bluuebunny/crossref_metadata_2025 huggingface dataset:
An investigation of seabed acoustic penetration and scattering via broadband echo soundings
Authors: Boyle Frank A. , Chotiros Nicholas P. , Pace Nicholas G. , Bergem Oddbjorn , Pouliquen E.
Published: July 2002

Recent experiments were conducted in the Mediterranean Sea with the intention of developing a better understanding of the physics of acoustic penetration into and scattering from the seabed. Two shallow water sites were studied, one near Viareggio, Italy and the other off the coast of Sardinia. One site consisted of a fine mixture of silt and sand, with a large amount of trapped gas, while the other included a layer of marine vegetation that produced a bright acoustic return ahead of the bottom echo. Echo soundings were collected with a broad beam impulsive sound source operating over a band between 1 and 15 kHz. The data acquisition system was configured to record the acoustic pulse before and after encountering the seabed, from which a calibrated measure of the seabed’s reflective character was obtained. The measured reflection coefficient was compared with model predictions, based on a poroelastic theory for the seabed. In general the measured reflection coefficients exceeded the predictions significantly. A likely hypothesis is that trapped gas bubbles act to enhance the reflection. A description of various signal processing techniques that were employed will be included, along with a discussion of relationships between the processed acoustic data and known structural features. [Work supported by ONR.]
A measurement method of an oblique-incidence absorption coefficient
Authors: Okoshi Masaru , Shiroko Norio , Kido Ken’iti , Suzuki Hideo , Ono Takahiko
Published: July 2002

There are two well-known methods for the evaluation of absorption of a material. One is the acoustic tube method (normal incidence) and the other is the reverberation room method (random incidence). In a real situation, however, oblique-incidence absorption coefficients are required. For example, it is necessary to have knowledge of absorption coefficients for various angles in order to evaluate the effect of the road surface on traffic noise. In the previous paper [H. Suzuki et al., ASA-ASJ Third Joint Meeting, Honolulu, Hawaii, J. Acoust. Soc. Am. 100, 2588(A) (1996)], a method of using the sound intensity technique was proposed. An absorbing material was placed on the floor of a semi-anechoic room. A line source with 16 loudspeakers with 100 mm diameter was used in order to try to generate a uniform sound field in the direction normal to the incidence-reflection plane. Measured results showed some irregular changes as a function of frequency. One possible reason for this irregularity was that the multi-loudspeaker source had a more complex directivity near the main axis than a single loudspeaker source. Results obtained by use of the single loudspeaker source are presented in this paper.
Acoustic stop bands for periodic arrays of metallic rods
Authors: Kushwaha M. S. , Halevi P.
Published: July 2002

Extensive band structure computations are performed for two-dimensional periodic arrays of rigid cylinders in air, with the Bloch vector being perpendicular to the cylinders. It is found that for any value of the period (of the square lattice), there is no acoustic gap for filling fraction f<30%; the magnitude of the gap, for f >30%, is inversely proportional to the period. Moreover, if the filling fraction exceeds 40%, a second gap, higher in frequency, opens up. Within these gaps (or stop bands) sound and vibrations are forbidden, irrespective of the direction of propagation. It is noted that for cylinders 2.9 cm in diameter and period of 10 cm (i.e., f=0.066), there is no acoustic gap for frequencies below 6.4 kHz. However, the density of states reveals prominent minima at 1.7 and 2.4 kHz. These frequencies do agree with those of the first two attenuation maxima observed in a recent experiment [Nature 378, 241 (1995)] and are indeed related to diffraction from the [100] and [110] planes. Thus, even with idealization, Sempere’s sculpture (investigated in the above reference) exhibits pseudogaps—not full gaps. A proposed application of the above-mentioned band structure calculations is presented in the following abstract.
Scattering from objects near the ocean bottom at very high frequency and geometrical diffraction theory
Authors: Werby M. F. , Sidorovskaia N. A.
Published: July 2002

For extremely high frequencies, the geometrical theory of diffraction is probably a sufficient, if not necessary, method to carry out predictions of scattering from ellipsoids or objects of more complicated structures for ka values on the order of a few hundred to tens of thousands. It is possible to extend the Helmholtz integral expression to the problem of ellipsoids near an ocean bottom by making use of a Green’s function with a reflection coefficient that characterizes the bottom. In carrying out the integral expressions, saddle point methods were experimented with to determine the reflected signal, and the Maggi transformation and principles from differential geometry to reduce the diffraction integral over the shadow side of the object to line an integral at the illuminated shadow region of demarcation. The latter two devices are necessary due to time constraints in evaluating surface integrals for ultra-high frequencies. Examples are presented. [Work sponsored by ONR, the Naval Res. Lab., and the Univ. of New Orleans.]
Mechanisms that cause changes to bottom reverberation fluctuation statistics
Authors: Deliman Linda S. , McCammon Diana F.
Published: July 2002

An investigation into the underlying mechanisms that cause bottom reverberation fluctuation statistics to deviate from the typically assumed Rayleigh distribution reveals that sediment ‘‘patchiness’’ is a major contributor. The investigation involved processing both shallow water data and simulated data. The shallow-water data, collected during a SACLANT Centre experiment, include low-frequency measurements made over chalky sediments with varying degrees of roughness using different pulse types (cw and FM) and pulse lengths (2.2 and 8 s). The processing produced both Rayleigh and non-Rayleigh statistics. The results indicate that sediment type, pulse type, and pulse length are not a factor per se in the deviations of the statistics from Rayleigh. However, sediment ‘‘patchiness,’’ defined as an ensonified area containing patches which contribute largely different backscattering strengths within the area, does cause the statistics to deviate from Rayleigh. The simulated data reinforce these conclusions and provide insight into the spacial and temporal extent of ‘‘patchiness’’ required to cause a deviation in the statistics.
Dispersion of circumferential creeping waves around a fluid-filled cylindrical cavity in an elastic medium
Authors: Hassan Waled , Nagy Peter B.
Published: July 2002

Ultrasonic inspection techniques using circumferential creeping waves can readily detect radial fatigue cracks on the fair side of thin airframe stiffener holes in empty aircraft wings. It is desired to utilize similar techniques in detecting fatigue cracks in the case of fluid-filled fuel tanks. The dispersion equation of a fluid-filled cylindrical cavity in an infinite elastic medium has been studied numerically. It was found that two distinct waves can propagate. One is the circumferential creeping wave that is attenuated by radiating energy into both the fluid and the surrounding solid. The other is the essentially nonattenuated halo wave caused by the resonances in the fluid within the cavity. The slower halo wave is the strongest of the two, but unfortunately it is not sensitive to the existence of any cracks at the surface because it propagates mostly in the fluid. It is the Rayleigh-type creeping wave that is important in the detection of surface breaking or near-surface cracks. It can be concluded that the adverse effect of fluid loading on the inspection technique is significant but not prohibitive. Additional finite-element calculations as well as experimental measurements are being conducted. [See NOISE-CON Proceedings for full paper.]
Time-reversed ultrasound
Authors: Fink Mathias
Published: July 2002

Time-reversal invariance is a very powerful concept in classical and quantum mechanics. However, simple experimental evidence of this concept is difficult to obtain. In the field of acoustic waves, where time reversal invariance also occurs, time-reversal experiments may be achieved simply with large ultrasonic piezoelectric transducer arrays. Time-reversal mirrors (TRMs) are made of such reversible acoustic retina, allowing an incident acoustic field to be sampled, recorded, time-reversed, and re-emitted. Time reversal of acoustic fields allows an optimal approach to focus pulsed ultrasonic waves through inhomogeneous media in the sense that it realizes the matched filter in both the spatial and temporal dimensions to the inhomogeneous propagation transfer function. In the field of acoustics, the ability to work directly in the time domain, with piezoelectric retina, allows pulse echo mode processing in an iterative mode. In multitarget media, such an iterative process converges on the most reflective target. In the case of an extended elastic target, automatic resonances of vibration modes can be achieved. TRMs are also innovative tools in the field of fundamental physics. They may be used to study inverse scattering problems, multiple scattering processes, chaotic scattering as well as scattering by turbulent media. Several experiments illustrating these areas will be presented. Applications of TRMs will also be described and include medical applications (kidney stone destruction and hyperthermia) as well as nondestructive testing of solid media. Experimental results illustrating these applications will be presented.
Mathemusical connections: The role of musical acoustics in teaching mathematics
Authors: Hinthorne Stephen G.
Published: July 2002

The purpose of this established liberal arts course, A survey of mathematics, is to change student attitudes towards mathematics through an innovative introduction to musical acoustics. This approach is particularly effective for those students who may not like mathematics or may have some phobia towards it. The topics and structure of the course are built around the quadrivium of mathematics attributed to Pythagoras. This includes four disciplines: arithmetic (today’s number theory); music; geometry; and astronomy. Added to these topics are: numeration systems; algebra; finance; and probability and statistics. Sequences of musical scale ratios associated with lengths of vibrating strings and the frequencies of related sounds in the ‘‘major’’ scale were invented by Pythagoras, Claudius Ptolemy, and musical theorists of the sixteenth-century Renaissance. The study of the development of these three scales and their comparison, together with Mersenne’s law of vibrating strings and some work with graphing sine functions, provide students with their first look at the mathematics of music. Music and acoustics provide the vehicle for students to gain a better understanding of the often troublesome areas of ratios, proportions, formulas, and graphs. This paper includes the course outline and structure, examples of topics and assignments, and student response.
Effect of aircraft maneuvers on sonic boom penetration into a flat ocean
Authors: Ferguson Tracie J. , Sparrow Victor W.
Published: July 2002

To fulfill its ongoing mission of evaluating the environmental noise effects of training operations, the U. S. Air Force Materiel Command has initiated research studies on the penetration of sonic boom noise into seas, bays, and lakes. All of the previous research on sonic boom noise penetration into water from supersonic aircraft have assumed steady flight. In realistic training, however, aircraft accelerate and decelerate and are involved in maneuvers such as climbing, diving, and turning. The present research will report the results of initial studies predicting the penetrating sonic boom noise created by maneuvering supersonic aircraft. The commercial package PCBOOM3 [Wyle Laboratories, Arlington, VA] is used to calculate the ray paths taken from the supersonic aircraft to the water surface. The sonic boom waveforms and respective incident angles on the surface are obtained from PCBOOM3 and are input to an algorithm for calculating the underwater sound field recently presented by Sparrow and Ferguson [AIAA Paper 97-0486 (1997)]. Given an arbitrary shaped incident waveform, the algorithm provides the resulting underwater sound field assuming the water surface is flat. [Work supported by Armstrong Laboratory, Air Force Materiel Command, USAF, under Grant No. F41624–96–1-0003.]
Three-dimensional sound: Distance identification of sounds synthesized using an atmospheric propagation model including turbulence
Authors: Gibian G. L. , Pepe K. M. , Koroljow W. S. , McBride W. E.
Published: July 2002

The perception that an electronically generated sound source originates from outside the head at a particular direction can be produced by filtering the sound with head-related transfer functions (HRTFs). The perception that the source is at a particular distance is more difficult to produce, particularly the first time an unfamiliar sound is heard. In this study, listeners tried to identify source distances when a 300-ms white noise burst was filtered with transfer functions from a physically based atmospheric model including scattering by turbulent eddies in the atmosphere (‘‘turbules‘‘). Distances of 200, 400, 800, 1600, and 3200 m were simulated using a fixed distribution of turbules. Listeners’ performance with and without inclusion of scattering by turbules was compared. Results indicate that: (1) when the subjects were given known sounds for reference or had become familiar with the test sounds, they correctly identified the distances about 80% of the time, despite changes in playback gain or reduced dynamic range, and (2) distance identification improved by about 10% when the effects of scattering were included. The improvement in distance identification was produced without the need for increased real-time computation over simple atmospheric absorption filtering. Larger improvements are expected with time-varying and spatial filters.
Auditory scene analysis-constrained array processing for sound source separation
Authors: Drake Laura , Rutledge Janet
Published: July 2002

In this work, techniques are developed and studied for the extraction of single-source acoustic signals out of multi-source cacaphony. Such extracted signals can be used in a variety of applications including: automatic speech recognition, teleconferencing, and robot auditory systems. Typical approaches fall into two categories: auditory modeling and array signal processing. The approach taken here is to combine these complementary techniques into an integrated one: auditory scene analysis-constrained array processing. Theoretically, this integrated approach should provide a performance gain since the information used by array processing, i.e., direction of propagation through a wave field, is independent of the signal structure information used by auditory scene analysis. One difficulty encountered by auditory scene analysis that can be overcome by array processing is the sequential grouping of spectrally dissimilar phonemes in a speech signal, such as a fricative followed by a vowel, or nasal.
Solving the inverse problem in nonlinear acoustics by backpropagation of the received signal
Authors: Hedberg Claes M.
Published: July 2002

The incentive for this work was to determine when and how the information in a nonlinear signal disappears when propagating. This may be done by backpropagating the evolution and comparing the back-traced signal with the original one. The Burgers equation is investigated for both the nondissipative and dissipative cases. For a nondissipative evolution the wave completely coincides with the original one when shocks have not formed, while only part of it coincides after shock formation. Although the wave no longer contains the original information, except for the original signals with all derivatives continuous, parts of the wave theoretically carry all the information. These parts grow smaller and smaller, and thus a problem lies in the numerical accuracy which sometimes causes several different signals conform, and in that way the wave in practice has lost information. A semi-analytical model has been developed which automatically determines the distance to be backpropagated for modulated waves and gives one or more alternatives for the original signal. This may be used as a signal processing tool. [Work supported by TFR, the Swedish Research Council for Engineering Sciences.]
Experimental investigation of modal structure of field generated by a pulse-source in a laboratory hydroacoustic waveguide
Authors: Egorychev Sergei A. , Kurin Vasily V. , Pronchatov-Rubtsov Nikolai V. , Virovlyansky Anatoly L.
Published: July 2002

This work contains the results of an experimental investigation of the modal structure of an acoustic field in a model oceanic waveguide. Due to limited propagation range, it is impossible to resolve modes either by arrival times of modal pulses or by Doppler spectra of the received signal. The experiment was done in a laboratory two-layer waveguide where 32 normal waves had propagated at ranges up to 260 cm. As a result of intermodal dispersion, original short pulses radiated at 200 kHz were destroyed, and different parts of the received signal carried information about different modes forming the field in the waveguide. The beginning of the received pulsed signal was formed by the low-number modes, while the area near the back front of the pulse was constructed by high-number modes. Selection of modes was done using Doppler spectra of the received signal by a receiver uniformly moving along the waveguide and information about interference structure was read in the beginning, in the middle, and in the end of each received pulse. Such signal processing allowed to ‘‘impoverish’’ the original modal spectrum and, afterward, to analyze the modal structure of the acoustic field in the waveguide in more detail. [Work supported by RFBR Grant No. 95-02-04565.]
PiezogranTM: High sensitivity/low-cost composites for wide aperture arrays
Authors: Jessen Todd L. , Kahn Manfred , Chase Mark T.
Published: July 2002

A shift in US Navy emphasis from blue water to littoral regions has resultled in a need for improved acoustic devices. Wide aperture arrays will be hull-mounted, thereby making device weight critical. The Naval Research Laboratory has developed a unique, low-weight composite, transducer design, PiezogranTM, to address this need. Piezograns consist of discrete, polycrystalline piezoelectric-ceramic elements dispersed in an epoxy matrix between two pressure plates. These plates make a direct mechanical as well as electrical contact to the piezoelectric elements. This design approach uses a 1-3 composite structure for high-gain operation with the flexibility of variable element spacing and ceramic volume fraction while achieving both low tooling and fabrication costs. Independent tests of 3.8-cm diameter PiezogranTM prototypes have yielded high hydrostatic response (dh>150 pC/N) and capacitance of 1.18 nF. Ten-centimeter-square array panels have also been assembled and are currently undergoing testing. This presentation will focus on fabrication and assembly of the PiezogranTM transducers as well as material properties.
Vowel stability may constrain selective adaptation
Authors: Coady Jeffry A. , Aslin Richard N.
Published: July 2002

Current models of selective adaptation for vowels implicate acoustic or auditory mechanisms rather than phonetic mechanisms. However, inconsistencies have been reported for /i/, /■/, and /ε/ [Morse et al. (1976)] in that the middle vowel /■/ did not elicit significant adaptation effects. These inconsistencies may reflect category goodness effects [Miller et al. (1983); Samuel (1982)] in that adaptation is more robust for tokens further from the category boundary. In the present study, category goodness was determined precisely for /i/, /■/, and /ε/ by obtaining within-subject rating data. Two continua were then constructed for each subject with these three vowels as the endpoints. Vowel adaptation was then assessed for the two continua using each subject’s three best exemplars as the adapting stimuli. Despite the fact that each subject’s adapting stimulus was, by definition, the ‘‘best’’ exemplar of that category, only /■/ and /ε/ showed significant adaptation. The absence of adaptation for /i/ may reflect its status as a point vowel, suggesting that extreme vocal tract positions, and their acoustic correlates, are remarkably stable. Research in progress on the point vowel /a/ will reveal whether this hypothesis is correct. [Work supported by NSF SBR94-21064.]
Array processing techniques for broadband mode estimation
Authors: Wage Kathleen E. , Baggeroer Arthur B. , Preisig James C.
Published: July 2002

Normal mode representations are useful in many aspects of underwater acoustics, e.g., modal acoustic tomography uses precise measurements of the travel time of broadband pulses to invert for environmental parameters. A thorough understanding of the resolution of mode estimators is crucial for tomographic applications. Typically, acoustic modes are used for describing and analyzing the temporal and spatial structure of narrow-band signals. As source bandwidth increases, however, mode estimation algorithms can no longer ignore frequency variations of the modal wave numbers, which result in temporal dispersion and frequency-dependent mode shapes. To address the issue, this study establishes an array processing framework for estimating the modal composition of broadband receptions. Within this framework, the time/frequency resolution tradeoffs inherent in the processing of modal pulse arrivals are explored. Specifically, the effects of mode shape frequency variations on standard modal beamformers are investigated. In addition to characterizing the performance of spatial filters for mode separation, this study considers the problem of designing receivers to compensate for the dispersion (pulse-spreading) introduced by the waveguide. Techniques developed through theoretical work have been applied to data from the Acoustic Thermometry of Ocean Climate (ATOC) project. Some results of modal time series analysis for 1996 ATOC receptions are presented.
A spatio-temporal approach to acoustical imaging of laser-generated ultrasound
Authors: Candy James V. , Huber R. , Chinn Diane , Thomas Graham H.
Published: July 2002

In this paper an application of spatio-temporal array signal-processing techniques applied to broadband ultrasonic data gathered from a pulsed laser system is discussed. Using a laser source to heat a material specimen under test for flaws, a spatio-temporal processor capable of estimating the displacement field of the specimen is applied. The peak surface displacement is displayed as an image showing the initial source (displacement field) propagating throughout the material as well as any flaws (scatterers) that may be present within the specimen. Clearly, this method of imaging enables a unique methodology for nondestructive evaluation (NDE). Here, a pulsed laser generates an acoustic (ultrasonic) wave by heating the material and causing thermoelastic expansion. The resulting ultrasonic wave propagates throughout the material and is receied by an array of interferometers created synthetically. Assuming a spherically propagating wave field, the processor creates an image of the field by estimating the peak surface displacement at a given location. The resulting image displays valuable information about the material and its inherent flaws, providing an effective method of nondestructive evaluation. In this work, the results are shown of applying this method to data synthesized from a sophisticated thermoelastic simulation and specimen available in the laboratory are tested.
The combined effects of ototoxic drugs and noise
Authors: Rybak Leonard P.
Published: July 2002

Cisplatin is a widely used chemotherapeutic drug which causes ototoxicity. Previous studies from this laboratory using the rat have shown that cisplatin ototoxicity may be mediated by the production of free oxygen radicals. The ototoxic effects of cisplatin can be reduced by treatment with reducing agents such as diethyldithiocarbamate, methylthiobenzoic acid, ebselen, D-methionine, or sodium thiosulfate. The injury to the inner ear caused by cisplatin is potentiated by noise exposure. The greatest degree of cochlear damage in the guinea pig was observed when noise exposure precedes the administration of cisplatin [G. Laurell, Ann. Otol. 101, 969–976 (1992)]. These findings could be explained by an increase in oxidative stress produced by noise exposure, which could predispose to further injury by subsequent cisplatin administration. [Work supported by NIH.]
Effects of contralateral presentation and interaural intensity differences on the detection of mistuning
Authors: Gockel Hedwig , Carlyon Robert P.
Published: July 2002

Listeners were required to detect a mistuning imposed on the center (‘‘target’’) component of a harmonic complex consisting of the first seven harmonics of a 500-Hz fundamental. In the standard interval all components were frequency modulated in phase by a 5-Hz sinusoid. In the signal interval the frequency modulation of the target component was inverted in-phase, thereby introducing a mistuning proportional to the depth of FM. For monaural presentation, Carlyon [J. Acoust. Soc. Am. 95, 2622–2630 (1994)] reported a substantial elevation of thresholds in the presence of an unmodulated asynchronous interferer whose frequency was identical to that of the target component. He explained this by assuming that the interferer caused the target component to be perceptually segregated from the remainder of the complex, thereby impairing across-frequency comparisons. The present study investigated the effect of lateralization on the detection of mistuning. Preliminary results of experiment 1 suggest that in the absence of an interferer, presenting the target component contralaterally to the remainder of the complex raises thresholds, but that the task is still possible. Results from experiment 2 indicate that a contralaterally presented interferer impairs performance, a result which is interpreted in terms of auditory streaming.
Smart foam for active/passive plate radiation control.
Authors: Gentry C. A. , Guigou C. , Fuller C. R.
Published: July 2002

A new type of noise suppression device known as ‘‘smart foam’’ that integrates a distributed piezoelectric polymer actuator (the active component) between individual layers of sound-absorbing foam (the passive component) is discussed. Smart foam acts as a hybrid active/passive surface coating as it adaptively modifies the acoustic impedance of a vibrating surface. The potential of such a device to reduce low- and high-frequency sound radiation from a simple acoustic source (piston) has been demonstrated. In this experimental study, an array of smart foam modules are implemented to yield a net decrease in the sound power radiated by a vibrating, baffled plate in an anechoic environment. A MIMO feedforward filtered-x LMS algorithm is used to minimize the error signals provided by microphones located in close proximity of the smart foam array for harmonic and broadband plate excitations. Two different reference signals, i.e., the voltage sent to the plate disturbance actuator and the signal from an accelerometer directly mounted on the plate, are investigated and compared in terms the sound attenuation achieved. The capability of smart foam for reducing plate sound radiation is demonstrated. Smart foam is relatively lightweight, which makes it a good candidate for interior aircraft noise control applications. [Work supported by NASA LaRC.]
Test of a piezoceramic actuator for the active control of power transformer vibrations
Authors: Micheau Philippe , Berry Alain , Masson Patrice , Charrette Francois
Published: July 2002

The sound radiated by large power transformers found in electric utility substations is an annoyance to nearby residents. Noise barriers are not effective at 120 Hz, the most prominent harmonic radiated by the transformers, and active control with loudspeakers and microphones is very sensitive to environmental conditions. Thus the active structural acoustic control is an attractive alternative because this strategy controls the sound at the source. However, efficient actuators are needed to generate antivibrations. This paper describes an experimental investigation conducted in order to select and test a piezoceramic actuator for the active control of large power transformer vibrations. It presents experimental measurements of transverse displacements on the various sides of a transformer, together with simulation and experimentation of a selected piezoceramic actuator bonded to a test structure and to the transformer. Vibration levels obtained with a (0.0075×2.5×1.5 in.) PZT piezoceramic actuator bonded to the transformer are generally larger than the primary vibration field at 240 Hz and at higher harmonics, but smaller at 120 Hz. Guidelines for the design of appropriate piezoceramic actuators to control power transformer vibrations are given. [Work supported by Hydro-Québec.]
Control of structural sound radiation using multiple detuned vibration absorbers
Authors: Wright R. I. , Charette F. , Fuller C. R.
Published: July 2002

The results of an experimental investigation into using multiple tunable vibration absorbers to minimize sound radiation from vibrating plates will be presented. The vibration absorbers consist of lumped mass-spring systems whose characteristics (i.e., resonance frequency) can be electronically varied. A multiple channel, gradient search algorithm, implemented on a DSP, is used to adapt the characteristics of each of the absorbers (i.e., detune them) so as to minimize the radiated sound at selected multiple positions in the farfield. Experimental results for the detuned case are presented and compared to those obtained by tuning the resonance frequency of the vibration absorbers to the disturbance frequency. Preliminary results indicate that attenuation of sound radiated from the plate can be significantly improved by detuning the vibration absorbers away from the disturbance frequency via a gradient search. Thus, an adaptive-passive approach to minimizing sound radiation has been demonstrated to obtain good attenuation with very little electrical power requirement while adapting. This result is in contrast to active structural acoustic control which typically requires a signicant control power. [Work supported by NASA LaRC and ONR.]
Comparison of native English speakers’ perception and production of the English vowel /i/
Authors: Frieda Elaina M.
Published: July 2002

This study examined the relationship between speakers’ production and perception of the English vowel /i/. Participants were 24 male native English speakers representing diverse dialects. First, subjects produced /i/ in two speaking conditions: In condition one, subjects were instructed to speak normally; in condition 2, subjects produced exaggerated/hyperarticulated speech. Second, subjects completed 30 trials using a method of adjustment procedure where they selected their perceptually ‘‘preferred or prototype /i/’’ by sampling from a grid of 330 synthetic vowel stimuli that differed in F1 and F2. Speakers differed substantially in their productions, as well as their prototype selections. Preliminary results demonstrate that subjects’ prototypes were generally higher and fronter in the vowel space (i.e., had lower F1 and higher F2 values) than the vowels they actually produced in the normal condition. Moreover, for individual subjects, there was a negative correlation in F2 values for these two conditions (r=−0.40,p<0.05). However, the F2 of the prototype and the hyperarticulated speech did not differ. In general, subjects both perceptually prefer and produce hyperarticulated vowels with extreme F2 values. [Work supported by NIH.]
Optimum microphone placement for array sound capture
Authors: Rabinkin Daniel V. , Renomeron Richard J. , French Joseph C. , Flanagan James L.
Published: July 2002

Microphone arrays can be used for high-quality sound pick up in reverberant and noisy environments. The beamforming capabilities of microphone array systems allow highly directional sound capture, providing superior signal-to-noise ratio (SNR) when compared to single microphone performance. There are two aspects in microphone array system performance: the ability of the system to locate and track sound sources, and its ability to selectively capture sound from those sources. Both aspects of system performance are strongly affected by the spatial placement of microphone sensors. A method is needed to optimize sensor placement based on geometry of the environment and assumed sound source behavior. The objective of the optimization is to obtain the greatest average system SNR using a specified number of sensors. A method is derived to evaluate array performance for a given array configuration defined by the abovementioned metrics. An overall performance function is described based on these metrics. A framework for optimum placement of sensors under the practical considerations of possible sensor placement and potential location of sound sources is also characterized.
Long-time semiclassical dynamics of chaos
Authors: Tomsovic Steven
Published: July 2002

Semiclassical methods, long used in most realms of wave physics, have recently experienced a surge of interest in quantum mechanics, especially in the context of highly chaotic systems. In this talk, discussion will center on a demonstration that accurate approximations of long-time quantum wave propagation can be constructed using the appropriate chaotic trajectories of the classical analog. It shows that pessimistic assessments of a breakdown time of τ=ln(1/ℏ)/λ in the approximation, where λ measures the rate of exponential separation of trajectories, are incorrect. Recent advances will be briefly summarized. [Support from the National Science Foundation under Grant PHY-9421153 is gratefully acknowledged.]
A broadband microphone array technique for correlated interference rejection
Authors: Ward Darren B.
Published: July 2002

A new method of broadband minimum variance beamforming is presented for use with microphone arrays in a reverberant environment. Conventional broadband algorithms (such as the well-known Frost algorithm) suffer from cancellation of the desired look-direction signal when employed in environments containing interference which is correlated with the desired signal. In this paper, a frequency invariant beamforming structure is used to perform spectral averaging. This reduces the correlation between the desired signal and the interference, thereby eliminating the signal cancellation problem. Such a technique is valuable for speech acquisition using a microphone array, since significant correlated interference will be present due to room reverberation.
Prediction of speech recognition by Fletcher’s articulation theory for normal-hearing and hearing-impaired listeners in noise-masked conditions
Authors: Rankovic Christine M.
Published: July 2002

Articulation theory is a computational model designed to predict speech recognition scores for nonideal listening channels. Fletcher’s original formulation [H. Fletcher and R. H. Galt, J. Acoust. Soc. Am. 22, 89–151 (1950)] has been overlooked, presumably due to its complexity and the availability of simplified versions such as ANSI S3.5-1969. The Fletcher and ANSI versions were compared on their ability to predict articulation scores using data collected by Rankovic et al. [J. Acoust. Soc. Am. 90, 2319(A) (1991); 91, 354–362 (1992)]. In those studies, consonant-correct scores were obtained from five normal-hearing and seven hearing-impaired subjects in the presence of octave-band noises centered on 0.5, 1, or 2 kHz and noise-masking patterns were obtained from all subjects in all conditions. The speech was amplified for the hearing-impaired subjects. For normal-hearing subjects, both models made good predictions of consonant-correct scores. For hearing-impaired subjects, the ANSI version overestimated scores for conditions with articulation indices (AIs) greater than about 0.5, whereas Fletcher AIs were smaller and predictions more accurate. Differences in the AI calculations that account for these results will be discussed. [Work supported by NIH.]
Effects of listening paradigm on speech recognition
Authors: Besing Joan M. , Koehnke Janet , Abouchacra Kim S. , Tran Tuyen V.
Published: July 2002

In everyday listening situations, binaural information enhances the ability to understand speech messages in multitalker environments. For applications to virtual auditory technology in such environments, the importance of preserving characteristics of the natural listening environment is not clear. The purpose of this study was to assess listeners’ ability to monitor target (T) messages in the presence of synchronous competing (C) messages in an anechoic and a reverberant environment in three modes: (1) through loudspeakers at ±45° and ±90° with the subject seated in the actual room; (2) through the same loudspeakers with the subject listening remotely to the stimulus presented to a manikin in the actual room; and (3) with the environment simulated to create virtual sources at ±45° and ±90° azimuth under earphones. Fourteen subjects listened to messages selected from four lists of 2034 ten-syllable sentences. In every listening condition, the T message was presented 40 times to the subject in the presence of 0, 1, 2, and 3 C messages; subjects recorded the T messages. Results demonstrate that the ability to understand the T messages decreases as the number of C messages increases. As expected, speech recognition is better in the anechoic environment than the reverberant environment when C messages are present. Performance is degraded whenever the subjects do not listen with their own ears. [Work supported by the U. S. Army Research Lab.]
Modeling temporal variations in the room impulse response
Authors: Tharakan Nancy T. , Kozloff Anthony , Thompson Charles , Mehta Vineet
Published: July 2002

The objective of this study is to determine the impact of a time-varying room impulse response on the rate of adaptation and accuracy of monaural and stereo echo cancellers. The variation in the impulse response results from secondary echoes generated by moving objects in a room. The time- invariant portion of the response is modeled using the method of images. The time-varying component will be modeled using a Pade approximant- based solution of the Kirchhoff integral equation. Simulation results are given for a room having a volume on the order of 2744 cubic feet. It is shown that for strong acoustic scatterers located closer than the critical distance to the receiver can induce up to a 15-dB residual error for monaural cancellers. Adaptation time requirements for acoustic echo cancellers will also be discussed.
Breakdown of ray perturbation theory due to chaos.
Authors: Tang X. , Tappert F. D.
Published: July 2002

The possibility of ocean acoustic tomography in its usual form is based on ray perturbation theory and the following assumption: Eigenrays computed in an unperturbed ocean model agree one-to-one with eigenrays computed in the actual perturbed ocean. Using a ray numerical model with an environment containing idealized mesoscale structure, the perturbation strength ε is gradually increased and it is found that the multiplicity of eigenrays increases exponentially due to ray chaos. When the multiplicity changes then ray perturbation theory, even to very high order, necessarily breaks down. Bifurcation diagrams show eigenray travel times versus ε, and at critical values of ε pairs of new eigenrays suddenly appear as triplications in accordance with catastrophe theory. Such plots dramatically display the breakdown of ray perturbation theory. The methodology that has been developed should be useful for understanding the limitations of ocean acoustic tomography. [Work supported by ONR.]
Rain and hail simulation and sound isolation tests on a wood and metal roof assembly
Authors: Cooper Russell
Published: July 2002

A series of acoustic tests were performed on a proposed roof construction for a new concert hall at The University of North Texas to evaluate its sound transmission loss properties. The test was conducted to determine if the wood and metal deck construction would be sufficient to isolate nearby highway traffic noise (airborne) and heavy rain and hail noise (structure-borne) from being heard in the concert hall. Four variations of the roof system were constructed and tested to determine the range of possible solutions. Simulation of rain and hail noise test methods and results are presented along with standard ASTM E-90 results and the recommended solution.
Ultrasonic wavefront distortion caused by human abdominal wall layers
Authors: Hinkelman Laura M. , Mast T. Douglas , Orr Michael J. , Waag Robert C.
Published: July 2002

The relative importance of the fat and muscle layers of the human abdominal wall in producing ultrasonic wavefront distortion has been assessed based on direct measurements. Specimens employed included 6 whole abdominal wall specimens and 12 partial specimens obtained by dividing each whole specimen into a fat and a muscle layer. In the measurements, a hemispheric transducer transmitted a 3.75-MHz ultrasonic pulse through a tissue section maintained at 37°. The received wavefront was measured by a linear array translated in the elevation direction to form a two-dimensional aperture with overall dimensions 92.16×46.08 mm2 and a measurement spot size of 0.72×1.44 mm2. Differences in arrival time and energy level between the measured waveforms and computed references that account for geometric delay and spreading were calculated. After correction for the effrects of geometry, the received waveforms were synthetically focused at 180 mm. The characteristics of the distortion produced by each specimen and the quality of the resulting foci were analyzed and compared. The results indicate that both fat and muscle layers contribute significantly to the distortion of ultrasonic beams by the abdominal wall. However, the spatial characteristics of the distortion produced by fat and muscle layers differ substantially, and the total distortion produced by the abdominal wall is not equivalent to the sum of the distortion produced by the layers.
Methods of integral equations in tasks of bio-organs sound scattering
Authors: Genis V. , Oboznenko E. I. , Oboznenko I. L.
Published: July 2002

Ultrasound scattering on biological, arbitrary-form structures is considered. Methods of integral equations by volume and surface integration are used in the task. The methods allow one to explore scattering from the surface and volume of arbitrary pathological insertions. A method of small perturbations is used. In the first approximation the weak-reflecting pathology is accepted. It is noted that pathology form and value appreciably affect echosignal and spectrum structure. In particular, echosignal spectrums for less dense and denser pathologies vitally are notable. The proposed method allows one to classify the pathologies of bio-organs. Sufficient agreement with theoretical models if shown experimentally on the weak-reflecting object (rubber).
Capabilities and limitations of ‘‘state of the art’’ modeling techniques for use in the design of electroacoustic transducers
Authors: Beltran Carlos I.
Published: July 2002

Several modeling systems exist to aid in the design and engineering of electroacoustic transducers. Finite element analysis (FEA) can be used to model magnetic circuits, electromagnetic circuits, thermal transfer capacity, mechanical and dynamic properties of structural parts, vibrational modes of diaphragms and other moving parts, piezoelectrics, acoustic impedance, and acoustic radiation. Several computer programs are available to model electroacoustic systems and components via lumped parameter methods derived from electrical-mechanical and electrical-acoustical analogies. Lastly, general purpose mathematics programs and spreadsheets can be used for a wide variety of problems. The capabilities and limitations of these modeling techniques are discussed using real world examples.
Measurement and analysis of loudspeaker component nonlinearities at low frequencies
Authors: Heed Thomas P.
Published: July 2002

In the constant displacement region, several displacement-dependent quantities control the behavior of electro-dynamic transducers. This research uses a tensile/compression loadframe to measure the displacement-dependent stiffness, B1-Product, and mechanical losses of various driver components. The individual data sets are curve-fitted by power series expansions. The power series coefficients are used to predict the relative low-frequency total harmonic distortion (THD). Then, the measured components are built into loudspeakers, and the predictions are compared to actual THD measurements. Experimentally, this method is very sensitive. Additionally, the effective stiffness of a transducer’s suspension is merely the sum of the power series coefficients representing the spider’s and surround’s stiffnesses. By comparing various spiders and cones, the effective stiffness of a suspension can be linearized, thereby reducing low-frequency THD. Finite-element analysis (FEA) simulation of nonlinear suspension behavior can facilitate such a linearization process. Lastly, power series coefficients can determine the relative variability of components (e.g., which of two spider designs is more variable). OEM manufacturers of transducers must maintain statistical process control of QTot, resonance, and THD. QTot, resonance, and THD are all dependent on displacement-dependent quantities. Therefore, reducing component variability is helpful in achieving an acceptable cpk for QTot, resonance, and THD.
A BEM approach to effectively modeling differential microphones
Authors: Yeager David
Published: July 2002

A flat frequency response, reasonably good signal-to-noise ratio, mechanical robustness, and low cost make omnidirectional electret microphones ideal for telecommunications applications. Differential microphones offer the potential additional advantage of improving the acoustic signal-to-noise ratio by rejecting unwanted background noise. Background noise rejection results from two independent acoustic mechanisms—directional selectivity and proximity. Modeling the acoustic performance of a differential microphone requires accurately representing these mechanisms, which means properly accounting for diffraction and damping effects. For these reasons, differential microphones are more sensitive to mounting conditions and housing geometry, and therefore, designing a differential microphone into a product can be a daunting task. This paper discusses some of the design principles that dictate noise rejection, and suggests methods for modeling differential microphones using numerical techniques such as the boundary element method (BEM). A coupled BEM model enables an acoustical interface between the microphone diaphragm and nearby cavities and ports. This is especially important near resonance. Damping plays a key role in microphone performance, and is an overriding factor in the directional behavior of a differential microphone. Techniques for representing these effects in a BEM model using SYSNOISE will be discussed. [SYSNOISE is a registered trademark of LMS Numerical Technologies, Belgium.]