bluuebunny/crossref_metadata_2025 huggingface dataset:
Layer model for assessing acoustic refraction effects in echo sounding
Authors: Phillips P. D. , Richner H. , Nater W.
Published: October 2005

Acoustic refraction effects in the atmosphere by wind-speed and temperature gradients are investigated using a simple multilayer model for plotting the ray path of a signal transmitted by an acoustic echo sounder (AES) and scattered back at various heights to be received monostatically. It is found that while nonperfect backscattering has negligible effect on the scattering cross section for acoustic waves in a turbulent medium, the effect on the ’’angle of arrival’’ and the Doppler shift is important. A purely horizontal wind can produce a Doppler shift in a vertically pointing monostatic AES which for a wind of 20 m/sec at 1 km can simulate vertical wind speeds of 50 cm/sec. For nonvertical sounding, this effect is of little importance. The generally accepted first-order approximation for the angle of arrival of the scattered signal appears, as a result of this more detailed analysis, to be incorrect. The problems of the height resolution of a horizontal scatter layer using a nonvertical AES are briefly considered in terms of the antenna characteristics as well as the length of the transmitted pulse.
One-dimensional model for acoustic absorption in a viscoelastic medium containing short cylindrical cavities
Authors: Gaunaurd G.
Published: October 2005

The equations of motion of linear viscoelasticity (Kelvin–Voigt Model) in cylindrical coordinates are reduced to an axisymmetric plane-stress situation valid for short hollow cylinders. The analysis models the deformations and oscillations occurring in the annular region conceptually constructed around one of the holes in a multiperforated rubber sheet to be used as an underwater acoustic absorber. The eigenvalue problem for the ring’s natural frequencies is solved in two cases: first, when both rims of the ring are subjected to a stress boundary condition; second, when the inner rim (r=a) is stress free and the outer one (r=b) is fixed. For incompressible materials with Poisson’s ratio very close to 0.5, the first few eigen-wave-numbers are numerically computed and plotted in nondimensional form versus the b/a ratio, in both cases, and the resulting sets of curves are shown to be interlaced, the ones from the first case being below those of the second. The transient vibration problem for the displacement ur is solved by the Laplace-transform method in the first case only. The finding that the frequencies at which absorption is maxima are the resonant frequencies of the cavity, is verified. We show a simple technique to estimate the amount of absorption present at these eigenfrequencies, and the estimates reasonably check with experimental meaurements within the limitations of this model (i.e., below critical damping).
Rotated Y-cut quartz crystal with two different electrodes treated as a one-dimensional acoustic composite resonator
Authors: Boersma F. , van Ballegooyen E. C.
Published: October 2005

In this paper a quartz crystal coated with electrodes of different materials is considered as a one-dimensional composite acoustic resonator. The piezoelectric properties of quartz are incorporated in the model. The model leads to a transcendental equation from which the resonant frequencies can be calculated. It will be shown that earlier develolped models can be derived from this more general model by making the proper approximations. The difference between the theoretical resonant frequencies, as calculated from the various models, is discussed. To express this difference a systematic error is defined which proved to be negligable for small mass loading, corresponding to a relative frequency shift <1%, and in the case independent of the electrode material. In case there is a larger mass loading, the discrepancy between the previous models and the model presented in this paper can be of the order of 5%–10%, depending on the properties of the electrode material, i.e., the acoustic impedance and on the ratio of the thicknesses of the electrodes.
Super-resolution ultrasonic imaging by combined spectral and aperture synthesis
Authors: Sato Takuso , Ikeda Osamu
Published: October 2005

In conventional sonar systems, high azimuth resolution is achieved by synthesis of the aperture, and range resolution by utilization of pulsed waves. When the aperture synthesis is done over a fairly large area, however, the duration of the pulse should be reduced to less than a few wavelengths in order to keep the range resolution within the azimuth resolution. The use of such short pulses makes difficult the precise detection of the amplitude and phase of the waves reflected from the object, mainly because of the sensitivity to noise caused by the short averaging time in the heterodyne detection process. In this paper a new method is proposed which realizes the impulselike range resolutions through the synthesis of detected signals for several frequencies of continuous waves. The fundamental idea is to synthesize the Fourier transform of a very short pulse by using several frequencies of ultrasonic waves. This method gives the synthesis in range, while the conventional synthesis gives the azimuth direction. The principle of the method is presented together with an analysis of the resolution, a noise analysis, and a discussion which includes a numerical example.
Parametric excitations of linear systems having many degrees of freedom
Authors: Nayfeh A. H. , Mook D. T.
Published: October 2005

The method of multiple scales is used to analyze parametrically excited, linear systems having many degrees of freedom and distinct natural frequencies. Explicit second-order expressions are obtained for the characteristic exponents which yield the transition curves. Various combination resonances are treated. The results are applied to the buckling of free-clamped columns under the influence of sinusoidally varying axial loads.
Real-time bispectral analysis of gear noise and its application to contactless diagnosis
Authors: Sato Takuso , Sasaki Kimio , Nakamura Yoichi
Published: October 2005

Real-time bispectral analyses of gear noises are carried out, intending to distinguish abnormal states from the normal one without stopping the machine. Firstly, a real-time bispectral analyzer which consists of bandpass filters, multipliers, and integrators is constructed. Then, the noises of gears are analyzed. The results show that when scorings have been grown on the gear surfaces the moduli of bispectra of the nosies are reduced markedly comparing to those of normal state, while conventional spectral analysis fails to distinguish the difference. And it is shown that this fact can be used to diagnose the conditions of gear surfaces. Finally, a proper stochastic model of gear noises based on physical considerations as well as the spectral and bispectral characteristics obtained by our study, is proposed.
Rejection of flow noise using a coherence function method
Authors: Chung J. Y.
Published: October 2005

A method has been developed for rejecting transducer flow-noise interference. The method makes use of coherence-function relations between simultaneous pressure measurements at three transducers in the signal field and extracts from the flow-noise background the power spectrum of the signal as received at each transducer. The signal in question can be periodic or stationary random. The coherence function relations are derived on the basis of a multivariate, random-process model. The theory indicates that for a single source or for a group of completely coherent sources, the three transducers can be placed at arbitrary locations in the signal field. For a group of sources that are not completely coherent, the transducers must be located close together relative to the distance from the group. In any case, however, the flow noises at the transducers must be mutually independent. Successful results were obtained in laboratory tests of the method. In these tests a 10–15-dB reduction in flow-noise interference was attained. The method can also be used to reduce other types of noise such as instrument electronic noise.
Directional characteristics of cylindrical receiving arrays with nonuniform hydrophone response
Authors: Queen William C.
Published: October 2005

In a recent paper [W. C. Queen, ’’The Directivity of Sonar Receiving Arrays,’’ J. Acoust. Soc. Am. 47, 711 (1970)] the problem of determining the directional characteristics of abitrary configurations of point-source hydrophones was considered and the results given for the special case of a cylindrical array. It was assumed that the array was transparent and consisted of onmidirectional, uniformly weighted elements. In this paper, we extend these earlier studies to include both the effects of nonuniform weights as well as nonomnidirectional hydrophone response. The question of optimal weighting with respect to maximum directivity index is considered and closed-form solutions obtained. These analyses are applied to the special case of an array formed on a cylindrical surface in a ring-stave configuration and results given in the form of beam patterns and directivity curves. Included are the omnidirectional case as well as the special case where the hydrophones exhibit cardioid response functions.
Auditory-filter characteristics derived from direct-masking data and pulsation-threshold data with a rippled-noise masker
Authors: Houtgast T.
Published: October 2005

The masker was ’’rippled noise,’’with a power spectrum (intensity on a linear frequency scale) shaped according to a sinusoidal function. The test signal was a pure tone. The masking effectiveness of the rippled noise depends on the position of its peaks and troughs relative to the test-tone frequency, but this dependence decreases for high-ripple densities (thus, for ripples with small peak-to-peak distances along the frequency scale). The results of masking experiments as a function of the position and the density of the ripple around the test-tone frequency allow an estimation of the degree of auditory frequency resolution in terms of a filter characteristic. Several masking paradigms were applied: direct masking, forward masking, and pulsation threshold. It was found that the degree of frequency resolution estimated both from pulsation-threshold data and from forward-masking data is substantially higher than that estimated from direct-masking data. The difference is about a factor of two when expressed in terms of the bandwidth of the auditory filter. This difference is interpreted as reflecting spectral sharpening by lateral suppression which apparently manifests itself only in threshold measurements where masker and test tone are presented nonsimultaneously.
Growth of masking and the auditory filter
Authors: Weber Daniel L.
Published: October 2005

The threshold for a sinusoidal signal (1, 2, and 4 kHz) centered in the ’’notch’’ of a broadband masker was determined as a function of notch width for five noise spectrum levels (10, 20, 30, 40, and 50 dB SPL). For narrow notch widths the signal-to-noise ratio at threshold remains constant as a function of level, which according to the critical-ratio hypothesis implies an auditory filter of constant bandwidth. For wide notch widths the signal-to-noise ratio at threshold increases as a function of level and this implies an auditory filter of increasing bandwidth. If the estimates of the filter bandwidth obtained for wide notch widths are used to predict thresholds for broadband noise, the corresponding signal-to-noise ratios will increase as a function of noise spectrum level. The predicted increase in signal-to-noise ratio is very small, however, and provides a good description of most available data. In fact, the predicted increase equals that observed by Reed and Bilger [J. Acoust. Soc. Am. 53, 1039–1044 (1973)]. The increase in filter bandwidth has significant consequences only when the signal and masker are widely separated in frequency; for other conditions, the assumption of a constant filter permits accurate predictions of performance.
On the dissociation of spectral and temporal cues to the voicing distinction in initial stop consonants
Authors: Summerfield Quentin , Haggard Mark
Published: October 2005

It has been claimed that a rising first-formant (F1) transition is an important cue to the voiced–voiceless distinction for syllable-initial, prestressed stop consonants in English. Lisker [J. Acoust. Soc. Am. 57, 1547–1551 (L) (1975)] has pointed out that the acoustic manipulations suggesting a role for F1 have involved covariation of the onset frequency of F1 with the duration, and hence the frequency extent, of the F1 transition; he has argued that effects hitherto ascribed to the transition are more properly attributed to its onset. Two experiments are reported in which F1 onset frequency and F1 transition duration/extent were manipulated independently. The results confirm Lisker’s suggestion that the major effect of F1 in initial voicing contrasts is determined by its perceived frequency at the onset of voicing and show that a periodically excited F1 transition is not, per se, a positive cue to voicing. In further experiments, the relative levels and the frequencies at the onset of voicing of both F1 and F2 were manipulated. The influences on the perception of stop-consonant voicing that resulted were determined specifically by the frequency of F1 and not by its absolute or relative level or by the overall distribution of energy in the spectrum. The results demonstrate a complementary relationship between perceptual cue sensitivity and production constraints: In production, the VOT characterizing a particular stop consonant varies inversely with the degree of vocal-tract constriction, and hence with the frequency of F1, required by the phoneme following the stop; in perception, the lower the frequency of F1 at the onset of voicing, the longer the VOT that is required to cue voicelessness. In this way, the inclusion of F1 onset frequency in the cue repertoire for voicing, and the establishment of the cue trading relationship, reduce the problem of contextual variation that would be met were VOT alone, or some other amalgam of cues, the only basis of the voicing distinction.
Method for evaluation of the acoustical impedance of a black box, with or without mean flow, measuring pressures at fixed positions
Authors: Kathuriya M. L. , Munjal M. L.
Published: October 2005

The transmission-line or the impedance-tube method for the measurement of the acoustic impedance of any termination involves a search for various minima and maxima of pressure. For this purpose, arrangement has to be made for the microphone to travel along the length of the impedance tube, and this complicates the design of the tube considerably. The present paper discusses a method which consists in evaluating the tube attenuation factor at any convenient frequency by making use of measured SPL’s at two (or more) fixed locations with a rigid termination, calculating the tube attenuation factor and wave number at the required frequency of interest with or without mean flow (as applicable), and finally evaluating the impedance of the given termination by measuring and using SPL’s at three (or more) fixed locations. Thus, the required impedance tube is considerably smaller in length, simpler in design, easier to manufacture, cheaper in cost and more convenient to use. The design of the tube is also discussed. Incidentally, it is also possible to evaluate the impedance at any low frequency without having to use a larger impedance tube.
Experimental assessment of the Mindlin–McNiven rod theory
Authors: McNiven Hugh D. , Mengi Yalcin
Published: October 2005

The three-mode theory due to Mindlin and McNiven, [J. Appl. Mech. 27, 145–151 (1960)] governing axisymmetric motions in a circular rod, is appraised by comparing responses predicted by it with experimental data obtained by Miklowitz and Nisewanger [J. Appl. Mech. 24, 240–244 (1957)]. The problem studied involves a semi-infinite rod, made of 24S-T aluminum alloy, subjected to pressure applied to the end of the rod. The two sets of responses are compared at various stations along the rod. To make the comparisons meaningful, it was necessary to recognize that the pressure applied experimentally had a finite rise time, however short; to make an estimate from the responses of what that rise time might be; and then apply this time distribution of pressure in evaluating the theoretical responses. Comparison shows that the Mindlin–McNiven theory predicts the responses at stations farther than one diameter from the end of the rod. The matching is accurate not only at the head of the impulse but at points representing large times following the first disturbance.
Speed of sound in deuterium oxide relative to normal water as a function of temperature and pressure
Authors: Chen Chen-Tung , Millero Frank J.
Published: October 2005

The speeds of sound of liquid deuterium oxide relative to normal water were measured at 1 atm over the temperature range of 4°–60°C and at pressures up to 1000 bars over the temperature range of 5°–50°C. Three velocimeters were used to measure the sound speeds of deuterium oxide at 1 atm relative to normal water. These three sets of 1-atm data were fitted to a polynomial equation of temperature with a standard deviation of ±0.09 m/sec. The results of the high-pressure measurements were fitted to an equation of the form (with a standard deviation of ±0.07 m/sec) (UH2O−UD2O)−(U0H2O−U0D2O ) =AP+BP2+CP3+DP4, where UH2O and UD2O are, respectively, the speeds of sound in normal water and deuterium oxide (superscript zero denotes 1 atm). A, B, C, and D are temperature-dependent parameters which were determined by the least-squares method. Although the values of U0D2O obtained in this study are on the average 0.6 m/sec lower than the data of Wilson [J. Acoust. Soc. Am. 33, 314–316 (1961)], the pressure effect on the relative sound speeds from the above equation [(UH2O−UD2O) −(U0H2O−U0D2O)] agree with the work of Wilson to an average of 0.26 m/sec. One atmosphere adiabatic and isothermal compressibilities for deuterium oxide which are reliable to ±0.005×10−6 bar−1, were calculated from the sound-speed data. The results were compared with the data of various workers. The values of isothermal compressibility obtained in this study are systematically higher than the work of Fine and Millero [J. Chem. Phys. 63, 89–95 (1975)] (average 0.05×106 bar−1); however, they agree with the data of Millero and Lepple [J. Chem. Phys. 54, 946–949 (1971)] and Emmet and Millero [J. Chem. Eng. Data 20, 351–356 (1975)] to within ±0.02×10−6 bar−1.
Absorption of sound in air: High-frequency measurements
Authors: Bass H. E. , Shields F. Douglas
Published: October 2005

The absorption of sound in air at frequencies from 4 to 100 kHz in 1/12 octave intervals, for temperatures from 255.4° K (0° F) to 310.9° K (100° F) in 5.5° K (10° F) intervals, and at 10% relative-humidity increments between 0% and saturation has been measured. The values of free-field absorption have been analyzed to determine the relaxation frequency of oxygen for each of the 92 combinations of temperature and relative humidity studied and the results are compared to an empirical expression. The relaxation frequencies of oxygen have been analyzed to determine the microscopic energy-transfer rates.
Properties of feature detectors for VOT: The voiceless channel of analysis
Authors: Miller Joanne L.
Published: October 2005

Binaural adaptation was combined with a dichotic ear-monitoring task to estimate the output function of the voiceless detector sensitive to VOT and to determine how this function is modified by selective adaptation. On each test trial, a voiced stop was paired with one of a set of voiceless stops, which spanned a range of VOT values. The percentage of voiceless responses per pair, which reflects both the number of correct voicing judgments on voiceless targets and the number of voiceless intrusions on voiced targets, was measured before and after adaptation with a voiced stimulus and a variety of voiceless stimuli. Before adaptation, the percentage of voiceless responses per pair systematically increased with an increase in the VOT value of the voiceless stimulus of the pair. Adaptation resulted in a shift in the percentage of voiceless responses per pair in the predicted direction, with progressively larger shifts occurring for voiceless adapting stimuli with higher VOT values. These results indicate (1) that the output of the detector is graded signal whose magnitude is an increasing function of VOT value and (2) that adaptation depresses the entire output function from its preadaptation level, with the amount of depression determined by the extent to which the adapting stimulus excites the detector.
Factor analysis of tongue shapes
Authors: Harshman Richard , Ladefoged Peter , Goldstein Louis
Published: October 2005

A new analytic procedure PARAFAC has been applied to the description of the shape of the tongue in English vowels. The procedure models the data in terms of a unique set of possibly explanatory factors. It solves in parallel for factors in several data sets, simultaneously describing the differences among data sets in terms of different relative involvement of these common factors. Tracings were made of x-rays taken during the pronunciation of 10 English vowels by five speakers. The positions of the tongue in these 50 vowels were quantized in terms of 13 superimposed grid lines. PARAFAC analysis shows that the data can be described in terms of two factors. One factor generates a forward movement of the root of the tongue accompanied by an upward movement of the front of the tongue. The second factor generates an upward and backward movement of the tongue. Movements from front to back vowels involve decreasing amounts of factor one. Movements from high to low vowels involve decreasing amounts of factor two. Different speakers use the two factors to different degrees which may be associated with their individual anatomy. The correlation between the observed data and that predicted by the model is greater than 0.96.
Toward automatic identification of the language of an utterance. I. Preliminary methodological considerations
Authors: House Arthur S. , Neuburg Edward P.
Published: October 2005

A procedure is described for possible use in automatic identification of the language being spoken. The method is to assume that the gross linguistic classes of a language are characterized by certain statistical constraints, namely, they are probabilistic functions of a Markov chain; and that these gross classes can be identified automatically with usable accuracy. The present discussion is in the form of a feasibility experiment and deals only with artificial data (data not derived from acoustic signals). The results indicate that statistical models, as described, can discriminate amongst languages.
Operator formulation for maximum-likelihood estimation of acoustic emitter delay and Doppler
Authors: Fisher B. , Weill L. R.
Published: October 2005

A general, compact, and quickly derived solution to maximum-likelihood estimation of acoustic emitter delay and delay rate can be achieved by representing the signal propagation to each of M sensors as a matrix operator containing the unknown parameters. A general solution is first derived in which the emitted signal is unknown but nonrandom. This solution can incorporate the combined effects of propagation parameters such as wide-band Doppler, frequency-dependent attenuation, multipath time delays, and signal phase inversion due to some types of reflection. The result is simplified if only wide-band Doppler, single-path delays, and frequency-independent attenuation are present and, in the case of two sensors, it reduces to simple cross correlation. The matrix approach is also applicable to casses where the emitted signal is modeled as a random process.
Cochlear micromechanics—a mechanism for transforming mechanical to neural tuning within the cochlea
Authors: Allen Jont B.
Published: October 2005

A linear mathematical model is proposed which will account for the differences observed between mechanically measured data of Rhode (1971) for basilar membrane motion, and the responses of neural tuning curves (Kiang et al., 1974). We show that theoretical tuning curves may be derived from mechanical responses by forming the difference between the pressure across the basilar membrane and its displacement. Some ramifications of this proposal are discussed. We then propose a hypothetical physical model which could perform such a function.
Monaural and between-ear temporal gap detection: I. Single gaps
Authors: Pollack Irwin
Published: October 2005

Temporal gap thresholds were obtained for four signal combinations: monopolar (M) and random-polarity (R) pulse trains; and, presentation of signals before and after a temporal gap to the same (S) ear or to different (D) ears. When a monaural spectral analysis of the signal across the gap is possible (conditon M-S, gap sensitivity improves with longer and longer signals on each side of the gap, with extremely fine temperal gap sensitivity (∠1 μsec). Sensitivity decreases markedly for the other three conditions (∠103-fold for condition R-S; ∠104-fold for condition M-D or R-D). There, gap sensitivity decreases with longer signals about the temporal gap. The poorer sensitivity with longer random-polarity signals for monaural gap detection may be related to greater masking by the longer signals. The poorer sensitivity with longer signals for binaural gap detection may be related to binaural ’’precedence’’ effects. Strong asymmetrical effects upon gap detection were also noted.
Noncategorical perception of stop consonants differing in VOT
Authors: Carney Arlene Earley , Widin Gregory P. , Viemeister Neal F.
Published: October 2005

The discriminability of bilabial stop consonants differing in VOT (the Abramson–Lisker bilabial series) was measured in a same–different task, an oddity task, and a dual response, discrimination–identification task. Subjects showed excellent within-category discrimination in all three tasks after a moderate amount of training in a same–different task with a fixed standard and with feedback. In addition, discrimination performance continuously improved with increasing stimulus difference for both intra- and intercategory comparisons. Also, subjects were able to alter their identification responses so that well-defined category boundaries fell at arbitrary values determined by the experimenters. These results are not compatible with a strict interpretation of the categorical perception of stop consonants.
Perceptual switching in bilinguals
Authors: Elman Jeffrey L. , Diehl Randy L. , Buchwald Susan E.
Published: October 2005

Three groups of subjects, monolingual English speakers, monolingual Spanish speakers, and English–Spanish bilinguals, identified a series of naturally produced syllables which varied in voice onset time (from /ba/ to /pa/). The two monolingual groups differed substantially in their identification performance, with English speakers tending to label most of them as /ba/ and Spanish speakers tending to label most of them as /pa/. The bilingual subjects heard the test stimuli in both an English and a Spanish context, each designed to induce a particular language ’’set.’’ These subjects perceived a reliably greater number of the test items as /ba/ in the English context than in the Spanish. The magnitude of this perceptual switching effect depends on the listener’s degree of bilingualism.
Speaker identification by long-term spectra under normal and distorted speech conditions
Authors: Hollien Harry , Majewski Wojciech
Published: October 2005

Two experiments were carried out in which long-term spectra were extracted from controlled speech samples in order to study the effectiveness of that technique as a cue for speaker identification. In the first study, power spectra were computed separately for groups of 50 American and 50 Polish male speakers under fullband and passband conditions; an n-dimensional Euclidean distance technique was used to permit identifications. The procedure resulted in high levels of speaker identification for these large groups—especially under the fullband conditions. In a second experiment, the same approach was employed in order to discover if it was resistant to the effects of variation in speech production—at least under laboratory conditions. Talkers were 25 adult American males; three different speaker conditons were studied: (a) normal speech, (b) speech during stress, and (c) disguised speech. The results demonstrated high levels of correct speaker identification for normal speech, slightly reduced scores for speech during stress and markedly reduced correct identifications for disguised speech. It would appear that long-term speech spectra can be utilized to identify individuals from their speech—even in relatively large groups—when they are speaking normally or under stress (of the type studied); LTS does not appear to be an effective technique when voice disguise is employed. While this approach was utilized only in controlled laboratory experiments, it is suggested that it may have some merit for use in applied situations or as one of the features in a multiple-vector approach.
Analog of two-tone suppression in whole nerve responses
Authors: Dallos Peter , Cheatham Mary Ann
Published: October 2005

Some similarities are shown between single-unit two-tone suppression areas and those derived from the difference of compound action potential tuning curves obtained with simultaneous and forward masking.
Propagation of noise along a finite impedance boundary
Authors: Chessell C. I.
Published: October 2005

The purpose of this paper is to show that the essential properties of the propagation of noise over real soil surfaces can be modeled using the properties of fibrous absorbent materials and the appropriate theory of sound propagation along a finite impedance boundary. This work is an expansion of the original proposals of Delany and Bazley [J. Sound Vib. 16, 315–322 (1971)]. The model predictions are shown to be in good agreement with a variety of experimental ground absorption measurements. The application of the model to the measurement of aircraft reverse thrust noise is described.
General analysis of ocean eddy effects for sound transmission applications
Authors: Henrick R. F. , Siegmann W. L. , Jacobson M. J.
Published: October 2005

An analytical approach is used to obtain an approximate solution for deep-ocean mesoscale eddies, including depth-dependent effects. The solution is used in the development of an environmental-acoustics model which relates acoustically relevant quantities, such as sound-speed and current distributions, to eddy parameters. Parameters of the model are depth of influence, radius, rotational direction and maximum speed, and translational velocity. An application to a particular Gulf Stream ring is made, and the resulting current and sound-speed structures are shown to be in qualitative agreement with observations. Then, general results are presented for rotational current structure, maximum horizontal sound-speed change, and maximum SOFAR-axis elevation as functions of eddy radius and peak current speed. It is shown explicitly how these quantities change significantly with eddy size and strength. This model provides a basis for subsequent analytical studies of sound transmission through an arbitrary eddy or eddy field.
Models for the amplitude fluctuations of narrow-band signals and noise in the sea
Authors: Urick R. J.
Published: October 2005

The sound from a steady distant sinusoidal source in the sea fluctuates in amplitude because of multipath interferences, surface motion, internal waves, microstructure, and other propagation effects. Such fluctuations are found to follow a Rician or modified-Rayleigh distribution having as a parameter the fraction of random power in the received signal. At the output of conventional sonar processor—namely, a narrow-band filter, a squarer, and an integrator—the fluctuation statistics are determined by the propagation processes occurring in the sea between source and receiver. On the other hand, ambient sea noise is found, from analyses of field recordings, to have fluctuation statistics determined by the processor itself; ambient noise samples at the processor output obey a chi-square distribution having a number of degrees of freedom equal to twice the bandwidth-time product of the processor, as would be expected from a Gaussian input. The two distributions—Rician power for signals and chi-square for noise—while formally different, have remarkable similarities in the limits. In short, the output fluctuation statistics of narrow-band signals and Gaussian ambient noise can apparently be predicted from estimates of the degree of randomness introduced by the prevailing propagation conditions, and from a knowledge of the processor, respectively, provided these statistics remain stationary during the analysis period.
Vibrotactile frequency for encoding a speech parameter
Authors: Rothenberg Martin , Verrillo Ronald T. , Zahorian Stephen A. , Brachman Michael L. , Bolanowski Stanley J.
Published: October 2005

Frequency of vibration has not been widely used as a parameter for encoding speech-derived information on the skin. Where it has been used, the frequencies employed have not necessarily been compatible with the capabilities of the tactile channel, and no determination was made of the information transmitted by the frequency variable, as differentiated from other parameters used simultaneously, such as duration, amplitude, and location. However, several investigators have shown that difference limens for vibration frequency may be small enough to make stimulus frequency useful in encoding a speech-derived parameter such as the fundamental frequency of voiced speech. In the studies reported here, measurements have been made of the frequency discrimination ability of the volar forearm, using both sinusoidal and pulse waveforms. Stimulus configurations included the constant-frequency vibrations used by other laboratories as well as frequency-modulated (warbled) stimulus patterns. The frequency of a warbled stimulus was designed to have temporal variations analogous to those found in speech. The results suggest that it may be profitable to display the fundamental frequency of voiced speech on the skin as vibratory frequency, though it might be desirable to recode fundamental frequency into a frequency range more closely matched to the skin’s capability.
Experimental scheme for analyzing the dynamic behavior of electroacoustic transducers
Authors: Egolf David P. , Leonard Robert G.
Published: October 2005

Methods for determining dynamc properties of acoustic sources have received little attention in the literature. This paper describes the development and verification of a computer-aided experimental procedure for obtaining four parameters that characterize the dynamic behavior of electroacoustic transducers. This technique, herein called the ’’two-load’’ method, is first demonstrated on a simple electrical circuit. The reader is then introduced to the more difficult problem of applying the two-load method to an actual transducer. For this application, a small electroacoustic transducer, commonly used as a hearing-aid receiver, is employed. Essentially, the two-load method consists of applying two widely different acoustic loads to the transducer in questions and measuring, in turn, the response of the transducer to each of these loads. Practical application of this method is complicated by the fact that some laboratory instruments introduce inherent amplitude and phase distortion which msut be accounted for. Finally, an experimental scheme is devised to test the validity of the two-load method and the accuracy of data derived by this method. Experimental data and data derived by application of the two-load method agree quite well over the frequency range from 10 to 10 000 Hz.
Method for measuring i
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u acoustic impedance of marine sediments
Authors: Shirley Donald J.
Published: October 2005

As part of a program to measure acoustic properties of marine sediments in situ, a device has been developed to measure the acoustic impedance ρc of such sediments. With concurrent measurement of sound speed c, the bulk density ρ of relatively undisturbed sediments can be obtained. Acoustic impedance is obtained by measuring the electrical driving impedance at resonance of a small ceramic transducer element in contact with the medium. Relationship of the electrical impedance z to the acoustic impedance ρc as well as the effect of temperature on the measurement has been examined and will be reported.
Acoustical determinations of bubble-size spectra
Authors: Medwin Herman
Published: October 2005

The exact integration of the equations for acoustical cross sections of bubbles is considered to show when the dependence of the bubble number on radius can be determined by scatter and attenuation experiments in a liquid containing a broad spectrum of bubble radii.
Fiber-optic hydrophone
Authors: Bucaro J. A. , Dardy H. D. , Carome E. F.
Published: October 2005

We have demonstrated the feasibility of employing directly acousto-optic interactions in an optical fiber to produce a sensitive acoustic detector. Our technique utilizes the phase modulation of an optical beam in a submerged optical fiber coil by sound waves propagating in a fluid. Analysis of our results indicates that the sensitivity of this technique compares well with that of the best available hydrophone.
Pitch ratings of harmonic series
Authors: Pollack Irwin
Published: October 2005

When the components of a harmonic series are of equal amplitude, pitch ratings are determined almost exclusively by the fundamental frequency of the series. Only by extreme modification of the amplitudes of the several components was it possible to modify the pitch ratings assigned to the harmonic series in a direction opposite to that determined by the fundamental frequency.
Design of an auditorium where electroacoustic technology is fully available: Exhibition Hall of Yamaha Sportland TSUMAGOI
Authors: Yamaguchi Kiminori , Fujita Takashi , Kawakami Fukushi , Sotome Hiromi
Published: October 2005

The semiopen-type Exhibition Hall with more than 3500 seats was built in the Yamaha Sportland TSUMAGOI with the area of 1 700 000 m2. Since this hall is expected to be used for indoor sports and musical performances, it was required to make designs from the point of view quite different from the traditional auditoriums. In view of the control of sound field by the electroacoustical means, therefore, new tools have been adopted such as (1) the movable proscenium loudspeaker, (2) movable sound absorbing canvas, (3) artificial reverberation adding system, (4) multichannel public address console and delay/reverberation control console, and (5) other various types of electrical and spatial reverberation adding devices. This paper describes the idea of design for the Exhibition Hall meeting the requirements of the latest popular music involving electric musical instruments and explains the details of design and acoustic characteristics of those electronic equipments.
Intermodulation product pair 2f
l+f
h: Masking and growth
Authors: Erdreich John
Published: October 2005

To examine the question of whether the distortion product 2fl−fh is unique—whether a special nonlinear mechanism is responsible for its production—estimates of 2fl−fh and of 2l+fh amplitudes were obtained in three listeners. Threshold shifts measured with a tone-on-tone masking paradigm for each distortion product suggest that each have similar properties and are masked paradigm or each distortion product suggest that each have similar properties and are masked by other tones as if they had a physical realization in the cochlea. Their masking is not a function of stimulus fine-time structure. The 1 dB/dB growth of 2fl−fh was not changed by rendering it inaudiable as previously suggested. Possible explanations of the 1 dB/dB growth of 2 fl−fh compared with the accelerated growth of 2 fl+fh are examined.
Elastodynamic diffraction by a periodic rough surface (stress-free boundary)
Authors: Fokkema J. T. , van den Berg P. M.
Published: October 2005

The reflection of an elastic wave by a rough stress-free surface with a periodic profile has been investigated rigorously. The problem is formulated in terms of an integral equation for the particle displacement at a single period of the boundary surface. Numerical results pertaining to the reflection of either an incident P wave or an incident SV wave for a sinusoidal profile are presented.
Surface-wave diffraction and guidance
Authors: Rulf Benjamin
Published: October 2005

Diffraction of surface waves at a discontinuity of a surface impedance under oblique incidence is analyzed by means of singular integral equations. A closed form for the reflection coefficient is derived. The phenomena of nonradiation and total reflection of the surface wave are discussed. The results are used for the analysis of guided modes on a strip surface waveguide.
Fiber-optic detection of sound
Authors: Cole J. H. , Johnson R. L. , Bhuta P. G.
Published: October 2005

The response of an optical fiber in water to low-frequency acoustic waves was investigated experimentally and compared with analytical results. A single-mode optical fiber was used to pass a laser beam through a tank in which an approximately plane acoustic wave was produced. A change in the optical index of refraction of the fiber creates an effective path-length change for the optical beam which results in a phase shift of the optical beam with respect to a reference beam unaltered by the acoustic field. By mixing the phase-shifted beam with a reference beam of constant phase in a photodetector the phase variation at the acoustic frequency is detected. Theoretically, the sensitivity of a fiber-optic interferometer should be independent of frequency. Experimental results confirm this.
Rare earth iron magnetostrictive underwater sound transducer
Authors: Meeks S. W. , Timme R. W.
Published: October 2005

A low-frequency, resonant, longitudinally vibrating piston-type rare earth iron underwater sound transducer using rods of 27% terbium – 73% dysprosium iron (Tb0.27Dy0.73Fe1.95) is described. The performance of the rare earth iron transducer is compared with the performance of the same transducer with identical ceramic active elements. Data on transmitting current response, transmitting voltage response, free-field voltage sensitivity, coupling coefficient, efficiency, and linearity are presented. An equivalent circuit is presented which predicts the performance of the transducer in the presence of eddy currents and demagnetization. The predicted performance of a laminated transducer is compared with that of an unlaminated transducer. Advantages and disadvantages of the rare earth iron material are discussed.
Analysis of overtone modes in monolithic crystal filters
Authors: Tiersten H. F.
Published: October 2005

A previous treatment of overtone modes in trapped energy resonators is extended to the case of two-pole monolithic crystal filters. The asymptotic dispersion relations for the fundamental and odd overtone coupled thickness-shear and thickness-twist waves near cutoff along with the simple approximate edge conditions at junctions, which were employed in the earlier treatment of resonators, are applied in the analysis of two-pole monolithic crystal filters. Both thickness-shear and thickness- twist coupling directions are considered. The influence of piezoelectric stiffening, electrode mass loading, and electrical shorting is included in the analysis. A lumped parameter representation of the admittance matrix for the two-port device is obtained. Numerical results are presented for a few geometrical configurations.
Bias in bearing estimation resulting from source motion
Authors: Carter G. Clifford
Published: October 2005

A moving source is characterized by its velocity and range and bearing to a passive array. Source motion can cause biased estimates of bearing to result when the received signals are not preprocessed (or preprocessed inaccurately) to correct for source motion. The extent of bearing bias is shown to be a function source velocity (speed and direction relative to the array position) and range relative to the effective baseline. An example calculation is provided.
Review of the ARPA Speech Understanding Project
Authors: Klatt Dennis H.
Published: October 2005

In September of 1976, four speech understanding systems were demonstrated, signifying the end of a five-year program of research and development sponsored by the Advanced Research Projects Agency (ARPA). The best performance was displayed by the Harpy system developed at Carnegie–Mellon University. Harpy satisfied a set of design goals that were specified at the beginning of the program, including the gal of understanding over 90% of a set of naturally spoken sentences composed from a 1000-word lexicon. After defining the nature of the speech understanding problem, the four systems are described and critically evaluated. Based on this review, a structure for a next-generation speech understanding system is proposed and parts of it are considered as a possible model of the early stages of speech perception. The perceptual model addresses the issue of lexical access and includes a decoding network composed of expected spectral sequences for all word strings of English.
Sound scattering from thin shells in the Kirchhoff approximation
Authors: Brill Donald W. , Ugin?ius Peter , George Jacob , Chwieroth Frank S. , Überall H.
Published: October 2005

We calculate the acoustic backscattering cross section of a thin, air-filled elastic cylindrical shell using the Kirchhoff approximation. The spurious shadow-boundary reflections contained in the conventional formulationof the approximation are removed by the use of the stationary-phase method. Comparison with exact backscattering results based on the Watson–Sommerfeld transformation show a strong presence of creeping-wave contributions even for very thin shells.
Weak-shock solution for underwater explosive shock waves
Authors: Rogers Peter H.
Published: October 2005

The initial pressure wave measured at modest distances from an underwater explosion is often modeled as a spherical shock wave with an exponential decay. A closed-form analytical ’’weak-shock’’ solution for the subsequent propagation of such a wave has been obtained. The resulting simple formulas for peak pressure and decay constant as function of reduced range allow the prediction of the amplitude and initial slope of the wave given only the amplitude and decay constant of the original exponential shock and the density, sound speed, and parameter of nonlinearity of the water. The results are in good agreement with the Kirkwood–Bethe theory, existing measurements, and the widely used experimentally based semi-empirical similarity formulas. An expression which gives a close approximation to the shape of the waveform as a function of distance is also derived.
Minimum auditory movement angle: Binaural localization of moving sound sources
Authors: Perrott D. R. , Musicant A. D.
Published: October 2005

In the first experiment, subjects were asked to discriminate whether a sound was emanating from a moving or stationary source. The minimum audible movement angle (MAMA) thus defined was observed to increase as the source velocity increased. MAMA ranged from a low of 8.3° with the slowest velocity employed (90°/s) to a high of 21.2° with the fastest velocity (360°/s). In the second experiment, subjects were asked to localize where the moving source was, at signal on and offset. The results indicate that the apparent onset is displaced in the direction of motion and the amount of this displacement is directly related to source velocity. Less consistent results were observed with signal offset. The present results suggest that the binaural system is relatively insensitive to motion.