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Journal of the Acoustical Society of America

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Dec 1992

Volume 92, Issue 6, pp. 3039-3461

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Anisotropy of ultrasonic velocity and elastic properties in normal human myocardium

Edward D. Verdonk, Samuel A. Wickline, and James G. Miller

J. Acoust. Soc. Am. Volume 92, Issue 6, pp. 3039-3050 (1992); (12 pages) | Cited 6 times

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Measurements of ultrasonic quasilongitudinal velocity were made in the muscle fiber plane of excised human myocardium. Multiple adjacent planes across the left ventricular wall were interrogated to assess the transmural dependence of velocity. For each measurement plane, data were obtained in 2‐deg increments through the full 360 deg relative to the myofibers. An approximate 1.3% magnitude of anisotropy was observed with maximum velocity along the muscle fibers and minimum velocity perpendicular to the muscle fibers. The known transmural shift in myofiber orientation was evidenced in the anisotropy of velocity as angular shifts between plots obtained from adjacent transmural planes within the same specimen. Measured values of velocity and density were used to estimate the effective C33 and C11 elastic constants of a thin layer of normal myocardium.
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43.80.Cs Acoustical characteristics of biological media: molecular species, cellular level tissues
43.80.Ev Acoustical measurement methods in biological systems and media

20‐Hz pulses and other vocalizations of fin whales, Balaenoptera physalus, in the Gulf of California, Mexico

Paul O. Thompson, Lloyd T. Findley, and Omar Vidal

J. Acoust. Soc. Am. Volume 92, Issue 6, pp. 3051-3057 (1992); (7 pages) | Cited 17 times

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Low‐frequency vocalizations were recorded from fin whales, Balaenoptera physalus, in the Gulf of California, Mexico, during three cruises. In March 1985, recorded 20‐Hz pulses were in sequences of regular 9‐s interpulse intervals. In August 1987, nearly all were in sequences of doublets with alternating 5‐ and 18‐s interpulse intervals. No 20‐Hz pulse sequences of any kind were detected in February 1987. The typical pulse modulated from 42 to 20 Hz and its median duration was 0.7 s (1985 data). Most other fin whale sounds were also short tonal pulses averaging 82, 56, and 68 Hz, respectively, for the three cruises; 89% were modulated in frequency, mostly downward. Compared to Atlantic and Pacific Ocean regions, Gulf of California 20‐Hz pulses were unique in terms of frequency modulation, interpulse sound levels, and temporal patterns. Fin whales in the Gulf may represent a regional stock revealed by their sound characteristics, a phenomenon previously shown for humpback whales, birds, and fish. Regional differences in fin whale sounds were found in comparisons of Atlantic and Pacific locations.
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43.80.Jz Use of acoustic energy (with or without other forms) in studies of structure and function of biological systems
43.80.Ka Sound production by animals: mechanisms, characteristics, populations, biosonar

Structural design of hidden Markov model speech recognizer using multivalued phonetic features: Comparison with segmental speech units

L. Deng and K. Erler

J. Acoust. Soc. Am. Volume 92, Issue 6, pp. 3058-3067 (1992); (10 pages)

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A novel approach to speech recognition, on the basis of a multidimensional, multivalued phonetic‐feature description of speech signals, is presented and evaluated. The hidden Markov model (HMM) framework is used to provide the recognition algorithm, which assumes that the underlying Markov chain tracks the temporal evolution of the features. It is shown that this approach can naturally accommodate such coarticulatory effects as feature spreading and formant transition in the functionality of the recognizer, and can provide a high degree of acoustic data sharing that makes effective use of training data. Use of phonetic features as the basic speech units creates a framework where the Markov model’s state topology in the recognizer can be designed with guidance of detailed speech knowledge. Details of such a design for a stop consonant–vowel vocabulary are described. Experimental results on the task of speaker‐dependent stop consonant discrimination, evaluated from speech data from a total of ten male and five female speakers, demonstrate effectiveness of this feature‐based recognizer. Over the 15 speakers, the error rates were shown to be reduced by 23%, 37%, 42%, and 38%, respectively, compared with the conventional HMM‐based recognition methods using words, phonemes, allophones, and microsegments as the primary speech units.
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43.72.Ar Speech analysis and analysis techniques; parametric representation of speech
43.72.Ne Automatic speech recognition systems
43.70.Fq Acoustical correlates of phonetic segments and suprasegmental properties: stress, timing, and intonation

Consonant recognition by some of the better cochlear‐implant patients

Richard S. Tyler and Brian C. J. Moore

J. Acoust. Soc. Am. Volume 92, Issue 6, pp. 3068-3077 (1992); (10 pages) | Cited 5 times

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Fifty‐four of the better cochlear‐implant patients from Europe and the United States were tested on two consonant recognition tests using nonsense syllables. One was produced in an accent appropriate for their own language by a male and a female talker. Recorded tokens of /ibi, idi, igi, ipi, iti, iki, ifi, ivi, i1i, isi, izi, imi, ini/ were presented. With the French syllables, six patients with the Chorimac device averaged 18% correct (6%–29%). With the German syllables, nine patients with the 3M/Vienna device averaged 34% correct (17%–44%), ten patients with the Nucleus device (tested in Hannover) averaged 31% correct (19%–42%), and ten patients with the Duren/Cologne device averaged 27% correct (10%–56%). With the English syllables, ten patients with the Nucleus device (tested in the United States) averaged 42% correct (29%–62%), and nine patients with the Symbion device averaged 46% correct (31%–69%). An information‐transmission analysis and sequential information‐transfer analysis of the confusions suggested that different implants provided differing amounts of feature information. The place of articulation feature was typically the most difficult to code for all implants. In the second test a male and a female talker recorded the stimuli /ibi, idi, igi, imi, ini, i1i, isi, izi/ in a single manner that was appropriate for all three languages. Six patients with the Chorimac device averaged 27% (13%–48%), ten patients with the Duren/Cologne implant averaged 29% (15%–75%), ten patients with the Nucleus device (tested in Hannover) averaged 40% (25%–58%), ten patients with the Nucleus device (tested in the United States) averaged 49% (40%–60%), nine patients with the Symbion device averaged 61% (40%–75%), and nine patients with the 3M/Vienna device averaged 41% (29%–52%) correct.
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43.71.Ky Speech perception by the hearing impaired
43.71.Hw Cross-language perception of speech
43.66.Ts Auditory prostheses, hearing aids
43.64.Me Effects of electrical stimulation, cochlear implant

Electromagnetic midsagittal articulometer systems for transducing speech articulatory movements

Joseph S. Perkell, Marc H. Cohen, Mario A. Svirsky, Melanie L. Matthies, Iñaki Garabieta, and Michel T. T. Jackson

J. Acoust. Soc. Am. Volume 92, Issue 6, pp. 3078-3096 (1992); (19 pages) | Cited 41 times

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This paper describes two electromagnetic midsagittal articulometer (EMMA) systems that were developed for transducing articulatory movements during speech production. Alternating magnetic fields are generated by transmitter coils that are mounted in an assembly that fits on the head of a speaker. The fields induce alternating voltages in a number of small transducer coils that are attached to atriculators in the midline plane, inside and outside the vocal tract. The transducers are connected by fine lead wires to receiver electronics whose output voltages are processed to yield measures of transducer locations as a function of time. Measurement error can arise with this method, because as the articulators move and change shape, the transducers can undergo a varying amount of rotational misalignment with respect to the transmitter axes; both systems are designed to correct for transducer misalignment. For this purpose, one system uses two transmitters and biaxial transducers; the other uses three transmitters and single‐axis transducers. The systems have been compared with one another in terms of their performance, human subjects compatibility, and ease of use. Both systems can produce useful midsagittal‐plane data on articulator movement, and each one has a specific set of advantages and limitations. (Two commercially available systems are also described briefly for comparison purposes.) If appropriate experimental controls are used, the three‐transmitter system is preferable for practical reasons.
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43.70.Jt Instrumentation and methodology for speech production research
43.70.Aj Anatomy and physiology of the vocal tract, speech aerodynamics, auditory kinetics

Intensity discrimination under backward masking

Christopher J. Plack and Neal F. Viemeister

J. Acoust. Soc. Am. Volume 92, Issue 6, pp. 3097-3101 (1992); (5 pages) | Cited 5 times

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The Weber fraction was measured for a 25‐ms sinusoidal pedestal presented 100 ms before, or 100 ms after, an intense narrow‐band noise. Consistent with the finding of Zeng et al. [Hear. Res. 55, 223–230 (1991)], the forward masker caused an elevation in the Weber fraction at medium pedestal levels. Surprisingly, however, a much larger midlevel elevation was observed in the backward masking conditions; in some cases, the Weber fraction was increased by over 20 dB by the backward masker. In both masking conditions, presenting a notched noise simultaneously with the pedestal reduced the magnitude of the midlevel elevation. These results indicate that it is possible to produce large masking effects on intensity discrimination in conditions where there is no possibility of the masker affecting the representation of the pedestal at the level of the auditory nerve. This suggests that there may be ‘‘central’’ processes underlying the original finding of Zeng et al. Despite the similarities in the results, however, it is not certain that the elevations seen in the forward and backward masking conditions were caused by the same mechanisms.
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43.66.Ba Models and theories of auditory processes
43.66.Dc Masking
43.66.Fe Discrimination: intensity and frequency

Frequency discrimination in forward and backward masking

Christopher W. Turner, Fang‐Gang Zeng, Evan M. Relkin, and Amy R. Horwitz

J. Acoust. Soc. Am. Volume 92, Issue 6, pp. 3102-3108 (1992); (7 pages) | Cited 2 times

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Frequency difference limens for pure tones preceded by a forward masker or followed by a backward masker were obtained across a wide range of signal levels. Relkin and Doucet [Hear. Res. 55, 215–222 (1991)] have shown that at a masker‐signal delay of 100 ms, the thresholds of high‐SR (spontaneous rate) auditory‐nerve fibers are recovered, while the low‐SR fiber thresholds are not. Therefore, forward‐masked frequency discrimination potentially offers a method to investigate the role of low‐SR fibers in the coding of frequency. It has been shown that when an intense forward masker is presented 100 ms before a pure‐tone signal, intensity difference limens are elevated for mid‐level signals [Zeng et al., Hear. Res. 55, 223–230 (1991)]. However, Plack and Viemeister [J. Acoust. Soc. Am. 92, 3097–3101 (1992)] have shown that a similar elevation in the intensity difference limen is obtained under conditions of backward masking, where selective adaptation of the auditory neurons would not be expected to occur. A condition of backward‐masked frequency discrimination was therefore included to investigate the role of interference resulting from adding additional stimuli to a discrimination task. For signals at 1000 and 6000 Hz, there was no effect of a forward masker upon frequency difference limens. For the backward‐masked conditions, an elevation of the frequency difference limen was observed at all signal levels, demonstrating that the effects of forward and backward maskers upon frequency discrimination are dissimilar and suggesting that cognitive effects are present in backward‐masked discrimination tasks. This difference between intensity and frequency discrimination supports the idea that intensity and frequency are encoded in different manners by the auditory system. The absence of an effect of forward masker on frequency discrimination suggests that the rate response of low‐SR fibers does not contribute useful information to a frequency discrimination task.
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43.66.Ba Models and theories of auditory processes
43.66.Dc Masking
43.66.Fe Discrimination: intensity and frequency
43.66.Hg Pitch

The ‘‘proportion‐of‐the‐total‐duration rule’’ for the discrimination of auditory patterns

Gary R. Kidd and Charles S. Watson

J. Acoust. Soc. Am. Volume 92, Issue 6, pp. 3109-3118 (1992); (10 pages) | Cited 5 times

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A principle of auditory perception that governs the detectability of changes in components in unfamiliar sequences of tones is demonstrated in four experiments. The proportion‐of‐the‐total‐duration (PTD) rule can be stated as follows: Each individual component of an unfamiliar sequence of tones is resolved with an accuracy that is a function of its proportion of the total duration of the sequence or ‘‘pattern.’’ An adaptive‐tracking frequency‐discrimination task was used in all experiments. Experiment 1 demonstrated that the PTD rule holds over a wide range of total pattern durations, numbers of components, and component durations. Experiment 2 demonstrated that the PTD rule governs discrimination performance despite variation in the relative durations of context and target tones. Experiment 3, using a variable temporal position for the target, confirmed that the PTD effect does not require that a listener be able to anticipate the temporal location of the target tone. Experiment 4, using two target tones, showed that the PTD rule applies to the proportional duration of individual components within patterns and not to the total proportional duration of nonadjacent components within the pattern. These findings are incompatible with performance limitations based on a fixed‐duration short‐term memory capacity and with versions of informational limitations in which the amount of information in a pattern varies either with the number of components or with the total pattern duration. The PTD rule appears to reflect the way listeners distribute their attention when presented with unfamiliar complex sounds that have no structural properties (other than proportional duration) that significantly increase the salience of individual components.
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43.66.Fe Discrimination: intensity and frequency
43.66.Ba Models and theories of auditory processes
43.66.Mk Temporal and sequential aspects of hearing; auditory grouping in relation to music

Detection of combined frequency and amplitude modulation

Brian C. J. Moore and Aleksander Sek

J. Acoust. Soc. Am. Volume 92, Issue 6, pp. 3119-3131 (1992); (13 pages) | Cited 21 times

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This article is concerned with the detection of mixed modulation (MM), i.e., simultaneously occurring amplitude modulation (AM) and frequency modulation (FM). In experiment 1, an adaptive two‐alternative forced‐choice task was used to determine thresholds for detecting AM alone. Then, thresholds for detecting FM were determined for stimuli which had a fixed amount of AM in the signal interval only. The amount of AM was always less than the threshold for detecting AM alone. The FM thresholds depended significantly on the magnitude of the coexisting AM. For low modulation rates (4, 16, and 64 Hz), the FM thresholds did not depend significantly on the relative phase of modulation for the FM and AM. For a high modulation rate (256 Hz) strong effects of modulator phase were observed. These phase effects are as predicted by the model proposed by Hartmann and Hnath [Acustica 50, 297–312 (1982)], which assumes that detection of modulation at modulation frequencies higher than the critical modulation frequency is based on detection of the lower sideband in the modulated signal’s spectrum. In the second experiment, psychometric functions were measured for the detection of AM alone and FM alone, using modulation rates of 4 and 16 Hz. Results showed that, for each type of modulation, d′ is approximately a linear function of the square of the modulation index. Application of this finding to the results of experiment 1 suggested that, at low modulation rates, FM and AM are not detected by completely independent mechanisms. In the third experiment, psychometric functions were again measured for the detection of AM alone and FM alone, using a 10‐Hz modulation rate. Detectability was then measured for combined AM and FM, with modulation depths selected so that each type of modulation would be equally detectable if presented alone. Significant effects of relative modulator phase were found when detectability was relatively high. These effects were not correctly predicted by either a single‐band excitation‐pattern model or a multiple‐band excitation‐pattern model. However, the detectability of the combined AM and FM was better than would be predicted if the two types of modulation were coded completely independently.
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43.66.Fe Discrimination: intensity and frequency
43.66.Ba Models and theories of auditory processes
43.66.Lj Perceptual effects of sound

Effect of multiple speechlike maskers on binaural speech recognition in normal and impaired hearing

A. W. Bronkhorst and R. Plomp

J. Acoust. Soc. Am. Volume 92, Issue 6, pp. 3132-3139 (1992); (8 pages) | Cited 52 times

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Speech‐reception thresholds (SRT) were measured for 17 normal‐hearing and 17 hearing‐impaired listeners in conditions simulating free‐field situations with between one and six interfering talkers. The stimuli, speech and noise with identical long‐term average spectra, were recorded with a KEMAR manikin in an anechoic room and presented to the subjects through headphones. The noise was modulated using the envelope fluctuations of the speech. Several conditions were simulated with the speaker always in front of the listener and the maskers either also in front, or positioned in a symmetrical or asymmetrical configuration around the listener. Results show that the hearing impaired have significantly poorer performance than the normal hearing in all conditions. The mean SRT differences between the groups range from 4.2–10 dB. It appears that the modulations in the masker act as an important cue for the normal‐hearing listeners, who experience up to 5‐dB release from masking, while being hardly beneficial for the hearing impaired listeners. The gain occurring when maskers are moved from the frontal position to positions around the listener varies from 1.5 to 8 dB for the normal hearing, and from 1 to 6.5 dB for the hearing impaired. It depends strongly on the number of maskers and their positions, but less on hearing impairment. The difference between the SRTs for binaural and best‐ear listening (the ‘‘cocktail party effect’’) is approximately 3 dB in all conditions for both the normal‐hearing and the hearing‐impaired listeners.
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43.66.Pn Binaural hearing
43.71.Gv Measures of speech perception (intelligibility and quality)
43.71.Ky Speech perception by the hearing impaired

Neural network models of sound localization based on directional filtering by the pinna

Chalapathy Neti, Eric D. Young, and Michael H. Schneider

J. Acoust. Soc. Am. Volume 92, Issue 6, pp. 3140-3156 (1992); (17 pages) | Cited 6 times

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Three‐layer neural‐network functions were developed to transform spectral representations of pinna‐filtered stimuli at the input to a space‐mapped representation of sound‐source direction at the output. The inputs are modeled after transfer functions of the external ear of the cat; the output is modeled on the spatial sensitivity of superior colliculus neurons. Network solutions are obtained by backpropagation and by a method that enforces uniform task distribution in the hidden layer of the model. Solutions are characterized using bandlimited inputs to study the relative strength of potential sound localization cues in various frequency regions. This analysis suggests that the frequency region containing the first spectral notch (5–18 kHz) provides the best localization cues. Response properties of model neurons were studied using input patterns modeled after auditory nerve response profiles to pure tones at various frequencies and sound levels. The response properties of hidden layer model neurons resemble cochlear nucleus types III and IV and their composites. Neurons in both hidden and output layers show the properties of spectral notch detectors. Although neural networks have limitations as models of real neural systems, the results illustrate how they can provide insight into the computation of complex transformations in the nervous system.
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43.64.Bt Models and theories of the auditory system
43.64.Ha Acoustical properties of the outer ear; middle-ear mechanics and reflex
43.64.Tk Physiology of sound generation and detection by animals
43.66.Qp Localization of sound sources

Analysis of dynamic behavior of human middle ear using a finite‐element method

Hiroshi Wada, Tetsuro Metoki, and Toshimitsu Kobayashi

J. Acoust. Soc. Am. Volume 92, Issue 6, pp. 3157-3168 (1992); (12 pages) | Cited 11 times

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Applying the general‐purpose finite‐element package program (ISAP), a three‐dimensional finite‐element method (FEM) model of a human right middle ear, which included ossicles, was made and the mechanical properties and boundary conditions of the middle ear were determined by a comparison between the numerical results obtained from the FEM analysis and the measurement results of the fresh cadavers, normal subjects and patients, which were obtained by our developed sweep frequency middle ear analyzer (MEA). The ‘‘Elastic’’ boundary condition consisting of linear and torsional springs at the eardrum attachments to the annular ligament was more appropriate for the actual condition than ‘‘fully clamped’’ one. Rotational axis of the ossicular chain was assumed to be a fixed straight line from the anterior process of the malleus to the short process of the incus, and a load of the ossicular chain and cochlea was simplified to be expressed by the stiffness of the cochlea. Vibration patterns of the eardrum and ossicles at the first resonance frequency, obtained under these assumptions, were in agreement with the experimental results obtained by means of time‐averaged holography and by using a video measuring system, except for the relatively large displacements at the tympanic ring.
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43.64.Ha Acoustical properties of the outer ear; middle-ear mechanics and reflex
43.64.Bt Models and theories of the auditory system
43.64.Yp Instruments and methods

Elasticity and active force generation of cochlear outer hair cells

Kuni H. Iwasa and R. S. Chadwick

J. Acoust. Soc. Am. Volume 92, Issue 6, pp. 3169-3173 (1992); (5 pages) | Cited 8 times

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The cochlear outer hair cell is described by a cylindrical membrane model, characterized by area and shear moduli for a passive elastic element and an active tension element dependent on the membrane potential. In passive experiments, these moduli are determined from the pressure‐strain relations. The area modulus obtained is 0.07 N m−1, similar to a lipid bilayer and the shear modulus is 0.007 N m−1. These moduli combined with previous active experiments show that the active tension is nearly isotropic and is about 1.6×10−2 N m−1 V−1, resulting in a 0.5 nN/mV force per cell. This implies that the receptor potential for acoustical stimulation produces an active force comparable to the acoustic force applied to the basilar membrane per outer hair cell. This finding supports the hypothesis that the outer hair cell acts as feedback motor in the fine tuning mechanism of the mammalian ear.
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43.64.Ld Physiology of hair cells
43.64.Kc Cochlear mechanics
43.64.Bt Models and theories of the auditory system

Place specific influences on the wave I to V interpeak latency of the auditory brain‐stem response

David A. Zapala, Herbert J. Gould, and Maurice I. Mendel

J. Acoust. Soc. Am. Volume 92, Issue 6, pp. 3174-3184 (1992); (11 pages)

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There is controversy over whether the wave I to V interpeak latency (I–V IPL) of the auditory brain‐stem response can be manipulated by cochlear processing. In this study, a forward masking paradigm was used to test the predictions of two contrasting models of I–V IPL generation. The paradigm was designed to determine if the I–V IPL can be affected by masking selected portions of the cochlear response region. The results from ten normal hearing subjects suggest that: (1) waves I and V can be masked semi‐independent of each other, and (2) the I–V IPL can be shortened or prolonged by masking the basal or apical portion of the cochlear response region respectively. These findings support the hypothesis that, at least in normal hearing subjects, wave V is biased to reflect more apical cochlear events than wave I. Additionally, they offer tentative support for anecdotal reports of shortened I–V IPLs in the presence of high‐frequency hearing loss.
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43.64.Ri Evoked responses to sounds
43.64.Qh Electrophysiology of the auditory central nervous system

Ocean acoustic signal processing: A model‐based approach

J. V. Candy and E. J. Sullivan

J. Acoust. Soc. Am. Volume 92, Issue 6, pp. 3185-3201 (1992); (17 pages) | Cited 2 times

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A model‐based approach is proposed to solve the ocean acoustic signal processing problem that is based on a state‐space representation of the normal‐mode propagation model. It is shown that this representation can be utilized to spatially propagate both modal (depth) and range functions given the basic parameters (wave numbers, etc.) developed from the solution of the associated boundary value problem. This model is then generalized to the stochastic case where an approximate Gauss–Markov model evolves. The Gauss–Markov representation, in principle, allows the inclusion of stochastic phenomena such as noise and modeling errors in a consistent manner. Based on this framework, investigations are made of model‐based solutions to the signal enhancement, detection and related parameter estimation problems. In particular, a modal/pressure field processor is designed that allows in situ recursive estimation of the sound velocity profile. Finally, it is shown that the associated residual or so‐called innovation sequence that ensues from the recursive nature of this formulation can be employed to monitor the model’s fit to the data and also form the basis of a sequential detector.
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43.60.Gk Space-time signal processing, other than matched field processing
43.30.Zk Experimental modeling
43.60.Pt Signal processing techniques for acoustic inverse problems

Annular array imaging with full‐aperture resolution

Stephen J. Norton

J. Acoust. Soc. Am. Volume 92, Issue 6, pp. 3202-3206 (1992); (5 pages) | Cited 1 time

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The problem of imaging with an annular array of transducers by employing all pairs of transducer elements around the circumference of the annulus as transmitters and receivers is considered. If θt and θr are, respectively, the angular locations of a pair of transmitting and receiving elements, then weighting the received signal with the positive weight 2‖sin(θt−θr)‖ before coherent summation results in an image point spread function of the form J1(R)/R. This corresponds to the point spread function of a full circular (area) aperture. Moreover, it is shown that the diameter of this synthetic aperture is twice that of the annulus. A more general weighting function is also derived that results in a point spread function of the form Jn(R)/Rn, n=1,2,..., which is shown to correspond to an apodized circular aperture of diameter twice that of the annulus.
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43.60.Gk Space-time signal processing, other than matched field processing
43.38.Hz Transducer arrays, acoustic interaction effects in arrays

Variable domain transform and the detection of frequency modulation pulses

Nico Roosnek

J. Acoust. Soc. Am. Volume 92, Issue 6, pp. 3207-3212 (1992); (6 pages)

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In this article a method is described for the detection of linear frequency modulation (FM) pulses with unknown frequency rate parameters in a noisy environment. The method, which is equivalent to the wavelet transform, enhances the signal towards noise. The technique is based on the variation of the domain after which a Fourier transform is applied, or for short called ‘‘variable domain transform.’’ The enhancement is in the order of pulse compression. The computational speed is of the order of that of the fast Fourier transform. The method is also applicable for other signal forms. A criterion is given for the type of signals that can be processed in that manner. An actual signal is analyzed and discussed.
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43.60.Gk Space-time signal processing, other than matched field processing

Coherent source location using a pattern diversity technique

Y. Sun, S. Roy, S. A. Kassam, and F. Haber

J. Acoust. Soc. Am. Volume 92, Issue 6, pp. 3213-3220 (1992); (8 pages)

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A high‐resolution pattern diversity method is presented for resolving multiple coherent plane waves impinging on a linear array. In this method, use is made of several different choices for the common directional response or array pattern to form an averaged covariance matrix. A full rank (equal to the number of plane wave arrivals) signal‐only covariance matrix can be achieved in the presence of multiple coherent sources, irrespective of the coherence structure. The underlying method does not require a uniform array, and can also be implemented using subarrays to effect the pattern diversity. The theoretical development of the method along with simulation results are provided.
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43.60.Gk Space-time signal processing, other than matched field processing

Excitation of thin beams using asymmetric piezoelectric actuators

Gary P. Gibbs and Chris R. Fuller

J. Acoust. Soc. Am. Volume 92, Issue 6, pp. 3221-3227 (1992); (7 pages) | Cited 6 times

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In this paper, an approximate analytical model is developed for the excitation of a thin beam by a single piezoelectric actuator bonded to the surface of the beam. The premise of this work is to investigate the excitation of beams by piezoelectric actuators on a more fundamental level than present work, and then use the asymmetric model to predict a wave response, rather than a modal response, on more complicated structure/actuator systems. It is determined that the single surface mounted piezoelectric actuator simultaneously excites both flexural and extensional motion in beams whose relative amplitudes are functions of beam/actuator geometry and properties. The model is then applied to the excitation of an infinite beam by two colocated arbitrarily driven actuators. It is shown that this configuration can produce any desired combination of flexural and extensional waves in beams by varying the degree of asymmetry between the actuators.
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43.40.Cw Vibrations of strings, rods, and beams
43.38.Fx Piezoelectric and ferroelectric transducers

Modal‐slowness analysis of plate vibrations

J. Robert Fricke and Arthur B. Baggeroer

J. Acoust. Soc. Am. Volume 92, Issue 6, pp. 3228-3238 (1992); (11 pages) | Cited 2 times

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A methodology for analyzing complicated plate vibrations is presented. It is based on the Radon transform and unravels the modal structure excited by transient broadband pulses. The difference in phase velocity between flexural and longitudinal plate waves is exploited to this end. Array data collected in the x–t domain is mapped into the slowness‐tau p–τ domain. In this domain, the flexural and longitudinal mode are isolated from one another and may be analyzed individually. Several analysis tools are developed and demonstrated for two plate models: a flat, finite‐width steel plate and the same plate with a small steel block in welded contact. Examples of the analysis results are estimates for modal waveform and dispersion, modal energy content, and modal transfer functions relating incident and scattered waves at an impedance discontinuity in the plate.
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43.40.Dx Vibrations of membranes and plates

Use of Lanczos vectors in fluid/structure interaction problems

R. Jeans and I. C. Mathews

J. Acoust. Soc. Am. Volume 92, Issue 6, pp. 3239-3248 (1992); (10 pages)

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The goals of any numerical computational technique used for the solution of structural acoustics problems in the exterior infinite domain should be ones of accuracy with rapid convergence, robustness, and computational efficiency. A computer program has recently been developed to achieve each of these three goals. Accuracy and robustness in the numerical representation of the integral equations used to represent the infinite fluid was attained through the use of boundary element implementations of the surface Helmholtz integral equations. The computational efficiency was resolved through the use of Lanczos vectors to model the deformation characteristics of the structure. The difficulty hindering the use of the boundary element formulation has been in finding a suitable numerical representation of the hypersingular integral operator contained in the differentiated form of the Helmholtz integral equations. This difficulty has hampered the convergence, accuracy, and robustness of many previous numerical integral equation implementations used for the modeling of the infinite fluid field. The authors have recently developed both collocation and variational techniques to overcome the difficulties previously encountered in the numerical implementation of the hypersingular integral operator.
The Cauchy singularity present in the integral formulation is made numerically amenable through the use of tangential derivatives in both the collocation and variational techniques. The variational approach has the advantage that the resulting added fluid mass term is symmetric and combines efficiently with a finite element approximation of the structural elastic response. Several different strategies making use of the Lanczos vectors have been investigated. The first involved the use of Lanczos vectors solely to characterize the structural response. This reduced form of the structural dynamical matrix was then substituted back into a Burton and Miller formulation of the acoustic problem, from which the normal surface pressure was obtained. The second strategy investigated involved forming the complex Lanzcos vectors of the dynamical matrix formed from the addition of a symmetrical added fluid matrix to the structural mass matrix. The size of resultant matrix equation set solved at each frequency for this strategy is determined by the number of Lanczos vectors used.
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43.40.Yq Instrumentation and techniques for tests and measurement relating to shock and vibration, including vibration pickups, indicators, and generators, mechanical impedance
43.20.Tb Interaction of vibrating structures with surrounding medium

The thermoelastic surface strip source for laser‐generated ultrasound

A. Aharoni, K. M. Jassby, and M. Tur

J. Acoust. Soc. Am. Volume 92, Issue 6, pp. 3249-3258 (1992); (10 pages)

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The vertical ultrasonic surface displacement generated by a laser‐induced, thermoelelastic, rectangular surface strip source is formulated. The main features of this source, ultrasonic rise times comparable to those of the generating light pulse derivatives, and a large amplitude, double‐pulsed waveform, are advantageous for ultrasonic spectral‐ and time‐domain measurements for nondestructive material evaluation. The analysis shows that, for laterally symmetric sources in the thermoelastic regime, only one tangential thermoelastic stress component contributes to the vertical displacement. Therefore, the strip source is equivalent to two, tangential line forces acting outward at the strip’s front and back edges. The leading‐edge rise time of the signal is virtually independent of the lateral extent of the source, which mostly affects the trailing portions of the ultrasonic pulses. Consequently, a particularly simple expression, which compares favorably with experimental results, is obtained for short strips (subtending small angles at the observation point). In conjunction with this formulation, the thermoelastic strip source is an important tool for quantitative, laser‐based, ultrasonic nondestructive material evaluation.
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43.35.Zc Use of ultrasonics in nondestructive testing, industrial processes, and industrial products
43.35.Ud Thermoacoustics, high temperature acoustics, photoacoustic effect

Evaluation of interference effect in the energy investigation of echoes scattered by an uncorrelated planar distribution of spherical targets

Zhigang Sun and Gérard Gimenez

J. Acoust. Soc. Am. Volume 92, Issue 6, pp. 3259-3270 (1992); (12 pages) | Cited 2 times

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In this paper, an evaluation function is proposed to determine the interference effect in the energy investigation of echoes scattered by a random distribution of spherical targets. The application of this function to an uncorrelated planar distribution of spheres leads to an analytical formula that has the advantage of being independent of either exciting signal of the sonar system or properties of particular spheres and therefore can be used to any kind of spherical isotropic targets. In the estimation of underwater target abundance by the echo squared integration method, this formula provides a fast way to evaluate the interference contribution of echoes.
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43.30.Ft Volume scattering
43.20.Fn Scattering of acoustic waves
43.30.Gv Backscattering, echoes, and reverberation in water due to combinations of boundaries

The short pulse method of isolation and identification of resonances: Comparison with a quasiharmonic method and application to axisymmetrical scatterers

P. Rembert, A. Cand, P. Pareige, M. Talmant, G. Quentin, and J. Ripoche

J. Acoust. Soc. Am. Volume 92, Issue 6, pp. 3271-3277 (1992); (7 pages) | Cited 3 times

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Several experimental methods using quasiharmonic insonification lead to a direct verification of the resonance scattering theory. They provide, for elastic cylindrical targets, the resonance frequencies and the mode number of each of them. Pulsed techniques already existing only allowed the resonance frequency determination. Recently, a short pulse method of isolation and identification of resonances (short pulse MIIR) has been initiated. It completes previous works and allows a complete comparison with theoretical and quasiharmonic experimental results. This method consists in digitizing the time signals that characterize the scattering from an elastic target insonified by a short pulse for different angular positions of the receiving transducer. For each acquired signal a resonance spectrum is obtained after a spectral amplitude analysis. From these data, computer processing allows the plotting of identification patterns and by the same way the knowledge of the mode of vibration at a given resonance frequency. Even if this method is not restricted to cylindrical scatterers, this paper presents the identification of resonances by means of the short pulse MIIR of targets as cylinders, shells and water‐filled shells and cavities. The accuracy and the possibility for separating overlapping resonances are also discussed.
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43.30.Gv Backscattering, echoes, and reverberation in water due to combinations of boundaries
43.30.Ft Volume scattering
43.20.Fn Scattering of acoustic waves

Deriving the equations of motion for porous isotropic media

Steven R. Pride, Anthony F. Gangi, and F. Dale Morgan

J. Acoust. Soc. Am. Volume 92, Issue 6, pp. 3278-3290 (1992); (13 pages) | Cited 21 times

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The equations of motion and stress/strain relations for the linear dynamics of a two‐phase, fluid/solid, isotropic, porous material have been derived by a direct volume averaging of the equations of motion and stress/strain relations known to apply in each phase. The equations thus obtained are shown to be consistent with Biot’s equations of motion and stress/strain relations; however, the effective fluid density in the equation of relative flow has an unambiguous definition in terms of the tractions acting on the pore walls. The stress/strain relations of the theory correspond to ‘‘quasistatic’’ stressing (i.e., inertial effects are ignored). It is demonstrated that using such quasistatic stress/strain relations in the equations of motion is justified whenever the wavelengths are greater than a length characteristic of the averaging volume size.
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43.20.Bi Mathematical theory of wave propagation
43.30.Ky Structures and materials for absorbing sound in water; propagation in fluid-filled permeable material
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