bluuebunny/crossref_metadata_2025 huggingface dataset:
Full wave simulations of pulsed focused fields in a nonlinear tissue mimic
Authors: Szabo Thomas L. , Clougherty Frances , Wojcik Gregory L. , Mould John C. , Carcione Laura M.
Published: July 2002

In order to simulate focused ultrasound fields in nonlinear tissue, a full wave pseudospectral finite difference model has been developed. Unlike models based on the KZK equation, this model can predict wide-angle propagation effects as well as tissue backscatter, essential for pulse-echo medical imaging. This simulation program is capable of including broadband pulse propagation, causal attenuation, nonlinearity (B/A), and inhomogeneities. Harmonics beyond the solver’s Nyquist limit are removed numerically and quantified. To facilitate comparison to data and to expedite calculation, the model has been adapted for circularly symmetric transducers. As input to the program, waveforms are taken from hydrophone scans close to the faces of 2.25- and 5-MHz focusing transducers. Simulations are compared to data taken in water and in front of and behind a tissue mimicking cylinders of tofu, which are sliced repeatedly to give transmitted and reflected pressure data versus thickness. Tissue inhomogeneities are simulated by voids in the tofu. Acoustic properties of the mimic are measured separately. B/A is extracted numerically from observed harmonic generation versus input pressure and range.
The acoustic field of a clinical pulsed Doppler ultrasound system near lung
Authors: Holland Christy K. , Flament Jeffrey T.
Published: July 2002

Several animal models have exhibited thresholds for petechial hemorrhage in lung exposed to clinical levels of diagnostic ultrasound. Extravasation of red blood cells into the alveolar spaces has occurred consistently near the visceral pleura (at the outer surface of the lung). In order to quantify the ultrasound exposure conditions under which damage occurs, the pressure field of an ATL HDI 3000 in Doppler mode was measured near ex vivo, aerated rat lung in a water bath. The pressure output of the ATL L10-5 linear array in pulsed Doppler mode (6.0 MHz center frequency) was interrogated near the lung with a Sonic Technologies 0.4-mm bilaminar membrane-type PVDF hydrophone. A Velmex 3-axis translation stage was utilized for automated positioning of the hydrophone. Virtual instruments were developed using labview v.5.0.1 on a Macintosh 8100 to sample the pressure waveforms from a LeCroy 9350CL digital oscilloscope via an IEEE 488 interface and National Instruments data acquisition boards. Images of the pressure field in the axial, transverse, and elevational planes near the surface of the lung will be compared to the same measurements in the free field (without the lung present).
Transducer elctronic data sheet implementation within traditional precision microphones and array microphone systems
Authors: Lally Michael J. , Dillon Mike J. , Rasmussen Gunnar , Andersen Peter Wulf
Published: July 2002

Consistent with the IEEE1451 standard development of smart sensors, microphones have been developed that can digitally report critical transducer characteristics. This new, standardized communication ability automates communication of calibration scaling and eliminates potential bookkeeping errors in large channel testing. Discussion includes the framework of the evolving standard, implementation in preamps/array microphone systems and practical examples of error reduction and time savings.
Acoustics at the end of the 20th Century—An overview of the state of the art
Authors: Crocker Malcolm J.
Published: July 2002

Recent years have seen rapid of advances in digital computers, miniaturization of electronics, and the development of new materials. These advances have led to improved knowledge in many scientific fields including a large number of different areas of acoustics. In many cases the developments have been synergistic; new experimental knowledge has led to improved theoretical models and approaches and vice versa. This paper begins with a brief introduction reviewing some of the technological developments that are helping to accelerate the increase in knowledge in different areas of acoustics. It then continues with a general review of the state of the art in a number of specific fields such as: use of finite-element and boundary-element methods in acoustics, computational aeroacoustics, sonochemistry, thermoacoustic engines, active noise and vibration control, sound-intensity measurements, techniques of speech coding and speech recognition, ultrasonics in medical acoustics, and cochlear mechanics. Finally, some projections about future scientific and technological developments in different areas of acoustics are made.a)The author is the editor-in-chief of the recently published Encyclopedia of Acoustics.
A radiosity-based model for simulating sound propagation in urban streets
Authors: Kang Jian
Published: July 2002

To study the fundamental characteristics of the sound field in urban streets resulting from diffusely reflecting boundaries, a theoretical/computer model has been developed using the radiosity technique. The model divides building facades and ground into a number of patches and then simulates the sound propagation in a street by energy exchange between the patches. Computations in a typical street with a single noise source show that with diffusely reflecting boundaries the sound attenuation along the length is significant, and the sound distribution in a cross section is generally even unless the cross section is very close to the source. The effectiveness of some architectural changes and urban design options on further increasing sound attenuation along the length, such as by strategically arranging buildings in a street or adding absorption patches on facades or ground has been analyzed. It has also been demonstrated that by replacing geometrically reflecting boundaries with diffusely reflecting boundaries, the sound attenuation along the length becomes greater, and the reverberation time becomes shorter. This suggests that from the viewpoint of urban noise reduction, it is better to design the street boundaries as diffusely reflective rather than acoustically smooth. [Work supported by the Lloyd Foundation.]
Occupational and leisure noise exposures as risk factors for adult hearing impairment: The Nord Trondelag (NT) Hearing Study
Authors: Hoffman Howard J. , MacTurk Robert H. , Reed George W. , Tambs Kristian , Holmen Jostein , Borchgrevink Hans M.
Published: July 2002

In the NT Hearing Study (1995–1997), air-conduction thresholds were obtained using sound attenuation booths. This report is based on 27997 subjects selected in a nested, case-control study to receive follow-up questionnaires on work and leisure noise exposures. Cases with low-frequency hearing loss (HL)—speech frequencies 500, 1000, and 2000 Hz—were defined by a pure-tone average (PTA) exceeding 30 dB in the better ear. A second case series with only high-frequency HL was defined by PTA exceeding 30 dB at 3000, 4000, and 6000 Hz. Controls were defined by PTAs less than 30 dB in both ears. Multivariate logistic regression analyses were stratified by sex and age. Family history of HL and recurrent ear infections in childhood were significant risk factors across all sex and age strata. Among men under 50, an odds ratio (OR) of 1.87 [95% confidence interval (CI): 1.53–2.28] for high-frequency HL was found for loud noise at work, an OR=1.35 (CI: 1.08–1.69) for use of noisy equipment, and OR=1.09 (CI: 1.02–1.17) for hunting/sport shooting. Low-frequency HL was related to occupational noise exposures in older (50+ years), but not younger men. Tinnitus was associated with occupational noise exposure at all ages.
Design and construction of a small thermoacoustic refrigerator
Authors: Underwood Tamra S. , Smith Dana M. , Muehleisen Ralph T.
Published: July 2002

While there has been much recent work on the development of medium to large scale (hundreds to thousands of watts) thermoacoustic devices, there has been relatively little work done on small thermoacoustic devices. Larger devices are often desired in order to get large cooling powers, but such devices can be expensive and difficult to construct. Small devices on the other hand, can be both inexpensive and easy to construct. To more carefully investigate small thermoacoustic devices, a refrigerator approximately 1 in. in diameter and 5 in. long has been constructed with a target of 10 W of cooling power and a temperature span of 25°C. To achieve higher cooling powers, several of the devices can be operated in parallel. The performance of the actual device and the comparison to the predicted values will be presented. [Work supported by the Office of Naval Research.]
Experimental investigation of the acoustical properties of narrow tubes filled with a mixture of gas and condensable water vapor
Authors: Gaitan D. Felipe , Raspet Richard , Hickey Craig J. , Slaton William
Published: July 2002

It is generally believed that using the latent heat of condensable vapors will increase the efficiency and energy density of thermoacoustic refrigerators. This phenomenon has been recently considered in the literature [Raspet et al., J. Acoust. Soc. Am. 105, 65 (1999)] in which the equations describing the acoustic properties of wet pores are described. In order to investigate some of the assumptions made in these equations (e.g., the boundary conditions at the walls of the wet pore), we have measured the acoustic damping in a small (1.3-mm-radius) tube filled with a mixture of gas and condensable water vapor. The explored parameter space included ambient pressure (0.1–2 bars), temperature (20–100°C), and acoustic frequency (2–10 kHz). Results of these measurements will be reported. [Work supported by the Office of Naval Research.]
Heat transport by acoustic streaming in a resonance tube
Authors: Mozurkewich George
Published: July 2002

The steady fluid flow known as acoustic streaming is capable of transporting heat advectively. This heat-transport mechanism has occasionally been applied for beneficially cooling a warm object, but in other contexts (such as thermoacoustic cooling) its effects can be detrimental. Gopinath and Mills have computed heat transport by streaming between two plane sections of a cylindrical resonator with insulating walls. In contrast, the experiments described here treat the interaction between streaming gas and thermally conductive cylinder walls. The walls of a horizontally oriented resonance tube were instrumented with a heater and more than 20 thermocouples to permit quantification of axial conduction through the solid walls and radially outward natural convection. By applying conservation of energy to the tube walls, the local, radially inward heat current, q(x), was deduced. The quantity q(x) is related to streaming transport within the tube. The measurements delineate regions where heat is added to and removed from the streaming gas. Although the magnitude of q initially increases with increasing acoustic amplitude, it quickly saturates. The fluid-dynamical implications of amplitude-independent q(x) will be discussed.
Pitch strength of regular interval stimuli as a function of duration and filtering
Authors: Yost William A.
Published: July 2002

The ability of listeners to discriminate between a random interval noise and a regular interval stimulus, iterated rippled noise (IRN), was measured as a function of duration and bandpass filtering. IRN is generated by delaying (by d ms) a noise and adding it back to the undelayed noise after attenuation (g) in an iterative process (where n is the number of iterations). A two-alternative, forced-choice adaptive procedure was used in which stimulus duration was varied adaptively. Different values of g, d, n, and filter condition were used. In general, the duration required to discriminate IRN from regular interval noise increased as the filter condition provided less low-frequency energy. Duration thresholds were higher for high values of attenuation. The smallest duration thresholds were obtained for d in the range of 2–5 ms. The results will be discussed in terms of temporal models of the pitch of complex stimuli. [Work supported by NIDCD and AFOSR.]
Infants’ discrimination of spectral slope
Authors: Tsang Christine D. , Trainor Laurel J.
Published: July 2002

Spectral slope is a global property of the spectral envelope representing the change in energy across spectral frequency. Spectral slope is a dimension for discriminating speech, voices and musical instrument timbres. The average spectral slope of speech and music is about −6 dB/octave. If spectral slope is important in timbre perception, enhanced sensitivity for spectral slopes around −6 dB/octave would be expected, and poor discrimination would be expected for spectral slopes very different from this value. Eight-month-old infants were tested across a range of spectral slopes (−16 to +16 dB/octave) to determine whether infants’ slope discrimination is best near −6 dB/octave. The stimuli were generated according to the following equation: dBi=blog2i, where i is the harmonic number, b is the spectral slope value, and dBi is the relative amplitude of harmonic i. The results show that infants were only sensitive to differences in spectral slope in a limited range between −10 and −4 dB/octave, despite the fact that local intensity cues were smallest in this range. This suggests that infants are using spectral slope as a dimension for discriminating timbre and that the auditory system is tuned to sounds with real-world spectral slopes. [Research supported by NSERC.]
The measurement of structural intensity on a plate with nonuniform thickness
Authors: Szwerc Richard , Chang Henry , Burroughs Courtney
Published: July 2002

A method to measure the structural intensity on a plate with non-uniform thickness is presented and demonstrated with experimental results. The method is an extension of the wave decomposition method, previously demonstrated on uniform thickness beams. It treats a plate as a superposition of beams, and relies only upon local rather than global vibration measurements. The average thickness in the local region of the measurement points is the plate thickness used in the calculations of the power levels.
The role of modulation spectrum amplitude and phase in consonant intelligibility
Authors: Aiken Steven J. , Jamieson Donald G. , Parsa Vijay , Allen Prudence
Published: July 2002

This paper considers an acoustic basis for speech intelligibility and evaluates various acoustically based speech intelligibility prediction algorithms. Earlier research indicates that speech intelligibility does not require preservation of spectral and temporal fine-structure, but is highly dependent on the preservation of the amplitude component of the modulation spectrum [R. Drullman, J. Acoust. Soc. Am. 97, 585–592 (1995)]. This study assessed the importance of the phase component of the modulation spectrum using a 21-alternative forced-choice consonant perception test. Temporal and spectral fine-structure were removed by modulating a white noise carrier with 50 Hz low-pass filtered speech amplitude envelopes in 4, 8, or 24 discrete bands. Modulation spectrum phase was distorted by imposing a random delay in each discrete band. Behavioral results are discussed in light of intelligibility predictions generated by the articulation index [N. R. French and J. C. Steinberg, J. Acoust. Soc. Am. 19, 90–119 (1947)] and the speech transmission index [H. J. M. Steeneken and T. Houtgast, J. Acoust. Soc. Am 67, 318–326 (1980)].
Cues to vocal-fold and vocal-tract lengths interact in talker-sex perception from vowel segments
Authors: Owren Michael J. , Bachorowski Jo-Anne
Published: July 2002

Even brief sounds can be richly informative about talker characteristics. Earlier acoustic analysis confirmed predictions that the best statistical classification of vowels by talker sex occurred using acoustic correlates of dimorphism in vocal-fold and vocal-tract lengths [J.-A. Bachorowski and M. J. Owren, J. Acoust. Soc. Am. (in press)]. In follow-up perceptual testing, 24 participants in experiment 1 classified talker sex from 180 short segments that balanced fundamental frequency (F0) and estimated vocal-tract length (VTL) cues. Responses were 98% correct, with a mean latency of 478 ms. However, latencies were longer for male talkers that combined high F0 and short VTL values, as well as for female talkers with low F0 and long VTL. In experiment 2, 24 participants heard stimuli composed of small numbers of waveform cycles. Classification ‘‘threshold’’ was 1.8 cycles, with male talkers being more accurately classified than females. These results show that indexical cueing is an inherent component of even the shortest possible vowel segments, and that the acoustic features most closely related to sexual dimorphism interact in predictable fashion in influencing perceptual processing. F0 and VTL, the particular acoustic cues in question, are fundamentally related to features also known to influence speech processing.
Spoken word recognition and the influence of lexical competitors at the beginning of a word
Authors: Duenas Nadia , Vitevitch Michael
Published: July 2002

Marslen-Wilson and Welsh (1978) suggest that the initial portion of a word activates multiple lexical candidates in memory, forming a ‘‘cohort’’ of competitors. Marslen-Wilson (1987) further states that the number of competitors in the cohort does not affect the speed and accuracy of spoken word recognition. Monosyllabic CVC words with the same number of lexical competitors (using a substitution-based metric to estimate competitor set size) but varying in the percentage of competitors sharing the same initial phoneme as the target word were presented in an auditory naming task and an auditory lexical decision task. Words with a smaller proportion of competitors sharing the same initial phoneme as the target word were responded to more quickly than words with a larger proportion of competitors sharing the same initial phoneme as the target word. These results suggest that the number of candidates activated in memory does affect spoken word recognition, and that the initial portion of a word is important in processing. The implications of these results for models of spoken word recognition are discussed. [Work supported by NIH-NIDCD Training Grant DC-00012.]
Normalizing inter-band scaling in minimum variance (MV) beamformer estimates of variable-bandwidth filtered wideband signals
Authors: Drake Laura A. , Katsaggelos Aggelos , Zhang Jun
Published: July 2002

Narrow-band minimum variance (MV) beamforming can be used to estimate a wideband signal when the signal is first filtered into several narrow-band signals. This paper identifies and solves a new problem for narrow-band MV beamforming of wideband signals. ‘‘Inter-band scaling’’ can occur when narrow-band MV beamforming is applied to a wideband signal filtered by a variable-bandwidth filterbank. This situation can occur when the wideband signal to be estimated is speech. Then, it is often desireable to use a filterbank with higher-frequency resolution in the low, than in the high-frequency bands since speech has more information in low frequency than high. In this case, the power out of the MV beamformer in each band can depend not only on the actual signal power in that band, but also on the bandwidth. The inter-band scaling normalization method presented here is tested with both white Gaussian noise, and a segment of a speech signal. The tests show that the method is effective and does not distort the speech signal. Finally, this method should be extendable to other problems (such as other adaptive array processing methods) that require estimates of statistical measures of variable-bandwidth filtered signals.
Constrained maximum likelihood estimation of the principal curvatures on the object surface in an ultrasonic tactile sensing
Authors: Teramoto Kenbu , Mori Noriko
Published: July 2002

The maximum likelihood estimation (MLE) provides a robust solution in matched-field imaging (MFI), which has evolved from various underwater acoustic applications successfully. In the identification of the principal curvatures during the medical or robotic applications of acoustic tactile sensing, matched field processing offers a reasonable detection of the reflected wavefront. Two major difficulties, however, are there in the principal curvature identification. The first reason is that the wavefront reflected by the elliptic paraboloidal or the hyperbolic paraboloidal surface cannot be described in the combination of the plane waves nor the spherical waves strictly. The second is that the principal curvature identification process becomes ill posed due to the nonlinear relationship between the principal curvatures and propagation time of flight. The MFI scenario, therefore, can solve the nonlinear optimization problem in order to identify the curvature. In this paper, the proposed identification algorithm seeks the unique KKT (Karush–Kuhn–Tucker) point in the augmented Lagrange function of the constrained likelihood function, which is defined over the observed signal field. Furthermore, several acoustical experiments show that the proposed tactile sensor can identify the principal curvatures of the following surface cases: (1) plane, (2) paraboloid, (3) elliptic paraboloid, and (4) hyperbolic paraboloid.
A real-time dual-microphone signal-processing system for hearing aids
Authors: Elledge Mark E. , Lockwood Michael E. , Bilger Robert C. , Goueygou Marc , Jones Douglas L. , Lansing Charissa R. , O’Brien William D. , Wheeler Bruce C.
Published: July 2002

A real-time dual-microphone-based signal-processing system has been developed that suppresses off-axis interference from multiple sound sources. The system employs optimal filtering methods to attenuate the amplitude of all signals other than that from a desired receive direction. The algorithm was implemented on a Texas Instruments TMS320C62 fixed-point digital signal-processor evaluation board installed in a PC. Input/output was accomplished via a 16-bit codec controlled by optimized assembly language routines. To simplify algorithm coding, the main algorithm was written in C and executes using approximately 55 processor time, allowing for future additions to enhance performance. It has a delay of 15 ms. The system can significantly improve the intelligibility of speech in noisy environments, as evidenced in off-line tests by a marked intelligibility weighted SNR gain [Link and Buckley, J. Acoust. Soc. Am. 91, 1662 (1992)]. Initial real-time testing with actual binaural signals supports the conclusion that this performance gain will be maintained. The results show that this system may be used as a main building block for hearing aids to be used in noisy environments. [Research supported by the Beckman Institute.]
Responses of Baleen whales to controlled exposures of low-frequency sounds from a naval sonar
Authors: Tyack Peter L.
Published: July 2002

Playback experiments were conducted from 8–27 January 1998 using a moored SURTASS-LFA sound source to study behavioral responses of gray whales migrating off Pt. Buchon, California. Shore stations operated for over 150 h during 18 days, tracking about 1200 migrating whales. When the source was moored inshore near most migrating whales, whales avoided exposure to 42-s sound stimuli in the 160–330-Hz frequency band repeated every 6 min at received levels near 120 dB (playback source levels 170–185 dB re: 1 μPa at 1 m). Responses to the source broadcasting offshore at source levels of 185–200 dB were greatly reduced. Playbacks to humpback whales singing in Hawaiian waters occurred in February–March 1998. 17 playbacks and 5 control follows of singing humpbacks were conducted from a small vessel. Singers continued to sing throughout 7/17 playbacks, and stopped during the remaining 10 playbacks. During 4/10 playbacks when singers stopped, the singer stopped when it joined with another whale. The remaining 6 cessations of song were considered possible responses to playback. Most of these whales resumed normal behavior before the hour-long playback ceased. There is no trend for the vocal response to playback to scale with exposure level; both the ‘‘vocal-response’’ and ‘‘no-vocal-response’’ categories ranged in exposure from 120–150 dB; re: 1 μPa.
Marine mammal research program for the Pioneer Seamount ATOC experiment
Authors: Costa Daniel , Calambokidus John
Published: July 2002

Aerial surveys and satellite and archival tags were used to test the effect of the ATOC sound source on marine mammals around the Pioneer Seamount 85 km west of San Francisco, California. Control surveys were flown at least 48 h after the end of any previous transmission cycle and experimental surveys were flown after at least 24 h of sound transmissions. Sound transmissions consisted of 20-min periods of 195-dB, that is, 1-μP transmission repeated very 4 h. Most commonly sighted species by group were: (1) mystecetes: humpback whale (372 sightings), (2) large odontocetes: sperm whale (337 sightings), (3) small odontocetes: Pacific white-sided dolphin (306 sightings), and (4) pinnipeds: California sea lion (167 sightings). Although there were no significant differences in the number of sightings when the sound source was on or off, both humpback and sperm whales were generally seen farther from the sound source during experimental versus control surveys (p<0.01). The highest intensity of sound measured on tags carried by elephant seals during transmissions ranged from 118–137 dB for 60–90 Hz compared to ambient levels of 87–107 dB (60–90 Hz). On a gross level, animals did not alter return track or go to the surface, and often continued to dive closer to the sound source if on the descending segment of a dive evident. Some animals with the highest levels of exposure showing no effects while two animals showing minor changes in diving pattern had exposure levels greater than 132 dB. [Work supported by the Strategic Environmental Research and Development Program through RPA and the Office of Naval Research.]
The influence of seismic survey sounds on bowhead whale calling rates
Authors: Greene Charles R. , Altman Naomi S. , Richardson W. John
Published: July 2002

During September in 1996–98, airgun arrays operated close to the coast of the Alaskan Beaufort Sea. It was of interest to know if the airgun pulse sounds affected migrating whales. Autonomous seafloor acoustic recorders were installed at various distances from shore and from the airguns. They recorded continuously for up to 22 days. Both the bowhead calls and the airgun pulses were clustered in time, requiring a greater quantity of data to isolate an influence than would be the case with nonclustered data. The consistent results from the three years were: (1) bowhead whales called frequently during their migration through the study area; (2) calling continued, although possibly at different rates, when the whales were exposed to airgun pulses; and (3) call detection rates at some locations differed significantly when airgun pulses were detectable versus not detectable. At least in 1996, patterns of call detection rates were consistent with aerial survey evidence that most bowheads were displaced from the area near the operating airguns. However, there was no significant tendency for call detection rates to change consistently at times when airgun operations started or stopped. [Work supported by BP Exploration (Alaska) and by Western Geophysical.]
150 years of free reed research: The acoustics of the reed organ and harmonium from Helmholtz to the present
Authors: Cottingham James P.
Published: July 2002

Free reed instruments have been known in Asia for thousands of years, but the Western free reed instruments such as the harmonica, accordion, and reed organ have only developed within the last 2 centuries. The birth of Hermann Helmholtz in 1821 coincided rather closely with the early stages of the development of these free reed instruments in Europe and North America, so that by the time of his work in acoustics these instruments were fairly common. A summary of acoustical research on free reed instruments from the time of Helmholtz to the present will be presented, with emphasis on the reed organ and the harmonium. The properties of these and other Western free reed instruments will be contrasted with those of the Asian free reed mouth organs: the sheng, sho, and khaen. Some current and future directions for free reed research will be discussed.
The history of free reed organ stops
Authors: Ahrens Christian , Braasch Jonas
Published: July 2002

Organ stops based on free reeds have been introduced by the end of the eighteenth century. Although their origin has not been proven, it is most likely that the technique of imported East Asian free reed instruments, probably the Japanese sho or the Chinese sheng, was adapted. Organ stops with free reeds became soon popular in Europe, especially in Germany and Switzerland and often replaced stops, like the clarinet for example, which were traditionally established with beating reeds. With the beginning of the ‘‘Orgelbewegung’’ in the 1920s free reed stops were abandoned, because they were not in accordance to the barock organ ideal that was maintained by the protagonists of the ‘‘Orgelbewegung.’’ The development of the use of free reed pipes is compared to the development of orchestra music and instruments of the orchestra, especially to the development of the clarinet and the improvement of clarinet playing techniques. Acoustical analyses of organ stops and orchestra instruments—mainly attack transients and Fourier spectra—were done and compared with the historic investigations of Helmholtz and Töpfer/Allihn, for an attempt to explain the evolution of free reed organ stops.
The impact of mass diffusion on SBSL dynamics in a variable-gravity environment
Authors: Wyatt Sean C. , Thomas Charles R. , Raymond Jason C. , Roy Ronald A. , Holt R. Glynn
Published: July 2002

Recent experiments performed on a KC-135 parabolic flight aircraft [Matula et al., J. Acoust. Soc. Am. 102, 3185 (1997); Young et al., ibid. 103, 3077 (1998)] indicate that the optical emission intensity of SBSL (air bubble in water) increases with decreasing gravitational acceleration. Wyatt et al. [J. Acoust. Soc. Am. 104, 1770 (1998); 105, 960 (1999)] argue that this effect is primarily due to small changes in the equilibrium radius for diffusively stable pulsations. This concept is further explored in a series of variable-acceleration experiments employing a water-filled acoustic resonator in which individual air bubbles are levitated. The critical role of mass diffusion is assessed by varying the dissolved gas concentration. Results are presented and compared with predictions from numerical simulations. [Work supported by NASA.]
Dynamics of laser-induced cavitation bubbles
Authors: Akhatov Iskander , Vakhitova Nailya , Zakirov Kamil , Mettin Robert , Lindau Olgert , Lauterborn Werner
Published: July 2002

Bubbles in liquids can be formed in a variety of ways. A convenient method to produce a single bubble in a liquid is to focus a short pulse of laser light into the liquid. A bright light-emitting plasma of obviously high pressure expands forming a gas-vapor bubble. After some time bubble expansion stops at some maximum radius. From thereon the bubble starts to shrink driven by the ambient pressure and leading to a violent collapse possibly followed by rebounds. A mathematical model that describes the evolution of such a laser-induced cavitation bubble is presented. The model takes into account heat conduction of the liquid and gas phases, evaporation and condensation processes of the liquid phase, gas diffusion from the liquid into the bubble, and compressibility of the liquid. A reasonable comparison of theoretical results with experimental data is obtained. [Work partially supported by Internationales Büro des BMBF (Contract RUS-133-1997) and European Commission (INCO-Copernicus, Contract ERBIC15CT980141).]
Numerical comparison of acoustic infinite elements
Authors: Cipolla Jeffrey
Published: July 2002

Numerical studies of various infinite element formulations are conducted: conjugated types, unconjugated, decay (unconjugated), and DAA1 elements of the Olsen/Bathe type. In addition, an ‘‘element’’ implementing the Bayliss et al. first-order condition is studied. All elements except the decay are implemented in separable coordinates. The element formulations are studied analytically, using axisymmetric modal impedances of a sphere. These studies reveal limitations, advantages, and appropriate methods of usage for the various formulations. Subsequently, the various elements are implemented in production-type software, and numerical comparisons are conducted. These studies reveal the relative accuracy of the formulations as well, but also permit discussion of numerical performance. Fully coupled, three-dimensional structural acoustic meshes are studied. Comparisons are made against a BEM method, and analytic solutions where appropriate. When used properly, infinite element methods can deliver accuracy comparable to the BEM for three-dimensional acoustic problems, at considerably reduced cost. However, this study reveals the superiority of some infinite element formulations.
Probabilistic phonotactics and levels of processing and representation in spoken word recognition: When is a word not a word?
Authors: Vitevitch Michael
Published: July 2002

Vitevitch and Luce (1998, 1999) hypothesize that two levels of processing—lexical and sub-lexical—are involved in spoken word recognition. Competition is observed at the lexical level: Words and nonwords with low phonotactic probability/sparse neighborhoods are responded to more quickly than stimuli with high-phonotactic probability/dense neighborhoods. Facilitation is observed at the sub-lexical level: Nonwords with high-phonotactic probability/dense neighborhoods are responded to more quickly than nonwords with low-phonotactic probability/sparse neighborhoods. Nonwords can be processed at either level, whereas words predominantly compete with each other at the lexical level. However, the present experiment demonstrates that words can be processed at a sub-lexical level. Target words varying in phonotactic probability/neighborhood density were presented in a same–different task with varying proportions of word and nonword pairs. When the stimulus set consisted mostly of word pairs, competition was observed: Words with low-phonotactic probability/sparse neighborhoods were responded to more quickly than words with high-probability/dense neighborhoods. When the stimulus set consisted mostly of nonword pairs, facilitation was observed: Words with high-phonotactic probability/dense neighborhoods were responded to more quickly than words with low-probability/sparse neighborhoods. The implications of these results for models of spoken word recognition are discussed. [Work supported by NIH-NIDCD Training Grant DC-00012.]
Wind-generated, low-frequency, ambient-noise source levels revisited
Authors: Carey William M. , Browning David G.
Published: July 2002

A central issue concerning high-resolution array processing is the limit due to natural sources of ambient noise and how can it be calculated. This question has been addressed by several investigators [W. Carey et al., J. Acoust. Soc. Am. 84, 201 (1988); D. Kewley et al., J. Acoust. Soc. Am. 88, 1894–1902 (1990); F. Jensen et al., Computational Ocean Acoustics (AIP, Woodbury, NY, 1994), pp. 517–547] and is still important to the calculation of absolute ambient noise levels. This paper revisits our previous discussion [W. Carey et al., J. Acoust. Soc. Am. 84, 201 (1988)] and reviews recent developments in the determination of wind-driven source levels. Source-level conventions used by several investigators are reviewed and source-level measurements for shallow- and deep-water are summarized with respect to frequency, wind speed, and measurement technique. The presentation of wind-driven noise sources in normal mode, fast field, and parabolic equation computer codes are considered with respect to the frequency-dependent noise source. The order (monopole, dipole, quadrupole) of the source of sound and the use of spatial correlation functions shall be discussed.
Investigation of generation of low-frequency fields produced by parametric acoustic array, located in the waveguides with rough surfaces
Authors: Gryaznova Irina Yu. , Gurbatov Sergey N. , Egorychev Sergey A. , Kurin Vasily V. , Kustov Leonid M. , Pronchatov-Rubtsov Nikolay V.
Published: July 2002

Under natural conditions the parametric acoustical arrays (PAA) work as a rule near the bottom or near the sea surface. If there are free surfaces in the area of interacting of the high-frequency waves (HFW) then the multi-lobe directivity pattern of the PAA is formed in the horizontal plane. However, these effects appear only in waveguides which are bordered smooth surfaces, because these boundaries do not violate conditions of coherence for interacting waves. If there is a rough surface in the area of interacting of HFW, it leads to limit the length of PAA. In this case a single-lobe directivity pattern on the low-frequency (LF) is formed [Gurbatov et al., Radiophys. Quantum Electron. 38, 82–86 (1995)]. In this paper the results of laboratory investigations of the LF fields angular distribution in waveguides with the rough surfaces are presented. It is shown that the rough boundaries leads to increasing of angular distributions width in composing with similar characteristics in boundless space, and the quality kind of field distribution in the horizontal plane depends not only on its mode structure, but on characteristics of rough waveguide boundaries. [Work supported by RFBR Grants No. 99-02-18354, No. 98-02-16402, INTAS Grant No. 11134.]
Auditorium and shell renovation at DeVos Hall, Grand Rapids, MI
Authors: Myers Joseph W. A.
Published: July 2002

DeVos Hall (Grand Rapids, MI) is a good multi-purpose hall that had some acoustic shortcomings, particularly in its function as home to the Grand Rapids Symphony, and was renovated to improve acoustic performance for orchestral use. The renovation combined modest modifications to room surfaces with a complete rethinking and reworking of the orchestra shell. Significant improvement was achieved at modest expense and without compromising the hall’s use for nonorchestral functions.
Decay mechanisms for large underwater bubble oscillations
Authors: McDonald B. Edward , Holland Charles
Published: July 2002

Acoustic signatures from centimeter-scale underwater bubble oscillations (primarily from lightbulb implosions) have been found to exhibit much higher decay rates than can be explained using viscous, thermal, or radiative mechanisms which apply to microbubbles. Where does the energy go? If one assumes energy transfer to shape oscillations (surface capillary waves) of large amplitude in subharmonic resonance with the breathing mode [M. S. Longuet-Higgins, J. Acoust. Soc. Am. 91, 1414 (1992)], the shape oscillations can drive fluid motions outside the bubble capable of exciting turbulent instabilities. Application of an appropriate eddy viscosity from mixing length theory to the viscous decay mechanism appears to offer a credible mechanism to explain the observed large decay rates. We give an analysis to show that energy is transferred from the breathing mode to surface capillaries fast enough to make the proposed decay mechanism viable. [Work supported by Saclantcen.]
Preliminary result of Central Equatorial Pacific Ocean Tomography Experiment
Authors: Fujimori Hidetoshi , Nakano Iwao , Nakamura Toshiaki , Kanaizumi Tomoyuki , Howe Bruce M. , Kamoshida Takashi , Yuan Gang
Published: July 2002

JAMSTEC (Japan Marine Science and Technology Center) and APL (Applied Physics Laboratory, University of Washington) started a joint program to directly measure part of the circulation cell of the Central Equatorial Pacific Ocean using acoustic tomography. The tropical and subtropical Pacific Ocean are suggested to connect via a shallow meridional circulation cell by Gu and Philander (1997). This subtropical cell (STC) has been hypothesized to the El Nino-Southern Oscillation phenomena in the tropics, which then, in turn, can affect the subtropical ocean via rapid atmospheric forcing. But no direct measurements exist and indirect measurements and modeling support the hypothesis. The observational centerpiece is tomography systems located in the Central Equatorial Pacific Region. JAMSTEC began the field work by deploying five acoustic transceivers in the observational region and this work will continue until January 2001 (two-year plan). Measurement data were sent to JAMSTEC in realtime using satellite telemetry system. These data were analyzed and compared with CTD observations. The time series fluctuation of the temperature was solved by this analysis.
Acoustic thermometry data compared with ocean models: The importance of ENSO in modifying the ocean interior
Authors: Spiesberger John L. , Hurlburt Harley E. , Johnson Mark , Keller Mark , Meyers Steven , O’Brien James J.
Published: July 2002

The difficulties in detecting anthropogenic changes in ocean temperature due to the industrial revolution are discussed using acoustic thermometry data taken between Oahu and seven SOSUS receivers at distances of 3000–4000 km. Measurements were made in late 1983, and over two 5-month intervals between 1987 and 1989. The travel times are dominated by interannual fluctuations. Two hydrodynamic ocean models are used to identify plausible oceanic features that could cause these variations. Modeled El Niños and La Niñas exhibit oceanic teleconnections between the Equator and mid-latitudes that lead to Rossby waves that propagate westward at mid−latitudes. Rossby waves are the dominant model features which affect the modeled acoustic travel times, and hence section-averaged temperatures in the eastern North Pacific. Modeled predictions of the travel times are significantly different than some of the data. The magnitude and rates of changes of the data are the same as those which result from the effects of the Rossby waves. The largely unknown large−scale variability of the ocean at interannual and decadal scales is a principal problem for detecting possible changes in temperature due to human activity. Effects from eddies, seasonal cycles, and internal waves are of second order.
Summary of some of the tools for mid-frequency analysis
Authors: Cuschieri Joseph M.
Published: July 2002

Over the past few years various attempts have been made to identify solutions/approaches that can address the mid-frequency regime. The mid-frequency regime in this paper is being defined as that frequency range which lies above the frequency of the first few modes (5 to 10), and below the frequency at which the modes start to strongly couple. The definition of strong coupling between modes is somewhat arbitrary and an alternative to define the high-frequency bound would be that, in broadband analysis (one-third octave), there are at least five or more, or a sufficient number of modes, within the one-third octave frequency band, such that the response of the system can be estimated in the one-third octave frequency band, using broadband type tools, to within ± 3 dB (also may be considered arbitrarily selected). In this presentation, a summary of some techniques and results that have been identified to address the mid-frequency regime, as defined above, will be presented, together with an overview of results that have been generated under a study to address structure-borne analysis techniques.
Comparison of robust matched-field processing algorithms in a real, shallow water environment
Authors: Kim K. H. , Hodgkiss W. S.
Published: July 2002

In recent years, a number of matched-field processing (MFP) algorithms have been proposed to mitigate the ever-present problem of mismatch. This paper compares the source localization performance of six processors—Bartlett, Midpoint, Minimum Variance (MV) Distortionless Response, White Noise Constrained, MV with Neighboring Location Constraints, and MV with Environmental Perturbation Constraints—in a real, shallow water environment. MFP was performed on an 84.4-m aperture, 16-element subset of a MPL vertical line array deployed from the R/P FLIP west of Point Loma in approximately 200-m water. A continuous wave source, emitting ten tonals from 53–197 Hz, was towed over both range-independent and range-dependent tracks at source ranges of 1–7 km and at source depths of 60–130 m. First, MFP results utilizing a detailed geoacoustic model of the region, water column sound speed profiles measured during the experiment, and a priori knowledge of array tilt, demonstrate general processor characteristics. Second, MFP results which assume: (1) reasonable though further mismatched geoacoustic parameters; (2) water column sound speed profiles based upon historical data; and (3) no a priori knowledge of array tilt, illustrate processor performance under various mismatch conditions likely to be encountered in realistic applications.
Application of current speech recognition technology to nonstandard domains
Authors: Kewley-Port Diane
Published: July 2002

The typical input to automatic speech recognition (ASR) algorithms is a word-length or longer acoustic waveform. The typical output consists of names of items identified from stored vocabularies. Most algorithms also use scoring procedures to find the top matches between the input and the stored information. These evaluation scores, probabilities or distance metrics, may also be output. There are many nonstandard applications of ASR technology that combine identification and evaluation scores. Examples to be discussed include speech training to improve intelligibility, spoken language proficiency of non-native speakers, and language training for adults with developmental disabilities. Most current ASR algorithms are fine tuned for species-specific properties of human speech and language. In addition, many incorporate psychophysical properties of the human auditory system in the initial signal processing. However, the underlying pattern recognition algorithms are also applicable to a wide range of animal vocalizations. Some commercial and laboratory systems will be discussed in relation to nonstandard applications of ASR. [Work supported by NIH, RO1 DC-02229, R44 DC-02213, and R43 HD-35425.]
Geoacoustic inversion techniques applied to field data
Authors: Lingevitch Joseph F. , Collins Michael D. , Dacol Dalcio K. , Nicholas Michael , Perkins John S.
Published: July 2002

Reliable and efficient techniques for solving geoacoustic inverse problems have been developed. A coordinate rotation technique can be used to identify the best resolved combinations of parameters for a given frequency and configuration of hardware [M. D. Collins and L. Fishman, J. Acoust. Soc. Am. 98, 1637–1644 (1995)]. Problems involving a vertical array of receivers at short range can be solved in a few seconds with an implementation of the self starter that is based on the method of undetermined coefficients [D. K. Dacol and M. D. Collins, J. Acoust. Soc. Am. (accepted for publication)]. These techniques are applied to data that were obtained in the Straits of Florida during the KWIX ’98 experiment. [Work supported by ONR.]
Estimation of anisotropic sediment parameters
Authors: Fredricks Andrew J. , Siegmann William L. , Collins Michael D. , Lingevitch Joseph F.
Published: July 2002

Parabolic equation techniques have recently been developed for solving geoacoustic inverse problems [R. J. Cederberg and M. D. Collins, IEEE J. Ocean Eng. 22, 102–109 (1997)] and for handling propagating problems involving anisotropic elastic layers [A. J. Fredricks, J. Acoust. Soc. Am. 101, 3182 (1997)]. These techniques are used to investigate inverse problems involving anisotropic elastic layers. Complications arise from the direction dependence of the two sediment sound speeds. For the particular case of transverse isotropy, both sound speeds need to be estimated in two different directions. A coordinate rotation technique [M. D. Collins and L. Fishman, J. Acoust. Soc. Am. 98, 1637–1644 (1995)] will be employed to estimate the resolvability of the direction-dependent wave speeds for different experimental configurations. [Work supported by ONR.]