bluuebunny/crossref_metadata_2025 huggingface dataset:
Feature enhancement of vibration signatures utilizing synchronous spectrum averaging
Authors: Bozich D. J. , Keller A. C.
Published: October 2005

Vibration response signals containing periodic components such as obtained from rotating equipment, gears, etc., can be effectively analyzed in the frequency domain. The averaging of spectra wherein the Fourier transform operation is synchronized with a particular periodic component, will reinforce that frequency component and all of its harmonics, while all nonsynchronous data will tend to be nonreinforced, thereby enhancing the synchronous spectral components. Features such as gear teeth and turbine blade passing frequencies, etc., and their related sidebands, can be extracted as signatures for baseline comparison and fault evaluation. This paper will discuss signatures obtained from averaged spectra and the related set of synchronous spectra; i.e., a synchronous average spectrum for each periodic component of interest.
Interpretation of sodar measurements: relevant properties of the atmosphere's planetary boundary layer
Authors: Thomson D. W.
Published: October 2005

Assuming a Kolmogorov spectrum of turbulence, the acoustic scattering cross section may be written as a function of incident wavelength, scattering angle, and the magnitude of the root-mean-square turbulent temperature and velocity fluctuations. In the “surface” layer SL (max ≃ 3–30 m height) of the atmosphere the statistical properties of turbulent velocity, temperature, humidity, and pressure fluctuations above many surface types are well understood for a wide variety of meteorological conditions. Hence, it is possible using sodar to derive unique, quantitative measurements of SL atmospheric turbulence. Detailed information about planetary boundary layer PBL (max ≃ 300 m–2 km height) turbulence is not required in order to make continuous sodar measurements of vertical profiles of turbulence related atmospheric structure and winds (from signal Doppler shifts). Sodar is, thus, now routinely used, e.g., for air pollution related, micrometeorological research, and wind-shear detection measurements. Our perception of many complicated PBL structural features and processes has been greatly expanded by the now familiar vertical-time section sodar records of various “layers” and thermal convection. However, except for “unstable” conditions of free convection, SL turbulence models are not generally applicable for the quantitative interpretation of PBL sodar data. In order to realize the full indirect sounding potential of sodar during “stable” atmospheric conditions, it is necessary to determine the relationship(s) between observed “macroscale” features such as multiple layers (with and without waves), intermittant “patches” of turbulence and “bursts” of vertical momentum or heat transport and the space-time dependent properties of the turbulence in such situations. This paper first reviews the application of contemporary SL and PBL models to the interpretation of sodar measurements. Functional relationships between sodar observed and model implicit parameters are summarized. Finally, the application of sodar for development and verification of selected atmospheric models is discussed. [Research supported by EPA Grant R800397.]
Atmospheric turbulence as observed by in situ measurements and Doppler Sodar
Authors: Weili A. , Baudin F.
Published: October 2005

The rate of dissipation by turbulent fluctuations and eddy flux of momentum are measured in the atmospheric boundary layer with the Doppler Sodar of the CNET and compared with hot-wire anemometer measurements. Different hypothesis and methods to calculate turbulence parameters will be first discussed with special emphasis on the possibilities offered by a triple Doppler Sodar. Then typical observations of eddy parameters and of their evolution will be presented. [Work supported by CNET/CRPE/CNRS.]
Experimental study of an echosonde interferometer for remote sensing of atmospheric temperatures
Authors: Brown Edmund H. , Wright Wayne M.
Published: October 2005

We have constructed a preliminary version of the combined echosonde interferometer suggested by Brown and Little [E. H. Brown and C. G. Little, Nature (to appear)]. Here, we give a description of this device (which for brevity we call an “echoferometer”) and present some of the first results obtained with it. The fact that the formation of an acoustic interference pattern from a pair of coherent sources can be combined with the scattering of sound to a receiver by turbulent refractivity fluctuations (as in an echosonde) is of interest in itself. Moreover, application of the effect to remote sensing of atmospheric temperature profiles would add an extremely valuable tool to meteorological research if sufficient sensitivity could be achieved. We include some possible schemes, such as correlation detection, for progress towards such a goal, together with a sketch of an engineering feasibility study.
Application of acoustic sounding to estimating fusion in an atmospheric boundary layer
Authors: Kerman Bryan R.
Published: October 2005

An acoustic sounder system can be configured to provide estimates of the vertical profile of velocity and dissipation rate of kinetic and potential energy. A method based on the concepts of boundary-layer similarity theory is outlined by which the characteristic velocity in a freely convective boundary layer is deduced from the inversion height and the asymptotic value of the dissipation rate of kinetic energy near the inversion. These values, in conjunction with the mean velocity, are used to establish the normalized heights, downwind and cross-wind distances, and cross-wind integrated concentrations. The normalized lateral and vertical standard deviations are then deduced from empirically established laboratory relationships for nonbuoyant particulates. Preliminary analysis indicate a similar methodology may be applicable to the stable boundary layer.
Sodar-derived CT2 and CV2 compared with simultaneous, in situ aircraft measurements: nighttime and daytime case studies
Authors: Thomson D. W. , Coulter R. L. , Warhaft Z.
Published: October 2005

Simultaneous measurements of CT2 and CV2 have been made using the Penn State Doppler sodar system and comprehensively instrumented research aircraft. In each experiment the aircraft was vectored using air-ground ratio to the altitude (ranging from ≃ 150 to 600 m) zone of interest on the basis of the received sodar signals. Measurements have been made in both “layers” (with and without detectable turbulence and “waves”), and in convective “plumes”. CT2, and CV2 have been derived both from the calibrated monostatic-bistatic Doppler sodar and airborne sensors. In addition to the normal complement of sensors for state parameters, winds, and position, the aircraft was equipped with special, spaced (at sodar λ/2) hot-wire probes capable of measuring turbulent temperature and velocity fluctuations to about 2.5-cm resolution. Using the combined sodar, turbulence, and meteorological measurements both the turbulent and larger-scale characteristics of the atmosphere in the sounder volume were well defined. Discussion of the results includes suggestions for further experiments involving remote sodar and in situ aircraft measurements. [Research supported by EPA Grant R800397 and ERDA Grant E(11-1)-2463.]
Processing of airborne acoustic noise sources using passive range determination techniques
Authors: Brooks G. W. , Huffnagle N. P.
Published: October 2005

This paper presents a continuation of the passive range measurement efforts reported at the 90th conference. Previous work included the modeling of passive range determination using a single point receiver based on moving source observables. The outcome as reported last year was a parameterization of a hypothetical system that had the capability to determine range to source passively. This year, we have expanded the work to include building a fully operational system and field testing it against various moving sources. In addition, a pseudo-Fourier analysis section has been incorporated to accurately and correctly sort out and classify moving sources. This, coupled with a priori selection information coded into the system, enables it to perform those required functions against preselected sources while ignoring nondesired signals. We have concentrated this year on enlarging our data base of acoustic signatures with an eye to determining the points of commonality between major source classes. This has led to vastly simplified signal processing techniques compared with previously tried methods. Specifically, we have found that within a source class, certain fundamental signature components cannot be masked by purely physical means without materially degrading the operation of the source. Masking efforts employing jamming techniques are not as effective as desired, since jamming means that the total acoustic output has been raised yielding a more visible source. Baffling of noise sources, while effective for high-frequency noise components, tends to be less than effective when used to suppress the lower components without an attendant loss in source mobility performance.
Coupling coefficients between acoustic wave normal modes and internal wave normal modes
Authors: Dozier L. B. , Tappert F. D.
Published: October 2005

Theories of acoustic signal propagation in randomly fluctuating oceans using normal modes as a basis require knowledge of the coupling coefficients, or matrix elements, that connect the acoustic wave normal modes and internal wave normal modes. We have computed these coupling coefficients numerically using programs that generate both sets of normal modes from model sound velocity profiles and Brunt—Vaisala frequency profiles. Statistical properties of these random coupling coefficients have also been obtained. [Work supported by ONR.]
Underwater sound wave propagation in the presence of a randomly perturbed profile
Authors: Besieris I. M. , Kohler W. E.
Published: October 2005

The propagation of acoustic signals over long ranges in the deep ocean is influenced significantly by the following factors: (1) a deterministic sound speed profile possessing a “sound channel”; (b) highly anisotropic spatial random fluctuations; (3) temporal fluctuations due primarily to internal waves. The wave equation, with a randomly perturbed sound-speed profile, is initially considered, and a scaling is introduced which is typical of the ocean environment. A space—time parabolic equation is then obtained for the random product of complex pressures at two spatial and temporal points. Finally, a transport equation is derived for a suitably chosen transform of the pressure correlation function. This transport equation is general in the sense that it can model the decorrelation of acoustic signals due to offsets in range, azimuth, depth, and time. [Work supported by ONR.]
Stochastic Green's function approach to acoustic propagation through a fluctuating medium
Authors: Vasholz David P.
Published: October 2005

A general, Green's function-based approach to the problem of acoustic wave propagation through a random medium is presented. Starting with the idea that stationarity should be preserved in the propagation from source to field point, the hypothesis that the moments of the stochastic Green's function G are invariant under arbitrary uniform translation of all time variables is proposed. Proceeding by analogy with quantum field theory the quantities 𝒢 and M, related to the first and second moments of G, are defined. Various properties of 𝒢 and M are discussed, and it is shown that they may be conveniently used to describe the principal physical effects induced by a fluctuating medium. An exactly soluble model which illustrates the general results is presented.
Ray-tracing technique adapted for rapid computation
Authors: Princehouse D. W. , Lacy C. E.
Published: October 2005

REVGEN provides a computerized ocean acoustic environment for the development of a modern active sonar system. [D, W. Princehouse, “Real-time reverberation generator,” J. Acoust. Soc. Am. 57, S68(A) (1975).] During an exercise, it is important that it run in as near to real time as possible. We have developed a method whereby the numerically difficult and time-consuming portion of ray tracing can be performed before the exercise starts, and the results used rapidly when needed. We work in the limit of geometrical acoustics and assume that the sound velocity profile depends only on depth, not range or time. While many ray-tracing techniques can be adapted, we have used those of NISSM II. [Henry Weinberg, Naval Underwater Systems Center, New London, CT 06320. NUSC Technical Report 4527.] A function of two variables is defined, range versus launch depth and vertex velocity. Knowledge of this function allows ray tracing to be performed. This function may be computed prior to its need, a lengthy process. By encoding it in an efficient fashion, however, it may be stored in computer memory for rapid access when needed. Examples of ray tracing, caustic detection, and multipath propagation will be shown.
Separation of individual multipath arrivals—an experiment
Authors: Ewart T. E. , Ehrenberg J. E.
Published: October 2005

During June 1976 an experiment was conducted in which acoustic pulses were transmitted from a fixed transmitter to a fixed receiver 1100 meters distant. A 2-msec pulsed tone and 4-msec Taylor weighted FM slide signals were sent at 2, 4, 8, and 16 kHz and the received signals were sampled every 15 msec. The sound velocity profile was such that two wholly refracted paths with a travel-time difference of about 0.5 msec were received. Analysis of the data shows distinctly different behavior in the phases and amplitudes of the two arrivals. The vertical arrival angle of each ray has been computed from the data received at two vertically spaced transducers and the fluctuations in the angle are discussed. The signal-processing techniques used to obtain the phase and amplitude estimates from the data are presented. The implication of this experiment toward understanding the relationship of multipath signals to the oceanographic environment (e.g., internal waves and fine structure) is discussed.
Acoustic phase fluctuations induced by internal waves in the ocean
Authors: Desaubies Yves J. F.
Published: October 2005

Acoustic phase fluctuations induced by internal waves in the ocean are calculated in the geometric approximation. The results compare favorably with available experimental data.
Amplitude fluctuations of signals received from cw sources towed at long ranges in the deep ocean
Authors: amsdale D. J. , Flowers K. D. , Frisk G. V. , Giellis G. R.
Published: October 2005

In April 1974, three cw sources (9.807, 110, and 262 Hz) were towed for 16 days in the Atlantic along a track from Antigua to the Grand Banks of Newfoundland. Data from these sources were recorded near Bermuda, Puerto Rico, and Antigua on both bottom and suspended hydrophones. A 3-h running average was subtracted from the amplitude data in order to remove long-term deterministic trends. The remaining fluctuations were analyzed in fixed time segments to determine the correlation time, the density function and the associated first four moments. The variate difference technique was used to estimate the degree of randomness of the fluctuations. Finally, the effects of range, range-rate, source frequency, and environment on the fluctuation statistics are discussed.
Sound attenuation by conglomerates of bubbles at low frequencies
Authors: Diercks K. J. , Huckabay W. B.
Published: October 2005

Experimental measurements of sound attenuation in the frequency range from 0.5 to 10.0 kHz are described. Small plastic bubbles simulating in volume that of an adult anchovy swimbladder were used. Both signal enhancements and attenuations ranging from +5 to −25 dB, respectively, were observed. Effects of bubble location and population density upon the frequency and magnitude of the maximum enhancement or attenuation are described and a simple mathematical model relating these parameters is presented. [Work supported by NAVSEA.]
Volume attenuation in the Labrador Basin
Authors: Fraser I. A. , Franklin J. B.
Published: October 2005

Volume attenuation coefficients for one-third octave frequency bands up to 400 Hz have been determined as part of an underwater sound propagation study conducted in the Labrador Basin in cooperation with the U. S. Naval Research Laboratory. Charge detonation depths, receiver depth and the sound channel axis depth were all roughly 200 m along the 1300-km track. A late summer maximum in the near-surface sound velocity and a water depth greater than 3000 m also helped to ensure nearly ideal measurement conditions. The deduced volume attenuation coefficients were greater than the Thorp predictions [W.H. Thorp, J. Acoust. Soc. Am. 42, 270 (1967)], with attenuation increasing toward both higher and lower frequencies from a minimum (0.002 dB/km) near 60 Hz. Comparisons with results from other areas will be made. Results on the low frequency dependence of propagation loss will also be discussed.
Low-frequency attenuation in the North Atlantic Ocean
Authors: Kibblewhite A. C. , Bedford N. R. , Mitchell S. K.
Published: October 2005

Several Pacific Ocean experiments have been reported in which the attenuation of underwater sound at low frequencies is apparently regionally dependent. An experiment conducted in the North Atlantic Ocean in 1973 has provided data for use in determining attenuation coefficients for frequencies below 300 Hz for that region. The methods used to obtain these coefficients, their values, and their relationship to previously reported data from other regions will be presented.
Excess attenuation in a stochastic environment: theory, model, and experiment
Authors: Schneider H. G.
Published: October 2005

A general scheme to explain the excess propagation loss in sound channels is outlined, in which the stochastic variability of the environment is combined with the usual deterministic effects on propagation loss. Then a method, previously reported by the author in Acustica [35 (1976)] to handle stochastic sound speed variations in ray tracing routines is applied to model the excess attenuation for the Hudson Bay data. Finally, results will be presented indicating that the excess attenuation in the sound channel is a function of both the variability of the environment and the bottom loss. [Work supported by NAVSEA.]
Variability of low-frequency sound absorption in the ocean: pH dependence
Authors: Mellen R. H. , Browning D. G.
Published: October 2005

Measured values of absorption coefficients in sea water below 1 kHz show a marked regional dependence; for example, the value in the North Atlantic is found to be about twice that in the North Pacific. Recent laboratory measurements have demonstrated that the absorption mechanism is a boric acid—borate relaxation. Since this boric acid reaction acts as a buffer in sea water, the absorption coefficient is strongly pH dependent. Over the range of pH values encountered in the oceans (7.5–8.2), the absorption is a monotonically increasing function of pH and has a range of values of roughly 4:1. Comparison of experimental absorption coefficients and predictions based on reported local pH values show reasonably good agreement. [Work supported by NUSC.]
Source-speed effect on fluctuation time scale
Authors: Rosenbaum I. J.
Published: October 2005

The time scale of acoustic fluctuations has been found to depend on source speed for a shallow-receiver shallow-sound source configuration in deep water off the coast of St. Croix, U. S. Virgin Islands Measurements were made of a 135-Hz cw source towed at different speeds on a tangential track through a point 7.5 miles from the receiving hydrophones. Higher source speeds consistently gave rise to shorter correlation decay times. As the data integration period was increased from 5 to 100 sec, the measured correlation decay times increased as well, covering the range of 4–462 sec for various source-speed—receiver-depth combinations. The position of the receiver with respect to the thermal layer depth also plays a significant role.
Approximate ray angle diagram
Authors: Cox Henry
Published: October 2005

The Ray Angle Diagram presents loci as a function of depth of the angles which selected rays make with the horizontal as sound propagates in the ocean. It provides useful qualitative information about deep ocean propagation and the vertical distribution of signal and ambient noise power. The loci are based on Snell's law. It is shown that an Approximate Ray Angle Diagram (ARAD) can be easily constructed directly from the Sound Velocity Profile since on an appropriate scale the loci of one minus the cosine of the ray angle versus depth are approximately mirror images of the Sound Velocity Profile. The approximations are shown to involve negligible errors for cases of interest in underwater acoustics. The use and construction of the ARAD are illustrated with examples. A simple graphical technique is presented for annotating the ARAD with range information from which rays may be plotted without the use of a computer.
Development of the Proposed Canadian Noise-Level Regulations For Vessels Engaged in Towing
Authors: Harford K. D. , Wakefield C. W.
Published: October 2005

The paper will describe the towboat noise-control experiments, noise-level survey, and general rationale behind the Proposed Canadian Noise-Level Regulations for Vessels Engaged in Towing (1975). The proposed regulations are based on what was determined to be the technically feasible noise levels for the large majority of British Columbia vessels at the time of the study with provisions for reduced levels for vessels constructed in the future when marine-noise-control practices will have advanced and become more widely adopted. The relative importance of airborne and structure-borne noise transmission aboard towboats and the significance of this relation to the general philosophy of towboat noise control will be discussed.
Recommended maximum noise and vibration levels on ships
Authors: Schwartz A , Munch P. , Naim L.
Published: October 2005

With the intention to incorporate it in a revised version of the “Regulations for the Prevention of Harmful Noise on Ships (1972)” a special “Notice to Shipowners and Shipbuilders” (SHN 4/76) was prepared by a technical committee and issued by the Ministry of Transport, Department of Shipping and Ports. The recommended maximum Noise Rating Levels (NR), for existing ships and for new ships, respectively, are navigating bridge and radio room 65/60, crew cabins 65/55, mess and recreation room 75/65, general machinery spaces 85/85, engine control room 75/70, and engine workshop 80/75. The maximum recommended acceleration m/sec2—overall values from 2 to 8000 Hz), for vertical and horizontal acceleration, respectively, are: crew cabins, mess and recreation rooms 0.4/0.2, general machinery spaces 2.0/1.0 and engine control room 0.6/0.3.
Airborne noise control in maritime structures—an overview
Authors: Nelson D. L. , Lewis D. P.
Published: October 2005

Similarities and differences of shipboard airborne noise problems and problems of noise in buildings, industrial plants, and transportation systems are discussed. Various types of shipboard environments are considered: these include machinery spaces, living compartments, office, command/control rooms, and deck stations. Criteria reflecting the different operational requirements are presented. Some special problems unique to ships are unusual noise sources such as active sonar ping and cavitating propellers, close proximity of personnel to major machinery, and ship structure as a significant sound-transmission path. Methods of noise control are subject to constraints not common to other branches of acoustics. These include space and weight limitations as related to craft performance, machinery operation, and maintenance. Materials and constructions used in land-based applications may be inappropriate in the maritime environment, because of fire hazards and contamination by oils. fuels, or salt water. The range of current activities in shipboard airborne noise control is outlined and the major deficiencies in the state of the art are discussed.
Cochlear wave mode transformations
Authors: Haggerty John W.
Published: October 2005

A three-dimensional “first filter” model, describing the transition from long to short cochlear traveling waves, contains no resonance. Amplitude maxima of middle- and low-frequency wave envelopes are calculated by this model to occur at the place where longitudinal wavelength equals 4πkb(x), where b(x) is the local width of the basilar membrane and k is the local (near unity) ratio of the longitudinal and lateral (dissipative) traveling-wave propagation velocities. These two wave modes share the same potential function and are thus tightly coupled; their phase relationship is established by both the curvature of the cochlea and the stiffening effects of the Arches of Corti near inner radius. The velocity-versus-boundary dimension methods used in constructing the above modeling are known collectively as normal-mode analysis. Normal-mode analysis also suggests many reasonable “second-filter” modes of vibration possibly operating within the Organ of Corti (which gelled material supports the bases of all hair cells). At high frequency, a vibration-absorber wave mode (squeeze of the Organ of Corti) can amplify the displacement of the traveling wave, especially at the inner hair cells, while simultaneously extinguishing (by well known force-balance processes) that traveling wave on the basilar membrane. At lower frequencies, where the Organ of Corti spans relatively less of the basilar membrane width, some constrained-sloshing wave modes in the gelled material also preferentially excite inner hair cells. Close agreement with published hydrodynamic cochlear wave observations is shown. Confirmation of any of these distinct (but theoretical) wave modes within the Organ of Corti awaits measurement of directional propagation velocities and elastic properties through the gelled material as it lies flooded and attached to the basilar membrane.
The critical band; its relation to cochlear nonlinearity and second filter
Authors: Duifhuis H. , Simons W. F.
Published: October 2005

The measurement of the effect of the bandwidth of a noise masker which masks a centrally located tonal probe, provides a possibility to determine the critical bandwidth. The result of such a measurement typically shows two types of asymptotic behavior: For narrow bands the masking effect is determined by the power in the band, for wide bands, however, by the power density. The intersection of the asymptotes defines the critical band. If we assume that the masking effect is determined by the signal-to-noise ratio of activity in the probe channel the above results can be predicted by the model on cochlear nonlinearity and second filter [H. Duifhuis, J. Acoust. Soc. Am. 59, 408–423 (1976)]. The predicted critical bandwidth then depends relatively strongly on the tuning disparity between first and second filter and relatively weakly on filter shapes. Also we predict certain observed differences between nonsimultaneous and simultaneous masking for the above stimuli.
Octave convergence in melodic perception
Authors: Idson Wendy L. , Massaro Dominic W.
Published: October 2005

Previous research indicates that the pitch of a single tone can be conceptualized as a bidimensional quantity, reflecting the overall pitch level of the tone (tone height) and its position in the octave (tone chroma). It has been suggested that the dimension of tone chroma is irrelevant to perception of the component tones of a melody. Supporting this argument, perception of a melody is disrupted when its tones are displaced by octave multiples. This has been interpreted as showing that melodic perception is based upon the magnitude of successive pitch intervals. However, octave displacement violates both interval magnitude and direction. In the current study, melodies were subjected to structural transformations in which the tones were displaced by octave multiples, either preserving or violating the direction of successive pitch intervals (melodic contour). Replicating previous research, when melodic contour was violated performance was severely disrupted. When contour was preserved, however, the melodies were identified as accurately when the tones were displaced by octave multiples as when the melodies were untransformed. These results demonstrate that the bidimensional model of pitch perception applies to melodies, and that contour plays a significant role in melodic perception. [Work supported by NIMH.]
Temperament of the musical scale in fixed pitch instruments: contemporary psychological adjustment to environment
Authors: Mulder George W.
Published: October 2005

The relation between just intonation and equal temperament has been a subject of concern for both musician and physicist. The Stevin monochord in 1596 seems to, have been the first solution in Western thought. However, Tsai-yu' Monochord in 1595 warrants attention in Eastern thought. As with Vicentino's solution in 1555, aesthetic considerations, tentative though they may be must be regarded. Through measurement and examined response, psychological adjustments to temperament are observed and recorded. The results of these perceptions qualifies the listener's attitude toward temperament of the musical scale. Environment and the status quo of aesthetic standard is observed the paramount determinant.
The paradox of pitch categories
Authors: Siegel W.
Published: October 2005

The categorical perception of pitch by musicians [e.g., W. Siegel and R. Sopo, J. Acoust. Soc. Am. 57, 511 (1975)] raises an interesting question: How can they tune their instruments when they cannot tell sharp from flat? Here, the results of two experiments on ELP, a musician with very fine absolute pitch, suggest a resolution of the categorization paradox. In a production task, which emulates tuning an instrument, he was very accurate at adjusting the frequency of an oscillator to a value corresponding to four musical notes, g♯ (415 Hz), a (440), c (523), and d♯ (622). However, in an identification task involving listening to single brief notes, he showed striking evidence of categorical perception. That is, he judged 400-, 500-, and 600-Hz tones to be accurate examples of the notes g♯ (415), c (523), and d♯ (622) on most of the trials.
Effects of coarticulation, syllable type, and stress on acoustic French vowel targets
Authors: Chollet G. F.
Published: October 2005

Modern computer speech-signal analysis techniques are applied to the study of the acoustic characteristics of surreptitiously recorded conversational speech. Vowel formant values are found to vary considerably more than previous studies have led us to believe. Several parameters that could be causing these fluctuations are dealt with here. The phonemic context appears to be an important cause of variation. A tendency to produce an open vowel in closed syllables would appear to be true for all French vowels, although slightly more pronounced for midvowels. Vowels in the last syllable of a breath group (the stressed syllable in French) are generally more stable than those occurring in unstressed ones. The phenomenon of vowel reduction in French, although less pronounced than in English, is nevertheless quite noticeable. Other possible causes of variation are discussed as well.
Explaining the intrinsic pitch of vowels
Authors: Ohala John J. , Eukel Brian W.
Published: October 2005

The reason why, other things being equal, the pitch of vowels correlates with vowel “height,” has long been a subject of speculation. The effect is usually attributed either to acoustic coupling between vocal tract and vocal cords or to the pull of the tongue on the vocal cords. To determine which of these hypothesized factors is responsible we observed the effect on the pitch of vowels when one of these factors (impedance of the vocal tract was held constant, while the other (tongue stretch) was varied over an exaggerated range, viz., by having eight speakers utter special test utterances when their jaw was propped open by small wooden blocks ⩽ 10 mm thick. (Lindblom and Sundborg have shown that under such conditions apeakers can achieve acoustically normal vowels by increasing the distortion—and presumably the stretch—of the tongue body with respect to the jaw and larynx-hyoid complex.) We found a small but consistent increase in the pitch interval between high and low vowels when subjects spoke with propped-open jaw, thus supporting the “tongue-pull” hypothesis. [Supported by NSF.]
Intensity and duration as correlates of F
Authors: Zee Eric , Hombert Jean-Marie
Published: October 2005

An experiment was conducted to determine whether intensity and duration are correlated with tonal differences. Taiwanese was chosen because it has five contrastive tones, which are high, mid, low, rising, and falling. In this dialect the syllable [si] occurs on all five tones. Preliminary results reveal that (1) there is a direct correlation between F0 and both duration and intensity; (2) falling tone has higher intensity and shorter duration than rising tone; (3) the rate of change of F0 is greater in falling tone than in rising tone; and (4) both in falling and rising tones, the rate of change of F0 is greater in the second half of the entire duration than in the first half. These results are discussed in terms of possible physiological explanations, such as the use of a higher subglottal pressure to produce a higher F0. [Research supported by NSF.]
Difficulty of producing different F in speech
Authors: Hombert Jean-Marie
Published: October 2005

Two studies (Chain and Ewan, 1973; Sundberg, 1973) have revealed interesting articulatory constraints on the production of certain F0 contours. One finding is that it takes longer to produce a rising F0 contour than a falling contour covering the same frequency range. The purpose of this paper is to investigate in more detail the articulatory constraints involved in the production of some typical F0 patterns found in tone languages. Ten subjects (five speakers of tone languages and five speakers of nontonal languages) were asked to imitate a total of nine fundamental frequency patterns: three levels (110 Hz, 130 Hz, and 150 Hz) and six contours (110–130 Hz, 130–150 Hz, 110–150 Hz, 150–130 Hz, 130–110 Hz, and 150–110 Hz). In order to control for the effect of vowel quality, three vowels were used: [i], [a], and [u]. Subject's accuracy at matching the various stimuli (in F0 and in duration) was taken as reflecting the difficulty involved in the imitation of these F0 patterns. The results suggest a greater difficulty for rising (versus failing) and for higher (versus lower) F0 patterns. [Research supported by NSF.]
Tone spreading and perception
Authors: Maddieson Ian
Published: October 2005

An experiment was designed to determined if perceptual processes contribute to the explanation of asymmetries in “tone spreading” in tone languages. The basic stimulus was a synthesized trisyllabic utterance [wawawa], whose fundamental frequency varied in the following manner. Each stimulus type contained two sections, one with a higher and the other with a lower F0. An abrupt change was located at 15 points 15 msec apart ranging from the beginning to the end of the middle syllable. In one series, H-V-L, the higher section occurred first, in the other, L-V-H, the lower section occurred first. Subjects were presented a randomized sequence of the stimuli from both series and asked to respond “first” or “last” according to whether the middle syllable sounded the same in pitch as the first or the last syllable. To elicit a “first” response, the portion of the middle syllable identical in F0 to the first syllable needed to be shorter for the H-V-L series than for the L-V-H series. This result demonstrates a perceptual phenomenon which possibly serves as the basis for the observation that high tones are more likely to spread than low tones. [Research supported by NSF.]
Rate estimates of sound-silence sequences in speech
Authors: Feldstein Stanley
Published: October 2005

Speech rate, a variable used frequently in psychological research, is usually indexed by number of words (spoken) per minute (WPM), the estimation of which is exceedingly time consuming. However, an automated temporal analysis of speech yields not words but a sequence of sounds and siliences. This study searched for a sequence parameter to index rate as well as does WPM. Eighteen women were each asked to tell a different 3-min story in each of two conditions: in one she was to talk rapidly, in the other, to talk slowly. The automated analysis yielded four measures: average duratuions of vocalizations and pauses (V, P) and proportionality constants of vocalizations and pauses (PCv, PCp). The PC's index the probabilities of continuing to vocalize or to remain silent within each successive time period sampled. Also, PCv/PCp was calculated and WPM were obtained. After within-subject variance was accounted for, the sequence-derived parameter most highly related to WPM was PCv/PCp (r=0.93). Moreover, PCv/PCp distinguished between the conditions about as well as WPM. Relations between the sequence parameters and perceived sequence rates are now being examined.
Flexible electret transducers
Authors: Miller G. K.
Published: October 2005

The availability of polymer electrets that can take on a wide variety of configuration has made possible many new applications for electrostatic transducers. Electret materials, poling techniques, and basic transduction schemes are reviewed. Several specific line transducers are then examined both analytically and experimentally. A coaxial cable electret transducer is shown to be useable underwater as a hydrophone, underground as a strain detector, and on a fence as a vibration detector. A long multilayered tape with an electret layer is shown to be a line transducer that can be used in an active ultrasonic perimeter intrusion detection system. [Supported in part by ONR and by RADC.]
Flexural disk peizoelectric polymer hydrophones
Authors: Powers J. M. , Sullivan T. D.
Published: October 2005

Piezoelectric polymer (poled polyvinylidene fluoride), a Japanese development recently introduced into high fidelity technology, shows promise as a hydrophone material because of its flexibility, ruggedness, low density, and potentially low cost. A family of lightweight flexural disk hydrophones which use piezoelectric polymer are described. Measurements on a 3.8-cm-diam unit gave a sensitivity of −199 dB re 1 V/μPa with good stability over ranges of 2 Hz–1 kHz, 0°–22°C, and 0–4 MPa static pressure. This particular unit uses two air-backed flexural disks with a stack of four pieces of 30-μm polymer film glued to each disk, resulting in a combined capacitance of 1 nFd. This hydrophone has a mass of 24 g in air and is neutrally buoyant in water. A theoretical analysis which shows the variation of hydrophone performace as a function of design parameters is presented, and the material properties of piezoelectric polymer and piezoelectric ceramic are compared. [Work supported by NAVSEA and NAVELEX.]
Temporal integration of tone glides
Authors: Collins M. J. , Cullen J. K.
Published: October 2005

Temporal integration of fixed-frequency signals is a well-established phenomenon. However, integration characteristics for speech-like sounds are not known. The present study investigates temporal integration, in a background of noise, for fixed-frequency signals and rising and failing frequency contours in the ranges 200–700 Hz and 1200–1700 Hz. These ranges were chosen because of their correspondence to first-and second-formant regions for speech sounds. Tone glides changed frequency linearly during the burst durations of 5, 10, 35, 50, 90, and 120 msec. Reference fixed-frequency signals were selected at the extremes and arithmetic centers of the glide ranges (i.e., 200, 450, 700, 1200, 1450, and 1700 Hz). The results for fixed-frequency signals are in agreement with previous investigations. The integration rate of the falling higher-frequency tone glide is similar to that of the fixed-frequency signals. However, other tone glides indicate a slower rate of integration than for fixed-frequency signals. The results are interpreted regarding time and frequency limitations on the development of the critical band, and possible relevancy to speech perception. [Work supported in part by NIH.]
Subjective loudness of short duration signals
Authors: Pavel Michael
Published: October 2005

It is known that the loudness of short signals with rectangular envelopes is proportional to the product of their power and their duration (i.e., their energy). This is consistent with the properties of a leaky integrator followed by a peak detector. As a result, it has been proposed that this combination be used as a basis for measurement of loudness. The present study was conducted to determine the effect of nonrectangular envelope shapes on loudness judgments. The test stimuli, whose overall bandwidth was within a single critical band, consisted of a pure tone (1000 Hz) whose amplitude was modulated by various slowly varying functions. The tested envelopes included combinations of rectangles, decaying exponentials and growing exponentials. The durations ranged from 25 to 2000 msec. The test stimuli were presented binaurally and were compared in loudness to rectangular standard tones of the same frequency and duration. The points of subjective equality were determined by an adaptive 2IFC method. As expected, the shape of the envelopes had little effect on the loudness of the very short stimuli (25 msec). Results for longer duration signals will be compared to predictions using an exponential integrator and a model proposed by Zwislocki. [J. Acoust. Soc. Am. 46, 431–441 (1969)].
Perception of duration: Effect of a second signal nearby in time
Authors: Pokorny R. , Elmasian R. , Galambos R.
Published: October 2005

Pairs of noisebursts (75 dB SPL, 150 msec apart) were presented to subjects who were instructed to judge the duration of one of them (the target) and ignore the other (the conditioner). The two noisebursts could be presented to the same ear (monaural) or to opposite ears (dichotic), and the conditioner could precede or follow the target (forward and backward paradigm, respectively). Subjects estimated target duration by changing the duration of a third noiseburst (also 75 dB SPL) presented 1500 msec after the pair. Five conditioner (5, 10, 20, 40, and 80 msec) and two target durations (20 and 25 msec) were used. In the forward monaural paradigm the average subject selected bursts of 17 and 26 msec as matching the 20 msec target when the conditioner was set at 5 and 80 msec, respectively. The matches showed smaller and more variable shifts in the monaural backward, dichotic forward, and dichotic backward paradigms. These perceptual duration shifts resemble a variety of other auditory illusions, such as loudness enhancement and pitch shifts (R. Elmasian and R. Galambos, J. Acoust. Soc. Am. 58, 229–234 (1975); 58 S83 (1975)] and may be related to them by a common mechanism.
Temporal resolution of components of word-length tonal patterns
Authors: Watson C. S. , Espinoza-Varas B. , Kelly W. J.
Published: October 2005

We have recently reported that frequency resolution, under conditions of high stimulus uncertainty, is much more accurate for late high-frequency components than for early, low-frequency components of word-length tonal sequences [Watson, Wroton, Kelly, and Benbassat, J. Acoust. Soc. Am. 57, 1175–1185 (1975)]. We have now replicated these experiments for the case of temporal resolution. Listeners were tested with high, medium, and minimal stimulus uncertainty until they performed at asymptotic levels of temporal resolution, defined by just-detectable proportional increments (Δt/t) in the duration of single 40-msec components of 10-tone sequences. The duration of late, high-frequency components is also resolved more accurately than that of early, low-frequency elements, although the differences are considerably less than for frequency resolution. Asymptotic just-detectable values of Δt/t range from about 0.75 to 0.15, from high to minimal levels of stimulus uncertainty, suggestive of a shift from detecting increments in tonal sequence duration to detecting increments in the duration of single test components. [Supported by Grant ♯NS-03856 from NINCDS.]
Sequential two-channel frequency difference limen
Authors: Ahroon W. A. , Pastore R. E.
Published: October 2005

The ability of trained observers to monitor the sequential inputs to two earphone channels was explored. In the independent same-different frequency discrimination tasks, the two temporally discrete standard stimuli were followed by two temporally discrete comparison stimuli such that the stimulus offset in the first channel was contiguous with the stimulus onset in the second. As with our earlier experiments [Ahroon, Pastore, and Wolz, J. Acoust. Soc. Am. 56, S45 (1974)], one-monaural and two-dichotic (selective-attention and divided-attention) listening conditions, temporal separation of the standard and comparison stimuli facilitated performance in the trailing channel in selective attention tasks, but not in the divided attention tasks. However, performance under both dichotic conditions was significantly poorer than monaural performance, with backward interference more disruptive than forward interference. In addition, the data give no support for a model based on an analogue to a real-time computer-interrupt handling system in two-channel discrimination experiments. [Research supported by a graft from NINDS.]
Ear dominance in dichotic chords and frequency discrimination in the right and left ear
Authors: Divenyi P. L. , Efron R. , Yund E. W.
Published: October 2005

In a series of experiments frequency discrimination thresholds were determinined for pure tones presented either to the right or to the left ear of experienced listeners. In some conditions the stimulus was monaural, whereas in others a tone of different frequency was simultaneously present in the contralateral ear. The difference between the frequency of the target in the discrimination task and that of the contralateral tone was 4–6%. Center frequencies of 1.2, 1.7, and 3.2 kHz were investigated. Results indicate that there is a small but reliable discrepancy between just-noticeable frequency differences obtained for the right and for the left ear, both in the monaural and in the dichotic conditions. Furthermore, the direction and the degree of asymmetry with respect to the frequency resolving power of the two ears was found to be strongly correlated with the direction and the degree of ear dominance for the pitch of dichotic two-tone complexes [see Efron and Yund, J. Acoust. Soc. Am. 59, 889–906 (1976)]. In other words, for a given subject and in a given frequency region, the ear which was more efficient in the frequency discrimination task was also the one which dominated the pitch percept of a dichotic tone pair in the same frequency region. Implications of the relationship between the two types of functional asymmetry of the auditory system are discussed in the paper.