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Journal of the Acoustical Society of America

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Jul 1989

Volume 86, Issue 1, pp. 1-456

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Determination of the acoustic nonlinearity parameter in biological media using FAIS and ITD methods

Xiu‐fen Gong, Zhe‐ming Zhu, Tao Shi, and Jian‐hong Huang

J. Acoust. Soc. Am. Volume 86, Issue 1, pp. 1-5 (1989); (5 pages) | Cited 8 times

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Two methods for determination of the acoustic nonlinearity parameter B/A in biological media are presented. One is the finite amplitude insert‐substitution method (FAIS), considering the influence of both the sound attenuation of samples and the diffraction of the transducer on the experimental measurement. The other is the improved thermodynamic method (ITD), based on the measurement of phase shifts in the acoustic wave due to the change of ambient pressure. The nonlinearity parameter B/A has been measured for various biological solutions and soft tissues using these two methods. Some results of dependence of B/A values on the concentration and temperature are also presented here.
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43.80.Ev Acoustical measurement methods in biological systems and media
43.80.Cs Acoustical characteristics of biological media: molecular species, cellular level tissues
43.25.Ba Parameters of nonlinearity of the medium

Effect of the pulse length of ultrasound on cell membrane damage in vitro

Lorenz O. Kober, Joachim W. Ellwart, and Hans Brettel

J. Acoust. Soc. Am. Volume 86, Issue 1, pp. 6-7 (1989); (2 pages) | Cited 2 times

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Suspended cells of a human lymphoblastic cell line were exposed to pulsed ultrasound of 775 kHz. The pulse lengths were varied between 16 and 1000 μs. The mark/space ratio was always kept at 1:1. Two ultrasound intensity levels were used: 3.6 and 6.4 W/cm2 spatial peak and temporal peak. After an exposure time of 5 min, cell membrane damage was measured cytometrically by a dye exclusion test. No membrane damage was observable at 16 μs, whereas, at pulse lengths of 1000 μs, about one‐third of the cells were damaged.
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43.80.Gx Mechanisms of action of acoustic energy on biological systems: physical processes, sites of action
43.80.Sh Medical use of ultrasonics for tissue modification (permanent and temporary)

The acoustic behavior of the fish‐catching bat, Noctilio leporinus, during prey capture

David J. Hartley, Karen A. Campbell, and Roderick A. Suthers

J. Acoust. Soc. Am. Volume 86, Issue 1, pp. 8-27 (1989); (20 pages) | Cited 5 times

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Many bats change the acoustic parameters of their echolocation calls in a deliberate manner during prey capture. Attempts to quantify these changes have been either of limited scope or subject to potentially severe errors due to an inadequate consideration of the directionality of both the bat and the recording microphone. Therefore, the echolocation pulses emitted by two N. leporinus have been recorded as they approached and captured stationary prey, with the microphone positioned in such a way that the structure of the pulses incident upon the target could be determined. The results of this study show that: (1) during the last 1.5 m of the approach, N. leporinus reduces the intensity of emitted pulses by 6 dB per halving of distance, so that the intensity incident upon the target is constant; (2) at a point in the pulse train that corresponds to the position of the hypothesized tracking phase of echolocation, N. leporinus selectively reduces the intensity of the frequency‐modulated (FM) fundamental so that the FM pulse component is predominantly second harmonic; and (3) a high degree of temporal overlap occurs between the FM component of emitted pulse and echo when N. leporinus is within 0.4 m of its prey.
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43.80.Nd Effects of noise on animals and associated behavior, protective mechanisms
43.80.Lb Sound reception by animals: anatomy, physiology, auditory capacities, processing
43.80.Jz Use of acoustic energy (with or without other forms) in studies of structure and function of biological systems

Ultrasonic reflection mode imaging of the nonlinear parameter B/A. II: Signal processing

Charles A. Cain and Hooman Houshmand

J. Acoust. Soc. Am. Volume 86, Issue 1, pp. 28-34 (1989); (7 pages)

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The nonlinear acoustic interaction between a reflected single‐frequency sinusoid and a broadband pump waveform propagating in the opposite direction produces phase changes in the probe proportional to the nonlinear parameter B/A in the spatial region of interaction. The instantaneous phase change along the received probe can be expressed as the convolution of the pump waveform with the spatial distribution of B/A along the propagation path over which the pump and reflected probe interact. In theory, the phase modulated sinusoidal probe can be processed (phase detection and deconvolution) to produce an ‘‘A‐mode’’ representation of B/A. If the pump is an intense unipolar impulse and the probe a swept‐frequency sinusoid, then the pump interacts with the probe at each point along the propagation path at a unique frequency. Thus the phase modulation that carries information about the spatial distribution of B/A can be extracted from the phase spectrum by a simple Fourier transformation analogous to the space to frequency mapping so basic to magnetic resonance imaging. If the impulsive pump is replaced by another swept‐frequency sinusoid, then the phase change in the probe due to B/A at a particular point along the propagation path will be spread out for the duration of the pump along the probe. Passage of the received signal through an appropriate matched filter restores spatial coherence to the phase information in the probe so that it can be processed as if the pump were a broadbanded impulse. This approach suggests a means of approaching the design of effective pump waveforms that can resolve a wide range of spatial frequencies in (B/A)(x).
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43.80.Qf Medical diagnosis with acoustics
43.80.Vj Acoustical medical instrumentation and measurement techniques
43.25.Ba Parameters of nonlinearity of the medium

Calculation of the steady‐state oscillations of a clarinet using the harmonic balance technique

J. Gilbert, J. Kergomard, and E. Ngoya

J. Acoust. Soc. Am. Volume 86, Issue 1, pp. 35-41 (1989); (7 pages) | Cited 7 times

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The harmonic balance technique is known as a time‐frequency simulation technique used for the study of large signal regimes of microwave circuits driven in forced oscillation. The technique can be adapted to self‐sustained oscillations, especially for wind musical instruments such as clarinets. The resonator (i.e., the instrument body) is the linear part, treated in the frequency domain, while the driving system (the reed) is the nonlinear part, treated in the time domain. The harmonic balance method is shown to connect the results of two known methods, so‐called weakly nonlinear (in frequency domain) and strongly nonlinear (in time domain). The advantages and disadvantages of the method are discussed.
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43.75.Ef Woodwinds

Quantization and measurement errors in the analysis of short‐time perturbations in sampled data

Neil B. Cox, Mabo R. Ito, and Murray D. Morrison

J. Acoust. Soc. Am. Volume 86, Issue 1, pp. 42-54 (1989); (13 pages)

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This paper provides an analysis of the effect of quantization and measurement errors on two algorithms for measuring short‐time perturbation about a slowly varying or periodically varying mean. While these algorithms can be applied in a variety of applications, they will be discussed in the context of their use in quantification of vowel perturbations associated with hoarseness. The algorithms are referred to as the relative average perturbation and the directional perturbation quotient in speech analysis literature. The analyses indicate that significant errors are present in published studies. Sampling conditions and analysis techniques to minimize these errors are described.
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43.72.Ar Speech analysis and analysis techniques; parametric representation of speech

Vowel representation: Some observations on temporal and spectral properties of the first formant frequency

Maria‐Gabriella Di Benedetto

J. Acoust. Soc. Am. Volume 86, Issue 1, pp. 55-66 (1989); (12 pages) | Cited 5 times

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Acoustic analysis of the vocalic portion of consonant–vowel–consonant (CVC) syllables (where V is one of the five vowels [I,q,1,a,v] of American English) spoken by three speakers (two males and one female) in the sentence frame ‘‘The — again’’ is presented. Results of acoustic measurements show that ambiguities between vowels, for each speaker, occur if the vowels are represented by the values of F1 and F2 sampled at the time where F1 reaches its maximum. These ambiguities occur primarily in the F1 dimension. Examination of the F1 trajectories of the vowels for which confusion occurs shows variations in the way F1 reaches its maximum among different vowels. In particular, if two different vowels such as [I] and [q] have the same maximum F1, then F1 for the lower vowel reaches its maximum value earlier. In addition, results show that the F1 onset frequency also might be important in determining vowel height. The implication is that the spectral characteristics at a particular ‘‘target,’’ represented by the time at which F1 reaches its maximum, are not invariant attributes of the vowel. The results support a hypothesis that time and/or frequency variations of the first formant must be taken into account if an invariant property is to be associated with a vowel.
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43.71.Es Vowel and consonant perception; perception of words, sentences, and fluent speech
43.70.Fq Acoustical correlates of phonetic segments and suprasegmental properties: stress, timing, and intonation

Frequency and time variations of the first formant: Properties relevant to the perception of vowel height

Maria‐Gabriella Di Benedetto

J. Acoust. Soc. Am. Volume 86, Issue 1, pp. 67-77 (1989); (11 pages) | Cited 5 times

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Perceptual experiments using consonant–vowel–consonant (CVC) syllables were carried out to examine the perceptual relevance of the first formant frequency (F1) trajectory in the perception of high vowels versus nonhigh vowels. Results show that stimuli characterized by a higher onset frequency and F1 maximum at the beginning of the vocalic portion are perceived as lower vowels than stimuli with a lower F1 onset frequency and F1 maximum toward the end of the vocalic portion. These findings are in agreement with the hypothesis, based on the acoustic analyses of Di Benedetto (1989), that stimuli with higher F1 onset frequencies and F1 maximum at the beginning of the vocalic portion characterize lower vowels. Results are similar for native speakers of different languages, leading to a suggestion that this phenomenon may have either an articulatory or an auditory basis. Possible interpretations based on an overshoot hypothesis or a formant time average theory were investigated through an additional perceptual experiment. Results of this last experiment agree with a weighted average time formant theory.
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43.71.Es Vowel and consonant perception; perception of words, sentences, and fluent speech
43.70.Fq Acoustical correlates of phonetic segments and suprasegmental properties: stress, timing, and intonation

Kinematic and electromyographic responses to perturbation of the jaw

Susan Shaiman

J. Acoust. Soc. Am. Volume 86, Issue 1, pp. 78-88 (1989); (11 pages) | Cited 6 times

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The task‐dependent organization of sensorimotor mechanisms during the production of speech was investigated using a perturbation paradigm. Six subjects received unanticipated jaw perturbations before and during tongue elevation for [1d1], in which the lips do not participate, and bilabial closure for [1b1], in which the tongue does not participate. A strain gauge system was used to monitor inferior–superior displacements of the upper lip, lower lip, and jaw, while hooked‐wire electrodes monitored muscle activity in various muscles of the lips, jaw, and tongue. Results indicated significant compensatory kinematic adjustments to jaw perturbations in the lips and/or jaw during [1b1], but no labial compensations during [1d1] (with the exception of one subject). EMG responses were inconsistent and not necessarily indicative of the kinematic findings. Individual subjects responded to perturbations reliably but differently, using different combinations of involved articulators to achieve bilabial closure and lingua–alveolar contact. The current study supports earlier research which suggests that the components of the motor system are flexibly assembled, based on the requirements of the specific task. That is, compensatory responses to sensory information occur only when such responses are functionally necessary.
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43.70.Aj Anatomy and physiology of the vocal tract, speech aerodynamics, auditory kinetics
43.70.Bk Models and theories of speech production

Directional sensitivity of sound‐pressure levels in the human ear canal

John C. Middlebrooks, James C. Makous, and David M. Green

J. Acoust. Soc. Am. Volume 86, Issue 1, pp. 89-108 (1989); (20 pages) | Cited 25 times

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Changes in sound pressures measured in the ear canal are reported for broadband sound sources positioned at various locations about the subject. These location‐dependent pressures are one source of acoustical cues for sound localization by human listeners. Sound source locations were tested with horizontal and vertical resolution of 10°. Sound levels were measured with miniature microphones placed inside the two ear canals. Although the measured amplitude spectra varied with the position of the microphone in the ear canal, it is shown that the directional sensitivity at any particular frequency of the broadband stimulus is independent of microphone position anywhere within the ear canal. At any given frequency, the distribution of sound pressures as a function of sound source location formed a characteristic spatial pattern comprising one or two discrete areas from which sound sources produced maximum levels in the ear canal. The locations of these discrete areas varied in horizontal and vertical location according to sound frequency. For example, around 8 kHz, two areas of maximum sensitivity typically were found that were located laterally and were separated from each other vertically, whereas, around 12 kHz, two such areas were found located on the horizontal plane and separated horizontally. The spatial patterns of sound levels were remarkably similar among different subjects, although some frequency scaling was required to accommodate for differences in the subjects’ physical sizes. Interaural differences in sound‐pressure level (ILDs) at frequencies below about 8 kHz tended to increase monotonically with increasing distance of the sound source from the frontal midline and tended to be relatively constant as a function of vertical source location. At higher frequencies, however, ILDs varied both with the horizontal and with the vertical location of the sound source. At some frequencies, asymmetries between the left and right ears in a given subject resulted in substantial ILDs even for midline sound sources. These results indicate the types of horizontal and vertical spatial information that are available from sound level cues over various ranges of frequency and, within a small subject population, indicate the nature of intersubject variability.
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43.66.Ba Models and theories of auditory processes
43.66.Pn Binaural hearing
43.66.Qp Localization of sound sources
43.64.Ha Acoustical properties of the outer ear; middle-ear mechanics and reflex

Intensity discrimination determined with two paradigms in normal and hearing‐impaired subjects

Christopher W. Turner, Jozef J. Zwislocki, and Paul R. Filion

J. Acoust. Soc. Am. Volume 86, Issue 1, pp. 109-115 (1989); (7 pages) | Cited 3 times

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The literature on auditory intensity jnd’s is ambiguous with respect to the relationship between the jnd’s measured with gated and continuous pedestals and with respect to changes in this relationship in the presence of loudness recruitment accompanying cochlear pathology. In an attempt to clarify these issues and to lay a foundation for systematic investigations of the dependence on the jnd’s on loudness functions, the jnd’s for pure tones with gated‐ and continuous‐pedestal paradigms of two groups of subjects, one with normal hearing and one with hearing loss of cochlear origin, were measured. The experiments were performed at 0.5, 2, and 6 kHz, and at a wide range of sensation levels (SLs) by means of an adaptive two‐alternative, forced‐choice (2IFC) procedure. The jnd’s obtained with the continuous‐pedestal method were smaller than those obtained with the gated‐pedestal method for both groups of subjects. They also had smaller intersubject standard deviations. When jnd’s of the two groups were compared on the basis of equal SLs, the group with hearing loss showed smaller jnd values than the group with normal hearing for both pedestal paradigms. When the comparisons were made on the basis of equal sound‐pressure levels (SPLs), both groups showed similar values for moderate and high SPLs. At relatively low SPLs, the group with hearing loss tended to have somewhat higher values.
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43.66.Fe Discrimination: intensity and frequency
43.66.Yw Instruments and methods related to hearing and its measurement
43.66.Sr Deafness, audiometry, aging effects

Perception of complex tone pairs mistuned from unison

Richard M. Warren, James A. Bashford, Jr., and Bradley S. Brubaker

J. Acoust. Soc. Am. Volume 86, Issue 1, pp. 116-125 (1989); (10 pages)

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Periodic sounds mistuned from unison may interact to produce pitch glides: When a broad‐spectrum complex tone having a fundamental frequency of 400 Hz or less and containing several harmonics above the 8th is mixed with itself after a slight change in the waveform repetition frequency (1 Hz or less), listeners hear a rising glissando when corresponding portions of the waveforms approach alignment and a falling glissando as they recede from alignment. Glissandi are unimpaired if harmonics below the 8th are absent, but if, instead, harmonics above the 8th are removed, only amplitude fluctuations are heard (not glissandi). When two broad‐spectrum complex tones with independent, randomly derived phase spectra are mistuned slightly from unison and mixed, complex repeated patterns other than glissandi are heard. These observations, along with others involving a variety of periodic sounds mistuned from unison, provide information concerning the nature of frequency domain and time domain mechanisms employed for the perception of iterated acoustic patterns.
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43.66.Hg Pitch
43.66.Ki Subjective tones

Comparison of discomfort levels obtained with pure tones and multitone complexes

Ruth A. Bentler and Chaslav V. Pavlovic

J. Acoust. Soc. Am. Volume 86, Issue 1, pp. 126-132 (1989); (7 pages)

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The relationship between threshold of discomfort (TD) estimates and the number of components in a complex signal has been investigated. The thresholds of discomfort were first obtained for 16 pure tones located at the center frequency of critical bands from 250 to 4000 Hz. Subsequently, thresholds of discomfort were obtained for 2, 4, 8, and 16 tone complexes. The pure‐tone components of the complexes were systematically selected from the same 16 pure tones. For each subject, the relative intensities of the components in the four complexes were determined in such a way so as to parallel the pure tone TD contour obtained for that subject. Data were obtained from 15 normal and 15 hearing impaired adults. The individuals in the latter group all had mild to moderate sensorineural hearing loss. Summation of discomfort (S) was defined as the difference between the threshold of discomfort for a pure tone presented in isolation and within the complex. The two groups demonstrated different summation values. For both groups, however, the summation was shown to be a linear function of the logarithm of the number of components in the complex: S=a+b log (n) where n is the number of components (2, 4, 8, 16). For the normal hearing group, a and b are 2.05 and 11.51, respectively, while for the hearing impaired group, they are 3.95 and 12.88, respectively. While the future digital hearing aids can easily regulate their limiting levels so as to accurately account for this summation, present day hearing aids may underestimate this effect. However, given current clinical practices (which somewhat underestimate the pure‐tone TD) this is probably not the case.
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43.66.Sr Deafness, audiometry, aging effects
43.66.Ts Auditory prostheses, hearing aids
43.66.Cb Loudness, absolute threshold

Realistic mechanical tuning in a micromechanical cochlear model

Paul J. Kolston, Max A. Viergever, Egbert de Boer, and Rob J. Diependaal

J. Acoust. Soc. Am. Volume 86, Issue 1, pp. 133-140 (1989); (8 pages) | Cited 2 times

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Two assumptions were made in the formulation of a recent cochlear model [P. J. Kolston, J. Acoust. Soc. Am. 83, 1481–1487 (1988)]: (1) The basilar membrane has two radial modes of vibration, corresponding to division into its arcuate and pectinate zones; and (2) the impedance of the outer hair cells (OHCs) greatly modifies the mechanics of the arcuate zone. Both of these assumptions are strongly supported by cochlear anatomy. This paper presents a revised version of the outer hair cell, arcuate–pectinate (OHCAP) model, which is an improvement over the original model in two important ways: First, a model for the OHCs is included so that the OHC impedance is no longer prescribed functionally; and, second, the presence of the OHCs enhances the basilar membrane motion, so that the model is now consistent with observed response changes resulting from trauma. The OHCAP model utilizes the unusual spatial arrangement of the OHCs, the Deiters cells, their phalangeal processes, and the pillars of Corti. The OHCs do not add energy to the cochlear partition and hence the OHCAP model is passive. In spite of the absence of active processes, the model exhibits mechanical tuning very similar to those measured by Sellick et al. [Hear. Res. 10, 93–100 (1983)] in the guinea pig cochlea and by Robles et al. [J. Acoust. Soc. Am. 80, 1364–1374 (1986)] in the chinchilla cochlea. Therefore, it appears that mechanical response tuning and response changes resulting from trauma should not be used as justifications for the hypothesis of active processes in the real cochlea.
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43.64.Kc Cochlear mechanics
43.64.Bt Models and theories of the auditory system

Dynamic spectral transforms: Properties of the canted spectral transform

Albert A. Gerlach, Kenneth D. Flowers, Wendell L. Anderson, and Edward L. Kunz

J. Acoust. Soc. Am. Volume 86, Issue 1, pp. 141-149 (1989); (9 pages)

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A generalized spectral transform is defined by extending the kernel of the conventional sectionalized Fourier transform (SFT). The generalized transform accumulates signal energy along narrow dynamic spectral channels that may be made to conform to the instantaneous‐frequency dynamics of a given signal. This property may be used to achieve optimum detection of a deterministically known signal or to estimate the spectral dynamics of an unknown signal over the temporal limits of the transform. As an initial step toward achieving the spectral transform, the canted spectral transform (CST) is defined by using a quadratic phase kernel. The statistical properties of the CST are derived and compared with those of the conventional SFT. In general, the use of shaded windows in the CST does not appear to be advantageous and can degrade the selectivity of the transform in estimating the signal frequency dynamics. Statistical distributions of the peak cant variable for an idealized signal in Gaussian noise provide a basis for determining the performance of the CST in practical applications.
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43.60.Gk Space-time signal processing, other than matched field processing

A new autoregressive method for high‐performance spectrum analysis

An‐Chen Lee

J. Acoust. Soc. Am. Volume 86, Issue 1, pp. 150-157 (1989); (8 pages)

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In this paper, a new method of autoregressive (AR) spectrum estimation is presented. It shall be called two‐sided autoregressive spectrum estimation, because an interpolative or smoothing model is postulated, as opposed to the predictive (one‐sided) model used in AR modeling. The matrix equations arising in the estimation procedures proposed in this paper exhibit a special structure. The exploitation of these structures leads to fast solutions that reduce the total number of computations by an order of magnitude compared with straightforward approaches. Also, special attention is directed to the constrained two‐sided AR model. Simulation examples show higher resolution capability of the proposed method relative to the least‐squares AR method.
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43.60.Gk Space-time signal processing, other than matched field processing
43.58.Kr Spectrum and frequency analyzers and filters; acoustical and electrical oscillographs; photoacoustic spectrometers; acoustical delay lines and resonators

Extended towed array processing by an overlap correlator

Stergios Stergiopoulos and Edmund J. Sullivan

J. Acoust. Soc. Am. Volume 86, Issue 1, pp. 158-171 (1989); (14 pages) | Cited 4 times

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A method of extending towed array measurements that provides an aperture greater than that of the physical array is presented. Such a technique can be used by matched‐field estimators to obtain information about the range and depth of a source and in other towed array applications requiring a very large aperture. The approach is to combine coherently the acoustic signals arriving at a moving array of hydrophones by making proper compensation through a factor that corrects for considerable fluctuations in phase irregularities in the tow path of the physical array as well as fluctuations in amplitude experienced during the coherent integration time. In this manner, the finite aperture of the physical array is exploited in a process that synthesizes the extended aperture of the method. The concept is based on an algorithm that we call an ‘‘overlap correlator,’’ which provides the phase correction factor by correlating overlapping space samples of the acoustic signal received at successive moments by the moving towed array. This is in contrast to the standard, passive synthetic aperture technique, which requires either highly accurate a priori knowledge of the source frequency or a maneuver in order to obtain a wavenumber or bearing estimate. The algorithm has been tested on numerical data generated by the SACLANT Undersea Research Centre’s normal mode model SNAP. The effects of space and time coherence of the signal and the random and systematic errors on the extended towed array measurements are examined and used to derive guidelines for experimental applications of this algorithm.
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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

Simulation of the transient and steady‐state sound propagation in rooms using a new combined ray‐tracing/image‐source algorithm

Michael Vorländer

J. Acoust. Soc. Am. Volume 86, Issue 1, pp. 172-178 (1989); (7 pages) | Cited 16 times

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A new method for the calculation of room acoustical impulse responses is described, which is based on two well‐known computer algorithms, the ray‐tracing and the image‐source models. With the new method, the procedure of sieving the ‘‘visible’’ image sources out of the enormous quantity of possible sources is carried out by examination of the histories of sound particles. From the obtained list of visible image sources, the impulse response of the enclosure is easily constructed. The new method combines the advantages of the ray‐tracing process, namely, the relatively slow increase of computation time with the length of the impulse response, with the accuracy inherent to the image‐source model, which is even sufficient to calculate the Fourier transform, i.e., the steady‐state transmission function of the room, or to convolve the impulse response with sound signals.
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43.55.Ka Computer simulation of acoustics in enclosures, modeling
43.55.Fw Auditorium and enclosure design

The insertion loss of finite length barriers on the ground

André L’Espérance

J. Acoust. Soc. Am. Volume 86, Issue 1, pp. 179-183 (1989); (5 pages) | Cited 4 times

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This paper presents a method for estimating the insertion loss (IL) of a finite length barrier on the ground. This method is an extension of the solution suggested by Jonasson [H. G. Jonasson, J. Sound. Vib. 22 (1), 113–126 (1972)] for estimating the sound reduction by an infinite length barrier on the ground. It uses diffraction theory, combined with a model for sound propagation over the ground, to calculate the diffracted field created by the diffracted path associated with each edge of the barrier. The diffracted paths are identified by simple geometrical considerations. The results obtained using this new method are compared with other theoretical methods and with the results of experimental measurements. It is shown that the new solution is as accurate as other more complex solutions.
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43.50.Gf Noise control at source: redesign, application of absorptive materials and reactive elements, mufflers, noise silencers, noise barriers, and attenuators, etc.

Active noise control in ducts: Some physical insights

Scott D. Snyder and Colin H. Hansen

J. Acoust. Soc. Am. Volume 86, Issue 1, pp. 184-194 (1989); (11 pages) | Cited 2 times

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The mechanisms of active noise control in a duct are examined. Acoustical measurements are used to determine directly the acoustic power flow associated with both primary and secondary sources as a function of secondary to primary source strength ratio and volume velocity relative phase angles. A complete analytical model is also developed which allows calculation of individual source power flows and total downstream power flow as a function of source strengths and relative phase angles for finite size sources. It is evaluated for monopole and dual secondary source arrangements, but can be extended easily to any number of secondary sources. The model considers a finite size primary source in the plane of the duct cross section and evaluates the effect that the secondary sources have on the primary source power output. Measurements of individual source output powers and total downstream acoustic powers agree well with theoretical predictions. It is demonstrated that, for the monopole system, sound attenuation is achieved primarily by suppression of the primary source acoustic power output, with a little remaining power being absorbed by the secondary source. For the dual secondary source system, it is shown that the power is primarily absorbed by the secondary sources, but that, at phase and amplitude values slightly different to optimum, noise reduction is achieved by a combination of energy absorption and primary source power suppression. The analysis also demonstrates the dependence of the achievable noise reduction on secondary source size and location with respect to the primary source.
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43.50.Ki Active noise control

Acoustic radiation from an insonified elastic plate with a line discontinuity

Cetin Seren and Sabih I. Hayek

J. Acoust. Soc. Am. Volume 86, Issue 1, pp. 195-209 (1989); (15 pages) | Cited 5 times

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This paper deals with the development of analytic models for the prediction of the acoustic radiated field from an infinite elastic plate with a single line force and line moment impedance discontinuity due to an incident plane acoustic wave. The solution is in the form of a Fourier integral with a kernel having ten poles. The integral is evaluated by three methods. The first uses the steepest descent path (SDP) method, leading to a solution that decays as 1/(k0r)1/2. The second method used conformal transformations and the modified saddle point (MSP) method, where all ten poles of the integrand are factored out. This second method yields a solution that has complementary error functions and an asymptotic series in (k0r). The third method employs a transformation of the integrand to effect an efficient and fastly convergent numerical integration algorithm. In general, the MSP asymptotic series solution and the numerical integration yielded numerically identical results. However, while the SDP solution predicted a similar directivity function, it predicted numerically higher values than the MSP solution by as much as 20 dB for observers located close to the discontinuity. The three solutions converged for higher values of k0r, the convergence being slower for higher frequencies, especially above the coincidence frequency.
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43.40.Dx Vibrations of membranes and plates
43.20.Rz Steady-state radiation from sources, impedance, radiation patterns, boundary element methods

Impedance characteristics of transducers and reciprocity calibration

Li‐Feng Ge

J. Acoust. Soc. Am. Volume 86, Issue 1, pp. 210-214 (1989); (5 pages)

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This paper investigates impedance characteristics of electrodynamic‐type and piezoelectric‐type transducers. The inherent relation between mechanical impedance of a mass load and electrical impedance of the transducing element of transducers is revealed. The substance of reciprocity calibration is just to utilize the impedance inversive property of both types of transducers so that one can determine the mechanical quantities (force and velocity) by measuring electrical quantities, which can be measured conveniently and accurately. According to the theory, for high‐frequency calibration, mass values in the reciprocity calibration formulas should be replaced by their equivalent masses rather than static mass values, because the mechanical impedances of mass loads will be remarkably changed due to the stress wave effect. Making the correction can improve calibration accuracy by about one‐half order of magnitude.
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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
43.58.Vb Calibration of acoustical devices and systems

A theoretical study of cavitation generated by an extracorporeal shock wave lithotripter

Charles C. Church

J. Acoust. Soc. Am. Volume 86, Issue 1, pp. 215-227 (1989); (13 pages) | Cited 43 times

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The intense acoustic wave generated at the focus of an extracorporeal shock wave lithotripter is modeled as the impulse response of a parallel RLC circuit. The shock wave consists of a zero rise time positive spike that falls to 0 at 1 μs followed by a negative pressure component 6 μs long with amplitudes scaled to +1000 and −160 bars, P+ and P−, respectively. This pressure wave drives the Gilmore–Akulichev formulation for bubble dynamics; the zero‐order effect of gas diffusion on bubble response is included. The negative pressure component of a 1000‐bar shock wave will cause a preexisting bubble in the 1‐ to 10‐μm range to expand to over 100 times its initial size, R0, for 250 μs, with a peak radius of ∼1400 μm, then collapse very violently, emitting far UV or soft x‐ray photons (black body). Gas diffusion does not appreciably mitigate the amplitude of the pressure wave radiated at the primary collapse, but does significantly reduce the collapse temperature. Diffusion also increases the bubble radius from R0 up to 40 μm and extends the duration of ringing following the primary collapse, assuming that the bubble does not break up or shed microbubbles. Results are sensitive to P+/P− and to the duration of the negative pressure cycle but not to rise time.
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43.35.Wa Biological effects of ultrasound, ultrasonic tomography
43.80.Sh Medical use of ultrasonics for tissue modification (permanent and temporary)

An optimal PE‐type wave equation

David H. Berman, Evan B. Wright, and Ralph N. Baer

J. Acoust. Soc. Am. Volume 86, Issue 1, pp. 228-233 (1989); (6 pages) | Cited 2 times

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A one‐way wave equation is presented with the following properties. (1) For low angles and small sound‐speed variations, it reduces to the standard parabolic approximation. (2) It allows a split‐step solution. (3) The rays associated with this equation are exactly the rays of the Helmholtz equation in a range‐independent environment. It is in the last sense an optimal one‐way wave equation. Results of the split‐step solution of this equation are presented and compared to normal‐mode calculations and results of another modification of the standard parabolic equation, which was given by Thomson and Chapman [D. J. Thomson and N. R. Chapman, J. Acoust. Soc. Am. 74, 1848–1854 (1983)].
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43.30.Bp Normal mode propagation of sound in water
43.30.Cq Ray propagation of sound in water

An exact ray theoretical formulation of the Helmholtz equation

Terry L. Foreman

J. Acoust. Soc. Am. Volume 86, Issue 1, pp. 234-246 (1989); (13 pages) | Cited 5 times

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Practical computational procedures for obtaining ray theoretical solutions to the inhomogeneous Helmhotz equation ∇2Ψ+k2Ψ=S(r,ω) resort to a well‐known approximation. A computational method is presented that enables one to trace rays without resorting to the ray theory approximation, provided a solution to the Helmholtz equation is available by independent means. In other words, given a solution to the Helmholtz equation, the exact rays for that case can be computed. This ray theory therefore serves, not as a computational method, but as a new method of displaying solutions to the Helmholtz equation. Exact ray diagrams are constructed for several cases using this technique. The resulting ray diagrams usually bear little resemblance to the corresponding classical ray diagrams. It is shown that the discrepancy is attributable to the nature of the classical ray theory approximation, which proves in most cases not to be a small perturbation. Some of the properties of the exact rays that distinguish them from their classical counterparts are: (1) The ray trajectories depend on the source frequency and configuration and on the boundaries; (2) the exact rays intrude into shadow zones impenetrable by classical rays; (3) the field is finite at caustics; and (4) the exact rays never exhibit multipathing, which is the hallmark of classical ray diagrams. The contrasts between classical and exact ray theory are demonstrated and explained.
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43.30.Bp Normal mode propagation of sound in water
43.30.Cq Ray propagation of sound in water
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