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Journal of the Acoustical Society of America

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Dec 1990

Volume 88, Issue 6, pp. 2527-2922

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Bacterial luminescence: A new tool for investigating the effects of acoustic energy and cavitation

Christopher McInnes, David Engel, and Roy W. Martin

J. Acoust. Soc. Am. Volume 88, Issue 6, pp. 2527-2532 (1990); (6 pages)

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An assay utilizing luminescent bacteria, Photobacterium phosphoreum, was adapted to assess the antibacterial effects of acoustic energy. Acoustic pressures up to 67 kPa in the 100‐ to 800‐Hz frequency range were applied to bacteria freely suspended in a liquid medium. Bacterial luminescence decreased after sonication, thus showing sensitivity to the effects of acoustic energy. This decreased luminescence was linearly related to exposure duration, appeared independent of acoustic frequency in this range, and was significantly heightened by the presence of cavitation. High‐frequency components of the acoustic emission were recorded from the sonicated fluid, and it was found that the decrease in luminescence due to sonication was directly related to the logarithm of the acoustic emission. Viability studies on exposed bacteria indicated a diminution of luminescence without bacterial death. The potential use of luminescent bacteria in assessing the biological effects of acoustic energy‐generating systems is discussed.
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43.80.Gx Mechanisms of action of acoustic energy on biological systems: physical processes, sites of action

Acoustical impedance measurements by the two‐microphone‐three‐calibration (TMTC) method

V. Gibiat and F. Laloë

J. Acoust. Soc. Am. Volume 88, Issue 6, pp. 2533-2545 (1990); (13 pages) | Cited 9 times

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A method is discussed for measuring the acoustical impedance of woodwind instruments that is relatively fast and insensitive to the ambient acoustical noise. The apparatus is essentially made of a measurement head, which is an approximately cylindrical cavity fitted with two microphones (in practice often three or more for technical reasons) and, at one of its ends, a loudspeaker; the other end is connected to the acoustical cavity under study (the instrument). The loudspeaker emits acoustical chirps and the signals provided by the microphones are Fourier transformed with the help of a microcomputer, which also extracts from the results the acoustical impedance as a function of frequency. A relatively high dynamic range is needed in order to match the high quality factor of the resonances of musical instruments, especially at low frequencies. To meet this requirement, it was found necessary to take into account the acoustical losses inside the head as well as the mutual perturbation of the microphones or various perturbations arising from geometrical imperfections of the cavity. This can be done automatically and without long calculations by using a calibration method, based on the successive use of three reference cavities, which are described and discussed in detail in this article. The method has been tested with cylindrical cavities and gives satisfactory results in terms of peak positions and heights. Impedance curves of clarinets have also been measured. The three‐calibration technique eliminates several sources of systematics and corrects automatically for various perturbations, such as perturbations of the acoustical field by the microphones, and allows the study of instruments of various diameters.
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43.75.Yy Instrumentation and measurement methods for musical acoustics
43.58.Bh Acoustic impedance measurement
43.75.Ef Woodwinds

On enhancement of spectral contrast in speech for hearing‐impaired listeners

H. Timothy Bunnell

J. Acoust. Soc. Am. Volume 88, Issue 6, pp. 2546-2556 (1990); (11 pages) | Cited 6 times

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A digital processing method is described for altering spectral contrast (the difference in amplitude between spectral peaks and valleys) in natural utterances. Speech processed with programs implementing the contrast alteration procedure was presented to listeners with moderate to severe sensorineural hearing loss. The task was a three alternative (/b/,/d/, or /g/) stop consonant identification task for consonants at a fixed location in short nonsense utterances. Overall, tokens with enhanced contrast showed moderate gains in percentage correct stop consonant identification when compared to unaltered tokens. Conversely, reducing spectral contrast generally reduced percent correct stop consonant identification. Contrast alteration effects were inconsistent for utterances containing /d/. The observed contrast effects also interacted with token intelligibility.
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43.72.-p Speech processing and communication systems
43.71.Ky Speech perception by the hearing impaired
43.66.Ts Auditory prostheses, hearing aids

Relating acoustic properties to perceptual responses: A study of Swedish voiced stops

Diana Krull

J. Acoust. Soc. Am. Volume 88, Issue 6, pp. 2557-2570 (1990); (14 pages) | Cited 1 time

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Perception models based on different kinds of acoustic data were compared with respect to their capacity to predict perceptual confusions between the Swedish stops [b,d,Δ,g] in systematically varied vowel contexts. Fragments of VC:V utterances read by a male speaker were presented to listeners. The resulting confusions were especially numerous between short stimulus segments following stop release, and formed a regular pattern depending mainly on the acute/grave dimension of the following vowel. The acoustic distances calculated were based on: (1) filter band spectra; (2) F2 and F3 at the CV boundary and in the middle of the following vowel; (3) the duration of the burst (=transient + noise section). Both the spectrum‐based and the formant‐based models provided measures of acoustic distance (dissimilarity) that revealed regular patterns. However, the predictive capacity of both models was improved by including the time‐varying properties of the stimuli in the distance measures. The highest correlation between predicted and observed percent confusions, r=0.85, was obtained with the formant‐based model in combination with burst length data. The asymmetries in the listeners’ confusions were also shown to be predictable, given acoustic data on the following vowel.
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43.71.Qr Neurophysiology of speech perception
43.71.Es Vowel and consonant perception; perception of words, sentences, and fluent speech
43.71.An Models and theories of speech perception

The influence of extraneous sounds on the perceptual estimation of first‐formant frequency in vowels

Brian Roberts and Brian C. J. Moore

J. Acoust. Soc. Am. Volume 88, Issue 6, pp. 2571-2583 (1990); (13 pages) | Cited 4 times

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The contribution of extraneous sounds to the perceptual estimation of the first‐formant (F1) frequency of voiced vowels was investigated using a continuum of vowels perceived as changing from /I/ to /q/ as F1 was increased. Any phonetic effects of adding extraneous sounds were measured as a change in the position of the phoneme boundary on the continuum. Experiments 1–5 demonstrated that a pair of extraneous tones, mistuned from harmonic values of the fundamental frequency of the vowel, could influence perceived vowel quality when added in the F1 region. Perceived F1 frequency was lowered when the tones were added on the lower skirt of F1, and raised when they were added on the upper skirt. Experiments 6 and 7 demonstrated that adding a narrow‐band noise in the F1 region could produce a similar pattern of boundary shifts, despite the differences in temporal properties and timbre between a noise band and a voiced vowel. The data are interpreted using the concept of the harmonic sieve [Duifhuis et al., J. Acoust. Soc. Am. 71, 1568–1580 (1982)]. The results imply a partial failure of the harmonic sieve to exclude extraneous sounds from the perceptual estimation of F1 frequency. Implications for the nature of the hypothetical harmonic sieve are discussed.
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43.71.Es Vowel and consonant perception; perception of words, sentences, and fluent speech
43.71.Qr Neurophysiology of speech perception
43.71.An Models and theories of speech perception
43.66.Ba Models and theories of auditory processes

Coarticulatory organization for lip rounding in Turkish and English

Suzanne E. Boyce

J. Acoust. Soc. Am. Volume 88, Issue 6, pp. 2584-2595 (1990); (12 pages) | Cited 1 time

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A number of studies, involving English, Swedish, French, and Spanish, have shown that, for sequences of rounded vowels separated by nonlabial consonants, both EMG activity and lip protrusion diminish during the intervocalic consonant interval, producing a ‘‘trough’’ pattern. A two‐part study was conducted to (a) compare patterns of protrusion movement (upper and lower lip) and EMG activity (orbicularis oris) for speakers of English and Turkish, a language where phonological rules constrain vowels within a word to agree in rounding and (b) determine which of two current models of coarticulation, the ‘‘look‐ahead’’ and ‘‘coproduction’’ models, best explained the data. Results showed Turkish speakers producing ‘‘plateau’’ patterns of movement rather than troughs, and unimodal rather than bimodal patterns of EMG activity. In the second part of the study, one prediction of the coproduction, model, that articulatory gestures have stable profiles across contexts, was tested by adding and subtracting movement data signals to synthesize naturally occurring patterns. Results suggest English and Turkish may have different modes of coarticulatory organization.
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43.70.-h Speech production
43.70.Bk Models and theories of speech production
43.70.Kv Cross-linguistic speech production and acoustics

Loudness relations for individuals and groups in normal and impaired hearing

Rhona P. Hellman and Carol H. Meiselman

J. Acoust. Soc. Am. Volume 88, Issue 6, pp. 2596-2606 (1990); (11 pages) | Cited 14 times

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Individual and group loudness relations were obtained at a frequency in the region of impaired hearing for 100 people, 98 with bilateral cochlear impairment. Slope distributions were determined from absolute magnitude estimation (AME) and absolute magnitude production (AMP) of loudness; they were also derived from cross‐modality matching (CMM) and AME of apparent length. With respect to both the means and the individual slope values, the two distributions closely agree. More than half of the measured deviations are less than 20%, with an overall average of −1.5%, meaning that transitivity is preserved for bilaterally impaired individuals. Moreover, over the stimulus range where cochlear impairment steepens the loudness function, both the group means and the individual slope values are clearly larger than in normal hearing. The results also show that, for groups of people with approximately similar losses, the standard deviation is a nearly constant proportion of the mean slope value giving a coefficient of variation of about 27% in normal and impaired hearing. This indicates, in accord with loudness matching, that the size of the slopes depends directly on the degree of hearing loss. The results disclose that loudness measurements obtained by magnitude scaling are able to reveal the operating characteristic of the ear for individuals.
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43.66.Ba Models and theories of auditory processes
43.66.Cb Loudness, absolute threshold
43.66.Sr Deafness, audiometry, aging effects

How much masking is informational masking?

Robert A. Lutfi

J. Acoust. Soc. Am. Volume 88, Issue 6, pp. 2607-2610 (1990); (4 pages) | Cited 17 times

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It is estimated that 22% of the masking observed in many traditional tone‐in‐noise detection experiments is due to uncertainty associated with trial‐to‐trial variation in the noise waveform.
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43.66.Ba Models and theories of auditory processes
43.66.Fe Discrimination: intensity and frequency
43.66.Dc Masking

Auditive and cognitive factors in speech perception by elderly listeners. II: Multivariate analyses

J. C. G. M. van Rooij and R. Plomp

J. Acoust. Soc. Am. Volume 88, Issue 6, pp. 2611-2624 (1990); (14 pages) | Cited 18 times

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In part I of this study [van Rooij et al., J. Acoust. Soc. Am. 86, 1294–1309 (1989)], the validity and manageability of a test battery comprising auditive (sensitivity, frequency resolution, and temporal resolution), cognitive (memory performance, processing speed, and intellectual abilities), and speech perception tests (at the phoneme, spondee, and sentence level) were investigated. In the present article, the results of a selection of these tests for 72 elderly subjects (aged 60–93 years) are analyzed by multivariate statistical techniques. The results show that the deterioration of speech perception in the elderly consists of two statistically independent components: (a) a large component mainly representing the progressive high‐frequency hearing loss with age that accounts for approximately two‐thirds of the systematic variance of the tests of speech perception and (b) a smaller component (accounting for one‐third of the systematic variance of the speech perception tests) mainly representing a general performance decrement due to reduced mental efficiency, which is indicated by a general slowing of performance and a reduced memory capacity. Although both components are correlated with age, it was found that the balance between auditive and cognitive contributions to speech perception performance did not change with age.
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43.66.Ba Models and theories of auditory processes
43.71.Es Vowel and consonant perception; perception of words, sentences, and fluent speech
43.71.Ky Speech perception by the hearing impaired

Summation bandwidths at threshold in normal and hearing‐impaired listeners

Maureen B. Higgins and Christopher W. Turner

J. Acoust. Soc. Am. Volume 88, Issue 6, pp. 2625-2630 (1990); (6 pages) | Cited 3 times

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The bandwidths for summation at threshold were measured for subjects with normal hearing and subjects with sensorineural hearing loss. Thresholds in quiet and in the presence of a masking noise were measured for complex stimuli consisting of 1 to 40 pure‐tone components spaced 20 Hz apart. The single component condition consisted of a single pure tone at 1100 Hz; additional components were added below this frequency, in a replication of the Gässler [Acustica 4, 408–414 (1954)] procedure. For the normal subjects, thresholds increased approximately 3 dB per doubling of bandwidth for signal bandwidths exceeding the critical bandwidth. This slope was less for the hearing‐impaired subjects. Summation bandwidths, as estimated from two‐line fits, were wider for the hearing‐impaired than for the normal subjects. These findings provide evidence that hearing‐impaired subjects integrate sound energy over a wider‐than‐normal frequency range for the detection of complex signals. A second experiment used stimuli similar to those of Spiegel [J. Acoust. Soc. Am. 66, 1356–1363 (1979)], and added components both above and below the frequency of the initial component. Using these stimuli, the slope of the threshold increase beyond the critical bandwidth was approximately 1.5 dB per doubling of bandwidth, thus replicating the Spiegel (1979) experiment. It is concluded that the differences between the Gässler (1954) and Spiegel (1979) studies were due to the different frequency content of the stimuli used in each study. Based upon the present results, it would appear that the slope of threshold increase is dependent upon the direction of signal expansion, and the size of the critical bands into which the signal is expanded.
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43.66.Cb Loudness, absolute threshold
43.66.Dc Masking
43.66.Sr Deafness, audiometry, aging effects

Limits of auditory pattern discrimination for patterns with various durations and numbers of components

Charles S. Watson, David C. Foyle, and Gary R. Kidd

J. Acoust. Soc. Am. Volume 88, Issue 6, pp. 2631-2638 (1990); (8 pages) | Cited 2 times

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In three experiments, listeners’ abilities to detect changes in randomly generated tonal sequences were determined for sequences or ‘‘patterns’’ ranging in total duration from 62.5 62.5 ms to 2 s. Experiment 1 utilized an adaptive‐tracking procedure, with n, the number of pattern components, as the dependent variable, and included a variety of spectral and temporal discrimination tasks with isochronous patterns. When the to‐be‐detected change was the only variation on a given dimension (e.g., the presence or location of a brief pause), patterns were discriminable when the absolute duration of the changed element, or pause, exceeded a critical value. However, when each pattern component varied on the dimension of the to‐be‐detected change (e.g., frequency), discriminability was strongly related to the number of tones in the pattern, and only weakly to the durations of either the target components or the total pattern. This dependence of discrimination performance on n was also demonstrated with anisochronous patterns in experiment 2. Experiment 3 revealed the same dependence of performance on the number of components per pattern as did experiments 1 and 2, but with Δf/f as the dependent variable, rather than n. The number of pattern components and the proportional duration of the target components, relative to total pattern duration, were confounded in these experiments. Additional research is therefore required to determine whether number or proportional target‐tone duration is the primary determinant of pattern discriminability.
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43.66.Mk Temporal and sequential aspects of hearing; auditory grouping in relation to music
43.66.Ki Subjective tones

Minimum audible movement angles as a function of sound source trajectory

Kourosh Saberi and David R. Perrott

J. Acoust. Soc. Am. Volume 88, Issue 6, pp. 2639-2644 (1990); (6 pages) | Cited 3 times

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Auditory resolution of moving sound sources was determined in a simulated motion paradigm for sources traveling along horizontal, vertical, or oblique orientations in the subjects’s frontal plane. With motion restricted to the horizontal orientation, minimum audible movement angles (MAMA) ranged from about 1.7° at the lowest velocity (1.8°/s) to roughly 10° at the highest velocity (320°/s). With the sound moving along an oblique orientation (rotated 45° relative to the horizontal) MAMAs generally matched those of the horizontal condition. When motion was restricted to the vertical, MAMAs were substantially larger at all velocities (often exceeding 8°). Subsequent tests indicated that MAMAs are a U‐shaped function of velocity, with optimum resolution obtained at about 2°/s for the horizontal (and oblique) and 7–11°/s for the vertical orientation. Additional tests conducted at a fixed velocity of 1.8°/s along oblique orientations of 80° and 87° indicated that even a small deviation from the vertical had a significant impact on MAMAs. A displacement of 10° from the vertical orientation (a slope of 80°) was sufficient to reduce thresholds (obtained at a velocity of 1.8°/s) from about 11° to approximately 2° (a fivefold increase in acuity). These results are in good agreement with our previous study of minimum audible angles long oblique planes [Perrott and Saberi, J. Acoust. Soc. Am. 87, 1728–1731 (1990)]. In summary, the results suggest that: (1) the ability to detect motion is essentially independent of the path traveled by the source, with one noted exception, sources moving within a few degrees of the vertical plane and (2) auditory resolution of sound sources in motion is a U‐shaped function of velocity with resolution degrading as velocities increase or decrease beyond an optical velocity range.
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43.66.Qp Localization of sound sources
43.66.Pn Binaural hearing

A single‐interval adjustment‐matrix (SIAM) procedure for unbiased adaptive testing

Christian Kaernbach

J. Acoust. Soc. Am. Volume 88, Issue 6, pp. 2645-2655 (1990); (11 pages) | Cited 4 times

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A new unbiased adaptive procedure is described that requires only half as many presentations in achieving the same precision as the well‐known two‐interval forced‐choice (2IFC) 2‐step procedure. The procedure is based on a yes–no task which avoids redundant presentation time. Furthermore, certain psychophysical studies can only be realized with yes–no tasks. Every trial contains randomly presented signals or noises and the answer is either yes or no. The outcome (hit, miss, false alarm, correct rejection) is taken into account by adjusting the signal level in a staircase manner. The adjustment matrix is set up to induce a neutral response criterion. Its convergence point can be adjusted at will. The single‐interval adjustment‐matrix (SIAM) procedure is compared to von Békésy and 2IFC transformed up–down methods using a Monte‐Carlo simulation. The SIAM procedure proves to be the fastest of the unbiased procedures. A test on four subjects verified these results. Implications for optimum track length and the number of reversals to discard are discussed.
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43.66.Yw Instruments and methods related to hearing and its measurement

Coding of spectral fine structure in the auditory nerve. II: Level‐dependent nonlinear responses

J. Wiebe Horst, Eric Javel, and Glenn R. Farley

J. Acoust. Soc. Am. Volume 88, Issue 6, pp. 2656-2681 (1990); (26 pages) | Cited 9 times

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Phase‐locked discharge patterns of single cat auditory‐nerve fibers were analyzed in response to complex tones centered at fiber characteristic frequency (CF). Signals were octave‐bandwidth harmonic complexes defined by a center frequency F and an intercomponent spacing factor N, such that F/N was the fundamental frequency. Parameters that were manipulated included the phase spectrum, the number of components, and the intensity of the center component. Analyses employed Fourier transforms of period histograms to assess the degree to which responses were synchronized to the frequencies present in the acoustic stimulus. Several nonlinearities were observed in the response as intensity was varied between threshold and 80–90 dB SPL. Response nonlinearities were strong for all signals except those with random phase spectra. The most commonly observed nonlinearity was an emphasis of one or more stimulus components in the response. The degree of nonlinearity usually increased with intensity and signal complexity and decreased with fiber frequency selectivity. Half‐wave rectification introduced synchronization to the missing fundamental. The strength of the response at the fundamental was related to stimulus crest factor. Signals with low center frequencies and high crest factors often elicited instantaneous discharge rates at the theoretical maximum of πCF. This suggests that the probability of spike generation approaches one during high‐amplitude waveform segments. Response nonlinearity was interpreted as arising from three sources, namely, cochlear mechanics, compression of instantaneous discharge rate, and saturation of average discharge rate. At near‐threshold intensities, fibers with high spontaneous rates exhibited responses that were linear functions of stimulus waveshape, whereas fibers with low spontaneous spike rates produced responses that were best described in terms of an expansive nonlinearity.
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43.64.Pg Electrophysiology of the auditory nerve
43.66.Hg Pitch
43.66.Nm Phase effects
43.66.Ba Models and theories of auditory processes

Neural correlates of psychophysical release from masking

John B. Mott, Lynn P. McDonald, and Donal G. Sinex

J. Acoust. Soc. Am. Volume 88, Issue 6, pp. 2682-2691 (1990); (10 pages) | Cited 2 times

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Responses of chinchilla auditory‐nerve fibers were measured for stimulus conditions analogous to those in which psychophysical release from masking has been observed in humans. The maskers were two equal power, narrow‐band noise stimuli with different amplitude envelopes. The neurons in the sample fell into three groups that resolved the maskers’ envelopes with varying degrees of accuracy. The boundaries of these groups were not sharply delineated by characteristic frequency (CF) but were dependent on the relationship between the masker level and the neurons’ thresholds at the masker frequency. For the neurons that best preserved the maskers’ envelope fluctuations, a neural release from masking was observed; rate‐based neural masked thresholds were higher for the masker with the least fluctuating envelope. The results suggest that neural and psychophysical release from masking arises because the probe evokes larger rate changes, relative to the background response to the masker, during periods of low masker energy. Between two otherwise equivalent maskers, the one with the periods of lowest energy will produce the lower masked thresholds because rate changes are larger and more detectable.
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43.64.Pg Electrophysiology of the auditory nerve
43.66.Dc Masking
43.64.Bt Models and theories of the auditory system

Signal detection in the presence of inaccurate multipath time delay modeling

Isabel M. G. Lourtie and G. Clifford Carter

J. Acoust. Soc. Am. Volume 88, Issue 6, pp. 2692-2694 (1990); (3 pages) | Cited 1 time

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In this paper, a new class of signal detectors is considered whose structure is similar to the log‐likelihood detector developed under low signal‐to‐noise ratio conditions, but with different choices to the sensor filters. In the presence of incorrect delay assumptions, the performance of the log‐likelihood processor is analyzed and interpreted. Based on this study, ad hoc detectors are proposed, and their performance compared to that of the log‐likelihood structure. It is shown that, for increasing misadjustment in the delay assumptions, the performance of the proposed ad hoc detectors is superior.
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43.60.Cg Statistical properties of signals and noise
43.60.Gk Space-time signal processing, other than matched field processing
43.30.Wi Passive sonar systems and algorithms, matched field processing in underwater acoustics

Sharpness applied to the adaptive beamforming of acoustic data from a towed array of unknown shape

Brian G. Ferguson

J. Acoust. Soc. Am. Volume 88, Issue 6, pp. 2695-2701 (1990); (7 pages) | Cited 1 time

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When a vessel towing a horizontal line array of underwater acoustic sensors changes course, the array responds by changing its shape so that the sensors no longer lie along a straight line. However, the spatial processing of the acoustic data still proceeds on the assumption that the array is straight with the positions of the sensors being known and invariant. When the array adopts a nonlinear shape so that the actual locations of the sensors no longer coincide with the assumed positions, the performance of the beamformer is observed to deteriorate. This degradation is most marked when an adaptive beamformer is used to optimize the array gain. The adaptive beamformer considered here is a constrained optimum beamformer that is based on the inversion of the observed (signal‐plus‐noise) cross‐spectral matrix of the sensor outputs. The adaptive beamformer responds to any deviation of the sensors from their nominal positions by suppressing the received signals so that the output signal‐to‐noise ratio of the beamformer, and hence the array gain, decrease in the direction of each signal. One method of overcoming the array shape problem is to monitor the geometrical configuration of the array’s sensors by instrumenting the length of the array with compasses and depth sensors. This paper examines an alternate method whereby data from the acoustic sensors themselves are used to infer the shape of the array. This narrow‐band acoustic technique, which relies on at least one acoustic source being present in the far field, evaluates a sharpness function for a variety of possible array shapes. The sharpness is calculated by summing the product of the beam output power squared and the sine of the beamsteer angle over all beamsteer directions from forward endfire to aft endfire. Using simulated data, it is shown that the estimated array shape matches the actual shape when the sharpness attains its maximum value. Also shown is the dramatic reduction in signal suppression when this acoustic technique is applied to the adaptive beamforming of real acoustic data.
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43.60.Gk Space-time signal processing, other than matched field processing
43.30.Wi Passive sonar systems and algorithms, matched field processing in underwater acoustics

On estimating the amplitude of harmonic vibration from the Doppler spectrum of reflected signals

Sung‐Rung Huang, Robert M. Lerner, and Kevin J. Parker

J. Acoust. Soc. Am. Volume 88, Issue 6, pp. 2702-2712 (1990); (11 pages) | Cited 8 times

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The Doppler spectrum of echoes from a sinusoidally vibrating scatterer has discrete spectral lines weighted by Bessel functions of the first kind. Because the signal and spectrum are complicated functions of the vibration amplitude, a number of different approaches have been tried in the past to estimate the vibration amplitude, given a received signal. Here, a new and simple relationship between the spread (or variance) of the Doppler spectrum and the vibration amplitude is derived. A method of estimating the vibration amplitude is proposed based on this relation and a noise compensation procedure is also demonstrated. The performance of the estimators is studied through simulations. High accuracy is predicted under proper sampling conditions even when the signal‐to‐noise ratio is poor. Slight deviations from single‐frequency oscillation, as would be caused by nonlinear or nonideal medium or source effects, are found to have little contribution to the total estimation error.
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43.60.Gk Space-time signal processing, other than matched field processing
43.20.Fn Scattering of acoustic waves
43.30.Es Velocity, attenuation, refraction, and diffraction in water, Doppler effect

Diffraction tomography of strongly scattering infinite cylindrical objects of arbitrary cross‐sectional shape

S. Jegannathan and Bhaskar Ramamurthi

J. Acoust. Soc. Am. Volume 88, Issue 6, pp. 2713-2719 (1990); (7 pages)

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A new method for imaging an infinite cylindrical object with arbitrary cross‐sectional shape, without any small‐perturbation approximations, is proposed. In this method, the well‐known integral equation equivalent to the inhomogeneous Helmholtz wave equation is discretized. The discretized equation expresses the scattered pressure field at any point in terms of the samples of the product of the unknown object function and the unknown pressure field present inside the object. Based on this relation, simultaneous equations are set up, relating these samples to the samples of the pressure field measured outside the object. By requiring the measurement points to lie on circles around the object, these equations are solved efficiently using the fast Fourier transform (FFT). From the product samples thus obtained, the samples of the unknown object function are extracted by means of a second set of FFT calculations. Details of illustrative computer simulations for the case of an elliptic cylinder are described.
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43.60.Gk Space-time signal processing, other than matched field processing
43.60.Pt Signal processing techniques for acoustic inverse problems
43.60.Rw Remote sensing methods, acoustic tomography

Signal processing of ideal echoes resonantly scattered by underwater structures

Edward McDaid and Guillermo C. Gaunaurd

J. Acoust. Soc. Am. Volume 88, Issue 6, pp. 2720-2735 (1990); (16 pages) | Cited 1 time

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A study is presented of the scattering of sound waves that results when penetrable underwater obstacles of various types are insonified by various incident transient signals. The targets are elastic shells, although the case of a rigid sphere was also studied for comparison. The deformation of the shell is described both exactly, by means of the three‐dimensional equations of elasticity, and approximately, by means of Donnell’s shell theory. The goal is to identify significant signal processing parameters that are useful at the source, the target, and the receiver, to properly and quantitatively estimate the nature of the backscattered echoes from submerged structures. These parameters play an important role in the ultimate goal of active target identification. The methodology presented has similarities to that most commonly used for impenetrable targets (i.e., perfect conductors)—in the radar literature. The analysis is first introduced here for targets away from boundaries, and in noiseless media. It is believed that the type of echo received from a sonar target should be known, before such return is distorted and corrupted by the environment. As presented here, this idealized situation is well suited to later account for the presence of noise in the medium, as well as for the proximity to environmental boundaries. The methodology already takes advantage of the machinery of statistical communication theory (viz., matched filters, ambiguity and correlation functions, etc.) to extract desired signals from signals mixed with undesirable noises. The work is naturally divided into descriptions and characteristics of: (a) the incident pulses, (b) the elastic targets themselves, and (c) the backscattered returns. The various quantities are illustrated at all stages with many computed displays.
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43.60.Gk Space-time signal processing, other than matched field processing
43.30.Gv Backscattering, echoes, and reverberation in water due to combinations of boundaries

Recovery of acoustic source extent by Fourier phase analysis of emitted signals

Wolfgang Sachse and Wilbur F. Pierce

J. Acoust. Soc. Am. Volume 88, Issue 6, pp. 2736-2742 (1990); (7 pages)

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This paper considers an algorithm for recovery of the size of an extended, finite source of elastic waves in a plate from the Fourier phase function of the signals it emits. In the forward problem, an extended source is modeled as an assemblage of point sources of variable strength arranged along a straight line. The Fourier phase spectrum of the first arrival signal from such a source is determined. It is shown that this phase function contains information related to the spacial extent of the source and that it can be processed, along with the phase function of a single point source, to recover the extent of the extended acoustic source. Applications of the algorithm to process synthetic signals from extended sources with three spatial distributions in the interior of the plate are given.
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43.60.Gk Space-time signal processing, other than matched field processing

An alternative mathematical description of the relationship between noise exposure and hearing loss

David A. Bies and Colin H. Hansen

J. Acoust. Soc. Am. Volume 88, Issue 6, pp. 2743-2754 (1990); (12 pages) | Cited 4 times

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Retrospective investigation of large populations has provided means for determining quantitative relationships between the hearing levels of non‐noise‐exposed populations and age, and between the hearing levels of noise‐exposed populations and age and noise exposure. In the latter case, noise exposures have been predominantly steady state over many years and no way of measuring the effects of noise alone, excluding the effects of age, has ever been demonstrated. In the following paper, attention is confined to the problem of developing a mathematical description of an existing set of empirically determined hearing level data; questions of audiology are not of concern here. It is shown that the mathematical analysis traditionally used to determine the contribution of noise exposure alone to hearing level is not unique; an alternative formulation is possible and indeed is demonstrated. Whereas the traditional formulation leads to the conclusion that noise‐induced hearing loss scales on the integral of sound pressure squared with time, and thus, to the equal energy hypothesis, the alternative formulation leads to the conclusion that noise‐induced hearing loss scales on the integral of pressure with time. Since either formulation adequately describes the data, and the equal energy hypothesis has never been adequately substantiated, use of the latter hypothesis to extend the findings of steady‐state exposures to application for unsteady exposures is not justified. The alternative formulation presented here is recommended for consideration.
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43.50.Ba Noisiness: rating methods and criteria
43.50.Qp Effects of noise on man and society
43.66.Ed Auditory fatigue, temporary threshold shift

On the prediction of propeller tone sound levels and gradients in an airplane cabin

Larry D. Pope

J. Acoust. Soc. Am. Volume 88, Issue 6, pp. 2755-2765 (1990); (11 pages) | Cited 1 time

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Theoretical and computational methods are presented that will permit a basic prediction of the spatial variation of sound‐pressure level in an airplane cabin arising from propeller excitation of fuselage sidewall structure. Analytical results focus on the calculation of the axial variation (or gradient) of the average mean‐square pressure in cross section for the case of a trimmed cabin of uniform cross‐sectional area terminating fore and aft in reflecting (and absorbing) partitions (i.e., bulkheads). Sample numerical predictions and qualitative comparisons with some available measurement data are presented.
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43.50.Lj Transportation noise sources: air, road, rail, and marine vehicles
43.55.Rg Sound transmission through walls and through ducts: theory and measurement

Vibration transmission through frame or beam junctions

James A. Moore

J. Acoust. Soc. Am. Volume 88, Issue 6, pp. 2766-2776 (1990); (11 pages) | Cited 3 times

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A model of vibration transmission through junctions of one‐dimensional frame members has been developed. The formulation is flexible in allowing from two to four frames at the junction where the frames lie in the same plane and intersect each other at 90° orientations. The extension to arbitrary orientations in the same plane was derived and the approach for arbitrary orientations in three dimensions is suggested. The model allows for torsional, longitudinal, and bending deformation about two axes. When the frames all lie in the same plane a decomposition into in‐plane and out‐of‐plane motions occurs that simplifies the solution procedure. Calculations for a frame construction representative of helicopter airframes indicate that out‐of‐plane bending and in‐plane longitudinal motion are strongly transmitted across frame junctions. Torsion and in‐plane bending are more weakly transmitted. The effect of junction asymmetries in producing transmission into otherwise unexcited motions is pointed out.
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43.40.At Experimental and theoretical studies of vibrating systems
43.40.Cw Vibrations of strings, rods, and beams
43.20.Mv Waveguides, wave propagation in tubes and ducts
43.20.Tb Interaction of vibrating structures with surrounding medium

Frame junction vibration transmission with a modified frame deformation model

James A. Moore

J. Acoust. Soc. Am. Volume 88, Issue 6, pp. 2777-2788 (1990); (12 pages)

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A previous paper dealt with vibration transmission through junctions of connected frame members where the allowed frame deformations included bending, torsion, and longitudinal motions [J. A. Moore, J. Acoust. Soc. Am. 88, 2766–2776 (1990)]. In helicopter and aircraft structures the skin panels can constitute a high impedance connection along the length of the frames that effectively prohibits in‐plane motion at the elevation of the skin panels. This has the effect of coupling in‐plane bending and torsional motions within the frame. This paper discusses the transmission behavior through frame junctions that accounts for the in‐plane constraint in idealized form by assuming that the attached skin panels completely prohibit in‐plane motion in the frames. Also, transverse shear deformation is accounted for in describing the relatively deep web frame constructions common in aircraft structures. Longitudinal motion in the frames is not included in the model. Transmission coefficient predictions again show the importance of out‐of‐plane bending deformation to the transmission of vibratory energy in an aircraft structure. Comparisons are shown with measured vibration transmission data along the framing in the overhead of a helicopter airframe, with good agreement. The frame junction description has been implemented within a general purpose statistical energy analysis (SEA) computer code in modeling the entire airframe structure including skin panels.
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43.40.At Experimental and theoretical studies of vibrating systems
43.40.Cw Vibrations of strings, rods, and beams
43.20.Ks Standing waves, resonance, normal modes
43.20.Fn Scattering of acoustic waves
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