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Journal of the Acoustical Society of America

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Dec 1980

Volume 68, Issue 6, pp. 1561-1918

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The audibility of Doppler distortion in loudspeakers

Edgar Villchur and Roy. F. Allison

J. Acoust. Soc. Am. Volume 68, Issue 6, pp. 1561-1569 (1980); (9 pages) | Cited 2 times

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The existence of Doppler distortion in loudspeakers was demonstrated more than 30 years ago, but the question of whether or not such distortion is of significance in music reproduction has not been settled. In this study the audibility of Doppler distortion in simple direct radiators is investigated both theoretically and experimentally through (1) an analogy to tape‐machine flutter, (2) an analysis of the effect of normal listening‐room conditions on Doppler distortion (for a given low‐frequency power the radiation‐load and reverberant‐field correction factors typically reduce this distortion to 0.36 of its value on the axis of a speaker radiating into anechoic half‐space), and (3) double‐blind listening tests. An eight‐member jury compared music reproduced in two modes, one of which was essentially free of Doppler distortion. In one experiment, two recordings were made of the output of a speaker, one with a sharp bass‐attenuation filter in the speaker input circuit and the other with the same filter moved to the microphone circuit. In a second experiment, on‐axis and off‐axis speaker output were recorded in an anechoic chamber, the speaker input circuits being compensated to make the frequency responses in the two modes equal. The theoretical analysis indicates that for any practical cone velocity Doppler distortion will be less audible by an order of magnitude than the flutter of a 15‐ips tape machine that meets the NAB Standard. The experimental results provided confirming evidence that it is inaudible.
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43.58.Ry Distortion: frequency, nonlinear, phase, and transient; measurement of distortion
43.38.Ja Loudspeakers and horns, practical sound sources

Transfer function method of measuring acoustic intensity in a duct system with flow

J. Y. Chung and D. A. Blaser

J. Acoust. Soc. Am. Volume 68, Issue 6, pp. 1570-1577 (1980); (8 pages) | Cited 3 times

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A new method of measuring the acoustic intensity of plane waves in a duct with flow is described. In this method, the acoustic transfer function between two locations in the duct is used to determine the acoustic intensity. In the case of no flow, the transfer‐function formulation reduces to a cross‐spectral relation similar to the relation used to measure acoustic intensity in three dimensions. In contrast to the relation in three dimensions, however, the cross‐spectral formulation for a duct is not limited in accuracy by the microphone spacing. The new method has been verified experimentally with a series of laboratory tests. Test results obtained by the transfer function method reveal that the net acoustic power transmitted along the pipe increases as the Mach number increases. The acoustic power radiated from the pipe opening, however, remains unchanged with increasing Mach number. This difference between the transmitted and the radiated power appears to be due to sound absorption caused by vorticity shedding at the pipe opening.
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43.58.Dj Sound velocity
47.60.-i Flow phenomena in quasi-one-dimensional systems
43.20.Mv Waveguides, wave propagation in tubes and ducts
43.28.Py Interaction of fluid motion and sound, Doppler effect, and sound in flow ducts

Influence of A‐weighting tolerances and frequency‐band limits on level measurements

G. S. K. Wong

J. Acoust. Soc. Am. Volume 68, Issue 6, pp. 1578-1583 (1980); (6 pages)

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The influence of type 1 A‐weighted tolerances and frequency‐band limits on level measurements is investigated theoretically and is verified experimentally. Three A‐weighting circuits, each of which satisfied type 1 tolerances, were tested with pulse trains with various crest factors; depending on the frequency‐band limits of the measuring system, erroneous level readings could be obtained. For example, at a constant repetition rate of 10 kHz and a crest factor of three, the measured levels obtained with the A‐weighting circuits spanned a range of 4 dB when the frequency‐band limit was increased from 30 to 100 kHz.
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43.58.Fm Sound level meters, level recorders, sound pressure, particle velocity, and sound intensity measurements, meters, and controllers

Hearing in Glires: Domestic rabbit, cotton rat, feral house mouse, and kangaroo rat

Henry Heffner and Bruce Masterton

J. Acoust. Soc. Am. Volume 68, Issue 6, pp. 1584-1599 (1980); (16 pages) | Cited 4 times

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Behavioral audiograms were determined for four species of Glires: one lagomorph (domestic rabbit, Oryctolagus cuniculus) and three feral rodents (cotton rat, Sigmodon hispidus; house mouse, Mus musculus; and kangaroo rat, Dipodomys merriami). Considerable variation in hearing ability was found among the four species with low‐frequency hearing limits ranging over 5‐1/2 octaves from 50 (kangaroo rat) to 2300 Hz (feral mouse) and high‐frequency hearing limits ranging from 49 (rabbit) to 90 kHz (feral mouse). Comparison of the characteristics of each audiogram with the audiograms of other animals of the same Order, Cohort, and Class provide further evidence for the validity of two relationships: (1) interaural distance is strongly and inversely correlated with high‐frequency hearing ability, and (2) good high‐frequency hearing is apparently incompatible with good low‐frequency hearing in most, if not all, land mammals. Furthermore, it is shown that cotton rats and feral mice possess the ability to perform frequency discriminations even at very high frequencies, indicating that there is probably no difference about the way in which they perceive high and low‐frequency sounds. Finally, it is shown that kangaroo rats are not unusual in their ability to localize brief sounds, indicating that these animals have not compromised this ability in their acquistion of their unusual low‐frequency sensitivity.
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43.80.Lb Sound reception by animals: anatomy, physiology, auditory capacities, processing
43.66.Gf Detection and discrimination of sound by animals

Modes of vibration and sound radiation from tuned handbells

Thomas D. Rossing and H. John Sathoff

J. Acoust. Soc. Am. Volume 68, Issue 6, pp. 1600-1607 (1980); (8 pages) | Cited 2 times

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Although handbells can vibrate in many different modes, only the two modes of lowest frequency are normally tuned by the bellcrafters. The frequency of the second mode is generally tuned to be three times that of the first. Each of these modes also radiates sound at twice the modal frequency, however, so that the radiated sound spectrum includes the second and sixth harmonics as well as the first and third. The amplitude of the second harmonic partial is proportional to the amplitude of the fundamental, and a similar relationship relates the sixth harmonic to the third. The first and third harmonics are radiated most strongly at right angles to the bell axis; the second and sixth harmonics are maximum along the axis. Small asymmetries in the bell often create warble due to beats between the two components of a split modal doublet. The tonal quality of a handbell depends upon the clapper strike point, the hardness of the clapper, and the force of the blow in addition to the design and construction of the bell itself.
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43.75.Kk Bells, gongs, cymbals, mallet percussion, and similar instruments
43.40.Ey Vibrations of shells

Relation between psychophysical data and speech perception for hearing‐impaired subjects. I

Wouter A. Dreschler and Reinier Plomp

J. Acoust. Soc. Am. Volume 68, Issue 6, pp. 1608-1615 (1980); (8 pages) | Cited 8 times

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In a heterogeneous group of ten hearing‐impaired adolescents, relations are studied between several psychophysical test results and speech reception thresholds in quiet and noise. To examine the results, data‐reduction techniques were used to extract the most relevant parameters. High correlations were found between the intelligibility of speech in noise on the one side, and vowel‐perception parameters resulting from INDSCAL analysis (frequency resolution parameters, critical bandwidth and critical ratio) and audiogram parameters on the other.
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43.71.Gv Measures of speech perception (intelligibility and quality)
43.66.Dc Masking
43.66.Sr Deafness, audiometry, aging effects
43.66.Fe Discrimination: intensity and frequency

Room acoustics for the aged

R. Plomp and A. J. Duquesnoy

J. Acoust. Soc. Am. Volume 68, Issue 6, pp. 1616-1621 (1980); (6 pages)

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This article deals with the combined effects of noise and reverberation on the speech‐reception threshold for sentences. It is based on a series of current investigations on: (1) the modulation‐transfer function as a measure of speech intelligibility in rooms, (2) the applicability of this concept to hearing‐impaired persons, and (3) hearing loss for speech in quiet and in noise as a function of age. It is shown that, generally, in auditoria, classrooms, etc. the reverberation time T, acceptable for normal‐hearing listeners, has to be reduced to (0.75)DT in order to be acceptable for elderly subjects with a hearing loss of D dB for speech in noise; for listening conditions as in lounges, restaurants, etc. the corresponding value is (0.82)DT.
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43.71.Gv Measures of speech perception (intelligibility and quality)
43.50.Qp Effects of noise on man and society
43.55.Br Room acoustics: theory and experiment; reverberation, normal modes, diffusion, transient and steady-state response
43.66.Sr Deafness, audiometry, aging effects

Task variables in the study of vowel perception

Winifred Strange and Terry L. Gottfried

J. Acoust. Soc. Am. Volume 68, Issue 6, pp. 1622-1625 (1980); (4 pages) | Cited 1 time

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Vowels produced in isolation by six speakers were identified less accurately than vowels coarticulated in /k/‐vowel‐/k/ syllables, whether tested by a key word task or by a rhyming task. Performance by naive listeners in the two tasks was highly correlated.
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43.70.Dn Disordered speech
43.70.Fq Acoustical correlates of phonetic segments and suprasegmental properties: stress, timing, and intonation

Identification of coarticulated vowels

Terry L. Gottfried and Winifred Strange

J. Acoust. Soc. Am. Volume 68, Issue 6, pp. 1626-1635 (1980); (10 pages) | Cited 2 times

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Previous explanations of vowel perception held that the most definitive information for vowel identity is the relatively constant formant frequencies in the steady‐state portions of vowels. Perceptual studies indicate, however, that vowels spoken in syllables with labial stop consonants are identified more accurately than vowels spoken in isolation. The present study investigated the nature and scope of this consonantal context advantage in the perception of ten American English vowels spoken by adult male and female speakers. Vowels in /p/‐vowel‐/p/, /b/‐vowel‐/b/, /k/‐vowel‐/k/, /k/‐vowel, and vowel‐/k/ syllables were identified much more accurately than isolated vowels. This is consistent with the hypothesis that dynamic acoustic information due to the coarticulation in syllables is important for vowel identification. Identification of vowels in /g/‐vowel‐/g/, /g/‐vowel, and vowel‐/g/ syllables was not better than isolated vowels and was significantly poorer than for other consonantal contexts. Acoustical analyses were performed to determine whether poor production of vowels could account for perceptual errors. Misproduced vowel targets could not account for the overall pattern of identification performance. Phonological factors were also considered but were found to be inadequate to account fully for the results.
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43.70.Dn Disordered speech
43.72.Ar Speech analysis and analysis techniques; parametric representation of speech
43.70.Fq Acoustical correlates of phonetic segments and suprasegmental properties: stress, timing, and intonation

Identification of vowels spoken in isolation versus vowels spoken in consonantal context

Marian J. Macchi

J. Acoust. Soc. Am. Volume 68, Issue 6, pp. 1636-1642 (1980); (7 pages) | Cited 2 times

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Strange et al. [J. Acoust. Soc. Am. 60, 213–224 (1976)] report that naive listeners misidentified approximately three times as many tokens of vowels spoken in isolation as the corresponding vowels spoken in consonantal context. On the basis of these findings, they argue that the superior intelligibility of vowels in consonantal context is due to coarticulatory effects: ’’acoustic information distributed over the temporal course of the syllable is utilized regularly by the listener to identify vowels’’ (p. 213). The present study tested naive listeners’ identification of eleven American English vowels spoken in isolation and in consonantal context with an experimental design comparable to that used by Strange et al. Here, however, listening tests were administered under high quality listening conditions, speakers and listeners were closely matched for regional dialect, and problems with the response alternatives for the vowels were minimized by having listeners identify the isolated vowels and /tVt/ syllables by rhyming them with English words. The results of the tests indicated that isolated vowels could be identified quite well; listeners misidentified only 2% of the vowels when tokens were blocked by speaker and only 8% when tokens were randomized across speakers. Further, the tests did not reveal any overall difference in identifiability between the isolated vowels and vowels in consonantal context.
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43.70.Dn Disordered speech
43.70.Fq Acoustical correlates of phonetic segments and suprasegmental properties: stress, timing, and intonation

Interaural time differences: Implications regarding the neurophysiology of sound localization

G. Linn Roth, Ravindra K. Kochhar, and Joseph E. Hind

J. Acoust. Soc. Am. Volume 68, Issue 6, pp. 1643-1651 (1980); (9 pages) | Cited 3 times

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See Also: Erratum

Show Abstract
Interaural time differences (ITDs) were measured from 400–7000 Hz on cats in order to provide quantitative data for use in physiological/behavioral studies on sound localization. ITDs derived from clicks and the initial portion of tone bursts showed a pronounced roughness and frequency dependence. This frequency dependence is most evident at higher angles of incidence and indicates that a single ITD will not always represent a single position on the azimuth. Controls demonstrate that most of the roughness in these functions was due to reflections off the surface supporting the animal and that the measured ITDs corresponded to predictions made by steady‐state theory. Measurements made with and without the pinnae in position indicate that they have relatively little effect on these ITD functions, particularly for frequencies below 2500 Hz and for small angles of incidence. In spite of acoustic limitations exemplified by the roughness and frequency dependence of these functions, ITDs generated by sound sources situated close to the midline provide reliable localization cues that are much better than those derived from sources well out on the azimuth. Finally it is noted that another ITD, the group ITD, can be ascribed to an acoustic signal. Calculations based on the measured steady‐state ITDs show differences between the group and steady‐state ITDs over a given range of frequencies. Differences between the group and steady‐state ITD can be significant, and it is argued that: (1) the group ITD can provide a localization cue to the auditory system that is distinct from the steady‐state ITD; and (2) it is possible these group ITDs are used by the nervous system to localize sound sources in realistic situations.
Show PACS
43.66.Qp Localization of sound sources
43.80.Gx Mechanisms of action of acoustic energy on biological systems: physical processes, sites of action
43.66.Mk Temporal and sequential aspects of hearing; auditory grouping in relation to music
43.80.Lb Sound reception by animals: anatomy, physiology, auditory capacities, processing

A multi‐parametric study of impact noise‐induced TTS

Claude Trémolières and Raymond Hétu

J. Acoust. Soc. Am. Volume 68, Issue 6, pp. 1652-1659 (1980); (8 pages) | Cited 1 time

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A series of three experiments with 15 normal‐hearing listeners measured temporary threshold shifts (TTS) as a function of various parameters of impact noise exposure. A criterion of 15 dB TTS at 4, 6, and 8 kHz, measured 3 min after exposure was adopted. Results indicated that; (1) the growth of TTS is a power function of the peak level of impact noises, (2) recovery from TTS appears to proceed linearly with the logarithm of the time in quiet, (3) the amounts of TTS3 are exponentially related to the logarithm of the decay time (or e duration: te) of the impact sound pressure envelope, (4) with constant total amounts of energy and a constant repetition rate, increasing the number from 60 to 1000 increases the final amount of TTS; the derived trading relationship involves a 12.7‐dB level change for a tenfold change in the number of impacts, (5) repetition rates ranging between 0.5 and 1 pps (pulse per second) are the most harmful to hearing and, (6) daily exposure to 100 impacts (te: 200 ms; repetition rate: 0.5 pps) should be limited to a 130‐dB peak level, in order to protect 90% of the exposed people. Results are discussed in terms of the possible use of the equivalent continuous noise level as an index in setting safe limits of occupational impact noise exposure.
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43.66.Ed Auditory fatigue, temporary threshold shift
43.66.Mk Temporal and sequential aspects of hearing; auditory grouping in relation to music
43.50.Qp Effects of noise on man and society

Cochlear micromechanics—A physical model of transduction

J. B. Allen

J. Acoust. Soc. Am. Volume 68, Issue 6, pp. 1660-1670 (1980); (11 pages) | Cited 20 times

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One of the basic questions which has persisted in the field of hearing theory is the still unresolved mechanical action of hair‐cell transduction. The fundamental problem that has historically plagued researches is the discrepancy between mechanically measured tuning of basilar membrane motion and neurally measured tuning. In this paper we show that the difference between these two measures appears to be accounted for by a specific, physically motivated, micromechanical model. This model gives rise to a spectral zero which we identify as the ’’second‐filter’’ of cochlear transduction. For high‐frequency fibers this zero resides at a fixed frequency ratio below CF (characteristic frequency) while for fibers having low‐frequency CF’s the zero appears to go to zero frequency faster than CF. In this paper we first present and analyze the assumed mechanical model. We then briefly discuss a possible specific physical realization for the nonlinearity of cochlea mechanics. The nonlinear model is based on dynamical variations in outer hair cell stereocilia stiffness.
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43.64.Bt Models and theories of the auditory system
43.64.Kc Cochlear mechanics

Analysis of the click‐evoked brainstem potentials in humans using high‐pass noise masking. II. Effect of click intensity

J. J. Eggermont and M. Don

J. Acoust. Soc. Am. Volume 68, Issue 6, pp. 1671-1675 (1980); (5 pages) | Cited 17 times

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Derived narrow‐band brainstem responses were obtained for click levels of 10–60 dB SL in normal hearing subjects. The amplitudes and latencies of the wave I, wave III, and wave V components in the derived BSER were studied as a function of click intensity. Characteristic differences were found between the input–output behavior of waves I and III on one hand and wave V on the other hand, especially for the low‐frequency narrow bands (center frequencies of 0.5 and 1.0 kHz). While the wave I and wave III (peak‐to‐succeeding trough) amplitude showed a small (20–30 dB) dynamic range with saturation effects, the wave V amplitude continued to increase across the intensity range studied. At the high‐frequency end (narrow‐band center frequencies of 4 and 8 kHz), wave V also showed saturation. It is suggested that this difference across center frequency (place of origin along the cochlear partition) is responsible for the dominance of wave V at low‐frequency stimulation (e.g., with tonebursts). The latencies of the three waves studied maintained their constant interwave delays across the observed intensity range in each narrow band. Quite large (up to 3.5 ms) increases in the narrow‐band latencies were found for decreasing click levels; this is comparable in value with those for the unmasked BSER although the mechanism seems to be different. The major contribution to the BSER which determines its latencies, originates at 60 dB SL from the 8‐kHz region but at low SL (10 and 20 dB) from the 2‐kHz region. At these low intensity levels, the contribution from the apical part of the cochlea, however, is still of the same size as that from the high‐frequency end.
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43.64.Ri Evoked responses to sounds
43.66.Dc Masking

Sound pressures in the basal turn of the cat cochlea

V. Nedzelnitsky

J. Acoust. Soc. Am. Volume 68, Issue 6, pp. 1676-1689 (1980); (14 pages) | Cited 19 times

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Techniques were developed for measuring sound pressure in the cochlea with calibrated, liquid‐filled, piezoelectric probe microphones. Sound pressures were measured in scala vestibuli and scala tympani in the basal turn in 25 cats for tones from 20–10 000 Hz. Control experiments indicated that intracochlear pressures were essentially uninfluenced by the measuring technique, and were conducted to the cochlea via the ossicular chain. Intracochlear pressures are linearly related to pressure at the tympanic membrane for tone levels at least as high as 105 dB SPL, and are relatively independent of depth of probe insertion in the scalae. The transfer ratio of sound pressure in scala vestibuli to that at the tympanic membrane increases in magnitude over the frequency range 50–1000 Hz to reach a maximum value of 15–30 dB, and decreases at higher frequencies, thus demonstrating that the middle ear provides a frequency‐dependent pressure gain. At frequencies below 40 Hz, the pressures in scala vestibuli and scala tympani are approximately equal and are both determined by the round‐window membrane compliance. At frequencies above 100 Hz, the round‐window membrane impedance is small compared to the acoustic input impedance of the cochlea, and the pressure in scala vestibuli considerably exceeds that in scala tympani; consequently, the pressure difference across the cochlear partition is approximately equal to the pressure in scala vestibuli.
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43.64.Kc Cochlear mechanics
43.80.Lb Sound reception by animals: anatomy, physiology, auditory capacities, processing

The effect of array errors on frequency‐domain adaptive interference rejection

P. N. Keating

J. Acoust. Soc. Am. Volume 68, Issue 6, pp. 1690-1695 (1980); (6 pages)

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An existing adaptive beamforming technique, known as frequency‐domain adaptive interference rejection (FAIR), is examined for robustness with respect to nonsystematic array errors. It is concluded that, for array position errors below ±10% of the mean spacing, the FAIR technique continues to be effective. However, for special situations where large errors are encountered or where the noise situation is so favorable that better interference rejection is worthwhile, error compensation might become necessary, and a self‐calibration scheme is briefly described.
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43.60.Gk Space-time signal processing, other than matched field processing
43.60.Cg Statistical properties of signals and noise
43.58.Ta Computers and computer programs in acoustics
43.30.Vh Active sonar systems

Binaural detection of sonar signals

John E. Kerivan

J. Acoust. Soc. Am. Volume 68, Issue 6, pp. 1696-1698 (1980); (3 pages)

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Standard methods of presentation of sonar signals to operators have not maximized the capabilities of the auditory system. This report describes a dichotic method of sonar signal presentation to operators based on the masking‐level‐difference (MLD) principle in hearing. Using a Bekesy tracking procedure, twenty‐six listeners of varied experience detected recorded broadband sonar targets and low‐frequency pure tones that were embedded in 100 Hz–4 kHz recorded and synthesized backgrounds. The signals were either homophasic or antiphasic and the backgrounds were always homophasic. Average detection thresholds for the antiphasic sonar signals were 5–7 dB better than their homophasic counterparts. Pure tone antiphasic detection thresholds were also 6–11 dB better than in the homophasic presentation. The sonar signals show similar MLD’s as those found with speech signals.
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43.60.Gk Space-time signal processing, other than matched field processing
43.30.Vh Active sonar systems
43.66.Pn Binaural hearing

Sidelobe suppression versus phase error characteristics of multistage modulation scanning sonar systems

J. F. Lynch, S. P. Pitt, and J. G. Pruitt

J. Acoust. Soc. Am. Volume 68, Issue 6, pp. 1699-1703 (1980); (5 pages)

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The sidelobe level sensitivity of multistage modulation scanning sonar systems to internal electronic phase errors is examined by means of a ’’scanning function’’ approach and also by a hardware oriented analysis of a particular sonar system. It is shown that the later internal stages of such modulation scanning systems are the most sensitive to phase error. A general approach to calculating scanning sidelobe levels with phase error present is discussed.
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43.60.Gk Space-time signal processing, other than matched field processing
43.30.Vh Active sonar systems

On using pressure and pressure gradient sensor pairs in linear acoustic arrays

Robert H. MacPhie

J. Acoust. Soc. Am. Volume 68, Issue 6, pp. 1704-1710 (1980); (7 pages)

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This paper investigates the mapping capabilities of a linear receiving array of acoustic sensor pairs that measure both the pressure and pressure gradient along the array axis. It is demonstrated that unambiguous pressure maps of remote sources can be obtained even if the separation between sensor pairs is one wavelength (twice the spatial Nyquist sampling interval). The effect of additive noise—both correlated and uncorrelated at each sensor pair, but uncorrelated between sensor pairs, is studied and found not to be a major problem. In the second half of the paper an analysis of the cross correlations of the output voltages from the various sensor pairs reveals that each pair provides three distinct cross correlations. Accordingly from data obtained from an array with sensor pairs spaced three half‐wavelengths apart an unambiguous map of the source intensity distribution can be formed. Examples of such maps are provided.
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43.60.Gk Space-time signal processing, other than matched field processing
43.38.Ar Transducing principles, materials, and structures: general
43.30.Vh Active sonar systems
43.28.Tc Sound-in-air measurements, methods and instrumentation for location, navigation, altimetry, and sound ranging

Spectral structure of pressure measurements made in a combustion duct

J. H. Miles and D. D. Raftopoulos

J. Acoust. Soc. Am. Volume 68, Issue 6, pp. 1711-1722 (1980); (12 pages)

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The spectral structure of pressure measurements made in a ducted combustion test facility are studied. Dispersion and attenuation of acoustic plane waves may occur in the duct at low frequencies due to combustor emissions and affect the spectral structure. A model that considers the propagation of plane waves through a cloud of particles in a flowing gas, and which includes heat transfer between soot particles and the gas, is discussed. Experimental results are compared with theory.
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47.60.-i Flow phenomena in quasi-one-dimensional systems
43.20.Mv Waveguides, wave propagation in tubes and ducts
43.20.Hq Velocity and attenuation of acoustic waves
43.28.Py Interaction of fluid motion and sound, Doppler effect, and sound in flow ducts

Ultrasonic imaging system using orthogonal function wavefronts

Takuso Sato, Osamu Ikeda, and Tomoji Azuma

J. Acoust. Soc. Am. Volume 68, Issue 6, pp. 1723-1728 (1980); (6 pages)

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A new ultrasonic imaging system that employs the orthogonal function wavefronts derived from the a priori knowledge of the second‐order moment function about a class of objects is presented. First, it is shown theoretically that this new method has the ability to reconstruct images of sample objects of a given class with a higher signal‐to‐noise ratio (SNR) in a shorter time than the conventional beamforming methods. The assumptions are confirmed by numerical analyses and experiments carried out by using a prototype of an orthogonal function applied array imaging system.
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43.60.Gk Space-time signal processing, other than matched field processing
43.35.Sx Acoustooptical effects, optoacoustics, acoustical visualization, acoustical microscopy, and acoustical holography

Bispectral passive velocimeter of a moving noisy machine

Takuso Sato, Kimio Sasaki, and Mitsuhiro Taketani

J. Acoust. Soc. Am. Volume 68, Issue 6, pp. 1729-1735 (1980); (7 pages)

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A new method of determination of the velocity of a moving noisy machine by the analysis of a signal detected at a fixed point is proposed. In this system, it is assumed that the trajectory of the object is known beforehand and the special characteristics of a machine noise in the bispectral region are used so that a measuring system which does not suffer from any surrounding Gaussian noise is realized. The nonstationary change of the bispectral characteristics, along with the movement of the object, is detected from a set of short‐time bispectra in segmented intervals and the method of minimum mean‐square‐error fitting is applied to get reliable velocity. The principle, concrete construction, and experimental results are shown. The results show the superiority of the new method over conventional methods.
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43.60.Gk Space-time signal processing, other than matched field processing
43.58.Dj Sound velocity
06.30.Gv Velocity, acceleration, and rotation
43.50.Jh Noise in buildings and general machinery noise

Pressure developed within a cavity backing a finite panel when subjected to external transient excitation

R. W. Guy

J. Acoust. Soc. Am. Volume 68, Issue 6, pp. 1736-1747 (1980); (12 pages) | Cited 2 times

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An analysis concerning the response of a cavity‐backed panel when subjected to external excitation having arbitrary time and spatial variables is extended here to derive expressions for the pressure developed within the backing cavity. Two cases of practical interest are then considered as examples of application; a normally incident harmonic, and a normally incident ’N’ wave.
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43.55.Rg Sound transmission through walls and through ducts: theory and measurement
43.20.Bi Mathematical theory of wave propagation
43.40.Dx Vibrations of membranes and plates

Axisymmetric free vibration of thick annular plates

K. T. Sundara, Raja Iyengar, and P. V. Raman

J. Acoust. Soc. Am. Volume 68, Issue 6, pp. 1748-1749 (1980); (2 pages)

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Free vibration of annular plates has been studied using the method of initial functions. Numerical results of the natural frequencies are obtained for two typical support conditions. The present results depart from the classical plate theory for higher modes as well as for thicker plates, but is in good agreement with Mindlin’s improved plate theory.
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43.40.Dx Vibrations of membranes and plates

Sound channel propagation through eddies southeast of the Gulf Stream

J. C. Beckerle, L. Baxter, II, R. P. Porter, and R. C. Spindel

J. Acoust. Soc. Am. Volume 68, Issue 6, pp. 1750-1767 (1980); (18 pages)

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Acoustical signals at 270 Hz from SOFAR floats drifting in the region southeast of the Gulf Stream were recorded during most of 1975 from a near axis sound channel hydrophone near Bermuda. The amplitude levels received exhibit a large increase (12–18 dB) commencing about 24 July, following a long period (March to July) of relatively lower peak level amplitudes. A major part of the increase can be attributed to the influence of a large cyclonic eddy (Gulf Stream ring) that passed slowly between the SOFAR floats and Bermuda. Such an eddy produces a large sound speed anomaly that extends to depths below the axis of the sound channel. On 24 July, two SOFAR floats were known to have approximately the same sound transmission path through the edge of the large eddy. The sound transmission peaks occur when no ocean eddy is between the SOFAR floats and the receiver. Their spacing shows they occur at regular refraction caustics in the sound channel. When the sound transmission path passes through an eddy, these transmission focal distances are shifted to greater range and the signal level may be greatly enhanced. The decrease of caustic peak intensities with range is 5 dB per double distance, and this agrees with theory. Several different levels of peak acoustic intensity occur and these result from two float depths and oceanic thermocline oscillations.
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43.30.Bp Normal mode propagation of sound in water
92.10.Vz Underwater sound
93.30.Mj Atlantic Ocean
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