bluuebunny/crossref_metadata_2025 huggingface dataset:
Infant thresholds for short duration tone bursts
Authors: Olsho Lynne Werner , Marean G. Cameron
Published: October 2005

Thresholds for short duration tone bursts were measured for three-and six-month-old infants and for adults. An experimenter, blind to trial type, observed the infant's behavior. On each trial, the experimenter judged whether or not a tone had been presented and received feedback. The stimuli were 1000-Hz pure tones, with 16-ms rise/fall times and no steady-state duration. Eight tone bursts were presented at a rate of 2/s during a signal trial. Stimuli were presented monaurally via insert earphone. Thresholds were estimated using an adaptive procedure. Adults were tested using the same stimuli and procedure. Thresholds for six-month-olds averaged about 13 dB higher, while those of three-month-olds averaged about 24 dB higher than those of the adults. The size of the infant-adult difference for these short duration tones is nearly the same as that reported for longer duration stimuli [Olsho et al., J. Acoust. Soc. Am. Suppl. 1 80, S123 (1986)]. That the effects of duration on detection threshold are quite similar for infants and adults suggests that temporal summation is mature by three months of age in humans. [Work supported by NIH.]
Modeling speech production using dynamic gestural structures
Authors: Saltzman E. , Goldstein L. , Browman C. , Rubin P.
Published: October 2005

In the present computational model of speech production, an utterance is represented as an organization of primitive linguistic units, gestures, into a larger structure, a gestural score. Each distinct gesture is linked to a particular subset of vocal tract variables (e.g., lip aperture and protrusion) and model articulators (e.g., lips and jaw), and is associated with a set of time-invariant dynamic parameters (e.g., lip aperture target, stiffness, and damping coefficients). The values of the dynamic parameters and their activation intervals are computed as part of the gestural score for a given utterance using a linguistic gestural model that includes a gesture-based dictionary of English syllables and a flexible rule interpreter for manipulating dynamic parameters and inter-gestural phasing. The gestural score serves as input to our task-dynamic model of sensorimotor coordination. In this model, the evolving configuration of the model articulators results from the gesturally and posturally specific way that driving influences generated in the tract-variable space are distributed across the associated sets of synergistic articulatory components. Coarticulation effects of various sorts are automatically produced as a function of the spatial and temporal overlap of two (or more) gestures. Significantly, explicit trajectory. planning is not required and the model functions in exactly the same way during simulations of unperturbed, mechanically perturbed, and coproduced speech gestures. [Work supported by NIH NS-13617, HD-01994, and NSF BNS 8520709.]
A perception-based LSP distance measure for speech recognition
Authors: Paliwal K. K.
Published: October 2005

Because of their high correlation and reduced sensitivity to quantization errors, the line spectrum pair (LSP) frequency parameters have been used recently for efficient quantization of LPC information for speech coding. In the present paper, the LSP representation is used for speech recognition and a new perception-based LSP distance measure is proposed. This distance measure exploits the following two properties of the speech perception process [D. H. Klatt, Proc. ICASSP, 1278–1281 (1982)]: (1) The formant frequencies are the most important parameters for speech perception: and (2) the formant bandwidths and spectral tilt contribute very little to speech perception. The present distance measure uses these two properties in the form of weights in a weighted Euclidean distance measure. In order to derive these weights, the LPC power spectrum P(f) is computed for each speech frame and the weight for a given LSP frequency ft is taken to be proportional to [P(ft)]c. The perception-based LSP distance measure is studied here on a speaker-dependent speech recognition task. The optimum value of the exponent c is found here to be 0.15. The perception-based LSP distance measure results in a recognition score of 95.7%, while the recognition accuracy is found to be 89.9% by using the conventional Euclidean distance measure on LSP parameters and 94.1% by using the liftered cepstral distance measure [B. H. Juang et al., IEEE Trans. Acoust. Speech Signal Process. ASSP-35, 947–954 (1987)].
An estimation method of vertical sound-speed profiles in shallow water using multipath propagation
Authors: Hachiya Hiroyuki , Ohtsuki Shigeo , Okujima Motoyoshi
Published: October 2005

To observe changes in shallow water structure, a method of estimating the vertical sound-speed profile in shallow water is presented. “Shallow” means a water depth of less than about 1500 m in which sound can be propagated a certain distance by repeated reflections at both surface and bottom. In such water, the acoustic characteristics of both the surface and the bottom are important determinants of the sound field. The acoustical characteristics of shallow water in an acoustic transmission experiment that was conducted in Sagami Bay using a 5-kHz pulsed cw signal at a range of 10 km are examined. According to the results, bottom-and/or surface-reflected rays are stable in time and have identifiable ray paths. Using a shallow water model, a simulation to estimate the vertical sound-speed profile not from the travel times of the rays but from the travel time differences between the bottom- and/or surface-reflected rays was carried out. Travel time differences between individual rays could easily be measured by a clock, and estimated results agree well with given correct models.
The performance of an eigenvalue-based stabilization algorithm for high-resolution beamformers: Experimental tests with shallow water data
Authors: Gragg Robert F. , Pasewark Bruce H.
Published: October 2005

This work investigates the performance of high-resolution beamforming techniques as used with acoustic arrays in shallow water. The maximum likelihood method (MLM) is the high-resolution technique of principal interest and the maximum entropy method (MEM) is addressed briefly. The topics dealt with are the performance of beamformers with computer-simulated data, the performance degradation caused by system phase errors, and performance with actual field data. High resolution methods are found to suffer performance degradation in the form of instabilities in the beamform outputs. Results indicate that the proper pre-processing method (namely, the elimination of the lowest-lying eigenvalues from the cross-spectral density matrix) can stabilize maximum likelihood beamformers so that they outperform conventional beamformers at the tasks of resolving targets with similar bearings and tracking them through bearing crossings. No stabilizing method was found for maximum entropy beamformers. [Work supported by ONT.]
Chaos and hyperchaos in shallow water acoustics
Authors: Tappert Frederick D. , Goni Gustavo J. , Brown Michael J.
Published: October 2005

The ray equation for shallow water sound propagation in a “topless” ocean over a sinusoidal bottom can, under certain circumstances, be reduced to the area preserving standard mapping [F. D. Tappert and M. G. Brown, J. Acoust. Soc. Am. Suppl. 1 83, S37 (1988)]. For ray paths interacting with a flat sea surface and the ocean bottom, the Poincaré sections and the Lyapunov exponents evidence the chaotic behavior of the mapping. Numerical results show the sensitivity of the equations to changes in the environmental and initial conditions. The mapping dependence on the stochasticity parameter K (environment) and on the initial conditions is studied. For any nonzero value of K it can be shown that there exists a chaotic behavior (hyperchaos), in contrast to the critical value of K, Kc ≅ 0.97, for a “topless” ocean, under which no chaotic behavior is observed.
Generation of acoustic waves by an impulsive point source in a fluid/porous-medium configuration with a plane boundary
Authors: de Vries Sÿtze M. , de Hoop Adrianus T.
Published: October 2005

The space-time acoustic wave motion generated by an impulsive monopole point source in a fluid/porous-medium configuration with a plane boundary is calculated with the aid of the modified Cagniard technique. To describe the macroscopic propagation of linear acoustic waves through the fluid-saturated porous solid, four coupled partial differential equations are used that follow upon applying a volume averaging procedure to the two constituents, a perfectly elastic solid and an ideal fluid of the porous medium. The source is located in the fluid part of the configuration. Numerical results are presented for the reflected-wave acoustic pressure, especially in those regions of space where head waves occur. There is a marked difference in time response in the different regimes that exist for the wave speed in the fluid in relation to the different wave speeds in the porous medium (fast compressional, slow compressional, shear). These differences are of importance to the situation where the reflected wave in the fluid is used to determine experimentally the acoustic properties of the porous medium.
Auditory-perceptual analysis of selected syllables
Authors: Miller James D.
Published: October 2005

Analyses of consonant-vowel syllables (CVs) in terms of the auditory-perceptual theory of phonetic recognition will be presented. Examples of CVs will include a voiceless stop, a voiceless fricative, a nasal, and an approximant paired with monophthongal vowels. Spectral analyses are used to locate the formant peaks and to track these during the course of the syllable, producing a sequence of spectra, one for each ms of waveform. Formant and F0 information from these sequences is then converted into sensory and perceptual paths in the theory's auditory-perceptual space. This space contains subspaces, called perceptual target zones. The activation of a zone results in the output of a phonetic code. While the exact conditions for this activation are not yet known, it appears that certain aspects of the behavior of a perceptual path in relation to the perceptual target zones can determine the phonetic transcription of a syllable. [Work supported by NINCDS and AFOSR.]
Effects of attention on the phonetic importance of acoustic cues
Authors: Gordon Peter C. , Eberhardt Jennifer L.
Published: October 2005

The effect of attentional capacity on recognizing phonetic segments was studied by having subjects identify synthetic speech sounds while simultaneously performing a visual-manual tracking task. The difficulty of the tracking task was manipulated in order to vary the amount of attention available for speech perception. Within the speech stimuli, acoustic cues to segment identity were manipulated so that trading relations could be assessed under different levels of attention. The relative importance of the acoustic cues to perceptual identification changed, depending on the difficulty of the concurrent tracking task. It appears that acoustic cues that are easily encoded make an increased contribution to phonetic judgments when listeners are unable to pay close attention to a speech sound. In this way, attention to the speech stimulus is similar to other factors (e.g., environmental interference, stage of development, and hearing loss) that also affect the relative contribution of acoustic cues to phonetic perception. It also appears that the initial stages of speech perception can make use of general, modality-independent attentional resources. [Work supported by AFOSR.]
Some auditory-visual interactions are post-categorical
Authors: Pastore Richard E. , Layer Jody K. , Logan Robert J. , Tousman Stuart A. , Hsu Hanson
Published: October 2005

The McGurk effect recently has been variously conjectured to be based upon a precategorical interaction or integration of information from the auditory and visual representation of the linguistic event. The conjecture of precategorical interactions, which rejects the original notion of post-categorical interactions, may be in the form of a specialized module for speech or language, or may contribute to a logic-decision process, either of which leads to a relatively discrete perception of speech. While not disputing such precategorical interactions for language, the research to be described demonstrates post-categorical interaction of auditory and visual representations of phonemes. The stimuli are nonword CVC syllables. The set of auditory stimuli is edited from natural speech, while the set of visual stimuli includes the orthographic representations of the auditory stimuli syllables. Evidence for interactions in identification of initial phonemes is based upon both error rates and reaction times. [Research supported in part by an NSF grant to the first author.]
The perceptual distance between synthetic /s/ and /∫/ syllables in different vocalic contexts
Authors: Cheesman Margaret F. , Van Tasell Dianne J.
Published: October 2005

A nine-step, synthetic, /s/-/∫/continuum was crossed with a fivestep continuum of /i/ to /u/-like vowels to form 45 CV syllables. Thirteen listeners categorized the fricative portion of these syllables as /s/ or /∫/ in a 2AFC task. Identification judgments were strongly influenced by the vowel context; for example, more fricatives were identified as /s/ when they preceded /u/ than when they preceded /i/. The same group of listeners used a triadic comparison procedure [Levelt et al., Br. J. Math. Star. Psychol. 19, 163–179 (1966)] to provide estimates of the perceptual distances between a subset of the fricatives in different vowel contexts. Perceptual distances were smaller for within-category pairs than they were for across-category pairs. Relations between the perceptual distance and the acoustic similarity of stimuli will be discussed. [Supported by NIH Grant No. NS12125, the Bryng Bryngelson Research Fund, and an SSHRC doctoral fellowship to MFC.]
Phoneme processing in the perception of spoken Japanese words
Authors: Amano Shigeaki
Published: October 2005

Cohort theory, a speech perception model based on serial phoneme processing, is studied in respect to the lexical access of Japanese words. Six subjects made a lexical decision by pushing a key as soon as possible after hearing real Japanese words or nonwords that were 2-5 mora long and had various nonword discrimination points. A nonword discrimination point was defined as a point where no other real word candidates could be found in a cohort by serial phoneme processing. The results show that the reaction times measured from the nonword discrimination point are not constant. The nearer the nonword discrimination point is to the end of the word, the shorter the reaction times are. This indicates the existence of parallel phoneme processing. Moreover, the reaction times for short words are longer than for long words when they are measured from the end of the word. This shows that short words are buffered somewhere in the word perception process until the lexical decision is made. However, there is also evidence of serial processing. The reaction times from word onset gradually increase along with time to the nonword discrimination point. These results suggest that phoneme processing is quasiserial. Parallel or buffered processing should be incorporated into cohort theory along with serial processing.
The role of attention in speech perception
Authors: Scharf Bertram , Dai Huanping , Miller Joanne L.
Published: October 2005

This study investigated the role of auditory attention during speech perception. The syllables /da/ and /ga/ were synthesized so that they differed in initial burst and third formant transition; the critical distinguishing information was in the vicinity of 2.5 Khz. Discrimination was first measured with a 1I, 2AFC procedure under six masking conditions. Performance was near chance (55% correct) when the masker was centered at 2.5 Khz, but increased as the masker moved away from this critical frequency region, reaching 100% with the masker at 1 kHz. Next examined was whether listeners' attention as they performed the task was focused specifically on the 2.5-Khz region, or spread across all frequency regions. In one condition, subjects were asked to discriminate the syllables when a weak 90-ms, 1-kHz tone was added to /da/ and, in the other condition, when a weak 90-ms, 2.5-Khz tone was added to /da/; subjects were not informed that the tones had been added. In both conditions, the masker was centered at 2.5 kHz. Performance was at 58% when the 1-Khz tone was added, but at 75% when the 2.5-kHz tone was added. In two control conditions, it was found that, when subjects were informed that the tones had been added, so that they could focus their attention on the relevant frequency regions, performance increased substantially (to 90% correct) for the 1-Khz condition, but only slightly (to 81% correct) for the 2.5-kHz condition. These results suggest that, when attempting to discriminate syllables, listeners focus their attention on the specific frequency region critical to the distinction. [Work supported by NIH.]
Some parameters affecting the perception of voicing for fricatives
Authors: Glicksman Laura B. , Stevens Kenneth N. , Blumstein Sheila E.
Published: October 2005

Measurements of voicing have shown that low-frequency periodicity can usually be found at VC and CV boundaries, specifically in the initial 30 ms or the final 20 ms of a phonetically “voiced” fricative. The purpose of this study is to determine the perceptual importance of voicing at VC and CV boundaries for fricatives. The VCV tokens with alveolar fricatives were synthesized with various fricative durations and various patterns of voicing within the consonant. Listeners identified the consonant as either [s] or [z]. Short fricatives are heard as [z] but, for durations longer than 70 ms, voicing must occur at either the VC or CV boundary. Voicing placed in the middle of the fricative did not elicit [z] responses. When voicing occurred at the boundaries, perception was influenced by fricative duration, and by amplitude and duration of voicing. Possible theoretical interpretations of the data will be discussed. [Work supported by grants from NINCDS to Brown University and to MIT.]
Using reaction times to explore the mechanisms of selective adaptation
Authors: Kat Donna , Samuel Arthur G.
Published: October 2005

Previous research has demonstrated that the identification shifts induced by selective adaptation are accompanied by reliable reaction time changes. For example, if a /ba/-/da/ test series is used, reaction times to tokens at the /b/ end of the series will be higher after /ba/ adaptation than after/da/adaptation; at the /da/ end, the reverse is true [A. G. Samuel, Cog. Psychol. 18, 452–499 (1986)]. The present study extends these results in three ways. First, a neutral adaptation condition (/a/) is used to provide an appropriate baseline for measuring absolute reaction time changes. Second, combination adaptors (/ba/-/da/) are used to look for reaction time changes in the absence of labeling shifts. Third, cross-series reaction time changes (/pa/,/ta/, and /pa/-/ta/ on a /ba/-/da/ series) are examined. The pattern of reaction time changes will be used to address the level of processing affected by adaptation, and to clarify the mechanism of the effects (contrast versus fatigue).
New solutions of the wave equation for a bowed string
Authors: Kojima Takashi
Published: October 2005

In 1860, H. v. Helmholtz concluded, by observing the movement of each particular point of a bowed string with his vibration microscope, that the highest point of the bowed string travels along upper and lower parabolic arcs. Since then, his conclusion has been accepted and little attention has been paid to the shape of the envelope of the bowed string. The experimental work of Kondo et al., which focused on this subject [6th I.C.A., N-2-4 (1968)], stimulated the present work. This work re-examines the general solution of the wave equation by giving some initial conditions and then, in addition to the normal Helmholtz solution, obtaining two new solutions, whose shapes are of an elliptic arc and of hyperbolic arc, respectively. These three solutions, parabolic, elliptic, and hyperbolic, coincide with each other when a specific parameter approaches infinity. One experimental result found in the literature seems to represent an elliptic solution.
Measurements of the response of gas-filled micropores to ultrasound by using PVDF transducers
Authors: Wu Junru , Nyborg Wesley L.
Published: October 2005

Techniques that use hydrophobic polycarbonate thin sheets containing randomly spaced, fairly uniform small holes, immersed in water to trap air bubbles, have been found useful in biophysical experiments. Using such sheets, the scattering from a stable array of small air-filled holes exposed to ultrasound has been investigated [D. L. Miller, J. Acoust. Soc. Am. 71, 471–476 (1982)]. A set of piezoelectric crystal transducers of high-quality factor were used during the earlier study, which allowed measurements to be made at a discrete set of frequencies. In the present work, utilization of PVDF transducers, which have flat frequency response in the range of interest, makes it possible to study the continuous frequency response of the trapped bubbles. Results of the new experiments will be presented and discussed. [Work supported by institutional funds of University of Vermont.]
The graduate program in architectural acoustics at the University of Florida
Authors: Kinzey Bertram Y. , Siebein Gary W.
Published: October 2005

The graduate program in architectural acoustics at the University of Florida offers a course of study that leads to Master of Architecture and Ph.D. degrees. The focus of the program is to address the theories and practice of acoustics in architectural settings. Core courses are offered in basic acoustical principles and the acoustical design of buildings. Students complete a major design project assuming the role of an acoustical consultant. They provide comprehensive acoustical recommendations for all aspects of a major building including room-shaping and finishes, mechanical system noise control, and the design of a speech reinforcement system. They gain experience in field measurement techniques, computer calculations, ray diagramming, and scale modeling. Additional study is pursued through individual research and thesis projects. Examples of previous projects include acoustical scale modeling, the acoustics of the case of an organ, measurements of new indices of acoustical quality in various auditoria, and studies of subjective musical quality. The program is supported by a dedicated laboratory and studio space and three faculty.
Graduate education in acoustics at Iowa State University
Authors: Holger David K.
Published: October 2005

Research and graduate education in a number of areas of acoustics and noise control are carried out in the Department of Engineering Science and Mechanics at Iowa State University. Graduate level courses in theoretical acoustics, signal processing, ultrasonic nondestructive measurements, and mechanical vibrations are offered by the department. Facilities for graduate instruction and research include an anechoic chamber with computer-controlled microphone scanning system, three two-channel FIT analyzers, an ultrasonic nondestructive testing laboratory in the Non-Destructive Evaluation Center, and several thousand square feet of laboratory space. Eight faculty members and more than 20 graduate students are conducting research in spatial transformation of acoustic fields, boundary integral equation methods in acoustics, acoustic intensity methods, ultrasonic transducer characterization, inverse problems in acoustic scattering, and ultrasonic nondestructive evaluation. Descriptions of some typical graduate student programs, selected graduate courses, and specific research projects will be included in the paper.
Graduate program for education in psychological acoustics at the Kyushu Institute of Design
Authors: Teranishi Ryunen
Published: October 2005

In Japan, the education of graduate students in universities is done mainly by colloquia and by research performance for their theses or dissertations. Therefore, lecture-type classes are few. In the Institute (a unique university in Japan, which has a department and a course mainly related to acoustics), the above-mentioned conditions are quite similar to other universities. However, the Institute has more acoustical subjects (nine in acoustics and three in musicology, total number of credits = 32) than other universities in Japan. Usually, these have only one or two, as minor subjects, in the electric or electronic engineering course or in the architectual engineering course. Subjects in the Institute program have been carefully selected with the intention of producing acoustic engineers with a deep understanding of nature and the characteristics of humans and the society. Accordingly, almost half of the subjects are related to psychological acoustics, and about one third to musicology. However, the Institute's aim is not to graduate professional musicians or psychologists.
A system for knowledge-based architectural acoustics
Authors: Yanagida Masuzo , Hidaka Takayuki , Yamaguchi Takahira
Published: October 2005

The concept of knowledge-based architectural acoustics is proposed showing a prototype of an intelligent system for room acoustic design as a practical example. The system is composed of two major subsystems, an acoustic field simulator and an expert system for room acoustics. The principal objective of the system is to assist acoustic designing. This system can also be used for training acoustic designers and for evaluating both design policies and design parameters to represent the physical characteristics and subjective aspects of sound fields, such as spaciousness and definition. The field simulator is utilized effectively by refining the calculation algorithms and checking whether the performance coincides with actual fields. The validity of the design parameters is examined by listening to the output sounds of the field simulator. Some examples of the system's performance on rooms with simple shapes are shown, and the principle of evaluating subjective design parameters is indicated.
Design of periodic signals for acoustic system identification using hard-limiting transducers
Authors: Johns Murray D.
Published: October 2005

Using periodic signals for underwater acoustic system identification enables signal averaging to be used to increase the signal-to-noise ratio of the received signal. The desired properties of such a signal are partially determined by the properties of the electroacoustic transducers used. This paper shows how to design signals to give the maximum receiver signal-to-noise ratio when the desired source signal is known but the source transducer has an ideal hard-limiting characteristic. In such cases, it is almost always better to generate a signal different from the desired one and to correct for the difference (using linear signal processing) in the receiver. Examples of such designs and the improvements to be gained will be given.
Directivity index of volumetric arrays using optimal amplitude shading
Authors: Williams Robert Bruce , Meacham M. Phillip
Published: October 2005

Anderson and Munson [J. Acoust. Soc. Am. 35, 1162–1168 (1963)] showed that the directivity index (DI) of an infinitely densely populated spherical shell array was about equal to that of a sphere. Extrapolating to discrete elements, this means the shell requires far fewer elements. They did not compute DI using amplitude shading, due to the impractical cost of such systems at that time. Today's technology removes that constraint. This work revisits the problem with shading, using an approach for choosing the amplitude shading coefficents that maximizes DI [H.S.C. Wang, J. Acoust. Soc. Am. 57, 1076–1084 (1975)]. Calculations have been made for the DI of shaded cubic volumetric arrays, forming beams perpendicular to one of its faces, in the presence of isotropic noise. Results show that for 27 and 125 element arrays with element matrix spacings of 1/2 wave-length, a full 10 log (number of elements) can be obtained for DI. Work is underway to investigate larger arrays and smaller spacings. The approach will also be extended to nonisotropic noise fields. [Work supported by NORDA and NOSC exploratory development programs.]
Stress-induced speech and speech recognition
Authors: Gordon John W.
Published: October 2005

The performance of speaker-dependent speech recognizers typically degrades when the speech input varies from its trained norm, particularly when the speaker is placed under stress. To evaluate such performance, a data base of isolated words, collected with the speakers under various simulated environmental conditions or emotional states, was used to train and test two commercial recognizers. For comparison purposes, the recognition performance of human subjects on the same data base was also obtained. The speech was transcribed, and the acoustic properties of the phonetic tokens from various speaking styles were analyzed; this facilitated the investigation of correlations between degradations in performance and speech variabilities. Results indicate the importance of direct representation of speech knowledge in speech recognition systems that are used in situations where changes in speech properties are to be expected.
Acoustical imaging using the propagation law of mutual intensity
Authors: Ueha Sadayuki
Published: October 2005

Vibrating objects are successfully imaged using acoustical holography where the radiated sound is used both for the object and the reference signals [Ueha et al., Appl. Opt. 14, 1478–1479 (1975)]. The same technique can be used both for partially coherent and incoherent objects. But the characteristics of the reconstructed images depend on the degree of coherence of the objects under study [Ueha et al., Opt. Commun. 18, 488–491 (1976)]. In order to treat reconstructed images qualitatively, this imaging technique is reviewed from the viewpoint of the propagation law of mutual intensity. In addition, the effects of the spectral bandwidth and signal averaging time upon the reconstructed images are discussed.
A noise source detection system using nearfield acoustical holography
Authors: Hayashi Takuro
Published: October 2005

A noise source detection system based on nearfield acoustical holography has been developed. This system consists mainly of a measuring section, a hologram section, and a reconstruction section. In the measuring section, acoustic waves radiated from sound sources are sampled simultaneously with both a scanning microphone in the measurement plane and a spatially fixed reference microphone. The hologram section obtains the hologram data set from the cross spectra between the two microphone signals at each sampling point. The reconstruction section processes the hologram data to reconstruct the acoustic wave field radiated from the original sound sources. In addition, three ways for acoustic image reconstruction, using evanescent wave components or plane-wave components, or both, can be selected at this reconstruction stage. Sound source identification tests for sound sources, such as loudspeakers, were carried out using this system. The results showed that a good spatial resolution could be obtained, and that the sound source locations could be identified satisfactorily.
Boundary element implementation of generalized nearfield acoustic holography for an axisymmetric geometry
Authors: Borgiotti Giorgio , Sarkissian Angie , Schuetz Luise , Williams Earl
Published: October 2005

Generalized nearfield acoustic holography (GENAH), which allows a three-dimensional reconstruction of the sound-pressure field from two-dimensional measurements of the pressure in the nearfield of a vibrating structure, is extended to nonseparable coordinate systems, in particular, a general axisymmetric structure. This technique allows a high-resolution imaging of the normal velocity at all points on the surface of the body. The Helmholtz integral equation formulation is used, with singular value decomposition of the matrix to treat the presence of evanescent waves in the nearfield. Symmetry principles are implemented to reduce computation times for the axisymmetric geometry.
Transient noise source location via computational and experimental surface acoustic intensity methods with an application to a punch press
Authors: Chun D. H. , Hodgson Thomas H.
Published: October 2005

This paper outlines a recent extension of the boundary element computational method to the case of transient acoustic energy flow from a complex mechanical structure. Results are compared with experiment by use of the fiber-optic surface acoustic intensity probe. The computational method presented combines an explicit and implicit method to reduce the numerical instability inherent in the method and to increase the efficiency of the solution for such problems. Also, a recent improvement of the fiber-optic surface acoustic intensity probe will be discussed to illustrate the validity of the experimental method, which eliminates the phase error in higher frequency measurements and increases the sensitivity and linearity. The application of these methods on a small punch press will be described to demonstrate the capability and accuracy of the methods. Importance of nonresonant frequency modes in the transient mechanism will also be discussed by analyzing the simulated frequency responses where resonant components have been eliminated.
Visualization of acoustic scattering from submerged objects
Authors: Gaumond Charles F. , Schuette Lawrence C. , Gatto Joseph G.
Published: October 2005

The scattering of harmonic plane waves from rigid spheres, rigid cylinders, and cylindrical shells was visualized using normal mode series solutions displayed on an IRIS-4D color graphic super-workstation. The temporal behavior of the pressure and the instantaneous energy density were studied for ka<10 in a square region 150×150 ka units in size. Particular attention was paid to processes that generate phase shifts in the forward direction. A video presentation of the temporal behavior will be shown.
Stability of marching methods applied to elliptic models of underwater sound propagation
Authors: Knightly G. H. , St. Mary D. F.
Published: October 2005

Recently [G. H. Knightly and D. F. St. Mary, in Computational Acoustics: Wave Propagation, edited by D. Lee, R. L. Sternberg, and M. H. Schultz (Elsevier, Amsterdam, 1988), pp. 397–407] an investigation of marching methods for the farfield elliptic equation of underwater sound propagation was begun. The elliptic model is obtained on transforming the Helmholtz equation, written in terms of the pressure p, by taking p(r,z) = x(r,z)H0(1)(k0r), and using the farfield properties of the Hankel function H0(1). Using a marching scheme to solve the resulting equation assumes that this elliptic problem has been cast as an initial value problem, and of course initial value problems for elliptic partial differential equations are in general not well posed and yield unstable numerical schemes. But such methods may be appropriate under certain restrictions on the parameters. Two marching schemes for computing solutions of the model are demonstrated and stability for these schemes is discussed.
Ray analysis near the critical angle
Authors: Tindle C. T. , Plumpton N. G.
Published: October 2005

The sound field due to spherical waves from a point source reflected from the interface between two fluids is re-examined. The conventional ray theory description in terms of a specular ray and a lateral wave is satisfactory at both short and long ranges but fails at intermediate ranges when the specular ray is near the critical angle. Ray theory with beam displacement leads to the formation of a caustic and gives good results beyond the caustic range but fails at shorter ranges. In the present work, ray theory with beam displacement is extended to allow for rays with complex wavenumber by associating rays with saddle points of a complex integrand. The field is calculated by the method of steepest descents and compared with the exact integral result. The field is smooth and continuous through both critical angle and caustic and in good agreement with the exact result. The method is based on eigenrays and should enable ray calculations to be extended to accommodate critical angles. The description of beam displacement caustics is similar to earlier treatments of refraction caustics in terms of complex rays.
Directional spectral and particle motion measurements of underwater seismic noise
Authors: Goodman Dean , Yamamoto Tokuo
Published: October 2005

Observations of underwater seismic noise were measured using arrays of ocean bottom seismometers (OBS) deployed in shallow waters on the continental shelf off New Jersey. To reduce uncorrelated noise on the seafloor, instruments were buried to 0.5-m depth with the aid of an underwater hydraulic pump. Array geometry and dimensions were tuned to measure microseismic noise in the period band of 2–6 s. Directional spectrums measured by maximizing cross spectrums between array elements for a given wavenumber indicate that slow microseismic waves having an apparent velocity of 200–300 m/s were traveling out of the east. Phase velocities determined from normal mode analysis agree closely with measured phase velocities. From particle motion analysis for single OBS elements, microseismic noise is shown to exhibit retrograde motion at the seafloor. These experiment results indicate that microseismic noise is due to Scholte-type interface waves. [Work supported by ONR, code OA.]
Velocity and attenuation from shallow-water experiments
Authors: Stoll Robert D.
Published: October 2005

During the summer of 1987 a series of shallow-water experiments were performed for the purpose of determining the seismoacoustic properties of the unlithified sediments near the seafloor. These experiments, which utilized a small explosive source and a string of receivers on the seafloor, were designed to maximize the excitation of shear and interface waves. By using two-axis, gimballed geophones spaced at 5-m intervals along a cable, a robust set of data defining motion in a vertical plane parallel to the direction of propagation was obtained. By analyzing the dispersion of surface waves and the amplitude and travel time of diving shear waves, the data were inverted to obtain shear and dilatational wave velocity and the attenuation of shear waves as a function of depth. One data set for a thick sand deposit in New York Harbor showed strong gradients in velocity and attenuation near the water-sediment interface in good agreement with a theoretical model which takes into account the rapid change in effective stress (overburden pressure) in the near-bottom sediments. [Work sponsored by ONR, code 1125 QA.]
Scalar correspondence with the overtone series
Authors: Blum Bennett S.
Published: October 2005

It is frequently stated that the overtone series produces the notes (starting on C) C, G, E, (flat) B♭, D, (flat) F♯, and (flat) A. These notes give (a mode of) the ascending melodic minor scale. Since this scale is widely used in western music, one is led to seek a relationship between the physics of overtones and the scale. Unfortunately, it turns out that the overtone corresponding to A is really closer to A♭. If a scale is constructed with the notes in the tempered scale closest to the tones in the overtone series, the results are C, D, E, F♯, G, A♭, B♭, C. This is not a scale in western music. Furthermore, the overtone associated with F♯ is almost as close to F as it is to F♯, and only tempered C, D, E, G are really close to tones in the overtone series. Thus while the overtone series may be applicable to intervals of second, third, fourth, and fifth, and to the major triad, it cannot be applicable to other intervals, the minor triad, or any scales.
Experimental developments for the measurement of violin bridge admittance
Authors: Boutilion Xavier , Weinreich Gabriel , Michael Nicholas R.
Published: October 2005

In a previous report [Weinreich and Yoder, J. Acoust. Soc. Am. Suppl. 1 81, S83 (1987)], a method for measuring a 2×2 admittance matrix of a violin bridge by comparing its motion with and without external loading was presented. Further investigations have shown that motion along the string direction cannot be neglected, so that a 3×3 matrix is required. Therefore, the phonograph pickup used for sensing bridge motion is replaced by three ultralight accelerometers. Their transverse sensitivity (3% to 5%) proves to have a critical effect on the validity tests for the measurements (symmetry of matrix, positiveness of dissipation). Therefore, a procedure to calibrate their sensitivity in all three directions was developed. Several possibilities have been tried as external loading: an ordinary mass, a mass attached with a wire, a spring. The first loads all three translational degrees of freedom; the others, if properly designed, only one. The approximations involved in each case will be discussed, and current results presented. [Work supported by NSF, CNRS, and French Ministry of Culture.]
Investigation of the acoustics of plucked string tones based on the analysis of their time-varying spectra
Authors: Chen Kwok-ping
Published: October 2005

Time-varying vibration patterns of selected tones of classical guitar, Chinese pipa, and ch'in are investigated. Based on Weinreich's theory [1977,1979] on horizontal and vertical frequencies and decay rates, partial amplitude envelopes of these tones are classified into four types. Using the normal finger-tip excitation method, it is found that certain envelope types appear more frequently in one instrument type than another. The results show that these instrument tones can be described in terms of the four types. The classical guitar open-string tone A-110 Hz is further invesitigated, using the pick-edge excitation method, playing at nodal position N which is L/N from the bridge. A fifth type envelope is found for the so-called missing modes [Young, 1800; Benade, 1976; Fletcher, 1984; Hall, 1987]. These modes, which are initially attenuated, subsequently rise with a more gradual attack to reach significant amplitudes.
Matched-mode processing corrections for array tilt and bottom type
Authors: Mercer James A.
Published: October 2005

In a related effort, Homer Bucker has shown that matched-mode processing for an unknown sound-speed environment can be significantly improved if correction factors for the mode-line amplitude functions can be determined. The correction factors are obtained when a source with known location is available to calibrate the system. This paper describes the results of applying the same techniques for simulated cases of unknown array tilt and bottom characteristics.
Self-consistent modeling of signal and noise in a three-dimensional environment
Authors: Perkins John S. , Kuperman W. A. , Ingenito F.
Published: October 2005

Previous propagation work is extended to model surface noise, shipping, and signal sources in a fully three-dimensional environment. The noise cross-spectral density matrix for a vertical array is computed as the sum of a local contribution and propagation from distant small patches of ocean surface. Propagation from any point to the array is made efficient by precalculating and storing the local normal modes for every distinct path. The usual cylindrical coordinate system for the propagation calculation is obtained, and various sound sources are added at any location and are propagated to the array. For example, a storm is simulated by increasing the source strength of the appropriate patches of surface. Results demonstrate that the vertical directionality on the array reflects the range-dependent environment.
An electronic sound cancellation system for air-conditioning ducts
Authors: Hyodo Hideki , Hamada Hareo , Miura Tanetoshi , Takahashi Minoru , Gotohda Ryusuke , Yoshimura Yasushi , Kuribayashi Taku , Akasaka Akio
Published: October 2005

Recently, research on an electronic sound cancellation system that is able to cancel noise using sound of the opposite phase has been conducted. This kind of noise control technique is referred to as active noise control (ANC). In order to realize a practical working ANC system, it has been considered desirable to apply the LMS (least-mean-square) algorithm to such a system's control, especially the VS (variable step)-LMS algorithm due to its high speed of convergence. The realization of an ANC system of the stand-alone type is presented. Experiments were conducted in actual air-conditioning ducts having a cross section of 500 × 500 mm by using the realized stand-alone type active noise control system with the VS-LMS algorithm. Below the cross-mode frequency, more than 15 dB of attenuation was obtained for broadband random noise, even in the presence of an air flow of 3.7 to 9.0 m/s.
A study of active control of sound transmission through a panel into a cavity
Authors: Jie Pan
Published: October 2005

A technique for controlling noise transmitted into the interior of a cavity involves use of point force actuators on the boundary structures. This paper is a study of the optimal actuator positions and the maximum theoretical reduction of the sound transmission through a panel into a rectangular cavity. Results obtained demonstrate that the noise reduction by the actuators depends upon the nature of the incident sound. Details of the manner in which the control forces act on panel-cavity system are also discussed.
Mechanisms of active sound control
Authors: Hansen Colin H. , Snyder Scott D. , Bies David A.
Published: October 2005

Active noise control applied to sound propagating down ducts has been successfully demonstrated by a number of researchers. The active control of small three-dimensional sound fields (such as those found in ear muffs) has also met with considerable success. However, application of active control techniques to large three-dimensional sound fields, such as those found in rooms and vehicle interiors, has only been partially successful, generally resulting in sound reduction in some locations at the expense of an increase in other locations. The lack of success is easily explained if the mechanism by which successful active control is achieved is fully understood. This paper presents the results of an experiment that demonstrates the mechanism by which successful active control is achieved in a duct and conclusions are made regarding the requirements for successful active control in large three-dimensional spaces, which would enable an overall, rather than just a local, reduction in sound level to be achieved. [Work supported by the Sir Ross and Sir Keith Smith fund and the University of Adelaide.]
Acoustic waveforms in a fluid-filled borehole with axially periodic thin rigid cylinders of finite length
Authors: Chang S. K.
Published: October 2005

The interference of acoustic waveforms in a fluid-filled borehole by a periodic structure in the hole is investigated theoretically. The structure consists of a series of finite length thin rigid hollow cylinders periodically located along and concentric with the axis of the hole. Numerical results are obtained using the Galerkin's method. The radial displacement field in the gaps between the sections of the rigid cylinders is expanded in a set of basis functions that satisfy the singularity condition at the edges of the cylinders. The time domain signals are calculated for a point source and point receivers distributed on the hole axis inside the periodic structure. The interference effects of the structure are observed by varying the period and the height of the hollow cylinderical sections. It is found that the interference is dominated by the open-cavity resonance of the hollow cylinders. The resonance can be avoided by choosing proper dimensions of the cylindrical sections. Without the cavity resonance, the interference is minimized and the periodic structure is almost acoustically transparent in the borehole.