bluuebunny/crossref_metadata_2025 huggingface dataset:
Effect of dc polarization on the CM and the apparent changing resistance in the guinea pig
Authors: Mountain D. C. , Geisler C. D. , Hubbard A. E.
Published: October 2005

The cochlear microphonic (CM) and the apparent changing resistance (CR) were measured in scala media of the guinea pig during dc polarization. In the majority of the animals the polarization effect on the cm fundamental was greater at high SPL than at low SPL. Polarization current levels which produced large changes in endolymphatic potential (EP) caused relatively small changes in CM. In anoxic animals, however, similar changes in EP result in much larger changes in CM. The phase of the CM is insensitive to EP under both conditions. The CR is also sensitive to dc polarization. At lower SPL's a polarizing current which increases the CM will decrease the CR and a current of opposite polarity will decrease the CM but increase the CR. At higher SPL the effect on the CR is much smaller and may even reverse direction slightly. We conclude that transduction theories involving only linear time-varying elements can not model the data. [This research was supported by the National Institutes of Health.]
Two-tone rate suppression in lizard cochlear-nerve fibers: Relation to auditory receptor morphology
Authors: Holton Thomas , Weiss Thomas F.
Published: October 2005

Apical and basal regions of the basilar papilla of the alligator lizard are morphologically distinct. Weiss et al. [Brain Res. 115, 71–90 (1976)] have shown that single apical and basal cochlear-nerve fibers have characteristically different tuning properties. We have investigated the phenomenon of two-tone rate suppression (TTRS), wherein the average discharge rate of a fiber to an excitatory tone is suppressed by the addition of a second (suppressing) tone of appropriately chosen frequency and level. Every apical fiber exhibited TTRS. No basal fiber demonstrated the phenomenon. We have studied the dependence of TTRS on the parameters of two-tone stimuli in apical fibers; the phenomenon is qualitatively similar to that reported in cat cochlear-nerve fibers. [Supported by NIH grants.]
Unmasking in the auditory nerve
Authors: Harris David M. , Dallos Peter
Published: October 2005

Unmasking as a psychoacoustical procedure was developed by Houtgast [J. Acoust. Soc. Am. 51, 1885–1894 (1972)]. It is a two-masker paradigm whereby a subject's masked threshold is determined with a probe tone (f1) preceded by two simultaneously occurring tones, one also at frequency f1, the other having variable frequency (f2). The f1 masker produces a forward masking effect on the probe and the degree of this masking is altered as the second masker influences the effectiveness of the first (f1) one. This paradigm has been applied to the study of single fiber responses and whole-nerve action potentials (AP) from the auditory nerve of the chinchilla. For fibers certain combinations of the maskers (f1=CF, and f2) yield more masking than the f1-masker alone. This occurs when the f2 masker is within the fiber's response area. When f2 is within the fiber's suppression area(s) the response to the probe tone is greater than with the f1 masker alone. That is, unmasking occurs under such conditions. The procedure applied to the AP yields a similar frequency dependence and thus makes possible the delineation of AP-suppression areas, in analogy with psychophysical and single unit two-tone suppression. [Supported by grants from the NINCDS.]
Some results from a preliminary study of British English speech rhythm
Authors: Hill D. R. , Witten I. H. , Jassem W.
Published: October 2005

The paper reports a statistical study that has been carried out in an attempt to gain insight into the nature and underlying causes of rhythm in spoken British English. Two sets of utterances were selected as specially suited to the study, being both widely available as published audio tapes and spoken carefully in an attempt to exemplify normally spoken British English for the benefit of foreign students of the language [Study Units 30 and 39 from M.A.K. Halliday, A Course in Spoken English: Intonation (Oxford University, Oxford, 1970)]. Segmental analyses were performed and the resulting data on segment durations together with additional (higher level) information were prepared for computer analysis. It was found that, although there was some tendency towards isochrony (equally timed rhythmic units), constraints of isochrony accounted for at most 10% of the durational structure of the rhythmic units. Much of the rhythmic structure (about 45%) is adequately accounted for by the mean durations of the constituent phonemes. Other factors play a minor role. [Work supported by the National Research Council of Canada.]
On the perception of contour tones
Authors: Zee Eric , Greenberg Steven
Published: October 2005

In certain Chinese dialects (e,g., Taiwanese and Shanghai), contour tones are frequently associated with increased vowel length relative to that of level tones. An experimental study explored the extent to which increases in the duration of the vocalic portion immediately preceding a terminal rise in fundamental frequency affects the subjective estimation of the degree of excursion and slope of the tonal contour. Toward this end, six synthetically generated variants of the vowel [i] were rated on their degree of “contourness”—defined as the amount of pitch change occurring over the final portion of the vowel. In each stimulus, the fundamental frequency of this terminal portion ascended linearly from 100 to 150 Hz over a 90-msec interval. Preceding the contour were vocalic segments of variable duration (0–160 msec) but of uniformly steady fundamental frequency (100 Hz). Preliminary results indicate a positive correlation between the length of the initial segment and the perceived degree of “contourness” associated with the terminal rise. [Research supported by NSF.]
Test of rhythm and intonation patterns: A pilot study
Authors: Koike K. J. , Asp Carl W.
Published: October 2005

A test of rhythm and intonation patterns (TRIP) was constructed by prerecording the nonsense syllable /ma/ in twenty-five different rhythm and intonation patterns. Twenty normal hearing children—ten 3 year olds and ten 5 year olds—imitated (reproduced) the pattern they heard. The stimulus-response pairs were judged by the tester and a panel; agreement was 96% and 100% on two children. The intrajudge agreement ranged from 90% to 98%. The test-retest agreement was 84% and 92% on two children. The reliability coefficient of measurement for individual test items was 0.85. For the between group comparison, the 5-year olds scored significantly higher (86%) than did the 3 year olds (60%). The 3 year olds had more difficulty on patterns that had a quick tempo, a pause, or more than four syllables. The falling to rising intonation patterns were more difficult than the rising to falling, for both “slow” and “fast” patterns. In comparing the test scores with receptive language scores, there was a significant correlation for 3 year olds, but not for 5 year olds.
Temporal control of utterance-initial and -final stop consonants
Authors: Westbury John R.
Published: October 2005

Velopharyngeal and oral closing and opening gestures for utterance-initial and -final stop consonants were monitored from lateral-view cinefluorographic (100 fps) data for one normal English-speaking male repeating nonsense disyllables in isolation. Utterance-initially, the coordination and durations of velopharyngeal and oral closures were affected by both voicing and place of articulation of stops. Closure durations were consistently greater for voiced than voiceless stops; and, oral closures were longest for velars, intermediate for alveolars, and shortest for labials. Velopharyngeal closure characteristically lagged oral closure for initial stops, by as little as 50 msec for labials and as much as 100 msec for voiced velars, but was invariably complete at least 60 msec prior to oral release. The temporal variability of utterance-initial closure durations was relatively stable across place and voicing categories, though somewhat greater than for utterance-medial stops. Utterance-finally, no consistent place-of-articulation or voicing-related trends were observed for control of release gestures. Instead, articulatory timing in the latter environment was idiosyncratic, and not surprisingly, temporal variability was substantially greater than for the same segments utterance-initially. The temporal control characteristic of pre- and postpausal stop consonants will be compared to control for stops occurring intervocalically, and will be discussed in terms of the production constraints imposed by voicing and place of articulation. [Work supported by NIDR grant DE-00853 and NSF grant BNS 77-07686.]
Variations in sentence duration due to lexical content, semantic emphasis, and text structure
Authors: Levine Arvin
Published: October 2005

Research on the relative durations of words within sentences leads to an algorithm that assigns fractions of the lexical duration, dependent, in a neutral context, on the grammatical function of each word. This algorithm accords, in general, with work by Klatt [J. Acoust. Soc. Am. 59, 1208–1221 (1976)]. One important difference is that we utilize duration information from the recorded isolation form. The durational effect of semantic emphasis can be accounted for in terms of word duration adjustments and additional silences surrounding specified words. Another extension of this algorithm predicts some of the deviation of an entire sentence (or utterance from the neutral context duration which we predict for it, by using elementary text structures (such as lists) that can be generally applied to paragraphs and other text units. Comparisons of predicted durations with observed durations, in utterances which were independently recorded for use in the Institute's computerized “Introduction to Logic” course [W.R. Sanders and R. Laddaga, J. Acoust. Soc. Am. 60 S76–S77(A) (1976)], are examined. [Work supported by NSF grant SED77-09698.]
Visual precueing of individual phonemes can facilitate speech perception
Authors: Johnston James C. , Streeter Lynn A.
Published: October 2005

Is the phoneme a psychologically accessible unit of speech perception? One approach to this question is to determine whether speech perception can be facilitated by advance information about individual phonemes. Naive, untrained subjects listened to monosyllabic words embedded in white noise. After listening to each word, subjects chose between two visually presented letters, one of which was the initial letter of the spoken word. (Each word began with a letter which has a reliable grapheme-phoneme correspondence.) On half of such trails, the letter alternatives were also presented as a precue before the spoken word. Results showed that precueing the alternative letters significantly improved performance. The improvement with letter precues (5.5% higher accuracy) was approximately as large as the improvement with whole-word precues, measured in a control condition (6.5% higher accuracy.) These results show that people are able to use the familiar English letter representations of phonemes as cues to improve perceptual processing. This phenomenon can be accounted for by a system of phoneme detectors whose thresholds are lowered by precueing the corresponding letter.
Cross-linguistic survey of vowel quality
Authors: Disner Sandra Ferrari
Published: October 2005

It cannot be presumed a priori that vowels in different languages which are transcribed with the same symbol have identical phonetic quality. Indeed, phonetics textbooks often note marked auditory differences along these lines, e.g., the more open quality of Dutch [ε] compared to English [e]. In order to determine precisely where in phonetic space the vowels of different languages lie, a cross-linguistic study was undertaken of the vowels of six related languages—English, German, Dutch, Norwegian, Swedish, and Danish. Each data set, drawn from published accounts, consisted of the vowels of at least six male speakers, and all were in similar contexts so that differences within languages could be compared with those between languages. As individual speaker differences tend to obscure the overall differences between languages, several different normalization procedures were used. Analysis of variance was used to test for significant differences between languages. Plotting of the normalized formant frequencies revealed that differences are at least in part related to the number of vowels in the system and to their phonological relationships. [Work supported by NSF.]
Speaker verification by human listeners over several speech transmission systems
Authors: McGonegal C. A. , Rabiner L. R. , McDermott B. J.
Published: October 2005

An experiment was performed to find out how well human listeners could determine whether or not two different utterances were spoken by the same speaker. The speech was coded in three ways: high-quality PCM (natural speech), linear prediction encoding (LPC), and ADPCM at 24 kbps. For this experiment all combinations of these three coding methods were used. A group of 16 reference speakers (customers—8 males, 8 females) along with 78 test speakers (16 reference and 62 imposters) were used in the experiment. Two test utterances were used: one for the male speakers and one for the female speakers. The 30 naive listeners who participated in the experiment were required to make same/different judgments. The results will be discussed in terms of the imposter acceptance error rates (miss rates) and customer rejection error rates (false-alarm rates) for the different test conditions.
Speaker identification: Effects of vocal disguise upon listener performance
Authors: Reich Alan R.
Published: October 2005

This research was designed to investigate the effects of vocal disguise upon speaker identification by listening. The experiment consisted of fixed-sequence pair discriminations. The listeners were asked to decide whether the two sentences were uttered by the same or different speakers and to rate their degree of confidence. The speakers produced two sentence sets utilizing their normal voice and five disguises. One number of each pair in the task was always undisguised; the other member was either disguised or undisguised. Two listener groups were trained for the task: a group of 24 undergraduates and a group of six doctoral students and professors of Speech and Hearing Sciences. Both groups of listeners were able to discriminate speakers with a moderately high degree of accuracy (92% correct) when both members of the stimulus pair were undisguised. The inclusion of disguised speech samples in the stimulus pair significantly interfered with performance (59%–81% correct depending upon the particular disguise). These results are similar to previous results utilizing spectrographic speaker-identification tasks [Reich et al., J. Acoust. Soc. Am. 60, 919–925 (1976)].
High resolution-cross sensor beam forming for a line array
Authors: Bucker Homer P.
Published: October 2005

The methods of high resolution spectral analysis can be used for high resolution cross-sensor beam forming. In this paper two high resolution techniques (maximum entropy and exact decomposition) are applied to the cross-sensor field. Sample calculations illustrate the effectiveness of these methods in the case of a realistic multiwave acoustic field plus narrow-band Gaussian noise.
Economical computation of array gain of large lattice acoustic arrays in anisotropic sea noise
Authors: Martin Gordon E.
Published: October 2005

The performance of an underwater acoustic array is dependent upon its gain in the presence of noise. While isotropic noise is often utilized for simplistic considerations of system performance, such analyses are of limited validity because ambient sea noise is significantly directional. Thus, it is necessary to evaluate the behavior of arrays having many elements and multi-dimensional configurations in realistic anisotropic sea noise. Such computations have been very expensive using very large computers and impossible or impractical with other computers. This paper describes a technique comparable to a multi-dimensional FFT for radically reduced cost for the computation of array gain of lattice-shaped arrays of one, two, or three dimensions. The computational aspects are discussed. Typical results are presented for various horizontal and vertical arrays and compared with the author's prior work reported at EASCON '75 and the Ninth International Congress on Acoustics, 1977.
Robust sequential detection using acoustical arrays in undefined noise
Authors: Dwyer R. F.
Published: October 2005

A technique is introduced which allows the design of a robust sequential detector in undefined noise based on quantiles. For each sampled output of M hydrophones, a set of m quantiles from the unknown noise field are estimated and used to partition the observation space into known probability regions which assure distribution-free performance. In addition, the effects of hydrophone noise variations and non-uniform spatial noise distributions are removed from degrading performance. It is assumed that the noise samples are both spatially and temporally independent and the signal is a plane wave arriving from a specific direction. Once the m quantiles are estimated a partitioned log likelihood ratio (PLLR) statistic is formed assuming a shift of scale alternative. The quantiles are estimated in real time using a stochastic approximation algorithm which allows the PLLR to adapt to slowly changing noise distributions in order to maintain its distribution-free performance. An efficiency measure for the sequential partition detector is calculated and it's shown to be bounded by Fisher's Information. The equations for the average sample number and operating characteristic function are given in terms of the number of quantiles estimated. For practical implementation, it is shown that only a small number of quantiles need be estimated to give high efficiency. For example, estimating only two quantiles gives an efficiency of 0.65 in Gaussian noise. Efficiencies as a function of m are given for Gaussian as well as non-Gaussian noise distributions. [Work supported by CNM.]
Maximum entropy array processing
Authors: Skinner D. P. , Childers D. G.
Published: October 2005

The maximum entropy spectral analysis technique is applied to the problem of bearing estimation using a high-resolution linear array. It is demonstrated via computer simulations that the spatial response to an individual point scatterer increases in width in the presence of nearby targets as well as with decreasing signal-to-noise ratio. It is also shown that while the spatial response to a single point target is typically much narrower for maximum entropy processing, the minimum angular separation required for the resolution of adjacent targets is only slightly less (roughly a factor of one-half) than that of a conventional beamformer even at high signal-to-noise ratios (26 dB). These experimental results are compared with theoretical derivations of the conventional linear beamformer array and the maximum entropy array for both beamwidth and resolution. The trends observed in the computer simulations are in general agreement with the approximate theoretical results.
Digital interpolation beamforming; concept and theory
Authors: Pridham Roger G.
Published: October 2005

For many applications, digital beamformers require an input rate which is significantly higher than that required for waveform reconstruction to achieve an adequate set of synchronous beams. Typically, this high rate is realized in the A/D conversion of the sensor outputs. A novel alternative, discussed in this paper, is to sample the sensors at a rate consistent with the Nyquist criterion and implement vernier beamformer delays by digital interpolation. This technique, which is referred to as digital interpolation beamforming, permits an efficient partitioning between A/D converter and cable bandwidth requirements and digital processing complexity. The basic structure of an interpolation beamformer is presented. It is shown that interpolation filtering and beamforming can be interchanged to minimize digital processing. Beam-pattern degradation due to interpolation error is derived and interpreted for the special case of a line array.
Digital interpolation beamforming; implementation and application
Authors: Mucci R. A.
Published: October 2005

Efficient methods of implementing a digital interpolation beamformer which utilize interpolation filters to reduce the rate at which the outputs of the array elements are sampled are discussed for both lowpass and bandpass signals. The bandpass case is an important application since the data can be sampled at a low rate consistent with the bandwidth of the signal. In both applications interpolation is performed efficiently using finite impulse response (FIR) digital filters. The advantages of an FIR filter over recursive filter implementations are presented. The interpolation filter thruput, which depends upon whether the interpolation is performed at the beamformer input or output, is shown to be related to the size of the array and beam steering requirements. However, the interpolation is usually performed most efficiently at the beamformer output for bandpass applications which require the modulation of the sampled signals. Efficient location of the modulation is also addressed. Examples of beampatterns are presented for digital interpolation beamforming and compared to those obtained with synchronous and nonsynchronous beamforming techniques.
Iterative algorithm and performance of amplitude adaptive beamformer in interference and noise environment
Authors: Wang H. S. C.
Published: October 2005

Following the concept of an optimum amplitude shading beamformer in an interference and noise environment reported earlier [H. S. C. Wang, J. Acoust. Soc. Am. 56, S18(A) (1974); 61, 1251 (1977)], an iterative algorithm for the realization of such a signal processor is 6formulated. This algorithm essentially follows the steepest descent path of the total received interference and noise power at the beamformer output subject to the constraint that the main beam signal response is maintained at a fixed level. It is demonstrated that, for an interference incident on the first minor lobe of a line array 4.4 wavelength long, the algorithm reduces interference power to below ambient noise level in fewer than ten iterations and improves output SNR by 11 dB within 100 iterations. If the interference incidence is within the main beam (6.5° from MRA), approximately 40 iterations are needed to provide a 17-dB improvement of output SNR. When the interference is very close (2.5°) to MRA, the algorithm can still improve SNR by 6 dB within 500 iterations. Numerous beam patterns are exhibited to demonstrate their metamorphosis during the course of iterative optimization. It is estimated that the effectiveness/cost ratio of such a processor would be substantially higher than conventional adaptive beamformers.
Accuracy of pulse arrival time measurement by a hard limiting receiver
Authors: Young J. W.
Published: October 2005

The accuracy with which the time of arrival of a pulse can be determined is a fundamental limitation on the performance of an acoustic positioning system. The theoretical accuracy of a linear matched filter receiver is well known. However, acoustic transponder navigation systems do not normally employ optimum receivers of this type. Because of the wide range of ambient noise conditions under which such devices must operate, a simple constant false alarm rate (CFAR) receiver which includes a bandpass limiter is commonly used. This paper presents an approximate theoretical analysis of the timing accuracy of a hard limiting receiver. The relationships among accuracy, false alarm rate, and input signal-to-noise ratio are considered. The model ia shown to give good agreement to measured arrival time statistics of an experimental receiver. A comparison of the performance of the hard limiting versus the matched filter receiver is also given.
Utilization of zero-crossing statistics for sonar signal detection
Authors: Higgins R. C. , Tufts D. W. , Kemp K. A.
Published: October 2005

The use of zero-crossing (ZC) statistics to improve signal detection performance in the presence of reverberation and ambient noise was theoretically and experimentally examined. The probability density of ZC intervals for the sum of two sinusoids of different amplitudes, different frequencies, and randomly chosen relative phases was calculated and experimentally measured—apparently for the first time. These results, extending the work of Longuet-Higgins, Rice, and Blachman, showed that the use of ZC statistics is not appropriate for processing against reverberation that is stronger than the desired signal. For the signal-plus-ambient noise case, several ZC statistics—mean frequency, mean ZC interval, and standard deviation of the ZC interval—were critically examined. It was found that the use of these estimated ZC statistics requires a significantly greater SNR for reliable detection than does the incoherent matched filter. [Supported by NUSC.]
Method to estimate surface ship range from a stationary hydrophone
Authors: Dixon W. C. , McGrath J. R.
Published: October 2005

An algorithm has been developed to calculate a ship's closest point of approach (CPA) to a stationary single hydrophone. The algorithm first tracks a selected spectral component of radiated ship's noise, yielding the component's Doppler-time history. Using this information initial estimates are then made of ship's speed (v), CPA, time at CPA (t0), component frequency at CPA (f0) and the Doppler shift. The frequency-time history is fitted with a curve using the Gauss—Newton nonlinear least square method. Final estimates of v, CPA, fo and t0 result as Δv, ΔCPA, Δf0 and Δt0 tend to zero. This technique is applied to recording of a supertanker passage.
Acousto-optic processing for real-time and adaptive memory applications
Authors: Berg N. J. , Udelson B. J. , Lee J. N.
Published: October 2005

Two different types of acousto-optic correlators, a real-time large time-bandwidth (∼3000) device and a memory correlator, have been developed. The real-time correlator utilizes successive interactions of a cw laser sheet beam with two “live” surface acoustic waves (SAW's) propagating in the same direction along long (14 cm) lithium niobate (LiNbO3) and bismuth germanium oxide (BGO) crystals that are placed side-by-side. The almost 2:1 SAW velocity ratio for these crystals permits a signal propagating in the LiNbO3 to pass one propagating in the BGO. Focusing the emerging SAW-modulated laser beam onto a nonlinear photo-diode results in an output proportional to the correlation integral. In the memory correlator, a SAW pattern is stored initially (as changes of the index of refraction in LiNbO3) by illuminating the LiNbO3 with high-intensity, short-duration pulses of green laser light during propagation of the (10 MHz) SAW signal to be stored. Subsequently, a live signal may be correlated with the stored signal by modulating a low-intensity laser sheet beam with both the live and stored signals, and focusing the emerging beam onto a photo-diode. Memory storage times of a few months have been demonstrated. Both device types have successfully correlated complex waveforms, such as FM chirps and Barker codes.
Some noise control problems peculiar to nuclear generating facilities
Authors: Putnam Robert A.
Published: October 2005

The ability to confidently predict the noise environment in a nuclear generating facility presently under construction is complicated by the design constraints of physical layout, the shape of enclosed volumes, and compliance with safety considerations. Three particular problems are discussed. First, compliance with seismic qualifications required that HVAC fan/coil units be rigidly connected to the floor, without resilient isolators. Calculation of the structure-borne noise transmitted into adjacent spaces is discussed. Secondly, proper control of HVAC duct-borne noise in critical safety areas involves selection of nonstandard materials. The rigid performance criteria are discussed. Thirdly, the reverberant field decay in the highly irregular spaces inside the reactor containment building presents a unique problem. The volumes to be considered are an elliptically domed circular room and two right circular cylinders, each having another right circular cylinder centered within it for the entire height of the room. Calculated results by two different methods are presented and the required assumptions for each are discussed.
Silencing boiler blowdown for pipe cleaning operations
Authors: Bradley Wayne E. , Strong W. F.
Published: October 2005

Prior to startup, many fossil fueled power stations clean the boiler piping to the turbine generator with high pressure steam. The steam is vented to the atmosphere in a series of several minute “blows” over a several week period. Steam conditions of up to 900 psi and 5000000 lb/hr are not unusual and sound levels from this activity can easily exceed 100 dBA at property lines. The traditional approach to the community noise associated with the blow has been to warn the neighbors, but the community reaction to the noise is still often significant. This paper describes the methodology and equipment used for quieting the blows. Sound level measurements of a number of steam blows, both silenced and unsilenced are presented and the resultant benefits are discussed.
Internal friction of metal spheres showing the effect of the anisotropy of the component metals
Authors: Mason W. P. , Marfurt K. J. , Boshere D. N. , Kuo J. T.
Published: October 2005

By using resonant vibrations of metal spheres, separation of contributions involving different amounts of shear, hydrostatic, and longitudinal strains can be obtained. Birch has shown that there is a 9–1 ratio between the Q of the radial mode to the Q of the torsional modes if there is no loss connected with the hydrostatic component (Assumption consistent with dislocation losses). We have measured this ratio for several cubic metals (Al, Fe, Ta, Nb, W, V, and α brass) and have found that this ratio varies systematically with the elastic anisotropy. It is shown that the difference in the elastic constants of adjacent grains introduces a shearing stress in the body of the material which adds a component of the internal friction proportional to the anisotropic factor minus unity. For the radial mode a much larger effect is to be expected than for the longitudinal vibration since the induced shearing strain in the grains is of the same order, while the ratio of the radial dissipation to the torsional dissipation is 19-th.
Generation of subharmonics in liquid helium in the presence of negative ions
Authors: Carey R. F. , Rooney James A. , Smith C. W.
Published: October 2005

Considerable evidence, both theoretical and experimental, indicates that the generation of subharmonics in liquids is associated with cavitation. To investigate details of this phenomena we have measured the acoustic spectra generated by a 1 MHz PZT4 transducer immersed in liquid helium as a function of displacement amplitude and temperature in the presence of ions. The threshold curves (subharmonic amplitude versus displacement amplitude of the fundamental) show onset and plateau structure which vary with temperature. Specifically, the subharmonic amplitude is increased by as much as an order of magnitude when the temperature is decreased below the lambda point. Ions produced by a tritium source in the liquid were injected into the sound field. Above the lambda point the subharmonic amplitude was enhanced by negative ions by as much as a factor of four. Below the lambda point the subharmonic amplitude decreased by as much as a factor of two. Although the thresholds were unchanged to within 10% by the presence of ions, the initial slope of the threshold curves was decreased by ions below Tλ but apparently unchanged above Tλ . These observations will be discussed in terms of the electron bubble model for the negative ion. [This work was supported by the Air Force Office of Scientific Research.]
Effect of ultrasonic cavitation inception on the electrical properties of the transducer
Authors: Shalis Edward V.
Published: October 2005

The appearance of vaporous cavitation occurring in a 24-ft3 tank, containing air saturated tap water, is described by its effect on the electrical properties of a four-element 4X-47 tonpilz transducer. The inception of cavitation on the face of the transducer implies the existence of weak points in the liquid. Therefore, the transducer is loaded nonsymmetrically and the amount of power that can be radiated into the liquid is severely limited. The change in the impedance may be measured as a function of power input and the onset of cavitation determined. Cavitation was induced at 31.8 and 41.3 kHz at a transducer voltage of 50 V, as determined by harmonic distortion measurements. The magnitude of the complex electrical impedance, and the phase angle between the transducer voltage and the current, change sharply in the cavitation region and level off as the input power is further increased. In the cavitation region, the electrical impedance slope was −7Ω/V. and the electrical resistance slope was −6.2Ω/V. [Work supported by NOSC.]
Forced oscillations of nonspherical bubbles
Authors: Hsieh D. Y.
Published: October 2005

The oscillation of a gas bubble in liquid under externally applied sinusoidal pressure field is studied. When the applied pressure amplitude is small and the forcing frequency is not in the neighborhood of the resonance frequency of the bubble, the familiar linear results are obtained. When the applied amplitude is not small, there is a tendency to generate both the subharmonic oscillations of the spherical mode and the surface waves on the bubble. The theoretical study is carried out by using both the normal mode analysis and the variational method. It is found that the variational method can lead to approximate solutions in a simpler way than other conventional methods. Experimental implication of the theoretical results is also discussed.
Continuum modeling of elastic wave propagation in laminated wave guides
Authors: Nayfeh Adrian H.
Published: October 2005

A continuum mixture theory with microstructure is developed for elastic wave propagation in laminated wave guides. The theory leads to simple governing equations for the actual medium which retain the integrity of the propagation process in each constituent but allow them to coexist under derived interactions. The utility of the resulting equations is demonstrated by studying the response of the medium to both harmonic and transient pulses. In the case of harmonic loadings the dispersion relations are found to correlate well with some derived exact solutions and experimental results. For transient loadings, solutions are derived by means of Laplace transform techniques. Analytical inversion of the transforms is possible only for the limiting cases of “weak” and “strong” coupling. The limit of strong interaction leads to the coalescence of both waves; in this case the laminated wave guide behaves like a single but anisotropic continuum whose properties are a geometrically weighted averages of the properties and volume fractions of the composites constituents. For the general coupling case, however, results are demonstrated by a direct numerical inversion of the transforms. [Work supported by NSF.]
Experimental study of the effects of water repellent treatment on the acoustic properties of Kevlar
Authors: Smith C. D. , Parrot T. L.
Published: October 2005

The effects of water repellant treatment of the acoustic properties of a bulk absorber have been investigated. The bulk absorber consisted of a synthetic fibrous material (Kevlar 29, style 1270) manufactured by DuPont. The propagation constant, γ, and the characteristic impedance Zc were measured over the frequency range 0.5 to 3.5 kHz. The measurements were conducted before and after a controlled application of water repellent treatment (“Zepel” DA-24M), also manufactured by DuPont. A comparison of test results indicated changes in both γ and Zc. These changes in γ and Zc generally increased with frequency. At the highest test frequency, the attenuation constant (real part of γ) and phase constant (imaginary part of γ) decreased by about 8% and 10%, respectively, when the treatment was applied. Corresponding changes in the real and imaginary parts of Zc at the test frequency of 3.5 kHz were about 6% and 25%, respectively.
Scattering of a sound wave from a grating of hollow elastic bars in a viscoelastic layer
Authors: Radlinski R. P. , Libuha J. J.
Published: October 2005

The problem of the scattering of a normally incident plane sound wave by an infinite grating of hollow elastic bars in a viscoelastic layer between two fluid haft spaces is solved by the method of partial domains. The solution in the layer is formulated in terms of inhomogeneous plane dilatational and shear waves. The attenuation is found to be always normal to the plane of the grating. The evanescent waves in the viscoelastic medium necessarily cannot propagate parallel to the grating as is the case for an elasticlike or fluid medium. The dependence of the grating transmissivity on parameters of the individual bars and the complex-valued, frequency dependent, material properties of the layer is analyzed quantitatively. By using measured viscoelastic material properties as input to the model, comparisons of predictions of transmissivity are made with experimental data. [Work supported by NAVSEA.]
A parametric sonar performance calculator
Authors: Pederson Thomas B.
Published: October 2005

Predicting the difference frequency source levels and beamwidths which will be developed by a given parametric sonar projector is a somewhat complicated procedure involving a number of calculations and the use of several curves or nomograms. We will introduce a slide calculator we have developed which will compute in a relatively straightforward manner these quantities for a parametric sonar operating in the unsaturated regime.
Acoustic/seismic vehicle locator
Authors: Brooks George W. , Reel Gwynn M. , Huffnagle Norman P. , Marietta Martin
Published: October 2005

We are investigating an automatic range measuring/target classification method based on the relative velocity differences of airborne acoustic and seismic waves. The technique appears to be a promising way to measure target range to an accuracy commensurate with present target interdiction techniques. In brief, assumptions on the relative velocities and their variances appear to be sound: errors of ±20% do not materially degrade interdiction probabilities. C. F. Richter and others have determined that the relation between the three seismic waves (Prompt-P, Shear-S, and Rayleigh-R) is virtually invariant for many media. One of the methods being investigated takes advantage of this invariance to measure range. A closed-loop system using a 512-element CCD serial analog delay as one element of a seismic/acoustic correlator shows promise as a viable method. Replacing the seismic/acoustic inputs by the P and R waves (polarized at right angles to each other and having wide velocity dispersion), this correlator would work as a single medium range system. Coupling this target range-measuring method with a Martin Marietta-developed target classification technique that also provides range measurement using Doppler and the amplitude increase function on the narrow-band spectra of predetermined targets, target interdiction by this multipath method leads to a system that is very immune to countermeasures and natural events.
Microphone coupler for noise source location
Authors: Botsford James H.
Published: October 2005

Reducing noise generation requires determining where sound waves originate. Machinery connected to other structures can form a complex vibrating noise source that is difficult to analyze. Measuring sound levels close to surfaces suspected of radiating noise can help in finding sound sources. But results are sometimes inconclusive because of sound reflections from the surface, especially where standing waves exist. Using accelerometers to detect surface vibration overcomes the sound reflection problem but is a complicated and time-consuming technique, requiring sophisticated equipment and calculations for definitive results. To overcome some of these difficulties, a flexible coupler was developed for the microphone of a sound level meter. The microphone fits into the throat of the flaring coupler and the mouth of the coupler is pressed against any surface suspected of radiating sound. The flexible mouth conforms even to curved surfaces and seals out extraneous noises. Sound radiated from the surface closing the coupler mouth is collected in the cavity, sensed by the microphone and indicated on the sound level meter. This useful device allows positive identification of vibrating surfaces responsible for sound generation.
Parameter identification in nonclassical linear mechanical systems
Authors: Michalopoulos C. D. , Wheeler L. T.
Published: October 2005

The identification problem for a system whose motion is governed by [A]{x̄(t)}+[B],{x̄(t)}+[C]{x(t)}={f(t)}, where [A], [B], and [C] are not the classical inertia, damping and stiffness matrices, is studied. Specifically, the eigenvalues of the associated reduced system of equations are determined by processing in the frequency domain the response {x} to a random excitation {f}, whose power spectral density is assumed constant (or reasonably flat). This is accomplished through the minimization of a mean-square error function involving the theoretical and measured power spectral densities of the response. It is shown that the level of the assumed white-noise excitation need not be known. [Work supported by NASA.]
Time difference pitch resolution in humans and animal echolocation capabilities
Authors: Johnson R. A.
Published: October 2005

The resolution capability of human subjects for time difference pitch stimuli was measured for delays between 1 and 10 msec. The stimuli consisted of wide-band, shaped, pulse pairs with random pair repetitions to avoid other periodicity effects. A PEST procedure was employed to determine the resolution thresholds at a 75% performance level. The resolution capability varied from 19 μsec at 1 msec to 53 μsec at 10 msec. Analogous echolocation experiments with bats and dolphins suggest that dolphin performance levels more closely parallel the human pitch resolution capabilities. Additional time difference pitch experiments were performed with Doppler tolerant pulse stimuli similar lo pulses used by bats. A particular value of Doppler compression was chosen to test for correlation processing. The result was a pitch shift which is not explained by correlation. [Work supported by the Naval Undersea Center now part of the Naval Ocean System Center.]
Speaker invariant characterizations of vowels, liquids, and glides using relative formant frequencies
Authors: Skinner Toby E.
Published: October 2005

Sperry Univac is developing a system for automatically recognizing words in conversational speech. The linguistically oriented procedure identifies sounds by their acoustic-phonetic correlates, forms an hypothesized sequence of sound segments for each phrase, and then matches lexical entries where similar sound segments occur. Problems to be overcome include the style of speech (exhibiting extreme reduction, coarticulation, and dynamic range), the restricted frequency range of the speech signal (telephone bandwidth), and the variety of speakers (including widely divergent males and females). To accomplish speaker independence in the acoustic-phonetic identification processes, relative (as opposed to absolute) formant frequency characterizations of vowels, liquids, and glides are being employed. While a particular sound has disparate formant frequencies when produced by different speakers, the relative relationships between the formant frequencies are nearly invariant. The nature of the relative formant frequency measures and the results achieved will be discussed.
Frequency perturbation and duration characteristics of pathological and normal speakers
Authors: Murry Thomas , Doherty E. Thomas
Published: October 2005

The purpose of this study was to determine if measures of speaking fundamental frequency and its perturbation, as well as selected durational aspects of speech may be used to discriminate normal from abnormal phonation, specifically, in five pairs of talkers matched for age. A spoken sentence and sustained /a/ provided the sample for the acoustic analyses. Fifty glottal waveforms from each subject were measured to determine the mean perturbation of the fundamental frequency for the sustained vowel. The best predictor variables for these subjects from the seven variables analyzed were found to be the vocalic duration of the sentence segment and the mean perturbation for sustained vowels.
Class of short-term integrators that produce time-intensity trades for noise bursts
Authors: Penner M. J.
Published: October 2005

Recent work on temporal masking and temporal resolution indicates that acoustic stimuli are combined to a degree determined by their temporal proximity. For broadband stimuli, the process of combining inputs has been modeled as resulting from an exponential integrator with a time constant of 2–4 msec. Such models seem to suggest that stimuli more than, say, 20 msec apart are treated largely as separate inputs. In contrast, the classical view, based on the results from time-intensity trades, holds that broadband noise stimuli are integrated together for 100–200 msec. This difference in the period over which stimuli are integrated therefore varies by about two orders of magnitude according to research in the two areas. We show one way to eliminate this apparent discrepancy by demonstrating that there exists a class of short-term integrators that can produce the time-intensity trade data. [ Work supported by NIH.]
Microcomputer-based system for speech synthesis from a large English text data base
Authors: Jaskie Cynthia A. , Russell Kenneth L. , Snider Donald L.
Published: October 2005

The basic goal for the described project is selective and interactive production of synthetic speech from a large data base such as a training manual. In a training environment at a sophisticated computer control console, an operator should learn more effectively and efficiently if both aural and visual senses are used and if his eyes and hands are free to concentrate on the panel before him. The system described includes an operator-directed control strategy to selectively retrieve that part of text which is to be spoken, a considerable amount of text preprocessing in non-real time to obtain synthesizer control parameters, and a microcomputerized cascade terminal analog speech synthesizer. The preprocessing of the English text utilizes an extensive set of translation rules, dictionary lookup of complete words, and morph lexicon lookup of parts of words to produce 123 distinct allophone designators with corresponding stress, pitch, and duration designators. Storage requirements for the preprocessed speech control parameters to be supplied to the real-time speech synthesized are approximately 120 bits/sec of synthesized speech. Example recordings of the synthetic speech produced by this system will be presented. [Work supported by the Naval Ocean Systems Center.]