bluuebunny/crossref_metadata_2025 huggingface dataset:
Sound pressure field into reverberant chamber containing structures excited acoustically
Authors: Nélisse Hugues , Berry Alain , Charron F. , Nicolas Jean
Published: October 2005

A method has been developed to obtain the sound pressure field for all points in a large reverberant chamber whether or not it contains a flexible structure excited acoustically by a so-called diffuse field. The approach is based on an integral equation method for the pressure in the chamber. The Helmholtz–Kirchhoff theorem is expressed using a pressure jump function across the surface of the structure [see, for instance, D. Ouellet etal., J. Acoust. Soc. Am. 89, 2131–2139 (1991)]. A modal expansion over modes of rigid cavity and in-vacuo modes of the structure is used. It leads to a set of linear coupled equations reflecting the existence of a fluid–structure interaction. Numerical resolution of the set of equations allows one to obtain physical quantities of interest, such as the distribution of pressure on each side of the panel and consequently the jump of sound pressure across the structure. Convergence of the solution, as well as its frequency range of validity, will be discussed.
Numerical predictions of acoustic radiation by unbaffled, finite plates from non-intrusive velocity measurements
Authors: Harbold Todd L. , Burroughs Courtney B.
Published: October 2005

The spatial distribution of the amplitude and phase of the velocity of vibrating rectangular flat plates is measured using a scanning laser Doppler vibrometer. These measurements are conducted in a nonanechoic environment with the plate in air and water, driven by a single shaker at frequencies of resonance and nonresonance. Using free-field Green’s functions and fast Fourier transform (FFT) algorithms, the free-field acoustic radiated field for the unbaffled plate is determined. In order to compute the acoustic radiation, the velocity field must be specified in the plane of the plate, both on and off the surface of the plate. Therefore, an iterative process is used to determine the velocity field off of the plate surface. Descriptions of the plate vibration measurements and algorithms used to compute the free-field acoustic radiated field are presented and discussed. Examples of results of measurements conducted at and off resonance are given.
Simultaneous measurements of in-plane and out-of-plane surface displacements of shell vibrations at several points by an array of miniature laser probe heads
Authors: Yang Ming , Berthelot Yves H. , Jarzynski Jacek
Published: October 2005

Experiments are being conducted to measure directly by laser Doppler interferometry the in-plane and out-of-plane components of the surface motion of a vibrating shell simultaneously at several surface points. The proposed design requires three Bragg cells to generate three coherent optical beams with different carrier frequencies. Three 1×4 fiber optic connectors are used to transmit each beam to four surface points. At each point, both the out-of-plane component and one of the in-plane components of the displacement vector are measured. The light scattered by the vibrating surface is Doppler shifted and collected into a large core multimode fiber situated close to the surface and terminated by a photodiode. The signals are acquired and digitized for direct phase demodulation on a computer by means of Hilbert transforms. [Work supported by ONR.]
Direct measurements of axial and circumferential in-plane surface displacements of shell vibrations with an automated scanning laser Doppler interferometer
Authors: Kil Hyun-Gwon , Jarzynski Jacek
Published: October 2005

The surface displacement of a vibrating shell has been directly measured by laser interferometry. The system is designed to measure the two in-plane components of the surface displacement: the axial and the circumferential components. The system is completely automated for scanning around the cylindrical shell both circumferentially and axially, for focusing at each point until the detected signal is stable and sufficiently above the noise level, and for recording and transferring the data to a computer. The data are analyzed in the frequency-wave-number domain with fast Fourier transforms and Prony decomposition to separate the extensional waves (fast waves, low wave numbers) from the flexural waves (slow waves, high wave numbers). The shell under consideration is 0.94 m long and 1.6 mm thick. It is freely suspended in air and it is excited radially by a shaker at various frequencies up to twice the ring frequency which is about 10 kHz. The optical probe head is very compact (about 4 by 4 cm) and it is designed to take full advantage of optical fiber technology. [Work supported by ONR.]
Experimental investigation of vibration excitation for structural sound radiation study
Authors: Couture Dany , Champoux Yvan , Lesage Claude
Published: October 2005

Over the last decade, several models have been proposed for modeling the sound radiation of vibrating structures. These analytical or numerical models are frequently considered for simple or semi-complex structures such as plates and shells. They were sometimes validated through simple experiments and the theories were often judged valid when only trends between theory and experimental results were obtained. The discrepancies were almost systematically associated with the limitations of the experimental set up. In order to use experimental results as a basis to challenge or validate the theories, accurate results must be obtained. The precise measurement of the structure excitation is deemed necessary. Laterally flexible rods are often used between the shaker and the structure to avoid the introduction of unwanted excitations or the inadvertent modification of the structure. A load cell that measures moments and longitudinal force has been developed and was used to study the influence of different excitation configurations. Alignment must be taken into account. The use of flexible rods will be examined and useful guidelines will be established.
Mode localization phenomenon analysis on infinite ribbed plates
Authors: Juang Ten-Bin , Pate Anna L. , Flatau Alison B.
Published: October 2005

An analysis of the mode localization phenomenon is demonstrated for disordered structural systems of infinite ribbed plates in air and in water. While for infinite ordered ribbed plates there exist frequency passbands for which vibrations propagate extensively throughout the structure, the introduction of disorder disables the propagation of vibrations to arbitrarily large distances and confines vibrations within a region close to the excitation force. Based upon the mathematical model built in a paper to be published, inertance transfer functions and vibrational responses due to line force excitation are implemented to investigate the passband and stopband characteristics and the localization phenomenon of multiple-rib plates in air and in water. Two facts are confirmed. First, water provides an energy path that reduces the decoupling effect between the bays of ribbed plates observed in the presence of air and consequently, diminishes the localization phenomenon. Second, passbands can merge at certain frequencies, as determined by rib size and dynamic properties, but have nothing to do with the number of ribs or the transmission span. For completeness, acoustic responses are presented for some specific cases of the structural system as well.
An acoustic analysis of the effects of cue deletions on the perception of stop consonants
Authors: Guadagno Roberto , Jamieson Donald G.
Published: October 2005

Cue deletion was used to examine the contribution of various acoustic cues to the identification of the English stop consonants (/b/, /d/, /g/, /p/, /t/, /k/). Stimuli were based on tokens spoken by four talkers (two men and two women) in a syllable-medial context (/(inverted vee) Cil /). Cues within five regions centered on the burst were examined: (1) the transition from the preceding vowel into consonant closure, (2) the closure interval, (3) the consonant release, (4) the initial transition from release to the following vowel, and (5) the final transition to the vowel. Each region either remained at the full normal amplitude or was attenuated fully (i.e., effectively replaced by silence). Four listeners who had normal hearing identified the 768 signals constructed from each of the possible combinations of attenuated and full-amplitude regions (i.e., 32 combinations ×6 stimuli ×4 speakers). Analyses of these identification responses demonstrates the relative importance of each region, and of each combination of regions, for consonant identification. Acoustic measures explain some aspects of the confusion errors which occurred.
Selective adaptation of context-conditioned voiceless fricatives
Authors: Cheesman Margaret F. , Greenwood Kristina G.
Published: October 2005

Two fricative-vowel continua were synthesized. One ranged from /si/ to /(sh)i/ and the other from /su/ to /(sh)u/. Listeners’ identification of the consonant portion of the syllables, as either ‘‘s’’ or ‘‘sh,’’ depended on both the frequency of the frication noise and on the vowel quality. Selective adaptation of these continua was investigated in three experiments. In experiment 1, the endpoints of the continua were used as adaptors, and the identification boundary shifted towards the adapting stimulus. In experiment 2, ambiguous frication noises (that were identified as /s/ before /u/ and /(sh)/ before the /i/) adapted the identification boundary in opposite directions, depending on which of the two vowels followed the noise and, therefore, on the perceptual identity of the consonant. In experiment 3, the isolated /i/ and /u/ vowels were demonstrated to be ineffective as adaptors. The selective adaptation effects observed in these experiments were not determined by the acoustical information in the consonant or the vowel alone, but rather the context-conditioned percept of the fricative. [Work supported by NSERC.]
Training Japanese listeners to identify English /r/ and /l/: A replication and extension
Authors: Pisoni David B. , Lively Scott E. , Yamada Reiko A. , Tohkura Yoh’ichi , Yamada Tsuneo
Published: October 2005

Monolingual native speakers of Japanese were trained to identify English /r/ and /l/ using a modified version of Logan, Lively, and Pisoni’s [J. S. Logan etal., J. Acoust. Soc. Am. 89, 874–886 (1991)] high variability training procedure. Both the talker’s voice and the phonetic environment were varied during training. Subjects improved in their ability to identify /r/ and /l/ from the pre-test to the post-test and during training. Generalization accuracy depended on the voice of the talker producing the /r/–/l/ contrasts: Subjects were significantly more accurate when words were produced by a familiar talker than when they were produced by an unfamiliar talker. Three months after the conclusion of training, subjects were given the post-test and the tests of generalization again. Surprisingly, accuracy decreased only slightly on each test, even though no training or exposure to /r/ and /l/ occurred during the 3-month interval. These results demonstrate that the high variability training paradigm is effective in modifying nonnative listeners’ phonetic perception and that these changes are extremely robust over time. [Work supported by ATR.]
Discrimination of broadband, multi-component, common-envelope signals
Authors: Ananthraraman J. N. , Krishnamurthy Ashok K. , Feth Lawrence L.
Published: October 2005

The EWAIF model was based on the listener’s ability to discriminate between narrow bandwidth, two-component complex tones. These tones shared a common envelope as described by Voelcker [Proc. IEEE 54 (1966)]. Later, the EWAIF model was applied to multi-component signals such as those used in profile analysis. Those signals were neither narrow bandwidth nor common envelope signals; however, the EWAIF model predicted a constant difference in pitch for just-detectable increments to the center component of otherwise flat-spectrum signals. Recently, Versfeld and Houtsma [Q. J. Exptl. Psychol. 43 (1991)] have applied to EWAIF model to a number of periodic signals, some of which are broadband. They report the failure of the model to predict the discriminability of these signals. Versfeld and Houtsma did not ensure that their test signals were common envelope pairs. Thus, the addition of envelope differences to their signals may have given their listeners additional cues upon which they could base discrimination. Signal pairs have been defined that are similar to those used by Versfeld and Houtsma, but that have common envelope properties. Recently the EWAIF model have been reformulated into what is better called an intensity-weighted average of instantaneous frequency (IWAIF) model. In this presentation, the relationship between EWAIF and IWAIF model predictions for discrimination of relatively simple, periodic signals are demonstrated and then the IWAIF model is applied to the discrimination of broadband complex signals. [Work supported by AFOSR.]
Critical bands for envelope cues
Authors: Berg Bruce G. , Southworth Curt , Turner Matthew
Published: October 2005

Previous studies show that temporal envelopes can be used to discriminate complex sounds. This study estimates the effective bandwidth over which the auditory system extracts an envelope. Listeners discriminate a stimulus consisting of n, equal-intensity tones centered at 1000 Hz, from a stimulus with an intensity increment of the 1000-Hz tone. A roving level procedure is used to degrade absolute intensity cues, while variation of the digital-to-analog conversion rate degrades pitch cues. Theoretically, only envelope cues are preserved. Across conditions, tones are equally spaced with separations of 10, 20, 40, or 80 Hz. Within conditions, the number of tones varies from 3 to 41. Thresholds increase as the number of components increases until a certain bandwidth is reached, after which a ‘‘Fletcher-type’’ breakpoint is observed with thresholds either decreasing or remaining constant. For individuals, the bandwidth at the breakpoint is roughly the same across conditions. Sizable individual differences are found, ranging from a factor of 1.5 to 4 greater than critical bandwidths estimated from tone-in-noise detection tasks. A model based on the power spectrum of the envelope is evaluated. [Work supported by ONR.]
Viscous effects in the acoustic levitation of gas bubbles
Authors: Hitt Darren L. , Prosperetti Andrea
Published: October 2005

Previous experimental work on the dynamics of acoustically levitated gas bubbles [Crum and Prosperetti, J. Acoust. Soc. Am. 73, 121–127 (1983)] has shown a discrepancy between the theoretical and measured values of the levitation number, Le = ρlg/‖∇p‖, in the nonlinear regime of oscillation. In an attempt to reconcile these differences, the existing inviscid theory is extended to include the effects of viscous drag forces in the dynamic equilibrium of the bubble. [Work supported by NSF.]
Development of an algorithm for evaluating finite amplitude waves resulting from arbitrary periodic excitation
Authors: Ginsberg Jerry H.
Published: October 2005

Without incorporation of weak shock theory or an explicit description of dissipation, a numerical evaluation of the distortion of finite amplitude planar waves generated by an arbitrary period excitation is extremely limited in its domain of validity. Prior descriptions of such waves using weak shock theory have apparently been restricted to the case of initially harmonic signals. The present study describes a computational algorithm combining the small signal version of Earnshaw’s solution with weak shock theory implemented in the form of the ‘‘equal area rule.’’ The major steps in this algorithm are: (1) development of a nominal waveform at a specified location based on equal increments in the phase variable, (2) identification of (possibly multiple) shocks and the corresponding instants at which the equal area rule is satisfied, (3) coalescence of overlapping shocks associated with each of the harmonics contained in the initial signal, and (4) reconstruction of the shocked waveform using equal time increments in order to permit evaluations using FFT’s. The presentation will describe the algorithm, and use a variety of test problems to illustrate the various manifestations of shocks.
On the instability of one-dimensional acoustic standing waves
Authors: Goldberg Zinoviy A.
Published: October 2005

The paper theoretically considers the instability of acoustical standing waves in a nondispersive medium in a plane-parallel layer excited at the border of the layer or by a distributed force. The instability is caused by the nonlinear resonance interaction that sets in motion the parametric temporal amplification of the spontaneously formed standing waves. Discussion will focus on the results of the deviation of the frequencies of the parametrically excited subharmonic waves from the natural frequencies of the layer. This phenomenon was noted by a number of investigators [L. Adler and M. A. Breazeale, J. Acoust. Soc. Am. 48, 1077–1083 (1970); N.-C. Yen, J. Acoust. Soc. Am. 57, 1357–1362 (1975); Z. A. Goldberg, in ProblemsofNonlinearAcoustics, XIth Int. Symp. on Nonlinear Acoustics (in Russian), Part I (SO AN USSR, Novosibirsk, 1987), pp. 154–158]. It is shown that this phenomenon is not connected with the borders of the layer but it is peculiar to waves running in opposite directions that replaced the standing wave. The cause of these results is the instability of the standing waves to waves with deviated frequencies.
Serial position effects in temporal pattern discrimination
Authors: Sadralodabai Toktam , Sorkin Robert D. , Montgomery DeMaris A.
Published: October 2005

Observers were presented with two successive 8-tone sequences; the task was to discriminate between the temporal patterns defined by the inter-tone intervals in each sequence (average inter-tone interval=50 ms, tone duration=25 ms, tone frequency=1000 or 2500 Hz). The observers had to indicate whether the two patterns on each trial were the same or different. A conditional-on-a-single-stimulus (COSS) technique [B. C. Berg, J. Acoust. Soc. Am. 86, 1743–1746 (1991)] was used to evaluate the salience of each temporal position. The probability of responding DIFFERENT is computed given the magnitude of ‖t1,i−t2,i‖, the absolute difference between the inter-tone intervals at each serial position (i=1,2,...,7). The temporal information in the first and last positions had the greatest influence on the observer’s response. These results have implications for models of temporal pattern discrimination. [Work supported by AFOSR.]
Effect of intense tone stimulation on the anatomy of the peripheral auditory system in bony fishes
Authors: Hastings Mardi C.
Published: October 2005

Results of current and previous studies by M. C. Hastings etal. [J. Acoust. Soc. Am. Suppl. 1 79, S80 (1986); J. Acoust. Soc. Am. Suppl. 1 80, S75 (1986); and J. Acoust. Soc. Am. Suppl. 1 81, S7 (1987)] and P. S. Enger [HearingandSoundCommunicationinFishes (Springer-Verlag, New York, 1981), pp. 243–255] to determine the effect of intense tone stimulation on the anatomy of the peripheral auditory system in bony fishes are summarized. In these studies different species of fish were exposed to pure tones at frequencies between 200 and 1000 Hz at different sound pressure levels for 1 to 4 h. The sensory epithelia in the saccule and legena were then examined using scanning electron microscopy to assess damage to hair cells. These experimental evaluations indicated (1) the possibility of a gross place-type mechanism for frequency discrimination and (2) a possible breakdown of the Weberian apparatus in goldfish (Carassiusauratus) at extremely high sound pressure levels because the primary site of hair cell damage switched from the saccule to lagena. A mathematical model for the dynamic motion of the swimbladder and Weberian ossicles has been developed based on the work of R. McN. Alexander [J. Exp. Biol. 38, 747–757 (1961)], which includes viscoelastic properties of the tissue. The model supports previous experimental observations and indicates that motion is no longer transmitted to the saccule when the system is excited by sufficiently high sound pressure levels. [Work supported in part by ONR and NIH.]
Statistical characteristics of bistatic surface scatter
Authors: Neumann Peter D. , Culver R. Lee
Published: October 2005

The statistical characteristics of the envelopes of bistatic surface reverberation data taken in January 1992 are compared against the Rayleigh distribution. Agreement with some slight deviations from Rayleigh behavior was found in the statistical characteristics of the data for all available wind speeds (1.4 to 7.2 m/s) at frequencies from 20 to 40 kHz. Additionally, the data were divided in time according to the dominant surface scattering mechanism, surface roughness or bubble layer, and the statistical characteristics of each of these portions examined for Rayleigh behavior. This type of analysis allows for an increased amount of data to be used as a result of using data not only from all of the transmitted pulses but also from a number of different times in the time series instead of at a single time as was done in the first part of the analysis. The statistical characteristics of these portions of the time series were also found to be Rayleigh with only slight deviations in the tails of the distribution. [Work supported by ONT.]
Low-frequency sound source for various acoustical oceanography and seafloor exploration applications
Authors: Donskoy Dimitri M.
Published: October 2005

A new design of a low-frequency sound source has been developed and tested. The oscillation of a rigid body and an arrangement for converting a dipole oscillation of the rigid body to a monopole pulsation are used in this radiator. The oscillation of the rigid body provides large displacement of a radiating surface without deformation of the material. Monopole pulsation of the entire source provides much higher acoustic radiation than dipole oscillation of a small (as compared with sound wavelength) body. The vibromotive force is generated by oscillation of a mass inside the rigid body. To prove the concept, a small version of the source has been built. It has a diameter of 0.32 m, a frequency of 35–150 Hz, and an acoustic power output of 25 W on a 80-Hz frequency. The full scale source will be capable of radiating different kinds of signals (cw, FM, or pulse) in the frequency range of tens to hundreds of Hz with a source level over 200 dB re: 1 μPa and efficiency of approximately 70%. A high source level in the low-frequency range can be achieved due to the lack of limitation in the displacement of the radiating surface. The source has a potential for high reliability and relatively low cost due to very simple construction, the use of highly reliable and inexpensive mass-manufactured units (like electric motors), and very small material deformation.
Dynamic mechanical properties of violin wood and the tone qualities of violins. III: Relation of dynamic mechanical properties to the tone and playing qualities of violins
Authors: Hutchins Carleen M.
Published: October 2005

Six violins constructed from the wood tested by E. R. Fitzgerald and M. A. Hutchins, and the plates tuned by C. M. Hutchins as reported in the previous two papers, have been played by several competent performers whose comments will be discussed in relation to plate tunings and final tone and playing qualities. The six violins will be demonstrated by members of the Ottawa Symphony.
The monopole cello
Authors: Veneklasen Paul S.
Published: October 2005

The conventional cello is one of the orchestral instruments whose sound is usually poorly projected from an orchestra, especially in its lower register. One of several reasons for this, in common with others of the string family, is that the foot of the bridge is connected to both top plate and back by a sound post constraining these points to vibrate in phase. As a result, in the lower frequency range, the instrument radiates as a bi-polar sound source. If the top plate and back can be coupled so as to vibrate in apposing phase, then the instrument would radiate as a mono-pole source with a potential substantial increase in sound output. This paper describes an exploratory attempt to induce such operation.
Passive localization and classification at low signal-to-noise via Bayesian inversion
Authors: Davis Nolan R. , Pitre Richard
Published: October 2005

The Bayesian inversion approach [R. Pitre and N. R. Davis, J. Acoust. Soc. Am. 92, 2418 (1992)] is employed to examine integration time dependence of aposteriori source location distributions and to perform spectral estimation. Model calculations are presented for low signal-to-noise conditions.
Model-based matched-filter processing of Doppler-shifted signals in a time dispersive ocean environment
Authors: Hermand J.-P. , Roderick W. I.
Published: October 2005

Dispersive multipath propagation in an ocean medium distorts wideband linear frequency-modulated (LFM) transmitted signals. As a result, the performance of correlation receivers is degraded if the receiver does not account for the Doppler of the incoming signal or the multipath (energy splitting) in the medium. In this study, results are presented that demonstrate that the performance of a conventional correlation can be improved if the reference (replica) channel is compensated in both Doppler and time dispersion. The model-based matched filter is generated by correlating the received signal with a reference channel that consists of the transmitted signal convolved with the impulse response of the medium and Doppler compensated. The channel impulse responses were predicted with a broadband propagation model using environmental (sound velocity) data. The data were collected during a February 1990 experiment conducted in deep water in an area west of Sardinia. The acoustic data set consisted of linear frequency-modulated signals, with a time-bandwidth product of 4000, transmitted from a moving source and received on a towed array. Comparison with conventional processing shows improvement (about 3 dB) in peak output signal-to-noise ratio for the propagation conditions encountered in the experiment. [Work supported by ONR.]
The abstractness of lexical representations
Authors: Myers James T.
Published: October 2005

This study attempted to address a fundamental question concerning lexical representations, namely, whether they are encoded in an abstract phonemic form, where only information that is lexically distinctive is recorded, or more concretely. To this end a same–difference task was given to native speakers of American English. Subjects found that the allophone of /t/ that occurs before /r/, as in ‘‘truck,’’ more similar to the distinct phoneme found in ‘‘chuck’’ than to the allophone of /t/ found in ‘‘tuck.’’ By contrast, subjects found the allophone of /t/ that occurs after /s/, as in ‘‘stuck,’’ more similar to the allophone of /t/ that occurs in ‘‘tuck’’ than the acoustically similar but distinct phoneme found in ‘‘duck.’’ It is hypothesized that while the second segment of words like ‘‘stuck’’ is truly an allophone of /t/, the first segment of words like ‘‘truck’’ is in fact the same phoneme as in ‘‘chuck.’’ The forms being compared are thus represented phonemically in both cases. [Work supported by NIDCD.]
On the perceptual integrality of duration and amplitude cues to stress
Authors: Turk Alice , Sawusch James R.
Published: October 2005

Traditionally, amplitude, duration and F0 have been proposed as independent cues to stress [D. Fry, J. Acoust. Soc. Am. 27, 765–768 (1955)]. However, there is psychoacoustic evidence that the perception of loudness depends on duration [B. Scharf, ‘‘Loudness,’’ in HandbookofPerception, edited by E. C. Carterette and M. P. Friedman (Academic, New York, 1978), Vol. 4, pp. 187–242]. Beckman proposed the intensity integral as a more appropriate cue to stress than either amplitude or duration alone [M. Beckman, StressandNon-stressAccent (Foris, 1986)]. The present study tested the perceptual integrality of duration and amplitude using Garner’s paradigm [W. R. Garner, TheProcessingofInformationandStructure (Erlbaum, Hillsdale, NJ, 1974)]. Preliminary results suggest that duration and amplitude are integral. Further, the integrality is asymmetric: Variation in duration interfered with responding to amplitude to a greater extent than amplitude variation interfered with responding to duration. These findings suggest that both duration and energy (the integral of amplitude over duration) may serve as auditory cues to stress. [Work supported by NIDCD Grant Nos. R01-DC00291 and T32-DC00036 to SUNY at Buffalo.]
Speech perception as a talker-contingent process
Authors: Nygaard Lynne C. , Sommers Mitchell S. , Pisoni David B.
Published: October 2005

The present study reports an experiment designed to investigate the nature of perceptual adaptation and memory representation for spoken words produced by familiar and unfamiliar talkers. To determine how familiarity with a talker’s voice affects perception of spoken words, two groups of subjects were trained to recognize the names of ten voices (five male; five female) over a 9-day training period. One group of subjects then identified words presented at four signal-to-noise ratios that were produced by the same set of talkers that they had learned to recognize during training. Control subjects identified the same words at the same signal-to-noise ratios but the words were produced by a set of new talkers that these subjects had not heard during training. The results showed that the ability to explicitly identify a talker’s voice improved intelligibility of novel words produced by the same talkers. Subjects who heard familiar voices in the word intelligibility task were better at identifying novel words in noise than control subjects who heard unfamiliar voices. The results suggest that speech perception may be a talker-contingent process whereby familiarity with aspects of the talker’s vocal source facilitates the subsequent phonetic analysis of the acoustic signal. [Work supported by NIH grant to Indiana University.]
Talker differences and perceptual normalization
Authors: Magnuson James S. , Nusbaum Howard C.
Published: October 2005

Recognition performance for speech is generally worse for utterances produced by a mix of several talkers compared to utterances produced by a single talker. This performance impairment can be attributed to those aspects of talker normalization used to determine the vocal characteristics of the talker each time the talker changes. The present study investigated the size and nature of talker differences that may affect normalization. Spoken words were generated by a text-to-speech system for matched pairs of synthetically defined talkers. All but two of these pairs differed only in average fundamental frequency. One remaining pair of talkers differed in perceived gender but both talkers had the same average pitch; the other pair differed in both gender and pitch. Response times in a speeded word recognition task were compared for blocks of stimuli produced by a single talker and blocks of stimuli produced by a mix of one of the pairs of talkers. The results are important for understanding how listeners use pitch differences between talkers during talker normalization.
A threshold estimate in a yes–no task using a maximum likelihood procedure
Authors: Green David M.
Published: October 2005

A maximum-likelihood procedure for estimating threshold in a yes–no task is presented. The absolute thresholds at six audiometric frequencies for a group of normal-hearing listeners were measured using this procedure. The average standard deviation of the threshold estimate is less than 5 dB with only 12 trials per threshold estimate. The total time taken to measure the audiogram was less than 3 min. The estimate of false-alarm rate is poor and strongly biased, but can be considerably improved by a simple modification that increases the testing time about 25%. [Research supported by National Science Foundation and National Institutes of Health.]
The growth of loudness of brief sounds
Authors: Florentine Mary , Buus So/ren , Poulsen Torben
Published: October 2005

The purpose of this paper is to review the literature on temporal integration of loudness and to derive the form of the loudness function for brief stimuli from existing data. The review of 19 studies reveals that the level difference between equally loud sounds of different durations varies with overall level. The variation is orderly across durations. For 1-kHz tones, the level differences between equally loud 5- and 200-ms stimuli appears to vary nonmonotonically with level. The available data indicate that it ranges from about 17 dB at low sensation levels to about 10 dB around 95 dB SL. For complex sounds, the level difference between equally loud 5- and 200-ms stimuli appears to decrease linearly from about 17 dB at low sensation levels to about 10 dB around 80 dB SL. The loudness functions derived for 5-ms stimuli are steeper than those for 200-ms stimuli. For the 1-kHz tones, the exponent of the best-fitting power function increases from 0.30 for 200-ms tones to 0.33 for 5-ms tones. [Work supported by NIH-NIDCD and The Technical University of Denmark.]
Microgravity investigations in the physics of fluids using acoustic techniques
Authors: Trinh Eugene H.
Published: October 2005

The Earth gravitational field strongly affects the motion of extended liquid surfaces, it establishes unavoidable density gradients, and it leads to the segregation of immiscible fluids having different density. Accessing a low gravity environment introduces the opportunity to investigate the effects of low-magnitude capillary and thermocapillary forces in systems such as two- and three-dimensional free-liquid surfaces (drops, bubbles, and floating zones) and stable multi-phase samples. Acoustic methods have already been shown to be effective in allowing the contactless positioning and manipulation of drops and bubbles in low gravity, as the other presentations in this session demonstrate. This successful implementation of acoustics to space-based research is currently represented by the Drop Physics Module flight facility for microgravity experiments. There is, however, additional opportunity for the application of advanced quantitative acoustic measurement techniques for the determination of the properties of fluid phases or for the management of fluids in low gravity. Experiments in microgravity require a variety of novel supporting capabilities, and acoustics could provide some viable initial approaches. On the other hand, the observation of strictly acoustic effects such as streaming, cavitation, and radiation pressure in the reduced gravity environment might also provide new clues useful for the understanding of these phenomena. [Work supported by NASA.]
Materials characterization of surfaces of water drops with surfactants by acoustic means: The need for a microgravity environment
Authors: Apfel Robert E. , Holt R. Glynn , Tian Yuren , Zheng Xiaoyu , Tao Shi
Published: October 2005

The goal of this laboratory’s ground-based and microgravity research is to determine the rheological properties of liquid drops of single or multiple components in the presence or absence of surface active materials by exciting drops into their quadrupole resonance and observing their free decay. The resulting data coupled with appropriate theory should enable one to understand better the physics of the underlying phenomena, providing a better foundation than earlier empirical results could. The space environment makes an idealized geometry available (spherical drops) so that theory and experiment can be properly compared, and allows a ‘‘clean’’ environment, namely, an environment in which no solid surfaces come in contact with the drops during the test period. Moreover, by considering the oscillations of intentionally deformed drops in microgravity, a baseline is established for interpreting surface characterization experiments done on the ground by this and other groups. Results of work at Yale and in the first United States Microgravity Laboratory (USML 1, summer 1992) will be presented. [Work supported by NASA through JPL Contract No. 958722.]
Bubble dynamics in low gravity and related experiments on Earth
Authors: Marston Philip L. , Trinh Eugene H.
Published: October 2005

A dual-frequency acoustic levitator containing water was developed for studying bubble and drop dynamics in low gravity. It was flown on USML-1 where it was used in NASA’s Glovebox facility. High-frequency (21- or 63-kHz) ultrasonic waves were modulated by low frequencies to excite shape oscillations on bubbles and oil drops ultrasonically trapped in the water. Bubble diameters were typically close to 1 cm and the low gravity alters biases present in related experiments in 1 g. Other observations concerned the agglomeration or coalescence of bubbles in response to acoustic Bjerknes forces, the response of bubbles to surfactant injection, oscillations of bubbles coated by oil, and ultrasonic cavitation. Observations suggest that shape oscillations tend to shift the coated bubble away from the oil–water interface of the coating. Supporting experiments in 1-g extending earlier results [Asaki etal., J. Acoust. Soc. Am. 93, 706–713 (1993)] will also be examined. [Work supported by NASA and by ONR.]
Improved acoustical measurements with MLSSA
Authors: Rife Douglas D.
Published: October 2005

Two fundamental properties of MLS techniques, namely, phaserandomization and energyconservation are exploited by MLSSA to achieve improved acoustical measurements. These include (1) the conversion of nonstationary interfering noise as well as weak nonlinear distortion into stationary noise and, as a consequence, (2) the possibility of correcting, as a post-processing operation, systematic errors in measured Schroeder decay plots whether due to interfering noise or transducer nonlinearities, (3) one-shot STI and RASTI measurements that properly account for contemporaneous background noise including nonstationary noise and, (4) the possibility of characterizing transducer nonlinear distortion by means of incoherency measurements that exclude, or at least attenuate, the contribution of the background noise. This last possibility is in contrast to traditional dual-channel coherence measurements that unavoidably lump together the effects of nonlinearity and noise. All of these results are demonstrated theoretically as well as experimentally using actual MLSSA measurements.
Development of a multi-beam array microphone for multi-channel pickup of sound fields and its applications to room acoustics
Authors: Ikezawa Ryou , Nishi Takashi
Published: October 2005

A new multi-beam array microphone (MAM for short) has been developed. The MAM is composed of four concentric rings, each of which has 12 microphones elements, and a common center microphone. The MAM synthesizes the output phase of each microphone by compensating for the path differences. It has 12 main beams that point in 12 different directions at intervals of 30 deg in the horizontal plane. These main beams are sharp over a wide frequency range. By utilizing these characteristics, the MAM can be applied to many fields such as precise acoustic measurement of sound fields, automatic talker locations, and beam steering. When reproducing the original sound field by this system, 12 loudspeakers arranged in a circle around a central point are connected to the corresponding 12 outputs of the MAM. A new acoustical parameter based on spatial correlation of the sound field (correlation between the 12 MAM outputs) is proposed by utilizing the MAM.
Maximum length sequence based testing of hearing aids
Authors: Schneider Todd , Jamieson Donald G.
Published: October 2005

An automated dual-channel maximum-length sequence (MLS) test system for the electroacoustic characterization of hearing aids has been developed. This test system applies a speech-shaped MLS acoustically to a hearing aid and measures the electro-acoustic frequency response. This method provides results that compare favorably to those obtained using the method employed by Kates [J. M. Kates, J. Rehab. Res. Develop. 27, 255–278 (1990)] and the ANSI standard method [ANSI S3.42 (1992)]. Test results show that MLS-based testing is significantly faster than noise-based testing. Two signal-biased MLS-based testing methods have also been developed. These methods apply a bias signal to force the hearing aid into a mode of operation where the frequency response, with adaptive or subtractive bias signal cancellation, is measured using a low-level, speech-shaped MLS. This method has proved valuable for the characterization of automatic signal processing hearing aids. [Work supported by ORTC and OMH.]
An LVQ2-trained connected-phoneme hidden Markov model for automatic phonetic segmentation and labeling
Authors: Kuo Y. Chun , Rutledge Janet C.
Published: October 2005

A connected-phoneme hidden Markov model (HMM) is proposed to perform automatic segmentation and labeling. Individual phonetic models are first created by a left-to-right HMM. The large connected-phoneme HMM is formed by grouping all these phonetic models together. Therefore, each state of this big HMM uniquely represents an English phoneme. The large connected-phoneme HMM is not trained by the Viterbi algorithm since the most probable state sequence dose not necessarily yield the correct segmentation and labeling. Learning vector quantization (LVQ2) is used to train the connected-phoneme HMM such that the phoneme confusions can be reduced. The proposed algorithm has two potential advantages over the existing speech recognition schemes. (1) With the aid of the unique representation of each state of the big HMM, more insight into speech characteristics can be gained, which is essential for the improvement of speech recognizers. Errors caused by insertion, deletion, and substitution can be properly analyzed and adjusted. (2) The computation load for the LVQ2 training is considerably less than the Viterbi training. Therefore, the LVQ2 training is also more suitable for a limited speech data base.
An alternative to the mechanical index as a means of assessing the safety of exposures to diagnostic ultrasound
Authors: Church Charles C.
Published: October 2005

The mechanical index (MI) is a means of providing the users of diagnostic ultrasound equipment with a real-time, on screen display of the potential for a given exposure to produce tissue damage by mechanical means. As originally conceived, MI was based on a calculation of the lowest rarefactional pressure Pr, needed to produce the growth and subsequent collapse of an optimally sized, pre-existing gas bubble such that the peak temperature attained upon collapse was at least 5000 K. While the assumption that stabilized microbubbles exist within the body may be necessary when assessing the safety of medical exposures to ultrasound, that assumption is unproven. If it is assumed that there are no pre-existing, stabilized gas bubbles in tissue, it is possible to calculate a ‘‘maximum threshold’’ rarefactional pressure amplitude Prmt, above which cavitation is highly probable. By combining the theory for spontaneous homogeneous nucleation and the rate of cavitation observed during clinical lithotripsy procedures, an estimate of this (frequency-independent) threshold may be obtained. Assuming that an ‘‘acceptable’’ cavitation rate in tissue is 10−20 events/m3/s and that an additional clinical safety factor of 2 is desirable, it is possible to show that Prmt=3 MPa. For comparison, at a frequency of 5 MHz, if MI=0.4, then Pr=0.9 MPa; if MI=1.0, Pr=2.2 MPa. [Work supported by NIH.]
Minimum audible angle for clicks with simulated echoes: Effects of azimuth and standard
Authors: Litovsky Ruth Y. , Macmillan Neil A.
Published: October 2005

Minimum audible angle (MAA) was measured in free field with a center position at 0°, and 25° or 50° on the right. Stimuli were 6-ms-wide band noise bursts presented as single bursts or noise pairs under precedence-effect conditions. In the single-noise condition, one burst was presented from the right or left. In precedence lead discrimination, a leading sound was presented from the left or right speaker and a lagging sound from a speaker at center. In precedence lag discrimination, the lagging sound was on the right or left and the leading sound was at center. The test stimulus was either presented in isolation or was preceded by a single standard noise at center. MAAs were lowest for the single condition, slightly higher for lead discrimination, and much higher for lag discrimination. Presence of a standard yielded a benefit in lag discrimination, especially at 0°. Thresholds at 50° were significantly higher than at 0° and 25°, especially in lag discrimination. The data can be described by a model in which judgments are based on apparent azimuth of the fused stimulus, and the lagging sound in a pair perceptually ‘‘pulls’’ the image along the azimuth dimension. [Work supported by NSF and NIH.]
Active control of finite beam volume velocity using shaped PVDF sensors
Authors: Guigou Catherine , Berry Alain
Published: October 2005

Active control of volume velocity using shaped PVDF sensors is analytically studied for finite beam with different boundary conditions. Controlling the volume velocity of a finite beam is equivalent of controlling the far-field pressure above the center of the baffled beam. The shape of the PVDF sensor is determined in such a way that the output signal of the sensor is directly proportional to the volume velocity. A general expression of the PVDF sensor shape is obtained and is shown to depend only on the beam characteristics (material properties and boundary conditions). However, the shape of the sensor is independent of the type and frequency of excitation. This type of shaped PVDF sensors detects only contributions from the odd modes, which are the only ones without a nonzero volume velocity and also the most effective radiators. When a simply supported beam is subjected to a harmonic input force, the output signal of the shaped PVDF sensor is minimized by a control point force. The far-field radiated power, the mean-square velocity, and the radiation efficiency before and after the control are extensively studied. The results demonstrate that such shaped PVDF sensors are effective to control the sound radiation from the most efficient radiating flexural modes.
Active control of nonlinear vibration of a panel structure
Authors: Maestrello Lucio
Published: October 2005

Experimental results show that one can convert the nonlinear periodic motion of a panel structure to a linear one by controlling the system about the fundamental orbit through a time-dependent, phase-shift forced perturbation. An aircraft fuselage panel structure is forced at resonance by plane acoustic waves. Once the nonlinear response is triggered, the oscillation evolves through a sequence of subharmonic and harmonic bifurcations. The radiated acoustic pressure also contains harmonics whose level increases with distance relative to the fundamental. Control is achieved by actively forcing the panel either with a piezoceramic element or with an electrical shaker. Results show that the energy is transferred from the fundamental to the harmonics and vice versa as the response changes from linear to nonlinear and back to the linear state. An analytical model using a control linearization technique of Duffing’s equation with periodic forcing is discussed.