bluuebunny/crossref_metadata_2025 huggingface dataset:
A simple two-projector procedure for producing impedance-controlled wave fields for transducer calibration
Authors: Forsythe S. E. , Van Buren A. L.
Published: July 2002

When calibrating hydrophones in an open body of water, it is sometimes desirable to establish a local impedance in the vicinity of the hydrophone that is accurately known. Three examples that are often used are (1) an impedance characteristic of a plane wave in open water (the so-called rho-c condition), (2) a pressure-only field with no local particle velocity (infinite impedance), and (3) a velocity-only field with no local pressure (zero impedance). This presentation describes a simple apparatus consisting of two spherical sources (or sources that appear spherical at the frequency of interest) and the description of the ratio of drive amplitudes required to produce the desired impedance condition. The useful ensonified volume, that region over which the desired impedance condition holds, is described. Examples of application to the calibration of pressure-sensitive and pressure-gradient hydrophones are given. [Work supported by ONR.]
Underwater facility for automated experimentation and measurement
Authors: Ranz Carlos , Cobo Pedro
Published: July 2002

The underwater tank at the Instituto de Acustica, CSIC (7.5×4.5×4.5 m), was built in 1970; it had facilities useful for the work expected at that time. The tank had two bridges. In both bridges only the vertical coordinate and the 360° turn were automated. In current experimental work, including calibration, the actual instrumentation and the sensors’ motion control, together with signal acquisition and manipulation, showed the need for a drastic change in the tank structure. The mechanisms of motion of both bridges are now fully automated and can be controlled via software. This control can imply positioning at a given field point, for any or both sensors, or along a specified trajectory, including mapping, in the four coordinates X, Y, Z, and angle. The new installation includes the simultaneous signal acquisition, the specific design of ‘‘calibration pages,’’ files management, storing of full working sessions, and remote control, e.g., through the web, of the whole tank, including operation and signal acquisition and management. The tank is expected to be inserted in the European facilities network. The new installation, funded by the PN-CYTMAR and CSIC, is described in this presentation.
Calibration and testing of a neutrally buoyant u-u intensity probe
Authors: Bastyr Kevin J. , Lauchle Gerald C.
Published: July 2002

The theory of operation and the methodology for calibration of an underwater acoustic intensity probe are explained. The intensity probe which is classified as a u-u probe, consists of two moving coil velocity sensors mounted separately in neutrally buoyant, collinearly oriented bodies. This two-velocity sensor probe is used to determine intensity in a manner analogous to that of pressure gradient hydrophones. The acoustic intensity is shown to be the imaginary part of the cross spectrum measured between the output voltages of the velocity sensors, scaled by a complex, frequency-dependent calibration constant. Due to the vector nature of intensity, these complex calibration constants contain magnitude and phase information. To establish the calibration constant, an experiment was conducted that employed the u-u intensity probe and a reference pressure transducer in a standing wave tube. The intensity in the tube is known, and measurements made of this field with the u-u probe are in agreement with the prediction. [Work supported by the Office of Naval Research, Multi University Research Initiative on Acoustic Transduction, Materials, and Devices.]
Calibration of a neutrally buoyant p-u intensity probe
Authors: McConnell James A. , Lauchle Gerald C.
Published: July 2002

Theory and test methods are presented for calibrating an underwater acoustic intensity probe containing two pressure hydrophones and one moving coil velocity sensor. The intensity probe is classified as a p-u probe and is neutrally buoyant in fresh water. In contrast to p-p and u-u probes, it determines intensity via direct measurement of the acoustic pressure and particle velocity. Specifically, intensity is computed by adjusting the time-averaged rms cross spectrum between the transducer output voltages by an appropriate calibration curve. Because intensity is a vector quantity, both a magnitude and phase calibration is required. This is accomplished by a straightforward technique involving a single reference hydrophone and a priori knowledge of the acoustic field where the calibration is performed. A simple experiment was conducted using a standard-wave tube to verify the effectiveness of the calibration technique. In this experiment, the pressure and velocity sensors were calibrated, then an intensity measurement was made. Results show that the calibration technique is highly effective. [Work supported by the Office of Naval Research and the Naval Air Warfare Center Aircraft Division via the DoD Small Business Innovation Research Program.]
An optical transducer for the study of the pressure field around a collapsing cavitation bubble
Authors: Emmony David C. , Alcock Robin D.
Published: July 2002

An optical reflection transducer using a critical angle technique has been built to study the pressures developed around a single cavitation bubble in water. The bubble was generated at different distances from a planar glass surface, and total internal reflection at the glass water interface gives the pressure through the change in refractive index in both the water and the glass. A simple hemispherical pmma block system has been built to prove the feasibility of the system. In this case the change of the critical angle is not linear, but the use of a modified equilateral glass prism gives the pressure as a function of time around cavitation bubbles with maximum radii of approximately 1.0 mm and gamma values down to 0.3 with a temporal resolution only limited by the photodiode detection electronics (∼10 ns). The transducer shows the development of positive pressure as well as the migration of the bubble to the surface and is capable of recording the rarefaction waves due to free surface reflection.
Description of nonlinear planar traveling waves in a gas filled tube
Authors: Bednarik Michal
Published: July 2002

This paper deals with problems for the description of nonlinear planar traveling waves in a gas-filled tube. These waves, which are influenced by the tube-wall boundary layer, can be described by both the Burgers equation and the Khokhlov–Zabolotskaya–Kuznecov’s equation with the inclusion of a hereditary integral known as the fractional derivate of order1/2 . Emphasis is placed on comparison of applicability of both the mentioned equations with respect to experimental data. The applicability of the Burgers model equation is limited by the cutoff frequency of the tube which is exceeded due to the growth of higher harmonic components that is caused by the nonlinear distortion of the primary harmonic shape of waves. The next part of this contribution is dedicated to the presentation of some new approximate solutions of the Burgers equation with a fractional derivate and a brief description of used numerical methods.
A*-wave spatial resonances on thin cylindrical shells: Experimental study
Authors: Martinez Loïc , Duclos Jean , Tinel Alain
Published: July 2002

An experimental study of a thin cylindrical shell in contact with two different fluids is proposed. Using surface wave analysis methods (SWAM), the A-wave and A*-wave attenuation and phase velocity are first measured. The experiments show that when the internal fluid has a sound velocity slower than the external fluid, internal reflection of energy inside the shell generates an A*-wave in multiple space and time positions. Using those results as a basis hypothesis, the 2-D Ksi(k,ω) wave-number-frequency response of the fluid filled target is performed. On this Ksi representation, the multiple space A*-wave echoes cause multiple space resonances on a Ksi-cut versus k. Those numerical results are investigated using SWAM. The so-identified multiple spatial resonances correspond very well to the earliest shell whispering gallery waves families found by Veksler et al. in terms of RST.
Order effects in the measurement of auditory thresholds during bimodal divided attention
Authors: Rawool Vishakha W.
Published: July 2002

This study investigated the effect of a visual distraction task on the detection of warbled tones presented at 0.5, 1, 2, and 4 kHz in two orders of presentation. Fourteen subjects responded to warbled tones in two conditions. In the first (attention) condition, the subjects pressed a button everytime they heard the tone. In the second (distraction) condition, they were asked to respond to the tones by turning their heads towards the speaker while solving a cardboard jigsaw puzzle as quickly as possible. The order of attention and distraction conditions was random. Following each correct response, visual reinforcement was presented. For seven of the subjects, the order of presentation of the stimuli was 4, 2, 1, and 0.5 kHz (order 1). For the remaining seven subjects, the order of presentation was 0.5, 1, 2, and 4 kHz (order 2). The same order was maintained in the attention and distraction conditions. Results showed a significant order, attention–distraction, and frequency interaction. The results suggest that distraction can worsen the thresholds at the beginning of the task, but it may cause improvement in thresholds over time. These results will be discussed with reference to dual task attention limits and intersensory facilitation.
Identification of brief auditory patterns
Authors: Kidd Gerald , Mason Christine R. , Chiu Chung-Yiu P.
Published: July 2002

The identification of brief nonspeech sounds poses a challenging problem for machine-based recognition systems, especially at low signal-to-noise ratios. In contrast, human observers are thought to be remarkably adept at such tasks. However, there are few systematic studies in the literature that have attempted to quantify human psychophysical performance, or the factors that influence performance, for the identification of brief nonspeech sounds in noise. This study examined human identification of brief sequences of tones arranged in six frequency patterns in masked conditions. The stimuli and methods have been described previously [Kidd, Jr. et al., J. Acoust. Soc. Am. 98, 1977–1986 (1995)]. Several acoustical factors were varied including pattern duration, signal-to-noise ratio, and range of frequencies forming the patterns. Identification performance declined as duration was shortened below 100 ms even when signal energy, or detectability, were equated. For brief patterns at low signal-to-noise ratios, increasing the range of frequencies comprising the patterns improved identification performance. These results provide insight into the limits of human performance in the identification of brief sounds and may prove useful in the design of algorithms for machine-based recognition. [Supported by ONR and NIH.]
Changing frequency after-effect of a linear frequency glide
Authors: Kayahara Takuro
Published: July 2002

After listening to a linear frequency glide in a single direction ten times, subjective frequency change of stationary sine wave in the opposite direction was observed (changing frequency after-effect). Adapting glides had a fixed frequency excursion (ΔF: 960–1040 Hz), five different durations (250, 500, 750, 1000, or 1250 ms), and glided either upward or downward in frequency. Test glides starting from 1000 Hz had three different durations (500, 750, or 1000 ms), and its ΔF increased or decreased according to subject’s response. ΔF at subjective stationary frequency was measured using a three-category (‘‘up,’’ ‘‘down,’’ or ‘‘no change’’ in frequency) double-staircase procedure. In the ipsilateral condition (presenting both adapting and test stimuli to the same ear), ΔF in the same direction with the adapting glides was obtained and maximum ΔF (12 Hz) was observed when the duration of each test glide was the same as adapting glides, although, in the contralateral condition, ΔF was very small or none. These results suggest that there is a feature extractor which is selective in changing frequency and its direction at the level before integration of interaural information, and this extractor changes its own property after a relatively short time exposure of a single direction frequency glide.
Active control of total structural intensity in a T beam: General case
Authors: Hayek Sabih I. , Won Jungyun
Published: July 2002

The active control of power flow in a T beam was presented at the 134th Meeting of the Acoustical Society of America. In that paper, the mechanical noise source was a shear force in the plane of the T beam. In this paper, the active control of structural intensity in the T beam is achieved for a general shear mechanical noise source that has in-plane and out-of-plane components. The reduction of the total power flow at a control point located midway in the vertical leg of the T beam is achieved through control actuators that are located at one end of the straight part of the T beam. The vector control actuators have components in three-dimensional space in various combinations of normal forces, shear forces, torques, and moments. The efficiency of the structural intensity reduction is considered for local versus global control. Various control strategies are compared which are also aimed at minimizing the cost function that represents the total power requirement for the control process versus the noise input power.
Vibration isolation of two elastic structures using active compliance at the isolation mounts
Authors: Jones Kenneth E. , Hwang Y. F.
Published: July 2002

A mathematical model of the vibration isolation of two elastically connected structures (via isolation mounts) employing active springs is presented. Two elastic structures are connected by an arbitrary number of springs, each made active by the inclusion of a piezoelectric pad at one of the spring and structure interfaces. Single-input–single-output feedback controllers at each spring support seek to minimize the force transmitted from one structure to the other. Solutions are based upon the formalism of Lagrange’s equations using the normal modes of the uncoupled structures as the generalized coordinates. The constraints by the springs and supports are enforced by Lagrange multipliers. A 2-D numerical example of two elastic beams is presented. One beam, driven by a harmonic disturbance force, is coupled to the second elastically supported beam through three active springs. A beam-relative displacement sensor colocated at each spring is the input to an arbitrary feedback filter which can incorporate practical amplifier characteristics and actuator dynamics. Results of the numerical simulation of the feedback control case are discussed.
A fully global hydroacoustic monitoring system for the Comprehensive Nuclear-Test-Ban Treaty—Plans and progress
Authors: Lawrence Martin W. , Galindo Arranz Marta
Published: July 2002

The first ever fully global hydroacoustic monitoring system is being planned and implemented for use in the verification of a new international treaty, the Comprehensive Nuclear-Test-Ban Treaty (CTBT). This system will provide hydroacoustic monitoring of all the world’s oceans for 24 h a day, every day of the year, into the indefinite future. This unique resource will utilize two types of station. One type will be based on a hydrophone at the SOFAR axis depth, cabled back to shore. The other will be based on a seismometer on a small island using detection of the T-phase signal. This latter station relies on a signal which has propagated predominantly through the ocean, but has been converted to seismic energy at the margin of the island. Although this network is being installed for monitoring of nuclear explosions, it will also be a unique resource for scientific investigation of various phenomena. Data will be available to all State Signatories to the CTBT.
Biomechanical effects of ESWL shock waves
Authors: Sturtevant Bradford , Lokhandwalla Murtuza
Published: July 2002

Impulsive stress in the repeated shock waves administered in ESWL is the mechanical stimulus of injury to the kidney. In order to better understand the mechanical origins of injury, the interaction of focused shock waves with simple, planar polymeric membranes immersed in tissue-mimicking fluids was studied. The ESWL shocks in uniform, noncavitating liquids do not cause damage, but after passing through tissue or simulated tissue they do. This result suggests that the acoustic inhomogeneity of tissue may contribute to injury from ESWL. Shocks with large amplitude and short rise time (e.g., in uniform media) cause no damage in noncavitating fluids, while long-rise-time, dispersed shock waves, though only moderately attenuated, do. A continuum model is described which incorporates the mechanical properties of the tissue and accounts for the effect on its strength of microscopic inhomogeneities. It is shown that when transient tensile stress is applied it takes a finite time for failure to occur, so if the pulse is not long enough the material survives, in qualitative agreement with the behavior described above. A definition of dose at failure is derived in terms of the stress applied to the membrane by each shock wave, the strain rate, and the material failure stress.
A stratified model for ultrasonic propagation in cancellous bone
Authors: Hubbuck Elinor R. , Leighton Timothy G. , White Paul R. , Petley Graham W.
Published: July 2002

A new model for ultrasonic wave propagation in cancellous bone is presented here. The model treats cancellous bone as a stratified medium of periodically alternating solid–fluid layers, and considers propagation of two compressional modes, analogous to the fast and slow waves of Biot theory, in terms of slowness surfaces. The behavior of the two modes is examined in vitro using bovine bone samples over a range of incidence angles. Two modes are seen when the cancellous structure is parallel to the propagation direction, but only one mode propagates normal to the structure, and the velocity of the fast wave increases with angle of incidence away from the normal. This behavior can be explained in terms of an angular dependent inertial coupling effect, in keeping with the theory of propagation in stratified media of periodically alternating fluid–solid layers. Quantitative agreement with theory is good. Although the structure being analyzed is essentially an oversimplification of the structure of cancellous bone, the agreement with theory suggests the stratified model offers potential for future research.
A new method of multibeamforming for a swath bathymetry sonar system and its MIMD parallel architecture
Authors: Li Haisen , Yao Lan , Xu Xinsheng
Published: July 2002

In a swath bathymetry sonar system, real-time multibeamforming is required in order to increase the coverage and the efficiency. To determine the direction of seafloor echo accurately, traditional beamforming must work at a high sample rate, the interpolation is used to reduce computing amounts, but when there are tens of receiving channels and beams, the interpolation filter algorithm will rapidly increase computing amounts to result in the real-time multibeamforming, this is very difficult. Combining with a swath bathymetry sonar system, a new method of quadrature beamforming is described in this paper. It combines time domain beamforming with phase shifting beamforming to process the available bandpass signal. To reduce sample rate, a mixing frequency technique is used. A quadrature component is obtained from a single-channel A/D converter utilizing Hilbert transformation and its coefficient is weighted with Hamming window function, therefore, amplitude and phase is highly accurate. Comparing with the traditional beamforming method, the method presented in the paper is characterized by high accuracy, less sample rate, and less computation complexity, easily realized with a high speed DSP device. The parallel pipeline architecture is given based on TMS320C30. Simulating and lake trial results show that the method is right and efficient. The method is universal for active sonar.
Novel techniques for manipulating single-bubble sonoluminescence
Authors: Hargreaves Kirk , Matula Thomas J. , Crum Lawrence A. , Moss William C.
Published: July 2002

Various acoustical methods are currently being utilized to manipulate the radial oscillations of single bubbles undergoing sonoluminescence. Previous measurements have shown that the addition of higher harmonics to the drive frequency [J. Holzfuss, R. Mettin, and W. Lauterborn (private communication)] can result in an increase in the light intensity by as much as 50%. The radial response is also dramatically affected by the addition of higher harmonics. Nonlinear bubble dynamics equations agree well with the measured radial response. This and related work will be described. [Work supported by NSF and under the auspices of the U. S. Department of Energy at Lawrence Livermore National Laboratory under Contract No. W-7405-Eng-48.]
Evoked otoacoustic emissions, suppression and comparison to loudness constancy, at speech frequencies
Authors: Weiler Ernest M. , Dou Hongwei , Tannen Robert S. , Sandman David E. , Dember William N. , Warm Joel S.
Published: July 2002

Spontaneous otoacoustic emissions (SOAE) reflect spontaneous activity of the outer hair cells of the cochlea. Suppression of transient evoked otoacoustic emissions (TEOAE) bears operational similarity to loudness adaptation determined by the ipsilateral comparison paradigm (ICP). The present study gathered these measures from 41 normal hearing adults (5 male, 36 female, 20–45 years old). There was no significant relationship between spontaneous SOAE, but there was a significant relationship (r =−0.36, p<0.05) between ICP loudness adaptation and suppression of TEOAEs. The relationship between these two active measures has implications for failure of adaptation or the apparent loudness constancy in the speech range revealed by simple adaptation measures. It has been previously noted that the ICP technique yields adaptation in the speech frequencies between 40 and 80 dB, where the simple adaptation apparently does not [Weiler, Sandman, and Dou, program abstract 4pPPa10, J. Acoust. Soc. Am. 101, 3177 (1997)]. Implications will be discussed in terms of the results uncovered in this study.
Use of temporal information in recognition of amplitude-compressed speech by older adults
Authors: Souza Pamela E. , Larsen Johanna J.
Published: July 2002

Amplitude compression alters the speech envelope and under some conditions can reduce the availability of temporal cues for speech recognition. This may be a particular problem for older listeners, who demonstrated temporal resolution deficits in some studies. Older and younger hearing-impaired listeners were tested on their recognition of VCV syllables processed using the signal-correlated noise (SCN) method [Schroeder, J. Acoust. Soc. Am. 44, 1735–1736 (1968)]. This method results in a time-varying speech envelope modulating a noise carrier, and preserves temporal cues while minimizing spectral information. Recognition scores were obtained for uncompressed SCN syllables and for the same speech tokens after digital processing with a syllabic compression algorithm. In control conditions, recognition was tested for VCVs that included both temporal and spectral information. Younger and older listener groups were matched for mean hearing thresholds up to 4 kHz. A high-pass masker was included to eliminate contributions from the basal end of the cochlea. When compared to younger listeners with similar hearing losses, the older listeners demonstrated poorer performance on the compressed SCN signals. These results suggest that the poorer temporal resolution abilities of older listeners may play a role in their understanding of amplitude-compressed speech.
A finite element algorithm for the study of nonlinear standing waves
Authors: Elvira-Segura L. , Riera-Franco de Sarabia E.
Published: July 2002

A finite element method (FEM) is presented for the study of nonlinear acoustic standing waves inside a chamber. The method is developed using the Galerkin–Bubnov weighted residual formulation and applied to solve the Lagrangian second-order wave equation, including thermal and viscous dissipation terms. A one-dimensional problem is studied in the frequency domain. Numerical data are compared with analytical results obtained by using a perturbation scheme for the cases of resonance and antiresonance in a rigid-walled tube with one-sided rigid or pressure release boundary condition. The FEM algorithm is shown to be well suited for the study of high-frequency standing wave fields in which the effect of absorption cannot be described by simple analytical expressions.
Future directions in speech information processing
Authors: Furui Sadaoki
Published: July 2002

Speech processing technologies, including speech recognition, synthesis, and coding are expected to play important roles in an advanced multimedia society with user-friendly human–machine interfaces. Speech recognition systems include not only those that recognize messages but also those that recognize the identity of the speaker. This paper predicts future directions in speech processing. It describes the most important research problems and tries to forecast where progress will be made in the near future and what applications will become commonplace as a result of the increased capabilities. The most promising application area is telecommunications. To solve various fundamental problems, a unified approach across speech recognition, synthesis, and coding is indispensable. Handling the common phenomenon of voice individuality from different aspects are: research on speaker adaptation in speech recognition, automatic speaker verification, voice conversion in speech synthesis, and the problems of very low-bit-rate speech coding quality variation from speaker to speaker. The methods that essentially solve such problems should be based on a common mathematical model. Research on the mechanism of speech information processing in our human brains, that is, how meanings of speech are understood and how speech is produced, is also crucial for epoch-making technological development in conversational speech understanding and natural speech synthesis.
New diagnostic system measuring orthogonal factors and calculating subjective attributes of sound fields in a real room
Authors: Sakurai Masatsugu , Aizawa Shinichi , Ando Yoichi
Published: July 2002

Based on the model of the auditory-brain system which consists of the autocorrelation mechanism, the interaural cross-correlation mechanism between the both auditory pathways, and the specialization of human cerebral hemispheres, a new diagnostic system was developed. The system works on a PC for Windows, thus there is no need for special additional devices. After obtaining the impulse response, four orthogonal factors including the SPL, the initial time delay gap between the direct sound and the first reflection, the subsequent reverberation time, and IACC are analyzed for the calculation of the scale values of global and individual subjective preference. In addition to the four factors, two more factors, τIACC and WIACC, in the interaural cross-correlation function are figured out for evaluating the image shift of the sound source and the apparent source width (ASW), respectively. Also, the effective duration, τe, and fine structures of autocorrelation function of sound signals including the value of first maximum, φ1, and its delay time τ1, are analyzed. Keeping the subjectively optimal condition, this system may be utilized for automatic control of sound fields by electroacoustic systems. The new system will be demonstrated at the presentation of this paper. [Work supported by the Ministry of Education, Grant-in-Aid for Scientific Research.]
Absorption by seating and audience—Whose figures, whose measurement method?
Authors: Barron Michael , Coleman Steven
Published: July 2002

Measuring the absorption coefficients of seating and audience is no easy matter because of the difference in size between auditoria and standard test chambers. An optimum test method remains still to be established. Two authors have published papers on this issue recently in the Journal of the Acoustical Society of America. The designer, though, is left confronted by contradicting figures and test proposals. Bradley (1992, 1996) has proposed a technique which involves measuring about five different configurations of seats to establish the absorption of infinite seat areas and absorption by the perimeter. Davies et al. (1994) deal with the perimeter absorption by testing with and without barriers placed over the front and sides of a test block of seating. Neither method is ideal; Bradley’s depends on extrapolation to the smaller ratios of perimeter to area found in real auditoria, while Davies’ method assumes that the barriers do not introduce any new problems of their own. Beranek in his new book (1996) has reanalyzed measured data from full-size concert halls to produce revised absorption figures. This paper seeks to find what common ground there is between the measurement methods and published data. It will draw on results of some tests on model seating in a model test chamber.
Single-bubble sonoluminescence: Acoustic emission measurements with a fiber-optic probe hydrophone
Authors: Gompf Bruno , Wang Zhaoqiu , Pecha Rainer , Eisenmenger Wolfgang , Guenther Ralf
Published: July 2002

In single-bubble sonoluminescence additionally to the short light pulses, the bubble emits in the collapse phase a sound wave which can be measured with a fiber-optic probe hydrophone [Staudenraus and Eisenmenger, Ultrasonics 31, 267 (1993)]. This type of hydrophone is an absolute ultrasonic wideband reference standard. Compared to piezoelectric hydrophones it allows measurements with higher spatial (0.1 mm) and temporal (10 ns) resolution. The intensity and the width of the emitted sound wave increase with increasing driving pressure, but vary only slightly with water temperature in contrast to the emitted light intensity. The total radiated energy of the sound wave is below 10% of the initial energy of the bubble. The results are compared with earlier measurements on transient cavitation bubbles and with new theoretical results.
Sonoluminescence: The effect of magnetic fields in the Planck theory
Authors: Prevenslik T. V.
Published: July 2002

Sonoluminescence (SL) observed in the cavitation of water may be explained by the Planck theory of SL that treats the bubbles as collapsing miniature masers having a Planck energy intrinsic to the dimensions of the maser cavity. Microwaves at frequencies proportional to the collapse velocity are created in the bubble wall molecules as the Planck energy at optical frequencies is absorbed. Because of the interaction between the microwaves and molecular dipole moments, the Planck energy accumulates in the electronic state of polar molecules with a rotation quantum state. As the bubble wall is polarized, intense electrical fields develop to charge the bubble surface molecules to visible photon levels. By this theory, SL can only be observed if the maser walls close to optical dimensions thereby allowing the bubble wall molecules to accumulate Planck energy to the 3–6-eV levels necessary to produce VIS-UV photons. Closure to optical dimensions is assured by imposing a net positive collision pressure. The consistency of the Planck theory of SL was assessed with data on the effects of applied magnetic fields on SL. The net pressure was taken as the difference between the stagnation pressure in bringing the liquid bubble walls to rest and the pressure generated by the magnetic field in the presence of a charged bubble wall. Consistency with the Planck theory of SL is found as the SL intensity with magnetic field is parabolic and the peak SL intensity is linear with collapse velocity.
Some consideration of processes of soliton formation from an initially sinusoidal wave form in a thin fiber of fused silica
Authors: Nakamura Akira
Published: July 2002

Soliton formation processes in a thin fiber of fused silica by computer simulation are discussed. It has already been found that tone-burst sinusoidal sound successively makes solitons from negative half-periods coming in order in the wave train. In this case, the first soliton made from the first negative period is the most stable with respect to the following solitons, because some disturbances caused by preceding positive pressure periods are delayed and superposed with following negative pressure periods. Then, the pressure profiles of these subsequent negative periods are changed. Therefore, nonlinear distortion is affected. In this paper the intention is to discuss the difference of soliton formation processes between two single sinusoidal pulses with preceding negative and positive periods. A preceding negative period makes a stable soliton without any disturbance generated from the following positive pressure part, while the disturbance from the preceding positive period gives change to the pressure profile of following negative period, then nonlinear distortion of this negative period is affected. As a result, frequency spectra of both are differing with the propagation. Discussion is made by operational treatment using a nonlinear distortion operator and linear propagation operator.
Fast spectral algorithm for modeling focused sound beams in a highly nonlinear regime
Authors: Khokhlova Vera A. , Averkiou Michalakis A. , Younghouse Steven J. , Hamilton Mark F. , Crum Lawrence A.
Published: July 2002

Numerical modeling of high-intensity focused sound beams is important for medical applications of ultrasound. The KZK equation, which may be modified to include relaxation in the time domain or arbitrary absorption and dispersion in the frequency domain, is widely used to model ultrasound beams in tissue. Numerical solutions are available in either the time or frequency domain, but they become time consuming at high intensities when thin shocks are developed. To speed up calculations, a known high-frequency asymptotic result for shocks can be incorporated in the frequency domain algorithm. In comparison with exclusively numerical schemes, the present approach substantially reduces the number of harmonics and thus the computation time [Yu. A. Pishchal’nikov, O. A. Sapozhnikov, and V. A. Khokhlova, Acoust. Phys. 42, 362–367 (1996)]. Focused cw beams radiated from Gaussian and piston sources are simulated in water and biological tissue. The accuracy of the asymptotic approach is verified via conventional numerical methods of solution. It is shown that the present asymptotic method provides accurate results with a considerable reduction in computation time. In combination with conventional numerical schemes, the developed asymptotic method provides an effective tool for investigating focused sound beams for a wide range of parameters. [Work supported by NIH PO1-DK43881, Fogarty Research Program, and CRDF.]
The vibroacoustical analysis of the teeming crane
Authors: Wiciak Jerzy W. , Panuszka Ryszard J. , Iwaniec Marek A.
Published: July 2002

The present article considers the vibration analysis of the teeming crane (lifting capacity Q=160 tons) and noise caused by working units of the crane. The vibroacoustical investigations and theoretical analysis have been applied to the main bridge with box-trapezoid cross section. In the article there is shown a comparison of the vibration acceleration and dynamic stress that have been measured in real structure with calculations obtained by means of the finite element method. The correct and incorrect (two-stage) reduction roll gear running also was considered. The significant influence of the incorrect reduction roll gear running on the dynamic parameters was noticed. The noise caused by working units of the crane has been presented for three operating conditions: when the tilting ladle was descending, when it was hoisting, and pass-by load running of the teeming crane. [Work supported by TUMM, Cracow, Poland.]
Interacting spectral and temporal properties in Jamaican Creole and Jamaican English vowel production
Authors: Beckford Alicia
Published: July 2002

The vowel system of Jamaican Creole has been described as exploiting a contrastive length distinction, wherein long vowels and falling diphthongs contrast with single vowels in identical phonological environments [Wells (1973); Mead (1996)]. For example, /pan/ on, pan, can /tan/ stand /pen/ pen /paan/ to grasp /taan/ turn /pien/ pain. However, the nature of the relative contributions of spectral and temporal differences remains unclear. The interactions between spectral (F1/F2) and temporal properties, and their roles in shaping a three-dimensional vowel space (F1×F2×length) were examined in an acoustic study of the vowel production of 24 speakers at different points along the creole continuum from Jamaican Creole to Jamaican English. Data suggest that Jamaican Creole and Jamaican English speakers utilize vowel duration differently. It will be demonstrated in this paper that, for the Creole speakers, vowel duration functions to enhance slight spectral distinctions among low central and upper-mid-back vowels /a,o,u/. [Mead, ‘‘On the phonology and orthography of Jamaican Creole,’’ J. Pidgin Creole Languages (1996); Wells, Jamaican Pronunciation in London (Blackwell, London, 1973).]
English vowel production by native speakers of Beijing Mandarin
Authors: Wang Xinchun , Munro Murray J.
Published: July 2002

Ten English vowels produced in a controlled elicitation paradigm by 15 Beijing Mandarin speakers and 15 native English speakers were randomly presented to trained English listeners who identified them in a forced-choice listening task. Although the non-native productions of the vowels with Mandarin counterparts were generally as intelligible as those of the native English speakers, the remaining vowels were produced with varying degrees of intelligibility. The results suggested a strong influence of the L1 vowel system on the production of L2 vowels. In particular, acoustic analyses revealed that the acccented English vowels tended to be ‘‘pulled’’ toward their Mandarin counterparts so that they fell between the English and Mandarin vowels in an F1–F2 space. Spectral overlap in the tense–lax pairs accounted for some intelligibility problems, while the dense distribution of second-language (L2) vowels in the area of the vowel space where the L1 vowel system is less crowded apparently increased the level of difficulty for the L2 learners in producing some vowels. The results are discussed in terms of error prediction models for L2 speech learning.
Effects of listener training on intelligibility of Chinese-accented English
Authors: Rogers Catherine L.
Published: July 2002

Previous research has demonstrated significant improvements in the intelligibility of synthetic speech following listener training [E. Schwab et al., Human Factors 27, 395–408 (1985)]. In the present study, the effect of listener training on second-language speech intelligibility was investigated. Chinese-accented words and sentences were presented to four groups of native English listeners—one training and one control group for each of two speakers. On word trials, listeners were instructed to select the word presented from two minimal-pair alternatives; on sentence trials, listeners were instructed to write down what they heard. Listener responses were scored as percent of intended words correctly identified. The two training groups participated in three 1-h training sessions. Auditory and visual feedback was provided to listeners during the training sessions only. Performance of the training and control groups was compared across pre- and posttraining test sessions. Improvement in minimal-pair word intelligibility following training was found across speakers for the training group, relative to the control group; a similar but smaller improvement in sentence intelligibility was found. Word results will be discussed in terms of improvement in discriminability following training for the twelve phonemic contrasts tested. [Work supported in part by NIH-NIDCD Grant No. 2R44DC02213.]
An instance-based model of Japanese speech recognition by native and non-native listeners
Authors: Yoneyama Kiyoko , Johnson Keith
Published: July 2002

Japanese has a short and long segment contrast (oji-san versus ojii-san, kita versus kitta, and kana versus kanna). Previous research has shown that this durational property is one of the characteristics of moras in Japanese. It has also been reported that Japanese learners generally have difficulty in acquiring this contrast, but after extensive language experience with Japanese, they do learn the contrast. This suggests that the contrasts and similarities are based on remembered instances of linguistic objects in various environments during a lifetime. This paper investigates whether an instance-based approach of speech recognition can replicate performance by native listeners and second language learners of Japanese. An instance-based model was trained with naturally produced utterances of koma spin and komma comma and then tested on a 24-step nasal duration continuum from koma to komma [T. Uchida, doctoral dissertation, Nagoya University (1996)]. The identification function produced by the model was very similar to identification responses produced by native Japanese listeners. Additional simulations trained with utterances produced by English native speakers (e.g., coma, comma) will be reported. These simulations will address the role of linguistic experience in second language acquisition. [Supported by NIDCD R29-016435-06.]
Training American listeners to perceive and produce Mandarin tones
Authors: Wang Yue , Jongman Allard , Sereno Joan A.
Published: July 2002

Auditory training of non-native speech contrasts is based on the assumption that the perceptual system of adults can be modified. Previous research has shown substantial improvements in the identification of segmental distinctions which are absent in the listeners’ native language after a simple phonetic laboratory training procedure. However, it is not known whether such a training procedure is applicable to the acquisition of non-native suprasegmental contrasts. In the present study, native speakers of English are trained to perceive Mandarin lexical tones, using the high-variability procedure shown to be efficient in previous studies concerning the generalization and long-term retention of the training effects. In 20 sessions lasting four weeks, ten American learners of Chinese are trained to identify the four tones in natural words occurring in different phonetic contexts, and spoken by different talkers. The results of the pre- and post-tests are compared to determine if suprasegmental contrasts can be obtained and improved by training, if the contrasts obtained can be generalized to novel words and talkers, if contrasts can be maintained long after training, and if learning gained perceptually can be transferred to production. The results will be discussed in terms of nonnative perceptual modifications at the suprasegmental level.
A CALL system for teaching the duration and phone quality of Japanese tokushuhaku
Authors: Kawai Goh , Hirose Keikichi
Published: July 2002

This paper describes a CALL (computer-aided language learning) system that teaches the pronunciation of Japanese tokushuhaku (long vowels, the mora nasal, and mora obstruents) to beginning learners of Japanese as a second language. The system detects mispronounced tokushuhaku durations and quality, including, for instance, shortened or lengthened tokushuhaku, dentalized mora nasals, diphthongized long vowels, and aspirated plosives. Learners are prompted by the system to read tokushuhaku minimal pairs aloud. Tokushuhaku durations are measured using speech recognition and compared with distributions of native speakers. Mistakes in phone quality are detected using a speech recognizer incorporating bilingual monophone models of both the learner’s native and target languages. HMMs for the two languages are trained separately on language-dependent speech data, but are bundled together during recognition so that the closest phone recognized indicates the non-nativeness of the utterance should the recognized phone not be Japanese. Knowing the phonetics and phonology of the learner’s native language can identify non-native articulatory gestures that result in Japanese pronunciation errors, thus allowing precise corrective feedback to the learner. The paper includes an overview of the system, its reliability, validity, and effectiveness in the foreigen language classroom.
Comparison of American English vowel production and identification by native speakers of Luso-Portuguese
Authors: Famoso Christina F. , Schwartz Patricia N. , DaSilva Adelia
Published: July 2002

Learning a new language requires mastering the phonetic as well as lexical and syntactical aspects of the new language. As part of a larger study of second-language learning, the identification of American English vowels by speakers of Luso-Portuguese and the identification of those speakers’ productions by native speakers of American English were explored. Four adult bilingual Luso-Portuguese/English speakers (two women and two men) participated in this study. The subjects performed two tasks. First, they identified ten English words (ten repetitions of each, presented in random order, of: heed, hid, head, had, hod, hawed, hood, who’d, hud, heard) produced by a native speaker of American English. Then recordings were made of the subjects as they produced four repetitions each, in random order, of these ten English words. These recordings were digitized using SoundDesigner II software with an audiomedia board on a Macintosh Quadra 800 computer. Listening tapes will be prepared, using SoundDesigner II software, for identification of the Luso-Portuguese speakers’ vowels by native American English speakers. The results of the identification of the Luso-Portuguese speakers’ vowels will be compared with the identification of American English vowels by the native Luso-Portuguese speakers. [Work supported by St. John’s University.]
Acoustic analysis of American English vowels by native speakers of Luso-Portuguese
Authors: Schwartz Patricia N. , Famoso Christina F. , DaSilva Adelia
Published: July 2002

The ability of individuals to learn a new language requires mastering the phonetic as well as lexical and syntactical aspects of the new language. As part of a larger study of second-language learning, the acoustical characteristics of American English vowel production by native speakers of Luso-Portuguese were explored, and these characteristics were compared with the identification of these speakers’ English vowels by native speakers of American English. Four adult bilingual Luso-Portuguese/English speakers (two women and two men) participated in this study. Recordings were made of the subjects as they produced four repetitions each, in random order, of 10 English words (heed, hid, head, had, hod, hawed, hood, who’d, hud, heard). The recordings were digitized using SoundDesigner II software with an audiomedia board; formant frequency analyses will be carried out using Signalyze 2.0 software implemented on a Macintosh Quadra 800 computer. The acoustical analyses will be examined in light of the identification of the Luso-Portuguese speakers’ English vowels by native speakers of American English (reported in another paper at this meeting). [Work supported by St. John’s University.]