bluuebunny/crossref_metadata_2025 huggingface dataset:
Horn loudspeakers for the home: Past, present, and future
Authors: Edgar Bruce C.
Published: July 2002

At the start of recorded sound, acoustical horns were an integral part of early home audio. As electrical reproducers were developed, horns were still required because of electromechanical driver and amplifier limitations. In 1928, the wider bandwidth of the Rice–Kellogg direct radiator loudspeaker and availability of higher-power amplifiers all but removed the horn loudspeaker from home audio systems. After 1928, horn loudspeakers were only found in theater PA systems, until the introduction in the late ’40s of the Klipschorn, which spawned a revival of horn loudspeakers in the 1950s. The introduction of high-power solid-state amplifiers and small bookshelf speakers in the 1960s removed horns from home audio systems. In the early ’90s, the reintroduction of low-power class A tube amplifiers spurred a new interest in new horn designs. The new horn designs utilize the Tractrix horn contour for high frequencies and hyperbolic-exponential horn contours for the bass frequencies. Also, the new horn design efforts have removed colorations in the sound which plagued many of the vintage horn systems. Future horn systems will feature lighter diaphragms, better magnets, and (maybe) room temperature super conducting voice coils for higher efficiencies.
Multipole expansion of loudspeaker as sound source
Authors: Melkas Kaarina
Published: July 2002

Higher-order vibration modes of a loudspeaker diaphragm are represented as multipole sources of a sound field. As each normal mode corresponds to a multipole series, the radiation induced by a vibration pattern containing many normal modes is the image of a linear mapping of the participation factors to the coefficients of a multipole series. In terms of linear algebra, the participation factors and the set of corresponding coefficients are written as vectors. The generated sound field is then computed as a superposition from the multipole sources with strengths given by the coefficients in the series. Correction factors are introduced to count for the finite extent and other geometry related effects of the source. The higher- order normal modes form separate loads for the driving circuit. This method allows one to apply lumped parameter models at frequencies where the loudspeaker diaphragm does not move as a piston anymore.
Experimenally observed acoustic characteristics of the King Songdok bell
Authors: Kim Yang-Hann , Yoon Doo-Byung , Jhin Young-Ok
Published: July 2002

The bells cast during the Silla dynasty were hung over various types of hollows or cavities in the ground. The sound pipe, called the eumtong, which opens through the top, appears in most of the bells. Another characteristic of these bells is that they ring with a very long beating frequency, sometimes for more than 3 s. The King Songdok bell (‘‘Emille’’ bell) cast during the Silla dynasty (771 A.D.) is the most famous one among these. Its height is 3.66 m, the diameter of bell mouth is 2.23 m, and its mass is about 20 000 kg. A very extensive experiment was designed and conducted. To visualize how the bell radiates sound, a cylindrical acoustic holograph was attempted. The total measurement points for the holograph are 32×12; 32 microphones at every 30 degree. The acceleration signals were also measured extensively at many points, which were 11×12. The acoustic characteristics of the inside cavity of the bell were also analyzed. This paper explains the role of the acoustic cavity in the ground, the sound pipe, and the source of long beating. Sound visualization based on the holographic measurement will demonstrate how its sound radiates in space.
The ‘‘exercise’’ of changeringing: Acoustics, athletics, and algebra
Authors: Edge Ronald D.
Published: July 2002

Ringing of changes on tower bells (the ‘‘exercise’’) involves acoustics, in the tuning of the bells, sound insulation, and propagation from the tower. Muscular power is required to ring, though even young girls can ring the light (treble) bells. At the same time husky teenagers ring the heavy tenors—all can join in. In England, 16th-century ringers developed the techniques to sound their bells (tuned diatonically) sequentially, using a rope and wheel to swing them through almost a full circle, mouth vertical. The period lengthens toward the balance point, and by using the rope to change the energy, the order in which the bells sound may be changed. Each bell is tuned for the strike note, the hum note (about an octave below), and partials a third (the tierce) and a fifth (quint) above. Methods such as ‘‘plain bob major,’’ ‘‘superlative surprise maximus,’’ or ‘‘grandsire caters’’ organize the ‘‘changes’’ (possible permutations of the bells rung). Mathematics is required to compose peals (5000 different changes) which are true (no repeats). Ringing in America died out after the Revolutionary War, but in the last 30 years, at least 25 new towers have been built here, and several pre-Revolutionary rings refurbished.
Effect of definition and jury testing experience on perceived generality and importance of selected sound quality attributes
Authors: Abouchacra Kim , Letowski Tomasz
Published: July 2002

People make judgments of sound every day. Sound judgments, like judgments of any other entity, can vary in their generality (general–specific) and the importance of the judged characteristic (primary–secondary). The development of sound quality criteria systems and jury testing protocols requires knowledge of both the generality and importance of specific sound assessment criteria. Smith and Letowski [J. Acoust. Soc. Am. 89, 1938 (1991)] reported intuitive generality and importance ratings of selected sound quality attributes measured on naive listeners and proposed a set of sound quality attribute definitions. The object of the present study was: (1) to determine whether listeners maintain their intuitive generality and importance ratings for selected sound quality attributes following exposure to the proposed definitions, and (2) to assess the effect of jury testing experience on the preservation of intuitive generality and importance ratings. The results support the notion that while some sound attributes have strong intuitive meaning (e.g., clarity), the others need clear definitions and experienced listeners (e.g., roughness) in order to be used successfully in sound quality studies. Some of the definitions tested in this study agreed well with the listeners’ intuitive meaning of the defined attribute whereas others did not. [See NOISE-CON Proceedings for full paper.]
Modeling judgments on sound quality by neural networks
Authors: Prante Holger U.
Published: July 2002

The current noise control methods are not sufficient for covering the ratings of different kinds of noise. They are mainly based on experiments with stationary and artificial test signals. New judgment models have to be developed to account for the time fluctuations of environmental sounds. During the last decade algorithms emerged in connectionism that not only find optimal mappings within stationary data (for example the backpropagation algorithm) but also within time series. One architecture for this task is called finite impulse response (FIR) neural network. Due to the architecture the networks include preceding time steps for the calculation of the current output. This contribution presents FIR neural networks to model sound quality judgments on environmental sounds. A listening test has been conducted for collecting training samples for the network and for testing its generalization abilities. The sound patterns and the subjects answers are pre-processed using factor analysis. The analysis of the subjects answers yielded 2 factors, which can be identified to represent the sound attributes annoying and powerful. The FIR neural networks are trained to estimate both attributes. The results of training and generalization show high correlation between the sound attributes and predicted judgments of the neural networks.
Detection of amplitude modulation, frequency modulation, and quasifrequency modulation by the budgerigar (Melopsittacus undulatus)
Authors: Lin Jian-Yu , Dooling Robert J.
Published: July 2002

In budgerigars, as in humans, the detection of amplitude modulation (AM) remains relatively constant as modulation frequency increases while detection of frequency modulation (FM) improves. The point at which FM and AM are equal defines the critical modulation frequency (CMF). The CMF is approximately half the size of the critical band in humans because phase information is lost outside the critical band. At small modulation indices, the power spectrum of FM is almost identical to the power spectrum of AM with the difference being the relative phase of the components. The power spectrum of quasifreqeuncy modulation (QFM) is exactly the same as AM even at high-modulation indices. In this experiment, two budgerigars were trained by operant conditioning to detect AM, FM, and QFM at several modulation rates at three carrier frequencies. Budgerigars show nearly identical thresholds for detecting modulation in FM and QFM tones at low-modulation rates and similar thresholds for detecting modulation in FM, AM, and QFM tones at higher modulation rates. These results argue for an insensitivity to phase differences in budgerigars when they fall outside the frequency bandwidths of the auditory system. [Work supported by NIH Grant Nos. DC-00198 and MH-00982 to RJD.]
Efficient representation of head-related transfer functions
Authors: Chateau Noel , Bronkhorst Adelbert W.
Published: July 2002

Head-related transfer functions (HRTFs) contain information that is vital both for the analysis of acoustic cues used in sound localization and for the application of virtual-sound synthesis techniques. As the HRTFs measured for a representative set of source positions constitute a large body of data, the application of suitable data reduction techniques is of crucial importance. An obvious method, which takes the properties of the human ear into account, is the averaging of the frequency spectrum within equidistant intervals on a logarithmic frequency scale. This method was perceptually evaluated in an experiment in which six listeners compared original (individualized) HRTFs, obtained for ten source positions, with HRTFs averaged within 1/3-, 1/6-, or 1/12-octave bands. Sound stimuli were bursts of pink noise with either a flat spectrum or with ± 2 or ± 4 dB roving within 1/3-octave bands. It appeared that differences could only be perceived for the HRTFs averaged within 1/3-octave bands. Neither roving level nor virtual source position had a significant effect on the results. It was furthermore found that data reduction of the HRTFs using principle component analysis is somewhat more efficient when HRTFs are first averaged within 1/6-octave bands.
Constrained optimization of directional transfer functions using subjective preferences
Authors: Runkle Paul R. , Wakefield Gregory H.
Published: July 2002

The synthesis of high quality virtual acoustic sources over headphones has been shown to depend on individual differences between subjects: Sources synthesized using one’s own head-related transfer functions provide more accurate localization cues than sources generated with HRTFs obtained from others. A method of perceptually optimizing a generic set of HRTFs to more closely approximate those of a given subject is discussed. In this paper, HRTFs are represented as a cascade of a common and a directional transfer function [Middlebrooks and Green (1990)]. The DTFs are modeled as low-order pole-zero systems, whose parameters are varied using an adaptive iterative method [Wakefield and Runkle (1992)] that incorporates subject preferences with a search algorithm. Unlike previous attempts where no assumptions were made about the subject’s criteria for judging the quality of prospective solutions, psychophysically based constraints have been imposed on the parameter space to facilitate rapid convergence. To assess the feasibility of using subjective feedback in the optimization of DTFs, subjects were asked to ‘‘move’’ a binaural source from an initial position to a nearby desired position. Issues of convergence and repeatability will be discussed. [Work supported by the Office of Naval Research in conjunction with the Naval Submarine Medical Research Laboratory.]
A new technique for the numerical solution of a class of integro-differential equations arising in elastodynamic crack propagation problems
Authors: Costanzo Francesco , Walton Jay R.
Published: July 2002

A strategy for the numerical solution of a type of integro-differential equation arising from the analysis of dynamic crack propagation problems is presented. Specifically, the problem of interest is that of a mode III crack dynamically propagating in a homogeneous linear elastic medium while the crack tip consists of a nonlinear rate-dependent cohesive (or failure) zone. The mode of propagation is general, that is, not restricted to steady-state or other special regimes. Furthermore, the presented solution technique is in no way restricted to mode III problems.
Temporal measures of tongue twisters produced by children
Authors: Schwartz Patricia N. , McGarr Nancy S.
Published: July 2002

The production of tongue twisters has been used to study the planning and organization of speech production in both adults and children. In general, temporal measures have differed as a function of age with older children (10+ yr) showing more adult-like productions than much younger children. In this study, five children, 8 years of age, were audio-tape recorded producing tongue twisters and control stimuli following Smith [B. Smith, Appl. Psycholinguist. 11, 131–144 (1990)]. Phonetic transcriptions of the children’s productions will be presented. Measures of VOT, stop consonant closure, and vowel duration will also be presented. Preliminary analysis of VOT, stop consonant closure, and vowel duration will also be presented. Preliminary analysis of VOT measures show that the twister VOT means were longer than the corresponding control VOT means, as have been reported for adults. These results will be compared to the data reported by Smith.
A course in acoustics of musical instruments
Authors: Burroughs Courtney B.
Published: July 2002

Graduate studies and a career in research and/or teaching in acoustics are often the result of an interest in acoustics started with the playing of a musical instrument. In teaching and research, there is a need for the application of acoustic theory to real sources, which often cannot be treated effectively with academic idealizations. Taking advantage of the interest in and familiarity with musical instruments possessed by most students in acoustics, and the need for application of acoustic theory to real sources, a course in the acoustics of musical instruments has been developed for graduate students in acoustics. This source is not a course in musical acoustics designed for musicians. Acoustic theory and mathematical models are applied to the analyses of the mechanisms of sound generation and radiation by musical instruments. A description of and philosophy behind this course are presented. The instruments treated are discussed. The principles revealed by each instrument are outlined.
Detectability of gaps in the modulation of a 4-kHz carrier
Authors: Sek Aleksander , Moore Brian C. J.
Published: July 2002

This experiment tested the idea that the auditory system contains a ‘‘modulation filter bank.’’ It was an analog in the modulation-frequency domain of an experiment in the audio-frequency domain, whose results were explained in terms of ‘‘ringing’’ in the auditory filter [M. J. Shailer and B. C. J. Moore, J. Acoust. Soc. Am. 81, 1110–1117 (1987)]. Subjects had to detect a gap in sinusoidal modulation (rates of 5, 10, 20, 40, and 100 Hz, modulation index =1) applied to a 4-kHz 70-dB sinusoidal carrier. The modulator preceding the gap ended with a positive-going zero crossing. The modulator was turned on at the end of the gap either at a positive-going zero crossing (standard phase) or at a negative-going zero crossing (reversed phase). For very small gaps [<0.1 times the period (P) of the modulator], detectability was very poor for the standard-phase condition but was good for the reversed-phase condition. For low modulation rates, and for a gap duration of 0.5P, the opposite pattern was observed. The results are consistent with what would be expected from ‘‘ringing’’ in a modulation filter bank. [Work supported by The Wellcome Trust and the MRC, UK.]
Time-reverse acoustic array focusing in the presence of a soliton-type internal wave
Authors: Khosla Sunny R. , Dowling David R.
Published: July 2002

Time-reverse or phase-conjugate arrays (TRAs or PCAs) have been shown to retrofocus sound to the location(s) of its origin even when the acoustic medium and its boundaries are unknown. This retrofocusing is degraded by temporal variations in the acoustic environment. In this study, analytical predictions are made for the focal properties of a linear vertical time-reversing array in a stably stratified acoustic medium having sharp thermocline separating regions with two different sound speeds. The medium is assumed to be quiescent except for the presence of a deterministic soliton internal wave traveling along the thermocline. The soliton wave shape and propagation speed are obtained from the K–dV equation and relevant shallow-water oceanic parameters. Soliton motion is incorporated in the necessary two-way acoustic propagation formalism with the single-scattering approximation. Depending on the acoustic frequency and source-array range, current results suggest there is ample opportunity for successful TRA deployment in the presence of soliton internal waves. [Work sponsored by the Office of Naval Research.]
Fourier spectra and formant patterns of German vowels produced at F0 of 70–850 Hz
Authors: Klinkert Andreas , Maurer Dieter
Published: July 2002

Previous studies of vowel synthesis and of natural vowels have indicated: (1) a correlation between the lower formants and F0 for F0>175 Hz; (2) intelligibility and spectral differences of vowels with high F0>500 Hz; (3) severe methodological problems of formant frequency estimation for F0>300 Hz; (4) formant number alterations (appearance of different numbers of relevant formants for one vowel); (5) formant pattern ambiguity (appearance of the same formant pattern for different vowels). The purpose of this study was to replicate these findings. A large sample of the German vowels /i,y,e,o/,ε,■,o,u/ produced in isolation as well as in CVC context was investigated. There were 18 674 recordings made of 35 men, 44 women, and 20 children. Isolated vowels were produced at different levels of F0, and with different intensities. A listening test was performed. Fourier spectra, LPC spectra and spectral constancy were analyzed. The results strongly support the five indications mentioned. In particular, for isolated vowel sounds the correlation between the formants and F0, the appearance of ‘‘one-formant back vowels’’ (only one apparent formant <2 kHz), and the formant pattern ambiguity for nonadjacent vowels proved to be very pronounced. Consequences for normalization approaches are discussed.
Product noise labeling and public information: Recommendations from the 1991 National Noise Strategy Conference
Authors: Berger Elliott H.
Published: July 2002

In 1990, a symposium ‘‘Combatting Noise in the ’90s: A National Strategy for the United States,’’ was initiated by Congressman Richard J. Durbin of Illinois and funded by the American Speech-Language-Hearing Association in cooperation with co-sponsoring organizations, the Acoustical Society of America and the American Academy of Otolaryngology—Head and Neck Surgery. The key participants were divided into nine working groups covering topics dealing with the effects of noise, the control of noise, and education and regulation. Two working groups, VI Consumer Noise Sources and Hearing Protection, and VII Public Information and Education, examined issues relating to the labeling of noisy products and also noise-reducing products, how to inform the public about the need for such ratings, and the meaning and application of such numbers. The products requiring rating and/or classification that were examined were categorized as guns, music reproduction equipment, equipment and appliances, toys, hearing protection, and building spaces. The author, who was the chair of Working Group VI, will review the recommendations of both working groups VI and VII in the context of current events.
Analysis of the complex acoustic signal for use in identifying outdoor propagation modes and effects of turbulence
Authors: Norris David E. , Thomson Dennis W.
Published: July 2002

In outdoor sound propagation; refraction, diffraction, and scattering may all affect the properties of the propagated signal. The wind and temperature profiles characterize the refractive effects, the boundary conditions determine the effects of diffraction, and the presence of atmospheric turbulence drives the scattering. Signal measurements recently conducted at Penn State Univ.’s Rock Springs micrometeorological field site were focused on measuring the properties of the received complex acoustic signal. A source was configured to radiate continuous tones at five frequencies ranging from 90 to 660 Hz. Ten- to fifty-min recordings were made at distances ranging from 46 to 460 m. Both the magnitude and phase of the received signal were evaluated. The results suggest that use of the complex signal, rather than considering only the sound pressure levels, is valuable in characterizing the propagation. Recorded phase differences exhibit strong coherence between the source frequencies, suggesting refraction dominated propagation. Recorded fluctuations in magnitude exhibit much weaker coherence and appear to be independent of source frequency. Observed travel time fluctuations were on the order of 0.2%. A model is presented that explains these fluctuations by considering the influence of atmospheric turbulence over a fixed propagation path.
Incorporation of source motion into the fast-field program
Authors: Wilson D. Keith
Published: July 2002

The fast-field program (FFP) has proven to be a useful tool for determining low-frequency sound fields generated by stationary sources in horizontally stratified atmospheres and oceans. Extension of the FFP to the case of moving sources is the topic of this paper. It is shown that the exact moving source solution involves a three-dimensional inverse Fourier transform (with respect to frequency and two horizontal wave numbers), that unfortunately is computationally infeasible. By shifting to source-following, Lorentz coordinates, however, the solution can be reduced to an approximate, one-dimensional, inverse transform involving only the horizontal, radial wave number. The result is an efficient computational method that works well at any subsonic Mach number, and includes accurate calculation of frequency Doppler effects in the presence of refraction. An example simulation of the noise field generated by a landing aircraft will be provided.
Auditory guided visual search with visual distracters
Authors: D’Angelo William R. , Bolia Robert S. , McKinley Richard L. , Perrott David R.
Published: July 2002

In order to study the interaction of the auditory and visual systems, a series of experiments has been designed. The first experiment was completed in 1994 and recently published [Perrott et al., Human Factors 38(4), 702–715 (1996)]. This initial experiment explored auditory-aided visual search with a single high-contrast target and a dark background. The results demonstrated a distinct advantage in visual target acquisition with the addition of a correlated directional audio cue. The objective of this experiment was to develop a function describing the effects of directional auditory cueing on searh times in visual fields of varying complexity. Visual conditions included 1, 5, 10, 25, or 50 distracters randomly distributed. The locations of the targets and distracters ranged between ±180° in azimuth and between −75° and +90° in elevation. The three auditory conditions included no auditory cue, a free-field cue, and a virtual cue. In each case, the stimulus was pink noise (20–20 000 Hz). Results will be presented which show the effect of the number of distracters on search times with and without auditory cues. [Work supported by AFOSR.]
Active control of noise transmission through aeroelastic plates
Authors: Frampton Kenneth D. , Clark Robert L.
Published: July 2002

This presentation investigates the application of feedback control toward the problem of decreasing noise transmission through aeroelastic plates. The modeling method is discussed for a plate with pinned boundary conditions, coupled to convected fluid loading on one side, and backed by a nonconvecting acoustic environment on the opposite. A static output feedback compensator is designed to minimize the sound transmitted to the nonconvecting medium. Results from the application of this control system are compared with levels of attenuation predicted by state feedback control. Results indicate that significant reductions in transmitted noise can be achieved; however, variations in the fluid convection velocity can alter the level of performance.
Finite-element analysis of a laminated plate including elastic, viscoelastic, and piezoelectric layers
Authors: Guertin Louis , Charron Francois , Berry Alain
Published: July 2002

A finite element formulation for the vibration of a laminated plate including elastic, viscoelastic, and piezoelectric layers is presented. The model takes into account the mass, stiffness, and damping of each layer. A simplified discrete-layer plate theory and Hamilton’s principle are used to formulate the equation of motion of the system. The continuity of the transverse stresses and the displacements are imposed at interlaminar layer interfaces. Only five degrees of freedom per node are necessary to describe the deformation of the plate. The discrete element is a quadrilateral shape with eight nodes and a total of 40 degrees of freedom. Extensional, bending, and transverse shear deformation are included in the formulation. One additional voltage degree of freedom per element is necessary for each piezoelectric layer. This finite-element formulation can be used to modelize the dynamic as well as the static response of a multilayered isotropic plate subjected to both mechanical and electrical excitations. The multilayer elastic–viscoelastic–piezoelectric element formulation will ultimately be used to study smart structures integrating piezoelectric layers acting as distributed actuators and sensors. Comparison with published experimental, analytical, and numerical results will be presented to validate the model.
Matching frequency glides with two steady tones
Authors: Schouten M. E. H. , Peeters W. J. M.
Published: July 2002

Subjects were asked to match sinusoidal tone glides, of various durations (30–210 ms) and sweep widths (50–600 Hz) around a center frequency of 1000 Hz, with two connected steady-state tones whose frequencies could be controlled independently. The expectation was that short glides would give rise to single perceptual events without any movement in a particular direction, so that the two matching steady-state tones would not show any frequency direction either; long glides, on the other hand, were expected to be perceived as rising or falling tones and matched accordingly. It was hoped that this would provide a basis for an explanation of diphthongs as unitary percepts. However, no effect of duration was found at all; on average, glides were matched with tones whose frequencies corresponded to those reached by the glides after one-third and two-thirds of their course had been completed, regardless of duration.
Echo components for aspect-independent classification
Authors: Gaumond Charles F.
Published: July 2002

Monostatic, backscattered data from a finite, ribbed cylinder are shown. These data from a nonspherical scatterer exhibit a great deal of angular dependence. This angular dependence complicates the design of a classifier which uses templates to test the similarity of a test signal with the known scatterer. The scattering data have previously been shown to primarily consist of components generated by specular reflection, elastic waves, and Bloch waves. The angular dependence of this data set is shown to be due to fading of each component from a physical mechanism, as well as the changing interference of the various components as their relative timing changes. The separation of the echoes into components is shown to decrease the angular dependence. Finally, it is shown that a modest number of templates can be derived from the angular-dependent components through the use of the Karhunen–Loeve transformation and the assumption of equal statistical weighting of each aspect angle. The ka range considered extends from approximately 3 to 10. [Work supported by ONR.]
Comparison of acoustic reflectance measurement techniques
Authors: Keefe Douglas H.
Published: July 2002

Acoustic reflectance measurement techniques have been investigated that are calibrated using pressure responses measured in one or more cylindrical tubes and viscothermal models of wall loss. Of clinical interest as a test of hearing is a single-microphone technique to measure the inherently nonlinear ear-canal reflectance at various stimulus levels. Calibration of the probe containing the sound source/receiver consists of measuring the source reflectance of the probe and the incident pressure response at each stimulus level. A single-tube technique relies on separability of the tube response into incident and reflected signals, while multi-tube techniques rely on solving an overdetermined matrix equation of responses. The matrix includes weighting factors that differ across tubes by reflectance or SPL to control for inadequate signal-to-noise ratios at isolated frequencies, and a regularized matrix inverse solution is compared to the pseudoinverse solution. The single-tube technique functions as well as the best performing multi-tube technique, and requires fewer calibration steps. However, there is an ambiguity between the tube length chosen in the tube response model and the measured phase of the source reflectance. This ambiguity is addressed by a priori knowledge of the phase response of the probe averaged across frequency.
Predictability of acoustic multipaths over basin scales
Authors: Spiesberger John L. , Tappert Frederick D. , Wolfson Michael A.
Published: July 2002

The stabilities and instabilities of acoustic multipaths have been studied over basin scales as part of the development of studying climatic temperature changes in the ocean. In 1983, such transmissions began from the Kaneohe source where the propagation was complicated by the interaction of sound with the bottom near Oahu, Hawaii. For years, these transmissions exhibited stable arrivals only for those multipaths that were quite steep at the axis of the sound channel, for example, about 15 deg. Models indicate that the steeper multipaths may form ‘‘bounded-chaotic seas’’ that are useful for thermometric studies because their upper turning depths are insensitive to fluctuations. The sensitive nature of these paths is a consequence of the interactions with the bottom and the long distance of propagation, i.e., 3709 km. The flatter multipaths could not be used for inferring temperature variations because they were unstable. It is now understood that these paths are sensitive to fine and meso-scales which vertically scatter the energy over a kilometer, the same scale as the waveguide. [Work supported by ONR.]
Application of phase screens to ray chaos
Authors: Mazur Martin A. , Gilbert Kenneth E.
Published: July 2002

In wave propagation in complex media, phase changes along the propagation path are often approximated by a series of abrupt phase changes at discrete planes called ‘‘phase screens.’’ It is shown here how phase screens can be used in an analogous way in ray-tracing problems. The total travel time of each ray is formulated as a sum over phase screen and free space contributions. Fermat’s principle is then applied to the travel time, yielding a discrete mapping. The mapping connects the ray position at one phase screen to that at each succeeding screen. Examples of the method are given for both nonchaotic and chaotic ray tracing problems. The ray-tracing solution is compared to the wave solution calculated by the parabolic equation. By letting the separation between phase screens go to zero, the connection between continuous propagation and propagation with discrete transitions at phase screens is shown. [Work supported by the Penn State Univ. Applied Res. Lab.]
The relationship between ipsilateral acoustic reflex thresholds and loudness judgments in young normal women
Authors: Rawool Vishakha W. , Pillion Joseph P.
Published: July 2002

This study examined the relationship between ipsilateral acoustic reflex thresholds (ARTs) and loudness judgments. Eleven young women served as subjects. The ipsilateral ARTs were recorded for 0.5-, 1-, and 2-kHz activators using two probe frequencies: 678 and 1000 Hz. The subjects were also asked to judge the loudness of warble tones (0.5, 1, and 2 kHz) presented at various sound pressure levels in the following categories: not audible, very soft, soft, comfortable, loud, and very loud. From this data following levels were derived for each subject: very loud, average loud, maximum loud, average comfortable, and maximum comfortable. Correlation analyses were performed to examine the relationship of these levels with the ARTs. For 1 kHz, the maximum comfortable loudness levels were significantly correlated with the ARTs obtained with the 678-Hz probe tone and the average comfortable loudness levels were significantly correlated with the ARTs obtained with the 1000-Hz probe tone. For 2 kHz, the ARTs obtained with the 678-Hz probe tone were significantly correlated with the sound pressure levels that were judged as ‘‘very loud.’’ For 500 Hz, none of the loudness judgments were predictable from the ARTs. Detailed analyses and possible implications of the findings will be presented. [Work supported by a Bloomsburg University Grant for Research and Disciplinary projects.]
A functional neuroimaging study of speech recognition in noise
Authors: Horwitz Amy R. , Dubno Judy R. , Vincent Diana J. , George Mark S.
Published: July 2002

Few studies have examined brain activity in response to auditory stimuli using functional magnetic resonance imaging (fMRI) due to the inherent loudness of the scanner noise. Here, the scanner noise has been incorporated into the design of a speech recognition in noise task. Normal-hearing subjects were scanned on a Picker EDGE 1.5-T MR scanner; structural and functional images were acquired for 15 coronal slices. In one active task, sentences included contextual information so that the final words were highly predictable from the other words in the sentence (PH); in the other active task, sentences had no contextual information so that the same final words were not predictable (PL). Using nonmetallic noise-reduction earphones, sentences were presented to subjects’ right ears for 1-min blocks, alternating with 1-min blocks with no speech stimuli. Using a multiple subtraction technique, activation during rest was subtracted from activation in response to both low-context sentences (PL-rest) and high-context sentences (PH-rest). The difference between activation in response to low-context and high-context sentences was also determined (PL–PH). This analysis identifies areas of brain activation associated with processing acoustic/phonetic information within words with and without sentence context. [Work supported by NIH/NIDCD.]
Differences between binaural sound-field thresholds and monaural audiometric thresholds
Authors: Murphy William J. , Franks John R. , Hall Stephanie L. , Krieg Edward F.
Published: July 2002

Pure-tone audiograms under earphones and noise-band open-ear sound-field thresholds were collected from more than 150 subjects during hearing protector testing. The pure-tone audiograms were administered with 5-dB resolution to normal-hearing subjects (≤25 dB HTL) with an audiometer using TDH-50 headphones in standard cushions. Sound-field thresholds were determined by presenting a third-octave band noise in a reverberant room for the same subjects at resolution of 1 dB [Franks et al., Ear Hear. 13, 2–10 (1992)]. The subject’s best pure-tone threshold for either ear at each frequency was converted to dB SPL for comparison to the sound-field thresholds. The transfer function between pure-tone, minimum audible pressure, to diffuse sound-field, minimum audible field, was calculated for each of the test frequencies 125–8000 Hz, including 3150 and 6300 Hz. A second experiment was performed with ten subjects where each received three pure-tone audiograms at 1-dB resolution, as well as three sound-field noise-band audiograms at 1-dB resolution. The ten-subject data set was used to test the robustness of the larger data set. The results may be useful for predicting the open-ear sound-field thresholds from a subject’s pure-tone audiogram and in establishing guidelines about test precision of sound-field audiometry.
Detection of amplitude changes in the presence of adjacent stimulation by cochlear implant recipients
Authors: Hawks John W. , Fourakis Marios S. , Arkis Peter N.
Published: July 2002

Recent results [Hawks et al., in revision for Ear Hear.] suggest that disambiguation of neighboring lax vowels by cochlear implant (CI) recipients using the spectral peak (SPEAK) speech coding strategy may be hindered by the lack of perceptually significant differences in the coding of F1 and F2 as represented in the electrode activation patterns. Since multiple adjacent electrodes are activated when coding these spectral prominences, it is important to know how much activation is necessary to elicit the percept of a spectral peak in the presence of adjacent stimulation. To investigate this issue, CI recipients and normal hearing adults were compared on a detection task. Continua of three-tone complexes were constructed with tone frequencies equal to the centers of the frequency bands encoded by three adjacent electrodes in the F1 region and again for the F2 region. Stimuli were varied in intensity of the center tone, incremented both above and below a fixed level for the bounding tones. An adaptive procedure was used to estimate subjects’ thresholds for detection of a change in center tone amplitude from the pedestal level. Results will be discussed in terms of their significance to CI processor coding strategies of spectral information for vowels.
A longitudinal study of native Japanese speakers’ production of English liquids
Authors: Riney Timothy J.
Published: July 2002

This study investigates English liquids |l| and |■|) produced by 11 Japanese students at two times during their first and fourth years at a bilingual university in Tokyo (T1, T2). Five native English speakers from the same university serve as the control group. Each speaker read the same word list at T1 and T2. The word-initial CV portion of 50 words that contained liquids were subjected to auditory evaluation to determine if (1) the liquids were produced more accurately at T2 than at T1, and (2) to what extent the liquids, |l| and |■|, were identifiable as |l| and |■|, respectively, at T1 and T2. For question (1), paired comparison judgments made by ten naive, native speakers of American English were obtained. For question (2), forced-choice identification by three trained listeners were obtained. The presentation of the results will separate the identification and paired comparison judgments for |l| and |■| as singletons, and for |l| and |■| in clusters in contrastive and noncontrastive environments. The findings are discussed in light of current models dealing with adult acquisition of second language pronunciation.
Wave chaos and travel time effects of ocean mesoscale structure at global ranges
Authors: Tappert F. D. , Tang X.
Published: July 2002

A range-dependent ray-trace model and the broadband UMPE model are used to model sound propagation at fixed bearing through a field of mesoscale baroclinic modes in order to study the effects of the mesoscale structure on travel time at long ranges. The ray model exhibits chaos at ranges beyond a few Mm as manifested by an exponentially increasing number of triplications of the wavefront. In addition, the ray model predicts a mesoscale travel time bias in the direction of later time (‘‘cold bias’’) of the last axial arrival that is about 50–150 ms/Mm. At center frequency 75 Hz and bandwidth 50 Hz, the full-wave UMPE model qualitatively confirms the ray-trace predictions of chaos, that is now called ‘‘wave chaos’’ because it represents apparently random and unstable behavior of finite frequency wavefields. Also, the UMPE model quantitatively confirms the above mesoscale bias of the last axial arrival. Theories based on the adiabatic approximation are used to explain the modeled mesoscale bias in magnitude and sign. The steeper, early raylike arrivals are relatively stable in the presence of mesoscale structure and should be useful for long-range tomography. [Work supported by ONR.]
First audiogram for marine mammals in the open ocean and at depth: Hearing and whistling by two white whales down to 30 atmospheres
Authors: Ridgway Sam , Carder Donald , Smith Rob , Kamolnick Tricia , Elsberry Wesley
Published: July 2002

In examining the potential impact of human-made sound on sea mammals, it was considered that whale hearing sensitivity might diminish with increasing ambient pressure. To test the effect of depth, two white whales made 885 dives to a platform at 5, 100, 200, or 300 m in the Pacific Ocean. At each stationing on the platform up to 12 min at a time, whales whistled when they heard a 500-ms tone from a hydrophone. With increasing depth, air density increase in the middle ear, sinuses, and nasal cavity changed each whale’s whistle response, but did not attenuate hearing as it does in the aerial ear (of humans and other land mammals tested in pressure chambers) due to middle ear impedance changes. The findings support theories that sound is conducted through whale head tissues to the ear without the usual ear drum/ossicular chain amplification of the aerial middle ear. These first ever hearing tests in the open ocean demonstrate that whales hear as well at depth as near the surface, therefore, zones of influence for human-made sounds are just as great throughout the depths to which whales dive, or at least to 300 m. [Work supported by Office of Naval Research N0001496WK30349.]
The use of CF/FM sounds in bats
Authors: Lei Yanling , Raichel Daniel R.
Published: July 2002

The manner in which bats process their echolocation calls involves two main components: (1) producing and transmitting the signals; and (2) receiving and analyzing them [Fenton and Brock, Bats, Facts On File (1992)]. The high-frequency signals constituting these calls consist of constant frequency (CF) sounds and frequency modulated (FM) sounds. CF components were found to be used for prey detection and identification; FM components were found to be utilized to update distance information. The Doppler shift also helps in detection of targets and gauging the speed of moving targets. Thus, combining the pulses and the resultant echoes enables the bat to determine object distances, sizes, and shapes. Bats were found to have their own call frequencies and are insensitive to a wide range of potential jamming sounds, which allows them to clearly distinguish their own calls amidst environmental cacophony. Experiments have shown that only when entire CF/FM simulations were used did the bats perform discrimination tests badly [J. D. Altringham, Bats: Biology and Behavior (Oxford U.P., New York, 1996)]. This indicates that the echolocation process may be essentially a series processing of CF/FM signals rather than parallel processing of both types.
Numerical modeling of biomedically applied transient high-energy sound pulses
Authors: Steiger Eckard
Published: July 2002

A finite difference in time-domain (FDTD) discretization is applied to simulate finite-amplitude sound pulse propagation in biomedical applications. A numerical model requires a method of accounting for correct representation, especially of shorter wave components. These are generated by nonlinear steepening during the propagation. Therefore, the FDTD discretization has to be of low-dispersive and low-dissipative nature. A high-order scheme suitable for the computation of pulsed sound including weak shocks is applied to model a Storz SL 10 shock-wave lithotripter. The SL 10’s electromagnetic source generates cylindrically diverging pulsed pressure waves. By reflection at a hemi-paraboidal brass reflector these waves are focused and steepening of the wave profiles causes weak shock-wave generation in the focal region. A high-order method to model curved boundaries (given by the reflector) is presented. Wave profiles predicted are compared with measured ones. Excellent agreements demonstrate the proposed method to be a flexible tool to simulate biomedically applied transient high-energy sound.
Design simulation of composite transducers
Authors: Cao Wenwu
Published: July 2002

Many advanced transducers, such as those for medical ultrasound imaging, are made of composite materials. These composites contain both piezoelectric active and inactive constituents and therefore have a nonuniform surface vibration profile. This nonuniform surface vibration will in turn influence many characteristics of the acoustic beam pattern produced by the transducer, such as the focusing ability, sidelobes, and the near-field pressure distribution. These complex transducer systems cannot be modeled properly using equivalent circuit models due to the 3-D nature of the transducer vibration. A 3-D finite-element modeling scheme will be given in this presentation. The modeling contains two stages: First, a finite-element method is used to study the nonuniform structural vibrations of the composite transducer and obtain the surface velocity and displacement distributions; second, this surface velocity and displacement information will be fed to the Helmholtz integral to derive the transducer radiation pattern. It is also possible to compute directly the near-field pressure distribution using FEM. Accurate predictions on the transducer behavior are obtained. In addition, stress and electric field concentrations inside the transducer can be calculated. Comparison with experimental results will be shown via an animation.
Optimal digital compensation networks for acoustic transducers
Authors: Titterton Paul J.
Published: July 2002

A digital compensation network is designed which (1) minimizes deviation from the user-determined optimal transducer response, (2) limits the sensitivity to physical perturbations of that response, and (3) ensures actuator authority limitations for known and unknown-but-bounded input sequences. H∞, H2, l∞, and l1 transfer function norms translate the design goals and specifications into a convex constrained optimization problem, which is solved using semidefinite programming. The methodology can be simply extended to treat transducer arrays and optimal beamforming. A mathematically modeled underwater acoustic transducer provides a numerical example for exploring the optimal trade-off between competing specifications. [This work is partialy supported by SRI International.]