shell that account for the interior loads. [Work supported by ONR.]
The acoustic ‘‘background’’ for scattering from heavily fluid-loaded thin shells.
Authors: Norris Andrew
Published: October 2005

Acoustic scattering from thin shells can and does display strong resonances below the coincidence frequency, and it is well known that these resonances are associated with supersonic compressional waves. There has been quite a bit of discussion in the literature about the ‘‘background’’ associated with these resonances, and it is known that they are not adequately explained by either a rigid or a pressure release boundary condition. Using asymptotic arguments based upon the small parameter (h/a), it is shown that the background is due to a purely flexural response. When the background response is subtracted from the full field the result is essentially zero except near the compressional resonance frequencies. In this sense, the compressional wave response is lightly fluid loaded. One implication is that the full response can be separated into the background, heavily fluid-loaded purely flexural response, plus an additional lightly fluid-loaded response caused by compressional and shear waves. These ideas will be demonstrated for cylindrical and spherical thin shells.
Axisymmetric impulsive excitation of fluid-filled elastic cylindrical shells−Propagation paths.
Authors: Brévart Bertrand J. , Fuller Chris R.
Published: October 2005

The transient response of an infinite fluid-filled cylinder to an axisymmetric radial impact applied on the shell wall was derived, using the Donnell–Mushtari shell equations coupled with interior acoustic field and computing a double Fourier integration in the wave number and the frequency domain. The radial displacement amplitude of the shell as well as the instantaneous energy in both media were evaluated at several distances from the input force and for different structural damping coefficients and shell thicknesses. The contribution of each generated wave, either fluid-like or structural-like waves, was extracted from the total response in order to determine the propagation paths of the impulse into the coupled system. The results reveal that most of the response of the shell is due to low-frequency energy carried by an ‘‘acoustically slow’’ wave, very close to a fluid wave in a rigid-walled tube. This study is a further advance in the understanding of noise propagation along practical piping systems as, for example, the impulsive force can model a constraint and disturbance applied to the pipe wall by a hanger. [Work supported by ONR/DARPA.]
Supralaryngeal patterns of declination: Labial and velar kinematics.
Authors: Krakow R. A. , Bell-Berti F. , Wang Q.
Published: October 2005

Recently, Bateson and Fowler [E. V.-Bateson and C. A. Fowler, J. Acoust. Soc. Am. Suppl. 1 84, S128 (1988)] and Bell-Berti and Krakow [F. Bell-Berti and R. A. Krakow, J. Acoust. Soc. Am. 89, 1916 (A) (1991)] have shown that declination effects are not restricted to laryngeal-respiratory behavior. Bateson and Fowler observed a decline in the extent, peak velocity, and duration of jaw opening and closing gestures, and, in this paper, a decline in peak velar height over the course of a sentence was observed. This study reports evidence of declination in another articulator, the lower lip. Consistent with previous findings for the velum, preliminary results show a decline in the height of the lip when comparing the first to the last peak in each sentence. Like the velum, the lower lip shows a decline within bisyllabic words, and across adjacent and nonadjacent bisyllabic words. Here, the co- occurring patterns of declination in these two articulatory subsystems are examined and how these patterns interact with stress is shown. [Work supported by NIDCD Grant No. DC-00121.]
A comparative intonational analysis of spontaneous and read speech.
Authors: Ayers Gayle M.
Published: October 2005

Many studies have investigated acoustic differences between spontaneously produced speech and prepared read speech and have found that no single aspect of the speech signal (average duration, average F0 frequency, frequency variation) conveys the spontaneity. Remez etal. suggested that spontaneity is expressed at the sentence level [J. Acoust. Soc. Am. 89, 2011–2012 (A) (1991)]. This study looks at the function and meaning of pitch range variations and intonational structure in spontaneous versus read speech. Narrative portions of a spontaneous conversation are compared with fluent readings by the same speakers of a text prepared from the conversation. Although the sentences in the two versions match lexically and syntactically, they do not always match in pragmatic and thematic function within the larger text. Symbolic intonation transcriptions were made and compared. While the two versions agree quite closely at the level of intonational phrasing, the spontaneous version has many more intermediate phrases and uses a wider variety of intonation contours. Also, while the overall pitch range is similar in the two versions, locally the use of expanded pitch range corresponds in only about half of the phrases, with the remaining half occurring in different locations. Pitch range variations mark the organizational structure of discourse, by signaling topic changes and focus prominence, and setting off parentheticals. Thus it might be more accurate to say that spontaneity is expressed at the discourse level. [Work supported by an NSF Graduate Fellowship.]
Intraspeaker/interspeaker variation in selected voice parameters and speaker identification.
Authors: Yonovitz A. , Smith Angela
Published: October 2005

Female speakers provided voice exemplars in both a speaking task (reading) and a sustained phonation into a reflectionless tube. Nine serial trials, spaced three days apart were obtained from each subject. The fundamental frequency and coefficient of variation were obtained from the speaking task (32.768 s). Jitter, shimmer, and spectral function were obtained from the sustained phonation and was based on the analysis of 100 consecutive cycles of the glottal wave. These data comprised the ‘‘known’’ exemplars. A tenth, ‘‘unknown’’ exemplar was also obtained. Multivariate linear discriminant analysis was utilized to determine the classification of both the known and unknown exemplars. The analysis correctly classified 69.1% of the known exemplars and 66.7% of the unknown. Speaking fundamental frequency provided the greatest discrimination and spectral function the least.
Phonation threshold pressure: A missing link in glottal aerodynamics.
Authors: Titze Ingo R.
Published: October 2005

Phonation threshold pressure has previously been defined as the minimum lung pressure required to initiate phonation. By modeling the dependence of this pressure on fundamental frequency, it is shown that relatively simple aerodynamic relations for time-varying flow in the glottis are obtained. Lung pressure and peak glottal flow are nearly linearly related, but not proportional. For this reason, typical power-law relations that have previously been proposed do not hold. Glottal impedance for time-varying flow must be defined differentially rather than as a simple ratio between pressure and flow. It is shown that the peak flow, the peak flow derivative, the open quotient, and the speed quotient of inverse-filtered glottal flow waveforms all depend explicitly on phonation threshold pressure. Data from singers are compared with those from nonsingers. The primary difference is that singers obtain two to three times greater peak flow for a given lung pressure, suggesting that they adjust their glottal or vocal tract impedance for optimal flow transfer between the source and the resonator.
Voice quality of Amoy falling tones.
Authors: Pan Ho-hsien
Published: October 2005

The two Amoy falling tones−high falling and low falling−have long and short versions. The short versions occur in syllables with ‘‘entering tone’’−i.e., syllables that are closed by final unaspirated stops [p, t, k, glottal stop]. The shorter duration and the final unaspirated stop are well known to be the major cues for identifying entering tone. In this study, voice quality is examined as a possible minor cue. Oral and nasal air flow were recorded in word pairs contrasting short- and long-falling tones. The oral air flow data were inverse filtered, and maximum ac flow, minimum ac flow, maximum flow declination rate, and durations of open and closed phases were measured. Preliminary results suggest that there is less nasal flow in the short versions by comparison to the long (perhaps reflecting a simultaneous glottal stricture in the syllable-final voiceless stop), but there seems to be no other phonation contrast.
Spectral parameters of a voice source pulse.
Authors: Ananthapadmanabha T. V.
Published: October 2005

This study reports systematic covariations between the time and frequency domain parameters of a voice source pulse. Open quotient, speed quotient, rate of closure, and the maximum negative amplitude of the voice source pulse are the time domain parameters. Frequency of the spectral peak, spectral roll-off, spectral level at fundamental frequency (F0), and twice F0 are some of the spectral parameters. A normalized representation for spectral parameters is proposed. Analysis methods to measure these normalized parameters reliably from the inverse-filtered signal are presented. The methods presented in this paper are generally useful in natural sounding speech synthesis and for studying variations in voice quality. Illustrative examples are presented of voice source variation with respect to phonetic and linguistic context and for different voice qualities. The methods may also be used in evaluating the voice quality of coded speech and in the diagnosis of voice disorders.
Robust pitch determination of noisy speech.
Authors: Rodríguez-Fonollosa José A. , Moreno Asunción
Published: October 2005

Most of the signal analyses made up to date have been based on the autocorrelation function or power spectrum. It is well known that these second-order statistics completely characterize a Gaussian process. However, in many applications where non-Gaussian processes or nonlinearities are present, analysis based on the autocorrelation (and hence, the power spectrum) fails to provide all the useful information about the process [C. L. Nikias and M. R. Raghuver, Proc. IEEE 75, 869–891 (1987)]. In this paper, it is shown that higher-order statistics are very useful in the study and characterization of the speech for two reasons: They can extract useful information about the statistics of voiced frames, and they can separate speech from noise. Taking advantage of these properties, a new pitch determination algorithm based on third-order statistics has been developed. Third-order statistics are quite insensitive to most noises (Gaussian, sinusoidal, car noise,...) because these noises have a symmetric probability density function, and therefore, their third-order cumulants are zero. The algorithm has been tested in noise-corrupted speech, at different levels of signal-to-noise ratio, and with different kinds of noise. The results show that this new algorithm gives a much better estimation of the pitch than the conventional autocorrelation method in all the cases. For example, the cumulant-based detector reduces the gross pitch error rate from 16.8% to 5.4% for speech sentences corrupted with car noise and a signal to noise ratio of 5 dB.
Harmful effects of underwater sound on fish.
Authors: Hastings Mardi C.
Published: October 2005

The acoustic impedance of aquatic animals nearly matches that of water; consequently, most of the sound energy from an underwater source will enter and interact with their bodies. Very intense sounds may even stun or kill fish. Several studies [see summary by K. S. Norris and B. Mohl, Am. Nat. 122, 85–104 (1983)] have reported easily observable damage such as stunning and rupturing of internal organs, caused by underwater sound. Available data indicate, however, that fish may suffer internal morphological and physiological damage as well as apparent physical damage when exposed to sufficient sound pressure levels for relatively short periods of time. Results of experiments relating underwater sound pressure level of pure tones at frequencies from 100 to 500 Hz with different types of damage to three species of freshwater fish are presented. The results show that fish may have internal morphological damage (i.e., destruction of the apical ciliary bundles of hair cells in the inner ear) in the absence of external physical or behavioral changes. Thus, harmful sound pressure levels may be much lower than those commonly reported.
A real-time computer music synthesis system.
Authors: Lent Keith
Published: October 2005

A real-time sound synthesis has been developed at the Computer Music Center of The University of Texas at Austin. This system consists of several stand alone processors that were constructed jointly with White Instruments in Austin. These processors can be programmed as general purpose computers, but are provided with a number of specialized interfaces including MIDI, 8-bit parallel, high-speed serial, two-channel analog input (18 bit A/D’s, 48-kHz sample rate), and four-channel analog output (18 bit D/As). In addition, a basic music synthesis language (Music 56000) has been written in assembly code. On top of this, a symbolic compiler (Patchwork) has been developed to enable the algorithms which run in these processors to be created graphically. And finally, a number of efficient time domain numerical models have been developed to enable the construction, simulation, control, and synthesis of many musical acoustics systems in real time on these processors. Specifically, assembly language modules for cylindrical and conical horn sections, dissipative losses, tone holes, bells, and a number of linear and nonlinear boundary conditions have been developed.
Analysis–synthesis of musical sounds on a desktop computer.
Authors: Beauchamp James W.
Published: October 2005

Recently, desktop computers have become fast enough for applications to run interactively that formerly were too slow for foreground processing. Several Unix-based programs have been ported to a NeXT computer with high quality digital audio input/output. These include programs for time-variant spectrum analysis, display and modification of spectral parameters, calculation of synthesis parameters (e.g., nonlinear synthesis) for optimum match of spectral data, and sound synthesis from spectral data. The current design combines these programs into a single NeXT computer application whose principal objective is to allow the user to analyze a sound, modify its parameters, resynthesize the result, and compare the original and synthetic sounds. To improve the analysis/synthesis response time there are two approaches: (1) use the NeXT’s internal 56001 DSP and (2) use a faster computer networked to the NeXT. The DSP approach is advantageous for repetitive operations such as digital filtering and FFT, whereas networked computers are better for operations which are likely to require frequent recoding. Examples of sounds synthesized from modified spectral data will be given.
Biological effects of lithotripter fields.
Authors: Carstensen Edwin L. , Child S. Z. , Hartman C. , Everbach E. Carr
Published: October 2005

Since the early 1980s, extracorporeal shockwave lithotripsy (ESWL) has grown to be one of the primary techniques used for the clinical treatment of both kidney stone and gallstone disease. The mechanisms of soft-tissue damage and stone breakup in the fields of spark-gap, piezoelectric, and catheter lithotripters, as well as those of diagnostic and therapeutic ultrasound devices have been studied. The results are consistent with the hypothesis that transient acoustic cavitation is the primary mechanism responsible for the observed bioeffects, and plays a role as well in the breaking of the stone. Thresholds have been established for bioeffects in mouse kidney (less than 3–5 MPa peak positive pressure), chick embryos (less than 10 MPa), and Drosophila larvae (less than 1 MPa). The extremely low thresholds for lung hemorrhage in mice for 10 lithotripter pulses (1.5 MPa) and short pulses of diagnostic ultrasound (less than 1 MPa) suggest that gas-filled tissues within the body are especially susceptible to lithotripsy damage. The efficacy of gallstone lithotripsy depends critically upon the mechanical properties of the stones themselves. These properties influence which destruction mechanism−cavitation or stress fracture−is dominant.
The problems of exposure and dose estimation in lithotripsy.
Authors: Coleman Andrew J. , Saunders John E.
Published: October 2005

Despite a decade of widespread clinical use, it is still unclear how to assess the effectiveness and safety of extracorporeal shock wave lithotripters. Two problems are of fundamental importance: (a) to establish the pressure field which occurs at depth in tissue during a typical treatment, i.e., the insitu exposure, and (b) to relate the insitu exposure to a useful acoustic dose by understanding how the field interacts with tissue or stone. The first of these problems has provided the impetus for the development of theoretical models of focused, high-amplitude fields from large aperture sources in which effects of diffraction, frequency-dependent attenuation, dispersion, and nonlinear effects are included. A relatively simple approach to this problem has been attempted using a one-dimensional solution of the KZK equation [Coleman etal., Ultrasound Med. Biol. 17, 245–255 (1991)]. This model successfully predicts some of the observed differences in the acoustic fields of preshock lithotripters (generally electromagnetic and piezoelectric types) and lithotripters where the shock is fully formed at the source (electrohydraulic type). An approach toward the second problem has been to examine the acoustic field generated by cavitation generated at the beam focus of a lithotripter using a focused hydrophone (1 MHz focused bowl of PZT-PC4). Some of the predictions concerning the dynamics of bubbles at the focus of a lithotripter made by Church [J. Acoust. Soc. Am. 86, 215–227 (1989)] are confirmed experimentally.
Lithotripsy shock-wave field characterization: Tools and techniques.
Authors: Schafer Mark E. , Lewin Peter A.
Published: October 2005

Accurate lithotripsy pressure field characterization presents a uniquely difficult acoustic measurement problem: how to design a sensor to survive in the destructive shock-wave environment without sacrificing bandwidth or signal fidelity. This presentation discusses the development, design, and use of reliable, wideband, quantitative shock wave sensors made of piezoelectric polymers such as polyvinylidene difluoride (PVDF). In design, there is a delicate balance between sensor survivability and acoustic performance (i.e., frequency response and directivity). In use, there is a balance between the need for statistically stable measurement data, and the calibration stability of the sensors after repeated shock-wave exposure. These points can be illustrated by selected measurement examples of lithotripter field distributions and shock-wave pressure-time histories. In addition, ongoing development work on a sensor designed for insertion into the body holds the promise of providing reliable invivo acoustic pressure measurements. These advances in sensor technology will contribute to improved lithotripsy treatment, in terms of understanding the mechanisms of stone destruction and tissue damage, reducing patient trauma, and regulating new lithotripsy technology. [Supported in part by Small Business Innovation Research (SBIR) Grant No. 1R43DK42747-01 from the National Institute of Diabetes and Digestive and Kidney Diseases.]
Lithotripter wave propagation. I. Theory and computation.
Authors: Christopher P. Ted
Published: October 2005

A model is described which accounts for the physical effects of diffraction, attenuation, nonlinearity, (planar) refraction, and (planar) reflection. The application of the model to the problem of computing the high-intensity fields of a Dornier HM-3 is discussed. The model’s novel time domain nonlinear algorithm is then presented as a critical development in making the HM-3 propagation computationally feasible. [Work supported by NIH.]
Target interpretation using inverse synthetic aperture sonar techniques.
Authors: Blake W. K. , Le T. D. , Peoples J. R.
Published: October 2005

A two-dimensional inverse synthetic aperture imaging technique is employed with acoustic backscattered signals from simple structures in water. This technique involves coherent signal processing of sequences of wideband acoustic pulses accompanied with continuous body rotation to form a two-dimensional ‘‘image’’ of the reflectivity distribution. Rotation angle aperture determines the lateral, or cross-range, resolution and the signal bandwidth determines the down-range, resolution. Image displays which show various angle down-range loci and cross-range down-range loci were generated. These displays divulge the existence of different classes of structural acoustic phenomena. The processing was applied to data collected on two targets. On one target, consisting of a pair of fluid-filled spherical shells, the method resolved the multipath nature, axisymmetry, and mutual shadowing of the acoustic reflections. On a cylindrical shell the method divulged details of wave behavior due to coincident membrane waves and end cap response.
An assumed modes approach for higher-order effects in circular cylindrical shells.
Authors: McDaniel J. Gregory , Ginsberg Jerry H.
Published: October 2005

Numerous investigators, notably Mindlin, have developed higher-order theories for beams, plates, and shells in which the variation of displacement through the thickness is represented in terms of series expansions. A major difficulty one encounters in those formulations is the derivation of hierarchal partial differential equations of motion accounting for the coupling between different orders of thickness dependence. The present work develops an alternative scheme that avoids the derivation of such equations by applying the assumed modes method in conjunction with Hamilton’s principle. An infinitely long cylinder provides the framework for the development. The basis functions representing the dependence on radial position consist of Legendre polynomials, which have been used in the past to analyze static deflections of thick cylindrical shells, while sinusoidal functions represent the variation along the circumference. The assumed mode representation leads to ordinary differential equations of motion governing the amplitudes of the basis functions. Because of the axisymmetric geometry, the modal equations decouple in the circumferential direction. When the shell is geometrically thin, h/a≪1, each radial mode couples to two adjacent modes above and below. The extent of this coupling increases with increasing h/a, but the mass and stiffness matrices for the modal amplitudes are banded in any case. [Work supported by ONR.]
The Ascension Island listening station.
Authors: Palmer D. R. , Georges T. M. , Wilson J. J. , Weiner L. D. , Paisley J. A. , Mathiesen R. , Pleshek R. R. , Mabe R. R.
Published: October 2005

Ascension Island is located in the South Atlantic Ocean 9100 km, or about one quarter of the distance around the globe, from Heard Island. It is the location of one of the U.S. Air Force’s missile impact locating system (MILS) sites. All HIFE transmissions were recorded on at least 8 MILS hydrophones and many more were recorded on 11 hydrophones, using both digital and analog systems. The hydrophones are located at the depth of the sound channel axis or somewhat deeper. Preliminary indications are that most, if not all, of the data are of high quality. Typical signal-to-noise ratios for the hydrophones south and southeast of the island were 20 to 30 dB in a 1-Hz band. Late, reverberative arrivals were observed for up to twenty minutes after termination of the direct arrival. These and other general characteristics of the Ascension data set will be discussed.
Vertical array resolution of the normal modes from the Heard Island signals.
Authors: Baggeroer A. B. , Lashkari K. , Miller J. , Chiu C. S. , Froger G. , Mikhalevsky P. N. , von der Heydt K.
Published: October 2005

A 32-element, 1.3-km vertical array was deployed off Monterey, CA from the R/V Point Sur during the signal transmission from Heard Island. The signals were received with an SNR on a single channel of approximately −10 dB re:1 Hz. Narrow-band filtering to 15 mHz improves this to +5 dB. The suspension system was designed to minimize array motion and array tilt and depth were monitored at two locations in the array. The multichannel data have been decomposed into normal modes using a time varying least-squares representation. Initial processing of the data indicates the first four modes can represent 75% of the energy of the average projection over a 180-s window. The time-varying structure of the representation and the effect of array tilt and Doppler spread will also be discussed. [Work supported by Dept. of Energy, Monterey Bay Aquarium Res. Inst. and the Commander, Naval Postgraduate School.]
Heard Island feasibility test: Long-range sound transmission from Heard Island to Krylov underwater mountain.
Authors: Burenkov S. V. , Gavrilov A. N. , Uporin A. Y. , Furduev A. V.
Published: October 2005

On 26–31 January 1991, the ‘‘Akademik Nikolai Andreev’’ research vessel anchored at Krylov underwater mountain (17.5° north, 30° west) received the low-frequency signals at a distance of 12.5 thousands kilometers from the source disposed at Heard Island. All transmitted signals were received by two omnidirectional hydrophones at the SOFAR axis and 200-m depth, sampled, and collected on the cartridge tapes by the data acquisition system. All three types of signals−continuous wave 57 Hz, pentaline, and phase coded (M sequence)−have been primarily processed in the time and frequency domains. The purposes of data processing were to determine what the intensity of signals are, the signal/noise (S/N) ratio and temporal variation of the signal phase and magnitude, what is the time structure of the multimode signal, and how time stable is this structure. On most of the transmissions the S/N ratio was about 10–15 dB in 1-Hz band at the carrier frequency 57 Hz, and then it decreased to 0–5 dB on last transmissions. Both temporal variation of Doppler shift and phase fluctuation dominantly depended on the nonuniformity of ship’s motion. GPS data of both ships were compared with the phase variance of some received signals. The distinction between GPS and phase estimations of distance variance is about 10–15 m on an average. The cross-correlation processing of the phase-coded signal enabled one to identify the periodical groups of pulse arrivals that follow with the period of transmitted M sequence. The time stability of multipath arrivals was 5 min and less.
Ocean acoustic thermometers and climatic change: Global monitoring systems.
Authors: Spiesberger John L.
Published: October 2005

Critical to understanding the climate system is a technique by which the large-scale thermal field can be observed in the ocean. Acoustic measurements at basin scales do provide a sensitive means to recognize and map changes in temperature at large scales [J. L. Spiesberger and K. Metzger, J. Geophys. Res. 96, 4869–4889 (1991)]. These types of acoustic measurements provide a perspective of large-scale processes orthogonal to that provided by other data sets and are essential components of future ocean monitoring systems. Practical acoustic monitoring systems could be deployed within the next 5 yr. Economically feasible systems will be discussed which provide real-time monitoring at basic-scale resolution in the global oceans.
Echolocation pulse rate and head azimuth of an Atlantic bottlenose dolphin in a detection task.
Authors: Sigurdson John E.
Published: October 2005

The interpulse intervals and the head azimuth of an Atlantic bottlenose dolphin were monitored continuously while the animal scanned a 10K sq yd area for spheres that were suspended 2 m below the surface during an echolocation detection task. The animal scanned the area from a stationary pen and was rewarded for correctly reporting the presence or absence of a sphere that was equally likely to appear at any position in the area. The recorded scan patterns provide a detailed quantitative description of the gross dynamics of echolocation in a two dimensional detection task and a basis for testing a variety of hypotheses. The overall pattern of scan on sphere-absent trials conformed to the spatial distribution of spheres previously encountered. On sphere-present trials, the interpulse intervals and movements of the head were correlated with the position of the detected spheres. The latter relationship was used as the basis for analyzing the number of pulses directed at a sphere as a function of its distance, the delay between receipt of an echo and the emission of a new pulse, and echo masking by subsequent pulses.
Potential benefits of varying the frequency-gain characteristic for speech reception in noise for hearing-impaired individuals.
Authors: Rankovic Christine M. , Zurek Patrick M. , Freyman Richard L.
Published: October 2005

Some commercially available hearing aids employ adaptive linear filtering in an attempt to improve speech reception in noise for hearing-impaired individuals. The potential benefit of this strategy depends on whether modifications to the frequency-gain characteristic can reduce the spectral spread of masking of the interfering noise. The purpose of our investigation is to address the finding that some hearing-impaired individuals benefit from this processing while others do not [e.g., J. N. Van Dijkhuizen, J. M. Festen, and R. Plomp, J. Acoust. Soc. Am. (in press)]. Hearing-impaired subjects were presented with nonsense syllables whose average spectrum is shaped to maximize audibility, together with a high-level octave-band noise centered on a low, mid, or high frequency. Frequency-gain characteristics are then imposed with the goal of reducing the overall level in the frequency band containing the noise. In addition to intelligibility scores, masking patterns are obtained for the noises presented in all listening conditions. Attempts will be made to account for individuals’ speech reception results in terms of their masking patterns via articulation theory. [Work supported by NIH.]
ISO estimations of noise-induced hearing impairment.
Authors: Lempert Barry L.
Published: October 2005

The 1990 (second) edition of ISO 1999, ‘‘Acoustics−Determination of Occupational Noise Exposure and Estimation of Noise-Induced Hearing Impairment’’ purports to provide a more accurate and generalized model of the relationship between hearing threshold level and occupational noise exposure than the 1971 (first) edition. While the 1990 standard states that hearing risk statistics are in agreement with those in the 1971 standard, this is not the case. ISO 1999 (1990) consistently underestimates hearing risk by about 10% at 90 and 95 dBA when compared to ISO 1999 (1971). While ISO 1999 (1990) is based upon Burns and Robinson (1970) and Passchier-Vermeer (1968) noise and hearing study databases, the mathematical expression of these data in ISO 1999 (1990) underestimates hearing threshold levels for these databases on average by more than 2 dB; a similar comparison to the NIOSH (1972) database shows that ISO 1999 (1990) underestimates hearing threshold levels by an average of 4.5 dB. Using the hearing threshold level index recommended by NIOSH to compute hearing risk, ISO 1999 (1990) underestimates risk for retirement age populations by an average of 12% when compared to Burns and Robinson, Passchier-Vermeer, and NIOSH databases. Further comparison of this new standard formulation to the original data upon which it is based, is recommended.
Validation of a computerized auditory model for the analysis of warning sounds.
Authors: Laroche Chantal , Hétu Raymond , Quoc Hung Tran
Published: October 2005

A computerized model was developed to predict the detectability of warning sounds in noisy workplaces. A previous study with 23 subjects having different degrees of noise-induced hearing loss allowed to quantify the relation between loss of auditory sensitivity and loss of frequency selectivity. This relation was then introduced into the model. This model takes into account the loss of auditory sensitivity and selectivity due to cochlear hearing loss and the use of hearing protectors. It is based essentially on loudness summation and excitation patterns. Before making the program available to health and safety personnel, a field study is being conducted in order to validate its predictions in terms of signal detection and recognition. The perception of three different warning sounds with well-defined acoustic characteristics (a buzzer, a siren, and a horn) is studied with different background noises from a steel and a bottling plant. At five different workstations, workers are asked to raise their hand when they hear a warning sound emitted by a loudspeaker. The time interval between two warning sounds varies between 10 and 50 s (an average of 30 s). The sound level is randomly set to one of five levels corresponding to S/N ratios between −5 and +15 dB within a 1/3 octave band. Based on previous studies, it is suspected that a signal-to-noise ratio of +15 dB is needed in at least 1/3 octave band in order (1) that the warning signal be detected, (2) that it attracts the attention of the workers, and (3) that it be recognized among other warning sounds. Results will be discussed with respect to the safety of workers in noisy workplaces. [Work supported by IRSST.]
Neural network applications to ultrasonic nondestructive evaluation.
Authors: Avioli Michael J.
Published: October 2005

Ultrasonics, between 500 kHz and 5 MHz, is used to inspect welded piping in electric power generating plants. Detection, characterization, and sizing of defects, possibly detrimental to the pipe’s structural integrity (e.g., cracks, lacks of weld fusion) are the tasks of ultrasonic inspection. These tasks are compromised by the similarities and subtle differences between ultrasonic responses from the nonconsequential geometric reflectors of the weld and pipe structure and other critical reflectors. This paper shows how neural networks can be used to discriminate integrity-threatening flaws from geometric artifacts. A sample application, using ultrasonic waveforms collected from welded pipes, is presented as part of the paper.
Automobile diagnostic expert system by noise and vibration.
Authors: Shin Joon , Oh Jae Eung
Published: October 2005

A diagnostic prototype expert system was built using characteristics of measured sound and vibration for the automobile. For the utilities of this system, a 1/3 octave filter (bandpass filter) and an A/D converter were used for data acquisition and then the data were analyzed using signal processing technique, statistical analysis, and pattern recognition using a hamming network algorithm. In order to raise the reliability of the diagnostic results by considering many operating variables and the condition of the automobile to be diagnosed, a fuzzy inference technique was applied combining several kinds of information. The results were displayed as a graphical method to help the novice in the diagnostic field. The validation of this diagnostic system was checked through a simple rotating simulator and automobile and it showed acceptable performance for a diagnostic process. Also, for the case of wrong decision making, this system included a learning algorithm and its knowledge base can be extended through that procedure. Therefore, as a result, it is expected that this system can be developed by implementing more simplicity and flexibility to the external environment.
The ANSI standard method for measurement of the intelligibility of speech over communication systems.
Authors: Hawley Mones E.
Published: October 2005

ANSI S3.2-1989 was adopted in December 1989 and replaces the standard adopted in 1961. The new standard is intended for people who want quantified results of comparisons among alternative systems. The standard is not intended for use with untrained subjects, for subjects without normal hearing or speaking abilities, for synthesis-by-rule (text-to-speech) devices, or for automatic speech recognition equipment. The standard is not intended for use to evaluate speaker recognition or identification systems. The standard is the result of many compromises, such as between the needs of military and civilian user groups, between the realism of testing in the field and the reproducibility of laboratory testing, between the desire for economy and the need for scientific validity. The new standard emphasizes the selection and training of talkers and listeners; the design, control, and reporting of test conditions; and the analysis and reporting of test results.