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Journal of the Acoustical Society of America

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Sep 1979

Volume 66, Issue 3, pp. 629-943

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An algorithm for the design of transformerless broadband equalizers of ultrasonic transducers

L. J. Augustine and J. Andersen

J. Acoust. Soc. Am. Volume 66, Issue 3, pp. 629-635 (1979); (7 pages)

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A new design strategy for broadbanding ultrasonic transducers was recently presented, which generates a lossless equalizer, matching a resistive source to a piezoelectric transducer with power gain displaying a maximally flat (Butterworth) characteristic. However, three undesirable restrictions are part of the existing algorithm; (i) the center frequency of the design is fixed, limiting its usage to certain passbands, (ii) the design requires the use of an ideal transformer for all source resistance and tranducer combinations but one, and (iii) the resulting element values are often impractical. The purpose of this paper is to present a generalization of the algorithm which broadens its applicability. The main result is an algorithm for the design of a lossless transformerless broadband equalizer with maximally flat power gain, and a delineation of the exact latitude on passbands and source resistances.
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43.38.Ar Transducing principles, materials, and structures: general
43.38.Fx Piezoelectric and ferroelectric transducers

A surface acoustic wave sonobuoy

P. Das and C. Lanzl

J. Acoust. Soc. Am. Volume 66, Issue 3, pp. 636-640 (1979); (5 pages) | Cited 2 times

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A transmitting hydrophone which remotely senses both absolute pressure and changes in pressure has been developed using a surface acoustic wave (SAW) oscillator. This oscillator is formed by placing a SAW delay line or resonator in the feedback loop of an amplifier. The pressure applied to the hydrophone is converted to a flexural loading of the piezoelectric plate which alters the surface wave velocity on the plate, consequently shifting the oscillator frequency. This frequency modulated signal is easily transmitted to a remote location for signal recovery. The piezoelectric plate thickness and length can be chosen to optimize the sensitivity of the SAW hydrophone. The surface wave element of the oscillator is a planar structure, therefore, it is simple and inexpensive to manufacture. Furthermore, the size and power requirements of the SAW hydrophone are small, typically less than 50 cm3 and 300 mW. This paper presents performance data of two SAW hydrophones using SAW oscillators at 44 and 77 MHz in conjunction with a versatile signal demodulation system. Measurements of sensitivity and frequency response were made for both delay line and resonator hydrophones manufactured on lithium niobate and S‐T quartz substrates, respectively, and compared with measurements obtained from conventional hydrophones in the identical set‐up.
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43.38.Fx Piezoelectric and ferroelectric transducers
43.30.Yj Transducers and transducer arrays for underwater sound; transducer calibration
43.35.Pt Surface waves in solids and liquids

Distribution of sonic plesio‐velocity in a compact bone sample

Sidney Lees, Paul F. Cleary, John D. Heeley, and Ernest L. Gariepy

J. Acoust. Soc. Am. Volume 66, Issue 3, pp. 641-646 (1979); (6 pages) | Cited 2 times

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Since compact bone is demonstrably anisotropic and inhomogeneous, the measured quantities yield a plesio‐velocity rather than a true sonic velocity, which makes it difficult to compare velocity measurements among several bone samples. Distribution of the plesio‐velocity for 11 pairs of 2 mm thick specimens from a single mature bovine compact bone sample using 100‐ns well‐damped sonic pulses emitted by a 10‐mHz transducer show that anisotropy exists in both axial and transverse directions and must be associated with the local structure and composition. Changes in the same specimens when wet, dry, and rehydrated show an increase in velocity and anisotropy of the dry over the wet state. The plesio‐velocity is greater in the axial direction, along the bone axis, than in the transverse plane. The change in plesio‐velocity from wet to dry bone is greatest along the bone axis and least in the transverse plane. Changes in dimension, when the bone specimens were dried, show the bone sample to be anisotropic in this parameter also, but in the reverse order from that for the plesio‐velocity. Shrinkage is least along the osteons and a maximum in the transverse plane.
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43.80.Cs Acoustical characteristics of biological media: molecular species, cellular level tissues
43.35.Cg Ultrasonic velocity, dispersion, scattering, diffraction, and attenuation in solids; elastic constants

Harmonic properties of the annular membrane

H. P. W. Gottlieb

J. Acoust. Soc. Am. Volume 66, Issue 3, pp. 647-650 (1979); (4 pages)

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Properties of a vibrating annular membrane are analyzed mathematically and numerically to investigate the relative harmonicity of the higher‐order modes compared with the fundamental mode. The harmonicity depends on the zeros of a cross‐product of Bessel functions. For a ratio of external‐to‐internal diameter up to about 1.2, the higher modes have frequencies which are very nearly exact integer multiples of the fundamental frequency. For circularly symmetric modes, the ratio of all higher radial mode frequencies to exact harmonic values is less than 1.0006 (a logarithmic cent), a ratio imperceptible to a listener. Angle‐dependent modes also give good harmonicity, especially if some lower radial modes are suppressed. Thus the modes of an annular membrane whose external‐to‐internal diameter ratio is less than 1.2 are much more harmonic than those of the full circular membrane of a conventional drum. The theoretical harmonicity is at least as good as reported for stringed melodic instruments (piano, violin). The analysis suggests characteristics for construction of an annular drum‐head which is supported on concentric rims of two cylinders; to sound the drum, a braced hoop might be used to excite radial modes, or strikers with grooved faces might be used to excite angular modes without exciting lower radial modes.
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43.75.Hi Drums
43.40.At Experimental and theoretical studies of vibrating systems
43.20.Ks Standing waves, resonance, normal modes

Performance of children aged 9 to 17 years on a test of speech intelligibility in noise using sentence material with controlled word predictability

Lois L. Elliott

J. Acoust. Soc. Am. Volume 66, Issue 3, pp. 651-653 (1979); (3 pages) | Cited 16 times

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Performance of children aged 9 to 17 years on the SPIN test (Speech Perception in Noise) is described. The 11‐ and 13‐year‐olds performed significantly poorer than 15‐ and 17‐year‐olds, and this difference occurred primarily for high‐predictability sentences presented at a 0‐dB signal‐to‐babble ratio. Performance of nine‐year‐olds was significantly poorer than performance of 11‐year‐olds. Possible reasons for these differences are discussed.
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43.70.Dn Disordered speech
43.71.Gv Measures of speech perception (intelligibility and quality)

Relation between voice‐onset time and vowel duration

Robert F. Port and Rosemarie Rotunno

J. Acoust. Soc. Am. Volume 66, Issue 3, pp. 654-662 (1979); (9 pages) | Cited 3 times

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As part of an investigation of the temporal implementation rules of English, measurements were made of voice‐onset time for initial English stops and the duration of the following voiced vowel in monosyllabic words for New York City speakers. It was found that the VOT of a word‐initial consonant was longer before a voiceless final cluster than before a single nasal, and longer before tense vowels than lax vowels. The vowels were also longer in environments where VOT was longer, but VOT did not maintain a constant ratio with the vowel duration, even for a single place of articulation. VOT was changed by a smaller proportion than the following voiced vowel in both cases. VOT changes associated with the vowel were consistent across place of articulation of the stop. In the final experiment, when vowel tensity and final consonant effects were combined, it was found that the proportion of vowel duration change that carried over to the preceding VOT is different for the two phonetic changes. These results imply that temporal implementation rules simultaneously influence several acoustic intervals including both VOT and the ’’inherent’’ interval corresponding to a segment, either by independent control of the relevant articulatory variables or by some unknown common mechanism.
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43.72.Ar Speech analysis and analysis techniques; parametric representation of speech
43.70.Bk Models and theories of speech production
43.70.Fq Acoustical correlates of phonetic segments and suprasegmental properties: stress, timing, and intonation

Considerations in applying clustering techniques to speaker‐independent word recognition

L. R. Rabiner and J. G. Wilpon

J. Acoust. Soc. Am. Volume 66, Issue 3, pp. 663-673 (1979); (11 pages)

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Recent work at Bell Laboratories has demonstrated the utility of applying sophisticated pattern recognition techniques to obtain a set of speaker‐independent word templates for an isolated word recognition system [Levinson et al., IEEE Trans. Acoust. Speech Signal Process. ASSP‐27 (2), 134–141 (1979); Rabiner et al., IEEE Trans. Acoust. Speech Signal Process.(in press)]. In these studies, it was shown that a careful experimenter could guide the clustering algorithms to choose a small set of templates that were representative of a large number of replications for each word in the vocabulary. Subsequent word recognition tests verified that the templates chosen were indeed representative of a fairly large population of talkers. Given the success of this approach, the next important step is to investigate fully automatic techniques for clustering multiple versions of a single word into a set of speaker‐independent word templates. Two such techniques are described in this paper. The first method uses distance data (between replications of a word) to segment the population into stable clusters. The word template is obtained as either the cluster minimax, or as an averaged version of all the elements in the cluster. The second method is a variation of the one described by Rabiner [IEEE Trans. Acoust. Speech Signal Process. ASSP‐26 (3), 34–42 (1978)] in which averaging techniques are directly combined with the nearest neighbor rule to simultaneously define both the word template (i.e., the cluster center) and the elements in the cluster. Experimental data show the first method to be superior to the second method when three or more clusters per word are used in the recognition task.
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43.72.Fx Talker identification and adaptation algorithms

Masking of filtered noise bursts by synthetic vowels

Steffi B. Resnick, Michael S. Weiss, and John M. Heinz

J. Acoust. Soc. Am. Volume 66, Issue 3, pp. 674-677 (1979); (4 pages)

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The present investigation assessed the simultaneous and temporal masking produced by computer‐generated synthetic vowels. The durations (100 and 200 ms) of each of four vowel‐like maskers were employed. The masker was presented at 70 dB SPL. The probe signals were three filtered noise bursts whose spectral distributions corresponded to regions of high spectral energy in three English stop consonants. Quiet and masked thresholds were determined using the method of adjustment. Data are reported for two experienced listeners who participated in all the listening conditions. The results were generally in accord with the results of masking experiments using nonspeech signals in that both the frequency specificity of masking and temporal masking effects were demonstrated.
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43.70.Dn Disordered speech
43.66.Dc Masking

Predicting vocal frequency from selected physiologic measures

Thomas Shipp, E. Thomas Doherty, and Philip Morrissey

J. Acoust. Soc. Am. Volume 66, Issue 3, pp. 678-684 (1979); (7 pages)

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Simultaneous physiologic measures were obtained on four young adult male subjects as they sustained phonation at seven frequencies within their modal‐to‐falsetto voice range. Data were analyzed using a multiple regression program to determine the contribution of each measure singly and in combination with other measures to the prediction of the resultant voice frequency. Results showed that by far the best predictor was cricothyroid muscle activity for both the pooled data model, and for each individual subject. The contribution of subglottal air pressure and thyroarytenoid muscle activity increased the variance explained by only 4% while the measure of vertical laryngeal position was a significant factor in only one subject’s predictive model. Partial models from the pooled data explained from 67% to 73% of the variance; whereas the obtained measures for individual subjects explained from 90% to 94% of the variance.
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43.70.Bk Models and theories of speech production
43.66.Hg Pitch

Speech synthesis from concept: A method for speech output from information systems

S. J. Young and F. Fallside

J. Acoust. Soc. Am. Volume 66, Issue 3, pp. 685-695 (1979); (11 pages)

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A synthesis method, called speech synthesis from concept, is described which has been designed specifically for providing speech output from information systems. It differs from conventional techniques in that data is passed from the information system to the speech synthesis system, not in the form of text or phonetic transcription, but in the form of an abstract structure called an input concept. The speech synthesis from concept system converts an input concept into speech by using a transformational grammar to generate a well‐formed English sentence and a word concatenation synthesizer to generate the actual speech output. The ’’top down’’ nature of this process reduces the computation required within the information system and enables high‐quality speech to be produced.
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43.70.Jt Instrumentation and methodology for speech production research
43.72.Gy Narrow, medium, and wideband speech coding

Exploring azimuth effects with an anthropometric manikin

Donald D. Dirks and Samuel Gilman

J. Acoust. Soc. Am. Volume 66, Issue 3, pp. 696-701 (1979); (6 pages)

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In these experiments, the effects of sound direction on the eardrum response of an anthropometric manikin (the KEMAR manikin) were investigated. Pure tones and pink noise (analyzed in 1/3‐octave bandwidths) over a wide frequency range were used as signals as the manikin rotated 360° with respect to a point source in a anechoic chamber. The simulated eardrum SPL was compared with the averaged human field‐to‐eardrum data reported by Shaw [J. Acoust. Soc. Am. 56, 1848–1861 (1974)]. It was concluded that the KEMAR manikin can be used up to frequencies of ?8.0 kHz, with (1) 1/3‐octave pink noise signals to measure a response equivalent to that obtained by averaging over a number of humans, and (2) pure‐tone signals to measure the response equivalent to that of a single human having average head and ear dimensions.
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43.66.Ts Auditory prostheses, hearing aids
43.64.Ha Acoustical properties of the outer ear; middle-ear mechanics and reflex
43.64.Yp Instruments and methods
43.66.Qp Localization of sound sources

Comparison of DT 48, TDH 49, and TDH 39 earphones

Hugo Fastl

J. Acoust. Soc. Am. Volume 66, Issue 3, pp. 702-703 (1979); (2 pages) | Cited 1 time

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The frequency responses of the earphones TDH 49, TDH 39, and DT 48 (the latter with and without free‐field equalizer) were compared by means of measuring hearing threshold curves. While the TDH 39 and TDH 49 give almost identical results (average deviations 2 dB), the TDH 49 and DT 48 show moderate (5 dB) differences. In the most usual application mode, i.e., TDH 49 without and DT 48 with free‐field equalizer, distinctly different threshold curves were found, indicating rather different real‐ear frequency resonses. Accordingly, using the latter two earphone systems, psychoacoustic experiments with broadband stimuli may lead to somewhat different data.
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43.66.Yw Instruments and methods related to hearing and its measurement
43.66.Cb Loudness, absolute threshold
43.66.Sr Deafness, audiometry, aging effects
43.38.Si Telephones, earphones, sound power telephones, and intercommunication systems

Evidence for direction‐specific channels in the processing of frequency modulation

R. B. Gardner and J. P. Wilson

J. Acoust. Soc. Am. Volume 66, Issue 3, pp. 704-709 (1979); (6 pages) | Cited 12 times

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Evidence is provided for the existence of at least three feature‐specific channels in the auditory system. Thresholds for the detection of small repetitive or nonrepetitive frequency changes were measured following various adapting stimuli using a 2IFC procedure in two subjects at 1 kHz. Thresholds for single linear upward frequency sweeps (up sweeps) were increased by a factor of 2 to 3 following exposure to repetitive (8 Hz) up sweeps but not following exposure to down sweeps or tone bursts; correspondingly, thresholds for down‐sweep stimuli were increased only by down sweeps. Sinusoidal FM test stimulus thresholds were elevated by both up‐sweeps and down‐sweeps and to a lesser extent by tone bursts. These results suggest the existence in the auditory system of channels specific to upward FM, downward FM, and probably repetition rate.
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43.66.Mk Temporal and sequential aspects of hearing; auditory grouping in relation to music
43.66.Fe Discrimination: intensity and frequency
43.66.Ba Models and theories of auditory processes

Predicting musical pitch from component frequency ratios

Martin Piszczalski and Bernard A. Galler

J. Acoust. Soc. Am. Volume 66, Issue 3, pp. 710-720 (1979); (11 pages) | Cited 1 time

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A computer algorithm has been developed for music applications that automatically estimates pitch from any small number of frequencies, not necessarily of equal amplitude. Low‐integer ratios are sought for each pair of frequencies. If an actual frequency‐pair ratio closely approximates a low‐integer ratio, the matching integers of this latter ’’ideal harmonic ratio’’ are accepted as possible harmonic numbers of the two actual frequencies in the tone. Hence, by comparing components to each other we can infer their likely harmonic numbers; if this is done successfully, then estimating the fundamental becomes a straightforward task. Therefore, the efficacy of the method lies in the cumulative effect of several component pair evaluations, all substantiating consistent harmonic assignments to the participating components. The method has proven to be useful in extracting pitch from spectral peaks taken from natural musical sounds, and it closely approximates some results of earlier experimental measurements of pitch perception. The method operates well with no fundamental present and for nonsuccessive partials (as well as inharmonic partials), and it is ’’phase insensitive.’’ Prior knowledge regarding the source of the signal (e.g., the musical instrument being played) is not required.
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43.66.Ba Models and theories of auditory processes
43.66.Fe Discrimination: intensity and frequency
43.66.Hg Pitch
43.75.-z Music and musical instruments

Central denervation hypersensitivity in the auditory system of the cat

George M. Gerken

J. Acoust. Soc. Am. Volume 66, Issue 3, pp. 721-727 (1979); (7 pages)

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Data are reported for seven cats with a total of 29 electrodes permanently placed in or near the cochlear nucleus, the superior olivary complex, the nucleus of the inferior colliculus, and the medial geniculate body. Detection thresholds for pulsate electrical stimuli were measured using an operant behavioral procedure. Electrical stimulation thresholds were measured prior to and following bilateral destruction of the cochleas in all animals. In addition, four of the animals were tested using a site‐of‐stimulation discrimination prior to and following the cochlear lesion. Finally, hearing loss was evaluated in all cats after the completion of the experiments. Electrical stimulation thresholds showed a mean reduction of 7.9 dB throughout the brain stem auditory system after cochlear destruction. The ability of the animals to perform the site‐of‐stimulation discrimination was not permanently impaired by the cochlear lesion. The data indicated the presence of increased sensitivity to electrical stimulation in most regions of the subcortical auditory system, although a lesser effect was found at the thalamic level. It was concluded that stimulation threshold provides an index relevant to the state of auditory neurons proximal to the electrode tip.
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43.64.Qh Electrophysiology of the auditory central nervous system
43.64.Tk Physiology of sound generation and detection by animals
43.66.Mk Temporal and sequential aspects of hearing; auditory grouping in relation to music

Temporal summation of pulsate brain stimulation in normal and deafened cats

George M. Gerken

J. Acoust. Soc. Am. Volume 66, Issue 3, pp. 728-734 (1979); (7 pages) | Cited 2 times

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Behaviorally measured, electrical‐stimulation thresholds were obtained from 11 electrodes permanently positioned in the auditory system and other brain loci. Number of pulses and interpulse intervals were varied to determine how detection thresholds were affected by stimulation parameters. Detection thresholds generally decreased with increased number of pulses and with shorter interpulse intervals. A method is presented to describe the parametric threshold data for each electrode in terms of three constants: a single‐pulse threshold which characterizes the sensitivity of the placement; a time constant of temporal summation; and a compression factor which describes the range of threshold variation. For three placements in the vicinity of cochlear nucleus, bilateral cochlear destruction permanently altered parametric thresholds. In particular, single‐pulse threshold was lowered by 9.2 dB; time constant of temporal summation was reduced by a factor of 100; and the compression factor was increased. Classic strength‐duration time constants were determined using behavioral methods and were shown to be equal in magnitude to the greatly reduced time constants for temporal summation in the deafened animals. This implies that capacity for temporal integration may be substantially reduced or lost in at least the lower level of the auditory system following deafness.
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43.64.Qh Electrophysiology of the auditory central nervous system
43.66.Mk Temporal and sequential aspects of hearing; auditory grouping in relation to music
43.66.Sr Deafness, audiometry, aging effects

The effects of aging on the stapedius reflex thresholds

Shlomo Silman

J. Acoust. Soc. Am. Volume 66, Issue 3, pp. 735-738 (1979); (4 pages) | Cited 1 time

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The acoustic reflex thresholds for broadband noise and 500‐, 1000‐, and 2000‐Hz activating signals were measured in a group of young normal hearing adults and a group of elderly normal hearing subjects. The results indicated that the acoustic reflex thresholds for tonal activating signals in the young subjects were similar to those in the elderly subjects. However, the acoustic stapedius reflex thresholds for broadband noise activating signals is significantly higher in the elderly subjects than in the young. These differences were explained in light of Bredburg’s findings [Acta Otolaryngol. Suppl. 236, 1–135 (1968)] regarding degeneration of outer hair cells as a function of aging.
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43.64.Ha Acoustical properties of the outer ear; middle-ear mechanics and reflex
43.66.Sr Deafness, audiometry, aging effects

On the application of modern control theory to improving the fidelity of an underwater projector

Claude J. Mazzola, John D. Birdwell, and Michael Athans

J. Acoust. Soc. Am. Volume 66, Issue 3, pp. 739-750 (1979); (12 pages)

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The paper presents a technique for controlling the radiated sound pressure from an electroceramic projector in water. This technique involves the implementation of a state variable feedback tracking algorithm. The state variables of the system are estimated with a Kalman–Bucy filter from the measured radiated pressure. The state variables are then fed back to effect an appropriate compensation of the system, which results in a least‐square minimization of the difference between the measured and desired pressure signatures. The algorithm has been tested with an actual projector. The results show that the measured pressure tracks the desired pressure reasonably well.
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43.60.Qv Signal processing instrumentation, integrated systems, smart transducers, devices and architectures, displays and interfaces for acoustic systems
43.30.Yj Transducers and transducer arrays for underwater sound; transducer calibration
43.38.Fx Piezoelectric and ferroelectric transducers

First‐order statistics for finite bandwidth multipath signals with and without frequency or phase modulation

Peter N. Mikhalevsky

J. Acoust. Soc. Am. Volume 66, Issue 3, pp. 751-762 (1979); (12 pages)

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The phase‐random model of multipath acoustic propagation is used to derive the first‐order probability densities for the time rate‐of‐change of the short time average mean‐square pressure ?, and the time rate‐of‐change of the level in decibels of the short time average mean‐square pressure ?. It is shown that the probability densities for the signal amplitude, and amplitude rate are insensitive to frequency or phase modulation of the signal by the source, but that the density for the time rate‐of‐change of the multipath phase ?, is sensitive to such modulation. Because the finite bandwidths of acoustic signals can be modeled by uniform frequency modulation, the analysis presented applies to this problem as well. It is shown that bandwidth effects can be neglected only if B≪2ν, where B is the signal bandwidth, and ν2 is the single path mean‐square phase rate. This inequality provides a useful definition of what is meant by ’’narrow band’’ as it applies to phase‐random multipath propagation. It is also shown that the density for ? depends only on ν, while the density for ? depends on ν and parameters of the modulation. A potentially powerful technique is developed for determining source bandwidth and parameters of the source modulation, minus the effects of oceanic fluctuations, from the received multipath signal. The analytical results are compared with a computer simulation, and data from experiments in the ocean with extremely favorable results.
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43.60.Cg Statistical properties of signals and noise
43.30.Bp Normal mode propagation of sound in water
43.20.Bi Mathematical theory of wave propagation

Development of temporal sampling strategies for monitoring noise

R. E. DeVor, P. D. Schomer, W. A. Kline, and R. D. Neathamer

J. Acoust. Soc. Am. Volume 66, Issue 3, pp. 763-771 (1979); (9 pages) | Cited 1 time

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This paper addresses the problem of the estimation of the long‐term (yearly) mean of the Community Noise Equivalent Level (CNEL) or day/night average sound level (LDN). Recent environmental noise standards have emphasized the significance of this problem. While it is possible to continually monitor the noise level, it is not necessarily desirable or practical. It is desirable to sample the level over a relatively short period of time and use this information to draw reliable inferences about the long term mean level. Examination of daily average noise levels (either in mean square pressure or in decibel units) shows that while the data may be stationary with respect to mean level over a several month period, they exhibit a strong pattern of autocorrelation in which positive correlation predominates. As a result, the sample sizes required to achieve a desired level of precision in the sample mean estimate are much larger than they would otherwise be if the data were uncorrelated serially in time. To assess the level of autocorrelation in the data, autoregressive‐moving average (ARMA) models are developed for the noise data via the Dynamic Data System (DDS) approach to time series analysis. These models are then used to derive estimates of the sample mean variance and therefore to establish sampling strategies. For the data examined, to obtain an estimate of the mean level within a 5‐dB range (±50% of the mean in mean square pressure units), sample sizes in the range of 20–50 consecutive daily averages would be required. If the daily averages were uncorrelated in time, only 5–15 consecutive daily averages would be required. The data used in this study were obtain from continuous monitoring at a number of sites in the vicinity of a busy Naval Air Station. Some data obtained from a large commercial airport were also analyzed and found to have even stronger positive autocorrelation, and therefore requiring even larger sample sizes for mean value estimation.
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43.50.Sr Community noise, noise zoning, by-laws, and legislation
43.50.Qp Effects of noise on man and society
43.50.Lj Transportation noise sources: air, road, rail, and marine vehicles

A method for modeling perforated tube muffler components. I. Theory

Joseph W. Sullivan

J. Acoust. Soc. Am. Volume 66, Issue 3, pp. 772-778 (1979); (7 pages) | Cited 16 times

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A simple method is presented for modeling perforated muffler components such as concentric resonators with perforated flow tube, and expansion chambers and reverse flow chambers with perforated inlet and outlet tubes. The theory includes mean flow, but is confined to those configurations having one acoustically long dimension. It is based on a segmentation procedure in which each segment is described by a transmission matrix. The four‐pole parameters of a component are then found from the product of the transmission matrices. The four‐pole parameters for configurations having through flow, cross flow, and reverse flow are presented. Because the product matrices are dimensionally small and because no inversion is needed, computational time is much lower than other methods such as finite element or finite difference. This allows rapid and economical modeling to be performed where iterative solutions are required because of dominating finite amplitude effects, for example.
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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.
43.20.Ks Standing waves, resonance, normal modes
43.58.Kr Spectrum and frequency analyzers and filters; acoustical and electrical oscillographs; photoacoustic spectrometers; acoustical delay lines and resonators
43.20.Mv Waveguides, wave propagation in tubes and ducts

A method for modeling perforated tube muffler components. II. Applications

Joseph W. Sullivan

J. Acoust. Soc. Am. Volume 66, Issue 3, pp. 779-788 (1979); (10 pages) | Cited 9 times

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This paper is the second of two papers describing a method for modeling muffler components having perforated flow tubes, as for example those found frequently in automotive applications. In the first paper the theoretical model was developed, whereas here the theory is applied to demostrate specifically the utility and potential of the model. The applications are idealized in the sense that effects which occur simultaneously in perforated‐tube mufflers are treated separately. These effects are (1) the nonlinear impedance of the perforation due to finite amplitude sound pressure, and (2) the change in impedance of the perforation with mean flow. The first effect is demonstrated with a straight through resonator, and the second with a cross‐flow chamber. Predicted results for transmission loss in both cases compared quite well with measurements. Crucial to the success of the modeling was a proper description of the perforate impedance. The confidence instilled by the results prompted an inquiry into the nature of the two muffler components. It was determined by modeling that, contrary to popular opinion, these devices can be very dissipative, even though they contain no recognizable dissipative materials. The controlling mechanism is the high resistivity of the perforation induced by the acoustic/flow environment.
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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.
43.20.Ks Standing waves, resonance, normal modes
43.20.Mv Waveguides, wave propagation in tubes and ducts
43.58.Kr Spectrum and frequency analyzers and filters; acoustical and electrical oscillographs; photoacoustic spectrometers; acoustical delay lines and resonators

Total reflection of SH waves in the presence of slip and friction

Maria Comninou, J. Dundurs, and Eugene L. Chez

J. Acoust. Soc. Am. Volume 66, Issue 3, pp. 789-793 (1979); (5 pages) | Cited 2 times

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The paper examines the total reflection of SH waves by a contact interface that can locally slip when the incident wave is strong enough to break friction between the two solids. The solution is constructed by correcting the fields for the welded interface and takes advantage of known results for moving dislocations. The formulation leads to singular integral equations of the Cauchy type which must be treated numerically. Because of the number of independent variables, the problem cannot be exhausted through a parametric study, and only a specific combination of materials is worked out in detail. The results allow one to form some judgment, however, about the location of slip zones, their extent, the form of the slip velocity, and the interface shearing tractions.
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43.40.Dx Vibrations of membranes and plates
43.20.Bi Mathematical theory of wave propagation
43.20.Fn Scattering of acoustic waves

Approximate dynamic analysis of Timoshenko beams and its application to tapered beams

Kosuke Nagaya

J. Acoust. Soc. Am. Volume 66, Issue 3, pp. 794-800 (1979); (7 pages) | Cited 1 time

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An approximate method of analysis for dynamic response problems of a Timoshenko beam is presented. The results for the beam are obtained by the addition of the solution for bending and rotatory motions and that for the shear motion by neglecting the inertia force of the shear motion. The result by this analysis compared with both exact results for Timoshenko and Euler–Bernoulli beams. As applications of this study, dynamic response problems of taper beams with moving load are solved by this method.
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43.40.Cw Vibrations of strings, rods, and beams

Scattering of sound in suspensions of spheroidally shaped particles

Avtar S. Ahuja

J. Acoust. Soc. Am. Volume 66, Issue 3, pp. 801-805 (1979); (5 pages) | Cited 2 times

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Particles are assumed to have oblate and prolate spheroidal, spherical, disklike, and needlelike shapes, with symmetry axes for oblate and prolate spheroids assumed to be either parallel or perpendicular to the acoustic field. The disk‐shaped and needle‐shaped particles are assumed to be either broadside or edgewise (or end on) to the acoustic field. Scattering cross section for sound propagation in particle suspensions prepared in viscous fluid media has been calculated. For large density differences (of about 150%) between suspension components, computations indicate substantial effects of particle shape and orientation on the scattering of sound in dilute suspensions. For an oblate spheroid oscillating broadside, the backscattered signal is significantly (up to 30%) greater than that for a sphere, or for an oblate spheroid oscillting edgewise. Departures from Rayleigh’s law of scattering (by about 10% over one frequency decade) occur for very dense particles suspended in a viscous fluid medium.
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43.35.Bf Ultrasonic velocity, dispersion, scattering, diffraction, and attenuation in liquids, liquid crystals, suspensions, and emulsions
43.20.Fn Scattering of acoustic waves
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