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Journal of the Acoustical Society of America

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Aug 1981

Volume 70, Issue 2, pp. 281-658

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Compact theory of a multielement broadside array of uniformly spaced echosonde antennas

S. Adeniyi Adekola

J. Acoust. Soc. Am. Volume 70, Issue 2, pp. 281-301 (1981); (21 pages)

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This paper develops a compact theory, which is an important extension of a single element theory previously formulated [S. A. Adekola, J. Acoust. Soc. Am. 62, 524–542 (1977)], for a multielement broadside array of echosondes (acoustic echo‐sounding antennas), uniformly spaced, and consisting of elements that are identical in all respects. The formulation enunciated allows for echosonde array design that gives low side‐lobe strengths, which can considerably minimize any ambiguous target detection of the weak echoes backscattered from the lower atmospheric structure, and an invaluable attribute arrived at, is the directivity‐pattern obtained, which can provide an improved target resolution required in acoustic echo sounding. A workable expression developed for the directivity, provides an inclination that is in good agreement with physical intuition, when expressed in a series form. The theory formulated here, permits an accurate echosonde‐array design when use is made of a computer, and some accurate computer‐generated directivity patterns are produced when an optimization procedure is adopted for the determination of interelement separation required for any particular echosonde array. A number of directivity patterns, illustrating numerous design examples, is presented. Results of several computations show, in general, that if a carrier frequency is specified, then the half‐power beamwidth reduces when the number of array elements is increased, whereas, if the size of the array is given, the 3‐dB beamwidth decreases when the carrier frequency is increased. As an example, when we use a carrier frequency fixed at 2250 Hz, the half‐power beamwidths calculated for a six‐ and twenty‐four‐element array are 6.137° and 5.094°, respectively, whereas the 3‐dB beamwidths obtained when the number of array elements is fixed at twelve, are 7.184° and 4.159° at carrier frequencies that assume values of 1750 and 3500 Hz, respectively. The question of how directivity patterns can be realized without high intensity side lobes that may produce an ambiquity of target resolution, and which may obscure the weak backscattered signal when use is made of the entire echosonde system for remote sensing applications, is investigated, and it is shown that, acceptable side‐lobe reductions can be produced, provided the interelement separation is less than a wavelength, and if a second large interference lobe is attenuated from the visible range of a directivity pattern. A comprehensive design table provided describes some optimum interelement separations which are determined for a number of array elements. For example, an optimum spacing of 0.907λa, where λa is the acoustic wavelength, is determined for an eight‐element array. When the optimum spacing just stated is employed, the directivity pattern computed when uniform, Gaussian and Lorentzian‐line shape distributions are assumed at the illuminating antenna aperture, shows that the side‐lobe attenuations realized are as low as −70 dB in each case.
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43.38.Ja Loudspeakers and horns, practical sound sources
43.30.Vh Active sonar systems
43.28.Tc Sound-in-air measurements, methods and instrumentation for location, navigation, altimetry, and sound ranging

Finite amplitude effects on the thresholds for lesion production in tissues by unfocused ultrasound

E. L. Carstensen, S. A. Becroft, W. K. Law, and D. B. Barbee

J. Acoust. Soc. Am. Volume 70, Issue 2, pp. 302-309 (1981); (8 pages) | Cited 2 times

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The phenomenon of thermal lesion production in tissues by unfocused ultrasound has been modeled under the assumptions (1) that damage to the tissues is the result of a chemical rate process and (2) that the time–temperature exposure of the tissue results from a competition between the rate of heat generation by absorption of ultrasonic energy and thermal diffusion. The model was tested by observing thresholds for color change in samples of excised bovine liver at 6 cm from a 2.25‐MHz, 1.27‐cm‐diam piston source and 11.5 cm from a 4.4‐MHz, 1.27‐cm source. The observations disagreed sharply with predictions of the linear model. However, after recognition of the fact that finite amplitude phenomena modify the effective absorption coefficients of the tissues, the model gives excellent predictions of observed thresholds simply by using intensity dependent values of the absorption coefficients in the numerical calculations. Thermoelectric and radiation force methods were used to demonstrate the nonlinear absorption phenomenon in excised liver. In contrast with the case of focused ultrasound, the thresholds for lesion formation by unfocused sources (1) occur at lower intensities, (2) are strongly dependent upon frequency, and (3) because acoustic saturation in the coupling medium imposes lower limits on tissue intensities for unfocused than for focused ultrasound, unfocused lesions are limited to relatively long exposure times.
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43.80.Gx Mechanisms of action of acoustic energy on biological systems: physical processes, sites of action
43.80.+p Bioacoustics
43.80.-n Bioacoustics
87.50.Y- Biological effects of acoustic and ultrasonic energy
43.25.Cb Macrosonic propagation, finite amplitude sound; shock waves

Heat generation by ultrasound in a relaxing medium

Wesley L. Nyborg

J. Acoust. Soc. Am. Volume 70, Issue 2, pp. 310-312 (1981); (3 pages) | Cited 10 times

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It is shown that at any point P in a continuous sound field the time‐averaged rate of heat generation by relaxational absorption per unit volume is given by αp02/ρc. Here p0 is the pressure amplitude at P while α is the absorption coefficient arising from relaxation when the coefficient of shear viscosity η is zero; ρ and c are, respectively, the density and velocity of sound in the medium. The result, a familiar one for plane traveling waves, is shown here to apply generally when η = 0.
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43.80.Gx Mechanisms of action of acoustic energy on biological systems: physical processes, sites of action
43.35.Fj Ultrasonic relaxation processes in gases, liquids, and solids
43.35.Ty Other physical effects of sound

Acoustic analysis of the Sonic Guide

Beverly A. Goldstein and William R. Wiener

J. Acoust. Soc. Am. Volume 70, Issue 2, pp. 313-320 (1981); (8 pages)

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The Sonic Guide is an eyeglass configured sonar‐type device available for the orientation and mobility of blind individuals. It operates by emitting toward and receiving from target objects modulated stimuli whose intensity and frequency components directly result from the distance and physical properties of the target. This investigation measured the sound pressure level output, via a sound level meter, of four Sonic Guide temples, at three volume control settings, for six octave frequencies (250–8000 Hz), at six distances (35.6 cm–4.6 m) for a total of 432 measurements. Means, standard deviations, and an ANOVA were computed to investigate the consistency of output across temples and the interactions among frequencies with volume control setting and with distance. Across temples, the Sonic Guide devices were found consistent at each volume control setting, at each frequency, and for each distance measured. Higher volume control settings produced proportionally greater output in the higher frequencies than in the lower frequencies. As distance increased, the peak of the response shifted upward in frequency, although overall, targets at shorter distances produced greater intensity at lower frequencies than did the more distant targets at higher frequencies. Implications for use of the Sonic Guide by hearing‐impaired individuals are discussed.
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42.66.-p Physiological optics
43.80.Vj Acoustical medical instrumentation and measurement techniques
43.38.Ar Transducing principles, materials, and structures: general
43.30.Vh Active sonar systems

An articulatory synthesizer for perceptual research

Philip Rubin, Thomas Baer, and Paul Mermelstein

J. Acoust. Soc. Am. Volume 70, Issue 2, pp. 321-328 (1981); (8 pages) | Cited 12 times

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A software articulatory synthesizer, based upon a model developed by P. Mermelstein [J. Acoust. Soc. Am. 53, 1070–1082 (1973)], has been implemented on a laboratory computer. The synthesizer is designed as a tool for studying the linguistically and perceptually significant aspects of articulatory events. A prominent feature of this system is that it easily permits modification of a limited set of key parameters that control the positions of the major articulators: the lips, jaw, tongue body, tongue tip, velum, and hyoid bone. Time‐varying control over vocal‐tract shape and nasal coupling is possible by a straightforward procedure that is similar to key‐frame animation: critical vocal‐tract configurations are specified, along with excitation and timing information. Articulation then proceeds along a directed path between these key frames within the time script specified by the user. Such a procedure permits a sufficiently fine degree of control over articulator positions and movements. The organization of this system and its present and future applications are discussed.
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43.70.Bk Models and theories of speech production
43.70.Jt Instrumentation and methodology for speech production research
43.72.Gy Narrow, medium, and wideband speech coding
43.60.Qv Signal processing instrumentation, integrated systems, smart transducers, devices and architectures, displays and interfaces for acoustic systems

Vowel height and the perception of consonantal nasality

Arthur S. Abramson, Patrick W. Nye, Janette B. Henderson, and Charles W. Marshall

J. Acoust. Soc. Am. Volume 70, Issue 2, pp. 329-339 (1981); (11 pages)

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By means of an articulatory synthesizer, the preception of the oral–nasal distinction in consonants was explored experimentally. This distinction was chosen because it is achieved by a very simple articulatory maneuver and because it is phonologically relevant in virtually every language. Lowering the velum in equal increments provided continua of CV syllables varying in size of velopharyngeal port which were divided perceptually into /d/ and /n/ categories by American English listeners. To test the hypothesis that the coarticulation of these nasal consonants with lower (more open) vowels requires a larger area of velopharyngeal coupling to give a nasal consonant percept, three oral–nasal continua incorporating the vowels /i/, /Λ/, and /α/, respectively, were presented for identification. The results were compared with those of A. S. House and K. N. Stevens [J. Speech Hear. Disord. 21, 218–232 (1956)] and A. S. House [J. Speech Hear. Disord. 22, 190–204 (1957)] obtained with steady‐state vowels and consonantal murmurs and with those of M. H. L. Hecker [J. Acoust. Soc. Am. 34, 179–188 (1962)]. Three conclusions emerged. First, the relationship between vowel height and the amount of velopharyngeal coupling needed for a nasal percept occurs in conditions where subjects are required to make linguistically relevant judgments. Second, the relationship can arise in conditions where vocalic coarticulation is present. Third, the relationship is not confined to vowels but can also be observed in the case of dynamically articulated consonants. One of the continua was also used for discrimation experiments, which yielded the classical pattern of high discriminability at the category boundary.
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43.70.Dn Disordered speech
43.70.Jt Instrumentation and methodology for speech production research
43.70.Fq Acoustical correlates of phonetic segments and suprasegmental properties: stress, timing, and intonation

Discrimination of speech by nonhuman animals: Basic auditory sensitivities conducive to the perception of speech‐sound categories

Patricia K. Kuhl

J. Acoust. Soc. Am. Volume 70, Issue 2, pp. 340-349 (1981); (10 pages) | Cited 10 times

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Chinchillas (Chinchilla laniger) were tested in a same–different task to determine the location of greatest sensitivity along a continuum of voice‐onset‐time (VOT). The procedure used was an up–down staircase technique which allowed the determination of the just‐noticeable‐difference in VOT (Δ VOT) for VOT’s on a continuum ranging from [dα] to [tα]. Results demonstrated that the animals were most sensitive to change (i.e., produced the smallest Δ VOT values) in the region of the phonetic boundary dividing voiceless‐unaspirated and voiceless‐aspirated sounds, in good agreement with the boundary value previously obtained in an identification task with this same species [Kuhl and Miller, J. Acoust. Soc. Am. 63, 905 (1978)]. The results support the notion that the mammalian auditory system provided a selective pressure on the choice of acoustic cues to represent the phonetic oppositions employed by the world’s languages. The data are discussed in terms of the original definition of categorical perception and current psychoacoustic explanations of the peak in sensitivity for two‐component stimuli varying in onset time.
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43.70.Dn Disordered speech
43.70.Fq Acoustical correlates of phonetic segments and suprasegmental properties: stress, timing, and intonation
43.66.Gf Detection and discrimination of sound by animals
43.80.Lb Sound reception by animals: anatomy, physiology, auditory capacities, processing

Influence of segmental factors on fundamental frequency in fluent speech

Noriko Umeda

J. Acoust. Soc. Am. Volume 70, Issue 2, pp. 350-355 (1981); (6 pages) | Cited 2 times

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The segmental factors which determine the fundamental frequency, F0, of vowels in stressed syllables in fluent readings were studied. The paper describes five characteristics of F0 in the following vowel for different preceding consonants: (1) the direction of F0 change at the vowel onset, (2) the frequency at voice onset, (3) the peak F0 during the vowel, (4) the amount of F0 increase, and (5) the time required to reach the peak. The two speakers show similar consistent tendencies for all the fundamental‐frequency characteristics. Our results are similar to those reported with isolated or carrier‐phrased utterences in general, but different in details. Evidence for the so‐called intrinsic fundamental frequencies for different vowels was not observed in our data.
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43.72.Ar Speech analysis and analysis techniques; parametric representation of speech
43.70.Jt Instrumentation and methodology for speech production research
43.70.Fq Acoustical correlates of phonetic segments and suprasegmental properties: stress, timing, and intonation

Relations between auditory functions in normal hearing

J. M. Festen and R. Plomp

J. Acoust. Soc. Am. Volume 70, Issue 2, pp. 356-369 (1981); (14 pages) | Cited 9 times

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The relations among a number of auditory functions were studied by concentrating on small interindividual differences in these functions. For this purpose a battery of 12 tests was applied to 50 normal‐hearing subjects. The tests included absolute threshold, auditory bandwidth measured with comb‐filtered noise in direct and in forward masking, psychophysical tuning curve both in direct and in forward masking, temporal integration measured with intensity‐modulated noise, forward‐ and backward‐masking curves, cubic‐difference‐tone strength, and lateral suppression. In all cases the test frequency was 1000 Hz. Among the relations found, are (1) a positive correlation between the shift of the steep edge of the tuning curve, away from the probe frequency, and the width of the auditory filter as measured with comb‐filtered noise, (2), an inverse relation between the width of the tuning curve and the width of the temporal window, and (3) a positive correlation between the width of the auditory filter and the strength of the cubic difference tone. Low correlations among tests were not caused by poor test reliability.
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43.66.Dc Masking
43.66.Mk Temporal and sequential aspects of hearing; auditory grouping in relation to music
43.66.Ki Subjective tones

The effect of occluded ear impedances on the eardrum SPL produced by hearing aids

Samuel Gilman, Donald D. Dirks, and Richard Stern

J. Acoust. Soc. Am. Volume 70, Issue 2, pp. 370-386 (1981); (17 pages)

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The purpose of the investigation was to determine the effect of eardrum and earmold impedance on eardrum sound pressure level (SPL) produced by a hearing aid receiver. Acoustic interactions between receivers and ears occurring at receiver resonances were explored and analyzed. Results with ear simulators showed that the effect of simulator eardrum impedance was increased in regions where ear and receiver impedance approached each other during resonances. Earmold acoustics were also significant factors at frequencies where their impedances were comparable to ear impedances. Resonant frequency shifts of 540 Hz in the region between 800 and 1500 Hz were observed for the range of simulator eardrum and earmold impedances employed, with SPL changes of 10 dB occurring at some frequencies. Results were different for each of the five receivers tested confirming the effect of the receiver characteristics on the eardrum SPL. A subsequent test was performed using real ears with results that agreed substantially with those obtained with the simulators.
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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
43.66.Sr Deafness, audiometry, aging effects

Practically perfect pitch

Gregory R. Lockhead and Robert Byrd

J. Acoust. Soc. Am. Volume 70, Issue 2, pp. 387-389 (1981); (3 pages) | Cited 6 times

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People who can identify piano notes with essentially no errors (perfect pitch) are much less capable in identifying musical notes produced by sine waves. Thus, frequency is not the only information these people use to identify musical notes; piano notes are complex waveforms or patterns, sine waves are not complex. Musically trained people who do not have perfect pitch ability have considerable difficulty identifying either sine waves or piano notes. As well as this quantitative difference, these two groups of musicians also differ qualitatively. The people in both groups are about equally able to judge octave levels, but people with perfect pitch are excellent in identifying the particular note, e.g., E, independent of its octave, while people without perfect pitch ability are not.
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43.66.Hg Pitch

Temporary threshold shifts produced by wideband noise

John H. Mills, Warren Y. Adkins, and Robert M. Gilbert

J. Acoust. Soc. Am. Volume 70, Issue 2, pp. 390-396 (1981); (7 pages) | Cited 4 times

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Groups of human subjects were exposed in a sound field to a wideband noise for 24 or for 8 h on consecutive days. The wideband noise was composed of octave bands centered at 0.5, 1, 2, and 4 kHz. For the 24‐h exposure, temporary threshold shift (TTS) increased for about 8 h and then reached a plateau or asymptote. TTS’s at asymptote (ATS) increased about 1.7 dB/dB increase in noise level above about 78 dBA. TTS produced by single‐octave band exposures were used to predict the TTS produced by the wideband exposures. Predictions were based on the ’’Intensity Rule’’ [W. D. Ward, A. Glorig, and D. L. Sklar, ’’Temporary threshold shift from octave‐band noise: Applications to DRC’s,’’ J. Acoust. Soc. Am. 31, 522–528 (1959)]. Predictions were acceptably accurate and the validity of the ’’Intensity Rule’’ for 24‐h exposures or 8‐h exposures is supported. There is a remarkable coincidence between the relation which describes ATS and noise level, and the relation which describes noise‐induced permanent threshold shift (in industrial workers) and noise level. This coincidence and animal data are used to support the hypothesis that TTS grows to an asymptote rather than a plateau, and that TTS at asymptote (ATS) produced by a given sound is an upper bound on any permanent threshold shift that can be produced by that sound.
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43.66.Ed Auditory fatigue, temporary threshold shift
43.66.Sr Deafness, audiometry, aging effects
43.50.Qp Effects of noise on man and society

Lateral position of sinusoids presented with interaural intensive and temporal differences

William A. Yost

J. Acoust. Soc. Am. Volume 70, Issue 2, pp. 397-409 (1981); (13 pages) | Cited 17 times

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The lateral position of pulsed sinusoids presented with interaural intensive or temporal differences was determined for a variety of frequencies, overall intensities, and durations. The lateral position of stimuli presented with interaural temporal differences was linearly related to interaural phase for interaural phases between 0 ° and ±90 °. Beyond ±90 ° the images often appeared at more than one position, usually on opposite sides of the head. The lateral position of stimuli presented with interaural intensive differences was linearly related to interaural intensity up to ±15 dB. These results were largely unaffected by frequency, duration, and overall intensity, although some of the small dependencies are described and discussed. The results are consistent with the assumption that lateral position at any one frequency is determined by mechanisms similar to coincidence networks or cross correlators. However, a single cross‐correlation mechanism cannot account for the results obtained across frequency.
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43.66.Pn Binaural hearing
43.66.Qp Localization of sound sources
43.66.Nm Phase effects

Identification of local and propagating distortion products from cochlear microphonic responses

Don A. Ronken and Donald H. Eldredge

J. Acoust. Soc. Am. Volume 70, Issue 2, pp. 410-425 (1981); (16 pages)

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Careful measurements show that sound pressures of 40 dB re 20 μN/m2 are sufficient to reveal two varieties of well‐behaved, nonlinear distortion products in the cochlear microphonic (CM). The first variety appears in the primary‐tone area and is designated as a local CM distortion product. The second type exists apical to the primary‐tone area and is identical to the mechanically propagated distortion seen in the phase‐locked responses of primary auditory nerve fibers. The existence of the propagating distortion product forces the conclusion that there must be a mechanical contribution to the local ’’CM’’ distortion product as well. The intrusion of nonlinear mechanical responses at such low levels (<40 dB SPL) indicates that the effective mechanical input to the hair cells may be nonlinear over most of the audible range. An important but unanswered question is the range over which the transducer characteristic of CM could be effectively linear, for that would determine whether CM could be used to probe mechanical nonlinear effects in the primary tone area.
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43.64.Nf Cochlear electrophysiology
43.64.Kc Cochlear mechanics
43.64.Bt Models and theories of the auditory system

Cochlear model including three‐dimensional fluid and four modes of partition flexibility

Larry A. Taber and Charles R. Steele

J. Acoust. Soc. Am. Volume 70, Issue 2, pp. 426-436 (1981); (11 pages) | Cited 3 times

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The WKB solution is developed for the analysis of a straight box cochlear model which includes four modes of partition displacement, simulating the motion of the bony shelf and arches of Corti, as well as the pectinate zone of the basilar membrane. The theory is similar to that previously used for the 1‐mode model with scalar quantities now replaced by 4‐vectors. Calculations are carried out for the guinea pig cochlea with stiffness computed mainly from the anatomy and assumed physiological values for the materials. Results show that the stiffness is such that the amplitude and phase of the basilar membrane response are not significantly altered from those given by the 1‐mode model. For primates and some other mammals, the bony shelf is substantially weaker than in the guinea pig and causes a much more rapid accumulation of phase along the basilar membrane. Thus, with anatomically and physiologically consistent parameters, the model yields good correlation in phase and amplitude with the in vivo measurements which have been made in the squirrel monkey by Rhode [J. Acoust. Soc. Am. 64, 158–176 (1978)] as well as in the guinea pig by Wilson and Johnstone [J. Acoust. Soc. Am. 57, 705–723 (1975)] and Rhode [Basic Mechanisms in Hearing (Academic, New York, 1973), pp. 49–63].
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43.64.Bt Models and theories of the auditory system
43.64.Kc Cochlear mechanics

Frequency spectra of cochlear acoustic emissions (’’Kemp‐echoes’’)

H. P. Wit, J. C. Langevoort, and R. J. Ritsma

J. Acoust. Soc. Am. Volume 70, Issue 2, pp. 437-445 (1981); (9 pages) | Cited 7 times

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Cochlear acoustic emissions were recorded with a sensitive microphone from within the ear canal of normal hearing subjects. Frequency spectra of these acoustic emissions were obtained with two different procedures: real‐time recording or calculation of the spectrum of time‐averaged emissions. The two procedures give different input–output curves for click‐evoked acoustic emissions. A phase‐locking mechanism is proposed to explain this difference. It is shown that the real‐time spectrum recording procedure can be used to measure tuning curves or to study the distortion product 2f1–f2. Experimental results indicate that sharply tuned emission generators are present in the human cochlea.
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43.64.Ha Acoustical properties of the outer ear; middle-ear mechanics and reflex
43.64.Kc Cochlear mechanics

Narrow‐band acoustic signals emitted by chinchilla ears after noise exposure

P. M. Zurek and W. W. Clark

J. Acoust. Soc. Am. Volume 70, Issue 2, pp. 446-450 (1981); (5 pages)

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Acoustic search in 21 ears of 17 chinchillas exposed to a variety of high‐intensity sounds revealed two instances of a spontaneous and continuous narrow‐band acoustic signal emanating from the ear. These signals, found in the 4–7 kHz region after exposure to an octave band of noise centered at 0.5 kHz, could be suppressed by external tones. The frequency selectivity and time course of the suppression effect suggest that these otoacoustic emissions originate from a vibratory source in the cochlea.
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43.64.Tk Physiology of sound generation and detection by animals
43.64.Ha Acoustical properties of the outer ear; middle-ear mechanics and reflex
43.66.Ki Subjective tones
43.80.Lb Sound reception by animals: anatomy, physiology, auditory capacities, processing

Adaptive focusing: A spatial distortion‐adaptive receiver

A. J. Claus and F. M. Labianca

J. Acoust. Soc. Am. Volume 70, Issue 2, pp. 451-459 (1981); (9 pages)

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Adaptive focusing is an array processing scheme in which the receiver is designed for adaptive detection of localized sources under conditions where the signal field distribution across the array aperture is distorted from that of a plane wave, but is unknown a priori. Complex propagation conditions, including slow medium fluctuations, coupled with source motion render the signal field non‐Gaussian. This property is brought into play in the design of the detector as a maximum‐likelihood receiver which actually exploits the slowness of the acoustic channel changes. The problem is to detect the signal in a spatially white Gaussian noise field. The receiver utilizes the spatial structure of the signal as it manifests itself in the maximum eigenvalue of the data sample covariance matrix. Temporal stability of the field distribution and signal‐to‐noise ratio are important parameters. These are incorporated into a Monte Carlo simulation which shows that for stable signals at threshold level in the output of a conventional beamformer degraded by 5 dB, adaptive focusing can perform within a fraction of 1 dB of a hypothetical ideal beamformer for which the signal field is known a priori. Decreased temporal stability coming about as a result of, say, increased source motion necessarily has an adverse affect on performance.
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43.60.Cg Statistical properties of signals and noise
43.60.Gk Space-time signal processing, other than matched field processing

Adaptive focusing: The likelihood ratio

F. M. Labianca and S. P. Lloyd

J. Acoust. Soc. Am. Volume 70, Issue 2, pp. 460-466 (1981); (7 pages)

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The problem of detecting a certain non‐Gaussian random acoustic signal field in a spatially white Gaussian noise field is solved within the formalism of classical detection theory. Two solutions of the problem are compared: the maximum‐likelihood or adaptive‐focusing (AF) receiver and the Neyman–Pearson or adaptive focusing optimal (AFO) receiver. Based on two criteria for comparison, it is found that the two receivers differ only slightly in performance with the AFO, of course, being the better. One of the criteria is based on the signal deflection ratio, and the other is the ROC (receiver operating characteristic) curve. The comparisons are done for a Monte Carlo simulation.
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43.60.Cg Statistical properties of signals and noise
43.60.Gk Space-time signal processing, other than matched field processing
43.30.Vh Active sonar systems

Uniqueness and linear independence of steering vectors in array space

Lal C. Godara and Antonio Cantoni

J. Acoust. Soc. Am. Volume 70, Issue 2, pp. 467-475 (1981); (9 pages)

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The study of array processing often involves the manipulation of steering vectors defined in array space. The question of uniqueness and linear independence of steering vectors often arises in the analysis of array processing algorithms and in their formulation. The paper examines the effect of array geometry on the mapping between directions in three‐dimensional space and steering vectors in array space. Conditions on array geometry for uniqueness and linear independence of steering vectors are established for linear, planar, and general arrays.
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43.60.Gk Space-time signal processing, other than matched field processing
43.30.Vh Active sonar systems
43.28.Tc Sound-in-air measurements, methods and instrumentation for location, navigation, altimetry, and sound ranging

Diffuse wave fields in solid media

Davis M. Egle

J. Acoust. Soc. Am. Volume 70, Issue 2, pp. 476-480 (1981); (5 pages) | Cited 7 times

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The diffuse‐field concept is applied to develop the power balance equations relating the time‐averaged energy densities associated with longitudinal and transverse waves to the diffuse‐field mode conversion coefficients and the power generated by sources in the medium. A solution for an impulsive source in a solid with stress‐free surfaces shows that, regardless of the initial partition of energy, the transverse‐wave energy density quickly increases to become the dominant energy form in the medium.
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43.55.Cs Stationary response of rooms to noise; spatial statistics of room response; random testing
43.20.Bi Mathematical theory of wave propagation
43.40.At Experimental and theoretical studies of vibrating systems
43.40.Le Techniques for nondestructive evaluation and monitoring, acoustic emission

Transverse‐vibration responses of nonuniform, internally damped cantilever beams. II

R. L. Kerlin

J. Acoust. Soc. Am. Volume 70, Issue 2, pp. 481-491 (1981); (11 pages)

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The Bernoulli–Euler theory of transverse beam vibration, suitably extended to take into account internal beam damping, is used to derive closed‐form expressions for the mechanical driving‐point impedance and force transmissibility of two types of nonuniform cantilever beams that are driven at their free ends by a sinusoidally varying point force. The two types of beams considered are ones: (1) truncated and of rectangular cross section with a linearly tapered depth and a breadth appropriately varied (hyperbolically) to maintain constant cross‐sectional area, and (2) composed of three stages, each of which is uniform but may vary arbitrarily from the others in cross section and proportionate length. Representative computations of the frequency dependence of impedance and transmissibility are plotted for beams having the same length and mass as an equally long and equally massive uniform reference beam. Significant attenuation or amplification of force transmissibility is observed, depending on the proportions of the beams and on whether their depth diminishes towards their fixed or free end. The measured resonance frequencies of a set of nonuniform aluminum alloy (6061‐T651) cantilever beams and the measured transmissibility of one other such beam agree closely with the predictions of theory.
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43.40.Cw Vibrations of strings, rods, and beams

Theoretical and experimental investigations of structureborne sound transmission through a ’’T’’ joint in a finite system

G. Rosenhouse, H. Ertel, and F. P. Mechel

J. Acoust. Soc. Am. Volume 70, Issue 2, pp. 492-499 (1981); (8 pages) | Cited 2 times

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See Also: Erratum

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The wave theory approach was used here for analysis of sound transmission through a ’’T’’ joint in a finite system consisting of rectangular beams. The analytical results were confirmed by experiments. A comparison of the sound transmission through a ’’T’’ joint in finite and infinite systems revealed completely different behavior. Various kinds of joints including a welded ’’T,’’ a ’’T’’ connected by screws, and a ’’T’’ incorporating a rubber layer were examined. Comparisons between the analytical and experimental results of structureborne sound transmission through the various joints showed that the analytical treatment developed is applicable for all types of ’’T’’ junctions described above.
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43.40.Cw Vibrations of strings, rods, and beams
43.40.At Experimental and theoretical studies of vibrating systems
43.20.Bi Mathematical theory of wave propagation

Radiating head flexure and its effect on transducer performance

John L. Butler, John R. Cipolla, and Willie D. Brown

J. Acoust. Soc. Am. Volume 70, Issue 2, pp. 500-503 (1981); (4 pages) | Cited 2 times

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The design of low Q underwater sound transducers often results in thin radiating head structures leading to flexural resonances that are close enough to the fundamental resonance to cause noticeable effects on it. An analysis of this flexure is presented along with a simple model that can be incorporated into transducer element and array interaction design programs. Results are presented for a particular case to illustrate the physical effects and the advantages (such as a reduction in the fundamental resonant frequency) and disadvantages (such as reduced performance above resonance) that are a consequence of flexural head motion.
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43.30.Yj Transducers and transducer arrays for underwater sound; transducer calibration
43.38.Ar Transducing principles, materials, and structures: general

Parabolic equation calculations versus North Pacific measurement data

J. S. Hanna and P. V. Rost

J. Acoust. Soc. Am. Volume 70, Issue 2, pp. 504-515 (1981); (12 pages) | Cited 1 time

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Calculations of transmission loss using a Parabolic Equation (PE) model are compared to measured transmission loss from a North Pacific propagation experiment. In the range‐dependent environment various combinations of two source depths, three receiver depths, and three frequencies permitted a thorough examination of the influence of PE model input parameters. Sensitivity of calculated transmission loss to variations in the input parameters is discussed and some omissions in the measurements are found to be significant when evaluating the model. When known measurement uncertainties are used to vary the model’s input, changes in the model’s predicted transmission loss are of the same scale as the differences seen in model‐to‐data comparisons. Repeatability of the measurement was determined by two comparable measurement events. Data‐to‐data statistical comparisons of those two measurement events are no better than the model‐to‐data comparisons. The data‐to‐data comparisons showed mean differences up to 1.5 dB with standard deviations between 1 and 3.5 dB. Model‐to‐data comparisons yield comparable means and standard deviations. It is concluded that the PE model permits comparisons with the data at a level of detail which demands more information from measurements than is available from even this unusually comprehensive data set.
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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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