bluuebunny/crossref_metadata_2025 huggingface dataset:
Application of the vector Bragg scattering wave-number condition to obtaining turbulent field variables
Authors: Goodman Louis , Oeschger John , Szargowicz Diane , Hebda Peter
Published: October 2005

Using the concept of the vector Bragg wave-number scattering condition, described in the previous presentation, the results of laboratory experiments on high-frequency broad bandwidth bistatic acoustic scattering from a buoyant turbulent plume are described. Estimates of a variety of characteristics of the turbulent field are made. These include a real time estimate of the spatial structure and wave-number spectra of the turbulent temperature field, and a simultaneous and real time estimate of the velocity field and the velocity variance field. Examination into the occurrence of organized structure will be made. Discussion of the appropriateness of classical turbulence theory to the results will be made with emphasis given to the issue of inhomogeneity and anisotropy.
Effect of rough surface scattering on linear beamforming
Authors: Tarng J. H. , Lin Y. F.
Published: October 2005

In this report, effects of surface scattering on beamforming technique operated by a linear array are investigated. The array pattern, synthesized by conventional beamforming, is illustrated as an index to show the scattering effects. Both the cases of a wave scattered by randomly rough and smooth surfaces are considered, which are solved by the image and Kirchhoff methods, respectively. It is found that the latter case gives stronger interference to the array pattern than the former one. In the randomly rough surface case, the array pattern is interfered more severely when the array focuses the main beam in the direction of the scattering surface. It is found that the pattern interference can be smoothed by the roughness of the surface. The rougher the surface is, the weaker interference is. It is also demonstrated that the longer range between the array and source may lead to stronger interference. The three-dimensional as well as two-dimensional problems are investigated. In the former problem, the sidelobe is interfered more severely and its level is much higher than those in the latter one. Here, a Gaussian correlation function of rough surface fluctuation is assumed.
Virtual acoustics at microgravity
Authors: Persterer A. , Opitz M. , Müller Ch. , Nefjodova M.
Published: October 2005

During the spaceflight of the first Austrian astronaut on board the Soviet space station MIR in October 1991, the experiment AUDIMIR was successfully performed as a part of the Austrian scientific program. AUDIMIR started from the assumption that, in the absence of gravity, orientation by acoustic cues would play a much more important role for a human being than on Earth. The experiment was designed to verify this assumption, and thus to verify the usefulness of binaural audio technology for communication systems of future spacecrafts and spacestations. The first goal of the experiment was the investigation of the accuracy of localization under conditions of microgravity. The second part should, for the first time ever, investigate the role of directional hearing as part of the human orientation system at microgravity by presenting moving sound sources to the subject. The whole experiment was performed in a virtual acoustic environment comprised of headphones and a digital signal processor. Visual influences were excluded by the astronaut having his eyes closed. The binaural processing of the test signals was done ‘‘live’’ during the actual experiment. This opened the possibility of adaptive control of the experiment procedure. To get optimum spatial performance, the subjects individual HRTF’s were used.
Naval and commercial underwater acoustic facilities
Authors: Reid Glenn N.
Published: October 2005

This paper will serve as the Chair’s Introduction and provide a listing of all underwater acoustic measurements facilities known to the Chair. It represents several years of searching for facilities that are in operation today starting with a report from the Director of Navy Labs. dated in 1979. Locations and points of contact will be provided.
Rigid body reconstruction of jaw motion in speech
Authors: Vatikiotis-Bateson Eric , Ostry David J.
Published: October 2005

Typically, during speech, the jaw rotates downward and translates forward during opening movements and follows the opposite path during closing; however, rotations and translations out of the midsagittal plane may also occur. In this study, Optotrak was used to examine jaw motion in six degrees of freedom−three orientations and three positions−relative to the occlusal (bite) plane and the condyle. Native speakers of English (2) and Japanese (2) produced CVCVschwa utterances in normal and fast rate as well as in loud speech conditions. As has been reported previously, the principle components of jaw motion were horizontal translation and rotation in the midsagittal plane. However, rotations out of the midsagittal plane are also observed. Yaw about the longitudinal body axis was typically about 3 deg and roll usually less than 2 deg. Magnitude of yaw and roll angles differed little across conditions. The remaining translational components−vertical and lateral−were small in magnitude and uncorrelated with either sagittal plane rotation or horizontal translation.
Experimental study of the aerodynamics of a larynx model
Authors: Mongeau Luc , Coker Cecil H. , Kubli Robert A.
Published: October 2005

The aerodynamic characteristics of the source of voicing during phonation were investigated using a larynx model. An experimental setup consisting of a uniform cylindrical tube partitioned by a flexible wall with an opening, which was made geometrically similar to the human glottis, was built. Two pushrods mechanically forced the opening to close and open with a cycle approximating that of human folds during voicing. A regulated air supply connected to a flexible hose filled with mineral wool simulated the action of the lungs, pressurizing the subglottal tube. Flow velocities within the jet formed at the orifice discharge and in the subglottal tube were measured using hot-wire anemometry. Dynamic subglottal and supraglottal pressure, glottal area, and mean flowrate were also measured. The results show how the dynamic flow resistance of the glottal opening, defined as the ratio of the instantaneous pressure drop across the opening and the instantaneous flowrate, compares with the steady-state resistance of the orifice, measured with the glottis at rest.
Observation of articulatory movements using a magnetic position-sensing device
Authors: Honda Masaaki , Kaburagi Tokihiko
Published: October 2005

A magnetic position-sensing device is one of the promising tools for measuring articulatory movements. The device (Articulograph AG100, Carstens Medizinelektronik GmbH, Göttingen, Germany) is evaluated in terms of measurement accuracy, effects of tilt and lateral displacement of the receiver coil, interference between adjacent coils, and temperature drift. The results show that measurement distortion is 0.93 mm on average in a 120-×80-mm rectangle on the midsagittal plane. If the tilt angle is 20 deg with respect to both the x and y axes, the distortion increases to 1.1 mm on the midsagittal plane and to 2.8 mm for a lateral displacement of 5 mm. The positions of four receiver coils placed on the tongue are monitored simultaneously with ultrasonic pulsed-echo imaging of the tongue contour. The magnetic and ultrasonic data are compared to reveal the accuracy of the magnetic data for measuring tongue movements. Also the relationship between positions on the tongue and the tongue shape was quantitatively investigated. Finally, simultaneous measurements of the tongue, jaw, lips, soft palate, and larynx movements are demonstrated.
Articulatory kinematics in VCV sequences
Authors: Löfqvist Anders , Gracco Vincent L. , Nye Patrick W.
Published: October 2005

The present study was designed to contribute information on tongue, lip, and jaw movements in sequences of vowel–stop–vowel sounds. The speech material consisted of VCV sequences, with all possible combinations of the vowels /i, a, u/ and the stop consonants /p, t, k/; stress occurred on the second vowel. Receiver coils were attached to the upper lip, lower lip, jaw, tongue tip, and tongue blade; their movements were tracked using an electromagnetic transducing system with three transmitter coils. Results from one speaker suggest the following. When plotted in an occlusal plane reference system, tongue movement trajectories are generally curved. In addition, when the vowels occur in the context of the consonants /t/ and /k/, the tongue trajectories commonly loop back across themselves, thus forming closed loops. Jaw movements showed both curved and linear trajectories for all consonantal contexts. Predictably, tongue movements towards and away from the point of tongue–palate contact showed influences from both the preceding and the following vowel. The temporal and spatial coordination of lip, tongue, and jaw movements in speech production will be discussed. This discussion will focus on the gestures underlying speech motor control and the blending and aggregation of these gestures. [Work supported by NIH.]
Acoustic harmonic radiation from cylinders using a new internal source density method
Authors: Stepanishen Peter R. , Ramakrishna Sudheer
Published: October 2005

Two-dimensional acoustic harmonic radiation from cylinders which are symmetric about an axis and vibrating with a specified normal velocity is addressed using a new internal source density approach. The approach is based on the use of internal monopole and dipole source density line distributions along the axis of symmetry of the cylinder. A least-mean-square error method is used to determine the source distributions by matching the normal velocity of the cylinder to the normal velocity field of the source distributions at the cylindrical surface. The surface pressure and exterior pressure field are readily obtained from the source distributions. Discretization of the resultant line integrals leads to sets of linear algebraic equations which are readily solved for the monopole and dipole source strengths. Numerical results are presented to illustrate the accuracy of the method for the case of circular and elliptical cylinders subjected to various specified continuous and discontinuous normal velocity boundary conditions and frequencies.
Transient propagation in a finite one-dimensional pipe: Direct temporal calculation and experiment
Authors: Dickey J. , Maidanik G. , Crouchley K.
Published: October 2005

The temporal response of a long (∼100-ft) pipe is calculated and measured. The calculation is based on a formalism that describes a complex of coupled one-dimensional dynamic systems [J. Acoust. Soc. Am. 89, 1–9 (1991)]. The flexural, longitudinal, and torsional waves in the pipe are considered as separate systems in the model and interact only at discontinuities; the ends of the pipe in this case. In the experiment, the pipe is subjected to an impact that excites all three types of propagation and the response is measured by an accelerometer mounted on the outside of the pipe, or, in the case when the pipe contains water, by a hydrophone inside the pipe. In the model, the excitation is simulated by forming initial wave packets in each of the three systems, and the relative amplitudes of the initial excitations in the systems are adjustable parameters. Other adjustable parameters include the reflection coefficients of a particular wave type, the coupling between wave types at the ends of the pipe, and the wave speeds and losses in the systems. The calculated response versus time is assessed in the flexural wave system (assuming that this is the only wave type which the accelerometer responds to), or in the fluid system when present, and compared with the experimental data and the model parameters are adjusted for an optimal fit. Once the model parameters are optimized, the adjustable parameters can be deduced and the model can be extended to more complicated situations.

The noninteraction of the mode types propagating in the systems is a basic premise in the model and is violated in the fluid filled case since the fluid wave is strongly coupled to the structural waves; nevertheless, the model shows qualitative agreement in this case and good agreement for the air-filled case. The model also includes the dispersive nature of the flexural wave propagation and demonstrates the complexity in the response which develops over time but which ultimately simplifies into a model pattern.
Examining a respiratory control model using an allometric analysis of speech breathing
Authors: Hogue David M. , Porter Robert J.
Published: October 2005

The value of examining various aspects of speech production from a nonlinear, dynamic systems point of view has been suggested by recent reports [e.g., J. A. S. Kelso and B. Tuller, J. Physiol. 246, R928–R935 (1984)]. Utilizing principles from physical biology the relationships between speech breathing, length of utterance, and upper airway resistance were examined. A simplified biological state equation served as the model [P. N. Kugler and M. T. Turvey, Information, NaturalLaw, andtheSelf-assemblyofRhythmicMovement (Erlbaum, Hillsdale, NJ, 1987)]. Speaking tasks were varied on reiterant versus meaningful speech, length of utterance, normal fundamental frequency speech, whispered speech, and monotonic speech. Upper airway resistance values were calculated as ratios according to differential airflow as measured by a Rothenberg mask. Duration and depth of inspiration and expiration were measured with a Respitrace, and acoustic information was obtained from digitized speech samples using the program CSPEECH. Subjects demonstrated strong relationships between utterance length and volume of expired air and between utterance length and upper airway resistance. Experimentally derived exponents for the volume and resistance components of the biological state equation closely estimate those predicted. This presentation is a followup to the study presented in Houston, Fall 1991; the current study has made more accurate measures of upper airway resistance and has additional conditions. [Work supported by the Dept. of Psychol. and Chancellor’s Fellowship.]
Interarticulatory coordination of mandibular, labial, and velar movements
Authors: Kollia H. Betty , Gracco Vincent L. , Harris Katherine S.
Published: October 2005

One approach to understanding the control of speech motor actions is to identify the manner in which individual articulators are coordinated. A number of recent investigations indicate that the relative timing of the lips, jaw, and larynx are functionally constrained during bilabial closing. The interpretation is that speech movements are not controlled independently, but are controlled in a global manner reflecting large scale vocal tract actions. A recent study [Kollia etal., J. Acoust. Soc. Am. 91, 2474 (A) (1992)] showed that when functionally related, lips, jaw, and velum behave in a manner that reflects this same complex sensorimotor organization. The present investigation then, is an attempt to refine observations of interarticulator coherence to include the velum. Lip, jaw, and velar kinematics were recorded optoelectronically and simultaneously with the acoustic signal during productions of the utterance /mabnab/. The spatial and temporal relations between the lips, the jaw, and the velum were examined. Evidence of temporal stability was found in the patterns of interarticulatory coherence. [Work supported by NIH Grant Nos. DC-00121 and DC-00594 to Haskins Laboratories.]
Norms for the hearing in noise test: The influence of spatial separation, hearing loss, and English language experience on speech reception thresholds
Authors: Nilsson Michael , Gelnett Donna , Sullivan Jean , Soli Sigfrid D. , Goldberg Robert L.
Published: October 2005

Norms have been developed for the hearing in noise test [Nilsson etal., J. Acoust. Soc. Am. Suppl. 1 88, S175 (1990)]. Speech reception thresholds (SRTs) were measured adaptively in the presence of spectrally matched noise for 150 young male and female adults. Speech was presented at 0-deg azimuth in all conditions, and noise was presented at either 0-, 90-, or 270-deg azimuth at 65 dB(A). Pure-tone thresholds were measured at 0.5, 1, 2, 3, 4, 6, and 8 kHz. Subjects were also characterized according to their early language acquisition experience with English in one of five categories ranging from ‘‘English only’’ to ‘‘no English in the home.’’ Average SRTs for normal-hearing, ‘‘English only’’ subjects (pure-tone thresholds at all frequencies tested of 15 dB HL or better) noise equaled 62.26 dB(A) (−2.74 dB S/N). Several factors significantly influence thresholds: (1) spatial separation between the speech and noise lowered thresholds an average of 7.42 dB; (2) unilateral, high-frequency hearing loss elevated thresholds in quiet by 3 dB; and (3) thresholds in quiet and noise were elevated by 3.34 dB in subjects with normal hearing but ‘‘no English in the home.’’ This elevation of thresholds is especially intriguing because it suggests a cognitive/linguistic factor in the ability to understand speech in noise.
Effect of distraction on auditory thresholds of adults in quiet and noisy backgrounds
Authors: Rawool Vishakha W.
Published: October 2005

This study investigated the effects of distraction on the auditory sensitivity of adults in quiet and noisy surroundings. Seventeen normal hearing adults in the age range of 18–30 years participated in the study. Thresholds of each subject were determined using warble tones for 0.5, 1, 2, and 4 kHz in the sound field for each of the following conditions presented randomly: quiet, quiet with distraction, noise (Speech noise, 60 dB SPL), and noise with distraction. The subjects were instructed to turn their head toward the speaker every time they heard a signal. For conditions with distraction the subjects were instructed to put a cardboard puzzle together as quickly as possible in addition to looking at the speaker when they heard the stimulus. Results indicated that for both quiet and noisy surroundings, distraction worsens the thresholds at 0.5 and 1 kHz, it has no effect at 2 kHz and the thresholds improve at 4 kHz. These effects will be discussed in reference to possible physiological mechanisms. [Research supported by the Grants for Research and Creative Projects and the Margin of Excellence Fund, Bloomsburg Univ., Bloomsburg, PA 17815.]
Time-domain simulations of stringed and percussion instruments
Authors: Chaigne Antoine J.
Published: October 2005

The tones of existing instruments are simulated in the time domain by means of standard finite difference methods. The string model previously applied to the guitar [A. Chaigne, J. Acoust. Soc. Am. Suppl. 1 88, S188 (1990)] and more recently to the piano [A. Chaigne, A. Askenfelt, and E. Jansson, J. Acoust. Soc. Am. 89, 1878(A) (1991)] has been extended to the violin, the xylophone bars, and tympani membranes. The general procedure, which is similar for each type of instrument, will be presented. It consists first in deriving a numerical scheme from the continuous differential equations governing the motion of the vibrating systems including the interaction with an exciter (hammer, bow, mallets,...). In a second step, the mathematical properties of these discrete approximations are evaluated with special emphasis laid on stability, accuracy, and dispersion problems. Simulated waveforms are computed with realistic values of the parameters derived from measurements on existing instruments. These waveforms as well as their spectral contents are then systematically compared with those of real tones. This time-domain approach is used in order to test the influence of various physical parameters on the transients of strings and percussive sounds in which the nonlinear effects are essential. The paper will be illustrated by sound synthesis examples of various instruments obtained with this technique.
Investigation of an in-the-ear technique for measuring the noise exposure of workers fitted with earmuffs
Authors: Burks J. Alton
Published: October 2005

With the availability of miniature microphones, an in-the-air (ITE) measurement location has become attractive for use in assessing the noise exposure level of workers. This technique is especially relevant in situations where a worker wears earmuffs to reduce his/her noise exposure since there are questions about the actual level of protection provided by these devices. Presently, the approach to this personal protection problem is either to ignore it, or to attempt to derate the manufacturer’s published attenuation data to obtain a more realistic estimate of an earmuff’s actual performance. The result in either case is that the actual level of protection is still unknown. However, with a miniature microphone mounted under the cup of the earmuff, near the ear canal entrance, there is an opportunity to extract the necessary information about the worker’s noise exposure level if one can only define what an ITE measurement under occluded conditions means. In other words, what is the relationship between the sound-pressure level (SPL) measured under occluded conditions at an ITE location and the equivalent SPL measured in the undisturbed sound field (without the worker present)? This paper provides an analytical description of this problem for a reverberant environment. Also, experimental data obtained on a group of ten male subjects with a microphone positioned in the intertragic notch suggest that an ITE measurement represents a conservative basis for estimating the noise exposure level of a worker wearing earmuffs.
Vibrating reed studies of magnetic vortices in superconductors
Authors: De Long Lance E. , Brill Joseph W. , Xu Zhigang , Drulis Henryk
Published: October 2005

Although high-temperature oxide superconductors have been studied since 1986, the static and dynamic behaviors of magnetic vortices generated by applied magnetic fields and electrical currents remain controversial. These phenomena are important for at least two reasons: (1) vortex motion generates dissipation causing degraded performance in practical superconducting devices; and (2) the high transition temperatures (Tc≊100 K), short coherence length (≤10 Å) and strong crystalline anisotropy of high-Tc oxides are predicted to cause novel structural rearrangements or melting transitions of the ‘‘vortex lattice’’ at temperatures T well below the superconducting/normal phase boundary Hc2(T). Recent mechanical oscillator studies of relatively isotropic, low-Tc superconductors reveal dissipation anomalies and frequency shifts similar to those observed in high-Tc materials, challenging current theories for the vortex state of superconductors. [Work supported by NSF and Research Corp.]
Acoustic anomalies in the resonant ultrasound spectra of La2−x SrxCuO4
Authors: Sarrao John L. , Lei Ming , Migliori Albert
Published: October 2005

Resonant ultrasound spectroscopy is a novel technique for the simultaneous determination of a solid’s elastic moduli. Measurements of the resonant ultrasound spectra of single crystals of La2−xSrxCuO4 for several values of x are reported. Anomalies associated with the nonstoichiometric (x≳0) doped superconductor have been observed. These anomalies indicate that the absence of short-wavelength translational invariance at the Brillouin-zone edge is being coupled into the zone center. Measurements of the stoichiometric (x=0) insulator will also be discussed. [This work was performed at Los Alamos National Laboratory under the auspices of the U.S. Department of Energy.]
A Maslov–Chapman wave-field representation for broadband, wide-angle, one-way propagation in range-dependent environments
Authors: Brown Michael G.
Published: October 2005

The Maslov, or Lagrangian manifold, technique provides a means of constructing a uniform asymptotic solution to the wave equation under conditions in which variables do not separate. This study exploits the assumption that there is a preferred direction of propagation to simplify the presentation of this subject given by Chapman and Drummond [Bull. Seismol. Soc. Am. 72, S277–S317 (1982)]. The one-way assumption does not require that a narrow angle (parabolic) approximation be made. The final one-way wave-field representation is easy to implement numerically, offering several advantages over the Chapman and Drummond formulation. The final wave-field representation correctly describes, in the time domain, direct and multiply turned ray arrivals, wave fields in the vicinity of caustics of arbitrary complexity, edge diffraction and head waves. It is valid in media with strong (nonadiabatic) range dependence. In the one-way formulation, all quantities required to compute the wave field at all depths at a fixed range (the preferred propagation direction) are computed concurrently; the technique is thus particularly well suited to the modeling of wave fields that are sampled using a multielement vertical array. Numerical results, including a comparison with the SLICE89 data set, will be shown. [Work supported by ONR and NSF.]
The Convolvotron: Real-time demonstration of reverberant virtual acoustic environments
Authors: Foster Scott H. , Wenzel Elizabeth M.
Published: October 2005

This demonstration illustrates some recent efforts to ‘‘render’’ in real time the complex acoustic field experienced by a listener within an environment using a very high-speed, signal processor, the Convolvotron, and headphone presentation. The current implementation follows conceptually from the image model. The filtering effects of multiple reflecting surfaces are modeled by a finite impulse response filter than can be changed in real time and is based on the superposition of the direct path from the source with the symmetrically located image sources coming from all the reflectors in the environment. Directional characteristics of the reflections are determined by filters based on head-related transfer functions. The demonstration scenario allows the listener to experience how sound quality is affected by the manipulation of various environmental characteristics. For example, while listening over headphones and ‘‘flying’’ through a three-dimensional visual scene, one can hear how the sound quality of four simultaneous sources changes as virtual walls are expanded and contracted in different room configurations. Other environmental parameters that can be changed include ceiling height, absorption characteristics of the walls, ceiling, and floor surfaces (e.g., wood, versus glass versus drapery), comparison of anechoic versus reflective environments, and location of the sound sources. Doppler effects and directional radiation patterns for the virtual sources are also implemented.
Strain measurements inside solids subjected to lithotripter pulses
Authors: Gracewski S. M. , Dahake Girish , Ding Zhong , Burns S. J. , Everbach E. Carr
Published: October 2005

To study the direct stress wave effects on the fragmentation of kidney stones or gallstones during lithotripsy, it is important to know the evolution of the stress state inside concretions subjected to lithotripter pulses. Knowledge of these stress fields can be used, along with measurements of the mechanical properties of the stone, to predict locations and modes of failure. A technique of implanting monocrystalline silicon strain gauges within plaster concretions to obtain information about these internal stress fields was developed. Concretions of simple geometries, i.e., disks and spheres, were chosen for this initial study so that the reflections from the boundaries can be more easily identified and compared with theoretical models. Experimental results were obtained for both spherically diverging and focused shock wave sources. Theoretical models based on geometrical acoustics were developed for a spherical wave front incident on a disk and on a sphere. Predictions from these models match well with the experimental results. In addition, the caustic surfaces for a spherically diverging wave incident on a sphere are present. [Work supported by NIH Grant No. DK39796.]
Limitations of diffraction correction techniques in the estimation of acoustic attenuation in biological media
Authors: Forster Fred K. , Cullen Gregory V. , Wang Tsu Hua
Published: October 2005

Numerous studies have addressed the accuracy of diffraction correction techniques for the measurement of the acoustic attenuation coefficient based on diffraction calculations or measurements in a lossless medium. While an often-sited limitation of the technique has been the effect of scatterer statistics, it is usually assumed that attenuation effects can be separated from diffraction in the Rayleigh–Sommerfeld diffraction integral. However, in this study, which utilized a high resolution pulse-echo system with a 30-MHz center frequency and an f/3 focused transducer, it was shown that the effects of attenuation on the diffraction cannot be separated. Using a graphite/silicon dioxide/agar tissue-mimicking phantom, measurements were made under lossless conditions that agree with calculations based on a current theoretical model [J. Cardoso and M. Fink, J. Acoust. Soc. Am. 90, 1074–1084 (1991)]. Using the same experimental model a simple test was developed that demonstrated attenuation cannot be separated from diffraction. It was also demonstrated that under these conditions using a lossless diffraction correction can lead to attenuation estimates that are less accurate than no correction at all when compared to a transducer translation method. [Work supported by NIH Grant No. AR 39818.]
Time-shift compensation of ultrasonic pulse focus degradation using least-mean-square error estimates of time delay
Authors: Liu Dong-Lai , Waag Robert C.
Published: October 2005

Focus degradation produced by abdominal wall has been compensated using least-mean-square error estimates of time shifts in measurements of ultrasonic pulses from a curved transducer that emits a hemispheric wave and simulates a point source. The pulse waveforms were measured in a two-dimensional aperture after propagation through a water path and after propagation through different specimens of human abdominal wall. Time histories of the virtual point source were reconstructed by removing the time delay produced by geometric path differences and including time shift produced by propagation inhomogeneities in the case of compensation, finding the amplitudes of the Fourier harmonics across the aperture, calculating the Fraunhofer diffraction pattern of each harmonic, and summing the patterns. This process used a least-mean-square error solution for the relative arrival time expressed in terms of the arrival time differences between neighboring points. Comparisons of the time histories in the source plane show that the −30 dB effective radius of the focus can be 90% greater for propagation through abdominal wall than through a water path and that this percentage is reduced to 37% by time-shift compensation.
Transfer function magnitude and phase in the awkward zone
Authors: Lyon Richard H.
Published: October 2005

All systems of finite extent are multimodal, but many have modes so sparsely distributed along the frequency axis that one can safely deal with them one mode at a time. This is the classic situation in the dynamics of machine structures. On the other hand, some systems have modes so densely spaced that statistical and asymptotic theories such as the geometrical theory of room acoustics provide powerful models. The low modal overlap theory of Bolt, and the high modal overlap theory of Schroeder for the transfer function magnitude in rooms represent these limits in room acoustics. There is a similar transition in behavior in the phase of transfer functions as one proceeds from low- to high-modal overlap. This process will be described in the discussion.
Convergence of the modal sampling method in the medium frequency range
Authors: Trentin D. , Guyader J. L.
Published: October 2005

At medium frequencies, several problems deal with systems with a large number of modes. This is encountered, for instance, in the aerospace field. In that case, the evaluation of the vibration behavior using classical modal methods must cope with resolution of huge linear systems and specific methods have to be developed. The modal sampling method, which has been presented in the case of a long beam in longitudinal motion [J. L. Guyader, J. Acoust. Soc. Am. 88, 2269–2276 (1990)], enables one to reconstitute the vibration response of a structure using a limited number of groups of modes, called a sample. The accuracy of the prediction can be improved by increasing the size of the sample. The modal sampling method is presented here in the case of a large, thin, rectangular plate where point masses can be added. The structure is excited by an harmonic point force or a plane acoustic wave and kinetic energy on the structure can be derived. A study of the convergence of the method is presented when the size of the sample is increased and results show the ability of the method to take into account structural heterogeneities. [Work supported by the Institut National de Recherche et de Sécurité and by the Centre National d’Etudes Spatiales.]
Numerical studies on the effectiveness of the CHIEF method in the medium frequency range
Authors: Butler Michael J. , Wilder John A.
Published: October 2005

It is well known that the boundary element method implementation of the Helmholtz integral equation produces nonunique solutions at certain frequencies in the medium frequency range. The spurious solutions can be eliminated by a straightforward application of the interior Helmholtz equation, a procedure often referred to as the combined Helmholtz integral equation formulation (CHIEF). This method has been in existence for 21/2 decades, and has seen increasing use over the years. However, its application has been limited, and speculation concerning its effectiveness persists. This study reports numerical investigations that have been conducted to further verify implementation of the technique, and to develop practical experience regarding its use. Radiation and scattering from finite cylinders and spheres have been examined up to ka=20. Automated selection of several interior point patterns has provided a measure of the method’s robustness, and comparisons with closed-form solutions serve to validate the technique.
An orthogonal modes approach to fluid-loaded elastic structures with arbitrary forcing functions
Authors: Giordano Jeffery A. , Koopmann Gary
Published: October 2005

A numerical scheme is presented wherein the structural equations for an elastic structure are coupled to the acoustic radiation equations and recast in canonical form. The acoustic impedance matrix, which may be found via boundary element methods or by the superposition method, is expanded in a power series on angular frequency prior to coupling it with the structural matrices. The resulting system of equations is then decomposed to find an orthogonal basis set which uncouples the original equations. Once the basis set has been determined, the structural response and acoustic radiation spectra may be easily reconstructed for any arbitrary forcing function on the structure. Results will be given for two water-loaded examples, a finite plate in an infinite rigid baffle and a spherical shell.
Acquisition of language models from text
Authors: Brown P. , Jelinek F. , Lafferty J. , Mercer R. , Roukos S.
Published: October 2005

Statistical language models that assign a probability of being generated to each possible string of words are of interest here. Grammatical language models may, moreover, assign probabilities to parses of the given string of words. Language models are used as components of recognition systems, speech and handwriting being current examples of interest. Another use of language models is in text understanding systems and machine translation systems, where grammatical information is very desirable. Because of the richness of language, language models are usually constructed for particular discourse domains, such as medical reports, office correspondence, legal documents, etc. The construction is on the basis of large amounts of sample text, and should be as automatic as possible. For grammatical language models, it is desirable to estimate not only their statistical parameters from the supplied text, but even the grammatical rules themselves. The talk will discuss various methods of language model construction, their relative performance characteristics, and some of the many open problems.
Spoken language acquisition by computer
Authors: Fallside Frank
Published: October 2005

Human beings evidently learn to produce and perceive speech ‘‘simultaneously’’−learning to produce classes of sounds that are agreed by existing speakers to fall within the spoken language and at the same time learning to perceive these classes of sounds in other speakers. By contrast in speech communication by machine, speech synthesis systems are designed quite separately from speech recognition systems, employing at present quite different techniques and importantly using labeled data. A structure for this acquisition of speech by machine (asm), has been given for the simultaneous synthesis and recognition of speech from unlabeled human speech [F. Fallside, Speech Commun. (in press) (1992)]. This employs a synthesizer and a recognizer that are trained by a coupled minimization to produce, recognize, and label the human speech classes. The method is apparently general−applying to any type of recognizer and synthesizer with adequate performance and to any level of speech representation−sub-word, word, and higher. A decomposition technique has been established that allows, in principle, the acquisition of speech to be built up for successively higher levels. The method offers one way of bringing together the specialist techniques of synthesis and recognition and in particular the use of prosody in each. Recent results will be given in the paper.
Connectionist models of language comprehension
Authors: McClelland James L.
Published: October 2005

Connectionist models provide mechanisms in which all of the different types of knowledge thought to be relevant to language understanding−syntax, prosody, semantics, context, etc.−can be brought together in a coherent, integrated manner to address the problem of extracting the intended meaning of an utterance. The approach is different from many other approaches in several respects, one of which is its treatment of the role of words in constructing representations of meaning. Words do not have meanings that show up as constituents of meaning representations; rather, words constrain the construction of learned representations sufficient to exhibit meaningful language use. Earlier work illustrating the ideas [M. D. St. John and J. L. McClelland, Artif. Intell. 46, 217–357 (1990)] and more recent work in collaboration with John Reger will be presented. [Work supported by NSF and NIMH.]
Structured networks for adaptive language acquisition
Authors: Miller Laura G. , Gorin Allen L.
Published: October 2005

In this research on adaptive language acquisition, connectionist networks are explored that map input messages into machine actions. In this work, the capability of such networks to learn associations between messages and meaningful responses to them as the set of machine actions increases in size and complexity is investigated. Specifically, how to reflect task structure in a network architecture is considered in order to provide improved generalization capability. A method for constructing networks from component subnetworks, namely a productnetwork, provides improved generalization by factoring the associations between words and action through an intermediate layer of semanticprimitives. A two-dimensional product network was evaluated in a 1000-action data retrieval system, the object of which is to answer questions about 20 attributes of the 50 states of the US. The system was tested by 13 subjects over a 2-week period, during which over 1000 natural language dialogues were recorded. The experiment was conducted using typed input with unconstrained vocabulary and syntax. During the course of performing its task, the system acquired over 500 words and retained 92% of what it learned. The results of this experiment demonstrate the scalability of previous results obtained on smaller, less complex tasks [Gorin etal., Comput. Speech Lang. 5, 101–132 (1991)] [Miller and Gorin, Proc. ICASSP 201–204 (1992)].
Suppression and the dynamic range of hearing
Authors: Plack Christopher J. , Viemeister Neal F.
Published: October 2005

The results from experiments that have examined intensity discrimination in the presence of notched noise indicate that spread of excitation is not necessary for the auditory system to maintain a large dynamic range. In these experiments, however, the notched noise and the pedestal were simultaneously present. It is possible, therefore, that the notched noise suppressed the pedestal, and increased the dynamic range by effectively reducing the stimulus level [A. R. Palmer and E. F. Evans, Hear. Res. 7, 305–323 (1982)]. In the experiment described here, spread of excitation was masked nonsimultaneously in order to avoid suppressive effects. The brief sinusoidal pedestal was presented in a 13-ms gap between two bursts of a masking complex. The masking complex consisted of two sinusoids at frequencies of 0.8f and 1.2f (where f was the pedestal frequency), each having a level 10 dB below the pedestal level, and a notched noise with a spectrum level 50 dB below the pedestal level. Detection thresholds were measured to ensure that the complex was effective in masking spread of excitation. Weber fractions were measured at two signal frequencies, 1 and 4 kHz, and at eight pedestal levels (40–110 dB SPL in 10-dB steps) at each frequency. The results indicate that, even at the highest pedestal level, the masking complex raised the Weber fraction only slightly. This suggests that the auditory system can maintain a large dynamic range in the absence of suppression and spread of excitation. [Work supported by NIDCD Grant No. DC00683.]
Nonlinear level dependence of the phase discrimination of ‘‘critically modulated’’ three-component stimuli
Authors: Long Glenis R. , Houtsma Adrianus J. M. , Narayan S. Shyamla
Published: October 2005

The effects of sensation level on discrimination of the relative phase of three component stimuli centered at 2 kHz was examined. The phase of the center component was varied in 180 steps between 0 and 3600. A 3I-2AFC paradigm was used with the zero phase stimuli as the standard. In order to maximize the range of performance, the stimuli were optimally chosen for each subject. The frequency separations between components were adjusted to be close to the critical bandwidth at 2 kHz, and the relative attenuation of the side bands were also varied. Maximum discrimination did not always occur for the QFM stimuli (relative phase shift of 900). The phase of optimal discrimination changed systematically as the stimulus levels increased from 20 to 70 dB SL. Interaction of distortion products generated by two adjacent components with the third component, or a level-dependent dispersion of the stimulus envelope due to changes in the traveling wave velocity near the characteristic place, could account for the results. A linear model of cochlear processing is not consistent with the data.
Nonlinear interaction of two simple waves in an open resonator
Authors: Benson Yulia J. , Makarov Sergey N.
Published: October 2005

The aim of the present work is to demonstrate a principally new approach to the problem of the definition of the boundary condition at the open end of the tube when the incoming wave is a pure sine with frequency corresponding to the first linear resonance and the outcoming wave is unknown. An account of interaction of those two oppositely directed waves leads to the modified Burgers’ equation for the initial sound wave. The method proposed allows one to describe the latter wave by the same equation. The numerical analysis of this equation by Adams’ method and predictor-corrector scheme permits one to calculate the unknown outcoming wave and to define the boundary condition at the open end as a superposition of the incoming and outcoming waves. It is shown that results obtained are in good agreement with those predicted by the Sturtevant’s experiment [B. Sturtevant, J. Fluid Mech. 63(1), 97–120 (1974)].
An electrical discharge steam bubble acoustic source
Authors: Olson Allen H. , Sutton Stephen P.
Published: October 2005

Electrical energy stored in a high-voltage capacitor bank is rapidly discharged into seawater, producing a steam bubble. The expansion and subsequent collapse of the bubble compresses the surrounding fluid and radiates acoustic energy. The source process is divided into three stages: heating, expansion, and collapse. The fluid is rapidly heated by the large electrical currents passing between the electrodes. The heated fluid expands as compressed steam with a rapid increase in volume. Inertia causes the steam pressure to eventually fall below hydrostatic and the steam bubble collapses. Experimental data are presented for 1.9-cm-diam electrodes separated by 1.0 mm in 20.0-ohm cm seawater. Field strengths as high as 90 kV/cm and discharge times of between 5 and 10 μs were achieved without electrical breakdown. A thermoacoustic efficiency of 1.5% was measured.
A theoretical investigation of a highly intense sound beam from a plane circular transducer
Authors: Berg Aina M. , Tjo/tta Jacqueline Naze , Tjo/tta Sigve , Nachef Sylvain
Published: October 2005

The experimental investigation reported by Nachef etal. (see preceding abstract) represents a good opportunity to assert the validity of the transformed beam equation and its numerical implementation [Naze Tjo/tta etal., J. Acoust. Soc. Am. 88, 2859–2870 (1990)], and their usefulness to simulate highly intense sound beams. Comparison between theoretical and experimental results shows that in spite of the fact that it does not account for higher-order nonlinear effects, the model reveals a remarkable ability to accurately simulate the fine structure of the sound field, at any range and level of excitation under consideration. The present theoretical investigation confirms the nonlinearly induced shift in the near-field structure of the beam (defocusing) experimentally discovered by Nachef etal. [Work supported by The Norwegian Research Council for Science and Humanities (NAVF), and Fondation Scientifique de Lyon et du Sud-Est, France.]
Finite-difference and Pestorius algorithms for solving nonlinear propagation equations with molecular relaxation included
Authors: Piacsek Andrew A.
Published: October 2005

An alternative to the Pestorius–Anderson algorithm based on FFT concepts for nonlinear wave propagation with dissipation is one that makes explicit use of finite difference schemes. In a relatively recent study, Dey [Adv. Comp. Meth. Part. Diff. Eq. 6, 382–388 (1987)] presented and examined various finite difference schemes for solving Burgers’ equation ut+βuux=δuxx. It is shown here that analogous schemes are applicable to a system of equations that arise when molecular relaxation is included into the formulation and which reduce to Burgers’ equation when the phase velocity increments associated with relaxation are set to zero. Codes based on these schemes are described and implemented for the propagation through a homogeneous medium of a waveform initially resembling a step shock. The choice of the numerical scheme and the discretization can cause anomalous distortions in the waveform; techniques for minimizing such or of discarding them are discussed. A general discussion is given of the pros and cons of using finite-difference techniques rather than algorithms based on the two-step process involving Fourier and inverse Fourier transforms. [Work supported by NASA Langley Research Center.]