bluuebunny/crossref_metadata_2025 huggingface dataset:
Formant frequencies of vowels produced under aided and unaided conditions by persons with residual high-frequency hearing
Authors: Tobey Emily A. , Ross Reginold G.
Published: October 2005

Remarkably intelligible speech has been reported for individuals with severe sensorineural hearing losses in the range 125 to 8000 Hz but normal hearing in the frequency region above 8000 Hz. In order to determine what factors may underlie the intelligible speech of six of these patients, we measured formant values of /i, æ, u/ in CVCs produced during an unaided speaking condition and when patients wore an aid designed to transpose low-frequency information upward to the region above 8000 Hz. The initial consonant in the stimuli varied between /t, d, r, s, sh, tr, str/ and the final consonant was /p/. First-by-second formant frequency plots revealed the vowels were near-normally distributed during unaided speaking conditions; however, values more closely approximated those of normal-hearing speakers when the aid was worn. Data were similar to those reported for a single subject [M. J. Collins, J. Acoust. Soc. Am. Suppl. 1 65, S69 (1979)] and suggested speakers may use residual high-frequency hearing to monitor vowel formant frequencies. [Work supported by NIH and Louisiana Lions Eye Foundation.]
Temporal properties of cleft palate speech in two rate conditions
Authors: D'Antonio Linda L.
Published: October 2005

Few studies have documented the temporal characteristics of the speech of individuals with repaired palatal clefts. Acoustically defined segment durations were compared in the speech of three groups of subjects; cleft talkers with high intelligibility, cleft talkers with low intelligibility, and noncleft talkers. CVC nonsense syllables were produced within a carrier phrase at habitual and rapid speaking rates. The acoustic waveforms of the speech samples were interactively displayed, segmented, and measured employing digital signal processing techniques. The results indicated: (1) cleft speakers produced longer segments than did noncleft speakers; (2) some segment types were more prolonged than others in cleft speech; (3) vowel environment had stronger effects (particularly carryover effects) on duration of adjacent segments for cleft speakers than for noncleft speakers; and (4) when speaking rate was increased, noncleft and high-intelligibility cleft groups reduced vocalic intervals more than consonant intervals. In contrast the low-intelligibility cleft group showed the greatest reduction in consonant intervals. Temporal abnormalities thus exist in the speech of some individuals with cleft palate, and appear to be associated with reduced intelligibility.
Room acoustics and listening
Authors: Berkeley David A.
Published: October 2005

The physical basis for sound propagation in rooms may be described by a time-domain model of reflection from the walls. Reflections arriving soon after direct source-to-listener transmission produce a perception of spectral “coloration” while later arrivals contribute “echo.” Directional perception and listener preference for rooms are heavily influenced by such underlying phenomena. These perceptual effects can be quantified and directly related to their physical bases through use of psychophysical testing and interpretation methods, such as multidimensional scaling, combined with computer techniques for creating and modifying sound fields measured in actual or synthetic enclosures. Examples of such approaches, including diotic/dichotic image models of small rooms, modification of concert-hall transfer functions, and manipulation of responses measured directly in the ear-canal of listeners, are used ot illustrate the current state of the art.
The localization and lateralization of sound by land mammals
Authors: Gourevitch George
Published: October 2005

During the past few years research on directional hearing in land mammals branched out considerably. For the first time physical measurements were made of interaural time differences between the ears of animals. Further measurements of interaural intensities were also determined. Localization acuity was examined in previously untested species, among which were the largest terrestrial mammal (elephant) and one of the smallest (kangaroo rat). Extensive investigations were conducted on sound localization and on binaural sensitivity for putative directional cues in nonhuman primates. The localization of different macaque vocalizations tending to occur in different social contexts was also investigated. Altogether this work indicates that in comparison to other terrestrial mammals man's directional hearing remains foremost; that in a number of instances localization is not as accurate as might be expected on the basis of head size; that the sensitivity of the binaural system does not necessarily increase to compensate for weaker interaural differences in a small headed animal and may be less than in humans; that in lower primates, localization may have an important communicative role in addition to that of orientation. [Supported in part by NSF grant BNS 79-15834.]
Mechanisms for directional hearing among nonmammalian vertebrates
Authors: Fay Richard , Feng Albert
Published: October 2005

Adaptations for directional hearing among the nonmammals are quite diverse, and include morphological and physiological mechanisms at both peripheral and central levels, as well as behavioral strategies. While some of these mechanisms, such as interaural differences in time and spectrum based on head size, operate to some extent among all terrestrial vertebrates, the nonmammals show several additional special features. Fishes are the most unique in that behavioral capacities to locate sound in space and to use directional tiltering to improve signal-to-noise ratio are based on the directional characteristics of hair cells themselves, and on the patterns of hair cell directional orientation both within and between receptor organs (saccule, lagena, and possibly the utricle). In addition, it is thought that fishes code the phase relations between particle acceleration and the sound pressure waveforms to solve 180° ambiguity problems. The terrestrial nonmammals are all faced with similar problems in sound localization, including a relatively small interaural distance and generally poor high-frequency hearing. It is well known, however, that anurans and birds localize sounds well. There is evidence that the anuran ear shows a complex, frequency-dependent directionality, and that the wide coupling of the two middle ear cavities via the mouth lead to acoustic interactions which enhance interaural time and intensity differences. This type of mechanism is thought to operate in some birds as well, particularly at low frequencies. Some owls may use more “mammalian” mechanisms for azimuthal localization in addition to a vertical asymmetry in ear position which gives rise to interaural cues for elevation.
Physical acoustics and measurements pertaining to directional hearing
Authors: Kuhn George F.
Published: October 2005

The physical mechanisms of acoustic wave diffraction and scattering by the head and torso, as well as the resonances excited in the external ear produce interaural time differences, interaural pressure level differences, and direction-dependent pressure spectra at the eardrum. The frequency dispersion of the diffracted wave causes the phase-derived and envelope-derived interaural time differences to be frequency dependent and to differ from each other. However, at high frequencies these time differences converge to one common value which equals the delay of the leading edge of a pulse. Interaural pressure level differences at high frequencies result primarily from the frequency-dependent attenuation of the waves diffracted into the shadow region of the head and from the directivity of the pinna. Spectral localization cues which are usually associated with the localization in the vertical, median plane result from the interference of the wave scattered by the torso and the direct wave to the external ear, at low frequencies; transverse wave motion in the pinna appears to produce the direction-dependent high-frequency pressure spectra at the eardrum. Mathematical models which describe the wave motion about the head and in the external ear will be described and compared to physical measurements. Potential effects on the localization cues for aided listeners and associated difficulties of these acoustical measurements with real microphones in the external ear will also be discussed. [Portions of this work were supported by the National Science Foundation.]
Localization in man: Importance of outer-ear structures
Authors: Wightman Frederic L. , Kistler Doris , Walczak Nancy Barker
Published: October 2005

To this day it is widely accepted that localization of sounds in auditory space is determined primarily by two cues, interaural differences in stimulus intensity and time of arrival. However, recent research forces us to appreciate the deficiencies of this point of view. It cannot, for example, explain localization in the median plane or localization with one ear, both of which are done with reasonable accuracy. Nor can it account for the fact that stimuli presented (over headphones) only with interaural differences, appear to originate from inside the listener's head. Acoustic measurements (ours and others) show that in addition to interaural time and intensity differences, stimuli in free space produce large direction-dependent spectral cues, resulting from the acoustic filtering action of outer-ear structures (pinnae, etc.). These cues are large, and occur in relatively low-frequency regions (below 5 kHz), even when the source is moved a small amount in the horizontal plane. When stimuli are computer synthesized such that both the spectral and interaural difference cues are preserved, headphone presentation produces a nearly perfect simulation of free-field listening. Results from several recent experiments conducted in our laboratory and elsewhere lead us to the conclusion that the filtering action of the outer ear is at least as important to our perception of auditory space as is the fact that we listen with two ears [Work supported by NSF.]
Propeller noise prediction
Authors: Zorumski William E.
Published: October 2005

Analytic propeller noise prediction involves a sequence of computations culminating in the application of acoustic equations. This paper describes the prediction sequence currently used by NASA in its ANOPP (Aircraft Noise Prediction) program. No attempt is made here to review the state of the art of noise prediction. Some elements of this sequence represent classic results while other represent the most recent publications. The elements of the sequence are called program modules. The first group of modules analyze the propeller geometry, the aerodynamics, including both potential and boundary layer flow, the propeller performance, and the surface loading distribution. This group of modules is based entirely on aerodynamic strip theory. The next group of modules deals with the actual noise prediction, based on data from the first group. Deterministic predictions of periodic thickness and loading noise are made using Farassat's time-domain methods. Broadband noise is predicted by the semi-emipirical Schlinker-Amiet method. Nearfield predictions of fuselage surface pressures include the effects of boundary layer refraction and (for a cylinder) scattering. Farfield predictions include atmospheric and ground effects. Comparisons are made to experimental data from subsonic and transonic propellers and NASA's future directions in propeller noise technology development are indicated.
Airframe noise using experimental, cross-correlation techniques and a theoretical, vortex model
Authors: Meecham W. C.
Published: October 2005

Correlation measurements have been made on large scale models placed in wind tunnels. The correlation has been taken between surface pressures and farfield radiated sound. The technique permits one to find the amount of sound radiated by a surface even when that sound is up to 20 dB and more below the ambient noise in the operating wind tunnel. These techniques have strongly suggested that the major source of radiated airframe noise is at the outboard edges of deployed aircraft flaps. Measurements have been made to determine the dependence upon flow Mach number, upon angle, and upon the various flap-surface positions. A theoretical model has been devised using a vortex (and its image) embedded in a potential flow about a corner. The theory has but one adjustable parameter; all computations can be made on a hand calculator. The radiated sound is found using the Curie theory. The effects of many vortices are included. The predicted, radiated sound matches the correlation-measured sound to within 3 dB for most of the measurements made. The predicted and measured Mach number dependence is as the six power for sound intensity and as the second power for the local, rms, hydrodynamic pressure. There is also some predicted, weak dependence upon the Strouhal number.
Cross spectra between temperature and pressure in a constant area duct downstream of a hydrogen fueled combustor
Authors: Miles J. H. , Wasserbauer C. A. , Krejsa E. A.
Published: October 2005

It was shown in a recent paper [J. H. Miles and E. A. Krejsa, J. Acoust. Soc. Am. 72, 2008–2019 (1982)] that knowledge of pressure-temperature auto spectra and cross spectra could be used in a model for sound propagation in a variable duct having acoustic and convected entropy fluctuations. The present exploratory study investigates the measurement of pressure-temperature cross spectra and coherence and temperature cross spectra and coherence at well separated points in a combustion rig. Data obtained near the inlet and the exit of a 6.44-m-long duct attached to a J-47 combustor fueled with hydrogen are presented.
Multiple scattering of elastic waves in discrete random media
Authors: Varadan V. K. , Ma Y. , Varadan V. V. , Adler L.
Published: October 2005

A multiple scattering theory for elastic wave propagation in a discrete random media is presented. A self-consistent multiple scattering formalism using the quasicrystalline approximation and a self-consistent paircorrelation function is employed to study the phase velocity and coherent attenuation of elastic waves by a random distribution of cavities and elastic inclusions embedded in an elastic matrix. Both uniform and Gaussian size distributions are assumed. The theoretical results obtained in this study are shown to be in excellent agreement with experimental observations.
Nearfield effects in acoustic scattering
Authors: Hackman R. , Werby M. F.
Published: October 2005

In a standard T-matrix calculation, the scattered radiation field is evaluated by assuming that both the source and receiver are a large distance from the scatterer. These conditions may not be satisfied in practice. We consider the effects of an incident spherical wave and of relaxing the farfield assumptions for the scattered wave amplitude for both rigid and solid elastic prolate spheroidal solids in water. The T-matrix formalism is used to calculate the scattered form functions and radiation patterns at a variety of distances. Special attention is given to nearfield effects on the measured widths and strengths of the resonances generated in the elastic solid.
Angular spectrum representation for acoustical measurements
Authors: Johnson Richard L. , Devaney A. J.
Published: October 2005

The transmitting and receiving responses of an electroacoustical transducer in a fluid are characterized using the angular spectrum of plane-waves representation. This representation provides for simple forward and backward propagation of the acoustical fields using Fourier transform methods. A circularly symmetric transducer may be characterized using one-dimensional transforms leading to a significant reduction in computation. Calibration to characterize a transducer requires array or synthetic array measurements in one of several geometries. Two such geometries, a planar nearfield array or scan and a spherical farfield scan, are described. An example of back-projecting measurements of the radiated field to determine the velocity distribution at the face of the transducer is given. Numerical results are shown which demonstrate the development of the radiated pressure field through the nearfield to the farfield. The angular system representation has previously been shown to be particularly suited for scattering problems [R. K. Johnson and A. J. Devaney, Appl. Phys. Lett. 41, 622–624 (1982)].
A vector-potential approach to radiation from forcelike sources: Further developments
Authors: Hadden W. James
Published: October 2005

As reported previously [J. Acoust. Soc. Am. Suppl. 1 68, S107 (1980)], it is convenient to represent the pressure field in response to a moving force as the divergence of a vector potential function; the associated fluid particle velocity field is proportional to the time divergence of the vector potential function. In this paper, it is shown that using the representation υ = ∇φ − c−2∂A/∂t for the irrotational velocity field in linearized fluid dynamic equations leads to separate wave equations in φ and A: The latter arises directly from the momentum equation and has a source term the force exerted on the fluid. The equation in φ is obtained from the continuity and thermodynamic state equations; its source term represents a mass-flow fluctuation. The applications of this method to viscous and thermally conducting fluids will be discussed.
A comparison of directional microphone performance in free field and on a manikin
Authors: Madaffari Peter L.
Published: October 2005

Four first-order gradient microphones were constructed. These microphones differed principally in their internal time delay, which formed a geometric progression for successive members differing by the ratio 4∛. The square of this ratio coincides with the ratio that provides the classical cardioid, super-cardioid, and hyper-cardioid polar patterns. It is also approximately the ratio between time delay as measured in a free field and over the ear of the KEMAR (KEMAR is a trademark of Knowles Electronics Inc.) manikin. Each microphone was measured in a simulated hearing-aid housing which was modified to provided a series of external port spacings varying by this same ratio. A 4×4 matrix of internal delay versus external port spacing resulted. The polar pattern of each entry of the matrix was measured in isolation and over the manikin's right ear. The progressive pattern change measured on the manikin is similar to that found in the free field, but offset approximately one matrix element.
Precise measurement of phase difference and amplitude ratio with an interchange reference method
Authors: Wong G. S. K.
Published: October 2005

A simple but very precise method for the assessment of small phase difference θ and amplitude ratio R of two coherent sinusoidal signals is described. The method utilizes a lock-in amplifier as a null detector, and a precision level decade ratio-transformer provides attenuation for one of the test signals. When each of the test signals is used in turn as the reference, one can compute R and θ: R = [(α1/α2)(β2/β1)]1/2, cos θ = [(α1α2)/(β1β2)]1/2, where α and β are attenuator readings. It is estimated that for θ = 10°, the uncertainties in R and θ are less than 10 ppm and 0.01 degree, respectively.
Response of an edge-supported, circular, foil electret earphone with radially-varying parameters
Authors: Busch-Vishniac Ilene J. , West James E.
Published: October 2005

In edge-supported, circular, foil electret earphones sound pressures are generated in the ear canal by volumetric changes produced by motion of the foil. For a given driving voltage the dynamic foil motion and sound pressure level increase as the static deflection of the foil center increases within limits. Beyond a critical central deflection the foil electret collapses onto the back electrode thereby reducing the earphone response. If the earphone parameters are uniform, the static ?nd dynamic foil amplitudes are parabolic functions of radial location. With the introduction of radially varying earphone parameters it is possible to broaden the region of the foil electret which experiences high amplitude static and dynamic motion, thus increasing the sound pressure level generated at a given driving voltage. Theoretical arguments limit the increase in earphone sensitivity which may be obtained in this manner to 6 dB. Numerical studies of the effect of a radially varying charge distribution or radially varying air gap thickness suggest that, in fact, the maximum obtainable sensitivity increase is 3 dB.
5–50 kHz cylindrical nearfield calibration array
Authors: Van Buren A. L. , Jevnager M. D. , Tims A. C.
Published: October 2005

In acoustic measurement tanks it is often impossible to directly determine the farfield radiation properties of underwater transducers because of limited space. The Trott nearfield calibration array (NFCA) concept solves the problem by allowing the farfield properties to be obtained by measurements made close to the transducer. We describe here the development of a 5–50 kHz synthetic cylindrical NFCA that is suitable for medium size measurement tanks such as the pressure tank at the Underwater Sound Reference Detachment of the Naval Research Laboratory. The NFCA is synthesized using a single line of 48 piezoelectric ceramic hydrophone elements spaced 2.54 cm apart and amplitude shaded in a special way. The transducer to be calibrated is driven electrically and rotated near the line about an axis parallel to the line. The response of the line is measured at equally spaced angular positions, thereby creating a virtual cylindrical array surrounding the transducer. The azimuthal farfield pattern of the transducer is obtained by computer processing the responses together with a set of complex shading coefficients. We evaluated the line experimentally by calibrating a large piston transducer. The resulting computed farfield patterus are in excellent agreement with corresponding patterns obtained from direct farfield measurements made in a lake. [Work supported by NAVSEA 63R.]
Psychophysics of vibrotactile stimulation
Authors: Verrillo Ronald T.
Published: October 2005

A large number of physical parameters of the vibrotactile stimulus have an effect upon the psychophysical response of subjects. Developing technology and a better, but still incomplete understanding of the physical characteristics of cutaneous tissue have made possible the systematic investigation of these effects. These investigations on the one hand have led to a more precise definition of the capacities and limits of cutaneous sensation and perception. These results are important to design considerations in the development of communication aids for the deaf and blind. On the other hand, when psychophysical results were compared to data from anatomical and physiological investigations they have elucidated the underlying neural mechanisms of vibrotaction. In some instances the psychophysical results have predicted the results of physiological experiments. In the present paper are considered the effects on vibrotaction of time and space variables, such as frequency and area of stimulation, gradients on the skin and temperature at the surface of the skin. Also discussed are responses to suprathreshold stimulation and finally the effect of subject variables, such as sex and age, on vibrotactile response characteristics.
Tactile pattern recognition and its perturbations
Authors: Craig James C.
Published: October 2005

A number of devices developed to present information to the skin, including speech information, have tried to take advantage of the spatial-temporal processing characteristics of the skin. To study these characteristics, tactile patterns were generated on an array of vibrators (six columns by 24 rows) and presented to subjects' fingerpads. When pairs of patterns, such as letters of the alphabet, were presented in close temporal proximity, considerable masking was produced. There was more interference in recognition when the masking stimulus followed the target (backward masking), than when the masker preceded the target (forward masking). Masking increased as the time between the target and masking stimulus decreased. Masking also increased if the masking stimulus and the target contained similar features. Using tactile stimuli derived from speech information, it was also found that reducing the amount of backward masking, by reducing the duration of the vowel portion of a CV, improved recognition of the consonant. The critical temporal variable in measures of both masking and temporal integration appeared to be the time between the onsets of the successive patterns. [Work supported by NIH and NSF.]
Research on the Tadoma method of speech communication
Authors: Reed Charlotte M. , Rabinowitz William M. , Durlach Nathaniel I. , Braida Louis D.
Published: October 2005

In Tadoma, speech is received by placing a hand on the talker's face and monitoring actions associated with speech production. Our initial research has documented the speech perception, speech-production, and linguistic abilities of deaf-blind individuals highly trained in Tadoma. This research has demonstrated that good speech reception can be achieved through the tactual sense: performance is roughly equivalent to that of normals listening in noise at an S/N ratio of 3 dB. It appears that the principal cues employed are lip movement, jaw movement, oral airflow, and laryngeal vibration, and that the errors which occur are caused primarily by inadequate information on tongue position. Our current research includes (1) learning of Tadoma by normal subjects with simulated deafness and blindness, (2) augmenting Tadoma with a supplemental tactile display of tongue position, and (3) developing a synthetic Tadoma system in which signals recorded from a talker's face are used to drive an artificial face. This research is expected to increase our understanding of Tadoma and its relation to other tactual communication methods, show that performance obtained with Tadoma does not represent the ultimate limits of the tactual sense, and provide a research tool for studying transformations of Tadoma. [Work supported by NIH and NSF.]
Lingual vibrotactile sensation magnitudes: Comparison of suprathreshold responses in men and women
Authors: Fucci Donald , Petrosino Linda
Published: October 2005

The psychophysical method of magnitude production was used to obtain suprathreshold sensation magnitude functions at a frequency of 250 Hz from the anterior midline portion of the tongue of 20 adult subjects (ten men and ten women). Results showed that the anterior dorsum of the tongue can produce suprathreshold magnitude functions compatible with those described in the literature, but with a steeper slope value. The steeper slopes can be related to the psychophysical method employed, the use of a small skin contactor, the use of a rigid surround around the contactor, and the physiological nature of the tongue itself. It was also determined that women produce slightly different lingual suprathreshold magnitude curves than men. The upper slope value for the women is similar to that for the men, but the lower slope of the magnitude curve is somewhat steeper indicating that if women and men are presented with the same stimulus amplitude, the women will perceive it as being slightly greater.
Misarticulating children's identification and discrimination of /r/-/w/
Authors: Hoffman Paul , Daniloff R. , Schuckers Gordon
Published: October 2005

Twenty children, ages 6:0 to 6:11, who misarticulate [r] were compared to age-matched normals in their ability to identify and discriminate seven synthetic stimuli representing a continuum between way and ray. Discrimination was tested among two-step pairs using the 4IAX paradigm. As a group, the misarticulating children showed poorer discrimination ability and less stable identification. However, there was considerable overlap with some misarticulating children demonstrating adultlike performance and some normal children showing aberrant identification and discrimination.
Perception of synthetic speech by children
Authors: Greene Beth G.
Published: October 2005

Many educational and commercial products that produce synthetic speech output are becoming widely available. However, there is little, if any, research on how adults or young children process (i.e., perceive, encode, retrieve) synthetic speech signals. Results of two experiments that studied children's perception of natural and synthetic speech indicated that subjects consistently showed a higher error rate when listening to synthetic speech than natural speech. In one experiment, children 5–8 years old listened to natural and synthetic speech tokens in a task that required them to point to a picture, from among four alternatives, that corresponded to the word heard. In a second experiment, children 8–10 years old listened to natural and synthetic digit strings and repeated the digit strings to the experimenter exactly as heard. Performance deficits were observed for the synthetic speech relative to the natural speech even for highly familiar material such as digits. These results have implications for the design, selection, and use of voice-response systems to be used in teaching and learning environments with young children and adults. Synthetic speech may require additional processing capacity and attentional demands that may well produce large decrements in other complex cognitive tasks that children are expected to perform routinely in educational settings. [Supported by NINCDS.]
Cross-language study of children's prosodic development-III: F0, intensity, and vowel durations in French 2-year-olds words
Authors: Allen George D.
Published: October 2005

This report continues the description of a corpus of utterances produced by 2- through 5-year-old speakers of French, German, and Swedish, describing in greater detail the utterances of one of the three language groups, the French children. These children produced a total of 188 analyzable utterances, ranging from a low of 17 for one child to a high of 59 for another. These 188 utterances contained a total of 357 F0 contours, of which 253 were falling and 104 were rising. The falling contours tended, on average, to be wider (in semitones) than the rising contours. Although nearly half of the contours were within the range of acceptable French, a substantial number were distinctly un-French in form. Approximately half of the intensity contours of these utterances were also appropriate, i.e., level or slightly falling. Several inappropriate rising intensity contours may have resulted from the children's attempts to control F0 by means of respiratory drive. Finally, the relative vowel durations in the last and next-to-last syllables of these utterances were observed to conform in general to previously described durational relationships for French. However, the tremendous variability within and between children in vowel durational ratios suggests that 2-year-olds have poor durational control. [Research supported by NSF.]
Formant frequencies and durations of /i/ and /I/ produced by native and non-native American speakers
Authors: Nábělek Igor V. , Maroonroge Sumalai , Silverstein Bernard
Published: October 2005

Durations and formant frequencies of /i/ and /I/ as produced by native and non-native American speakers were compared. The non-native speakers included one Thai female and one Slovak male. Their languages have only one vowel in place of the /i/ and /I/ and both speakers speak English with an accent. We wanted to determine which parameters of /i/ and /I/ are responsible for their incorrect pronunciation. The test words in which /i/ or /I/ were imbedded in /p-p/, /t-t/, or /k-k/ were uttered in the sentence “Say…/test word/…yet.” The preliminary data indicate that the durations of /i/ and /I/ were generally in the range of native American speakers, but that the Thai speaker overemphasized the difference in durations of /I/ and /i/. She also had the formant frequency ratios F2/F1 and F3/F1 for both /i/ and /I/ in the /p-p/ condition in the same range as the natives; her F2/F1 and F3/F1 for both /i/ and /I/ in the /t-t/ condition were too large. The Slovak had F2/F1 and F3/F1 smaller for /i/ and larger for /I/ than the native Americans in the /p-p/ condition; in the /t-t/ condition his F3/F1 was appropriate for /i/ but larger for /I/ than of the native Americans. Both Thai and Slovak speakers had all the formant frequency ratios in the correct ranges in /k-k/ conditions for which the native Americans showed the smallest differences between /i/ and /I/. These data indicate that the foreign accent of /i/ and /I/ of these two speakers is caused by spectral deviations from the American English, and not by durations.
Hearing vowels that aren't there: The perception of “unvoiced” /i/ and /u/ in Japanese
Authors: Shoji Atsuko , Beckman Mary
Published: October 2005

In descriptions of standard Japanese, linguists often refer to short /i/ and /u/ as being “unvoiced” in voiceless environments. A phonetically more accurate description, however, would have the vowels “deleted,” since spectrograms usually show only the spectral characteristics of the syllable-inital consonant. On the other hand, it could be argued that the term “deletion” does not reflect the subjective linguistic reality, because in most syllable types, phonotactic constraints prevent the deletion from neutralizing the /i/:/u/ contrast (e.g., dental fricatives cannot occur before /i/, so that a syllabic [s] always cues a missing /u/). The one exception is the palatal fricative [∫], which occurs with both short /i/ and short /u/. We elicited tokens of syllabic [∫] in words from minimal pairs, and presented them to native speakers to score on a five-point scale ranging from “definitely /u/” to “definitely /i/.” Although identification was far from perfect, the scores do not show the complete neutralization expected. Moreover, there was a significant dependency between the score and the frequency of the lower edge of the fricative's noise band, suggesting that there can be some spectral “coloring” cueing the missing vowel.
Measurement of acoustic intensity in reverberant sound fields
Authors: Elko G. W. , Tichy J.
Published: October 2005

The analysis of acoustic intensity measurement in propagating unbounded acoustic fields has been widely investigated in the last five years. There has, however, been very little published on the measurement of acoustic intensity in reverberant environments. It is the object of this talk to present some basic theoretical and experimental results that pertain to the nature of understanding sound energy flow in reverberant and specialized active acoustic fields. Measurements of a source in a reverberation room will be shown for distances of up to four times the critical distance from the source. The effects of bias and random errors in context to reverberant fields will be discussed and related to the experimental results presented.
Control valve noise in fluid process and transmission systems
Authors: Allen E.
Published: October 2005

This paper presents the results of indepth studies of noise associated with control valves in fluid process systems and/or transmission systems. Studies conducted provide verification of critical parameters and relative scaling effects. Discussion topics include equations for reliably predicting noise resulting from fluid flow thru control valves, effects of adjacent piping, and design criteria for eliminating or minimizing control valve noise. There will also be a comprehensive discussion of the technology—in hand—to minimize valve noise cost effectively via innovative design and application guidelines.
Characteristics of leaky Rayleigh waves on layered half-spaces
Authors: Chimenti Dale E. , Nayfeh Adnan H.
Published: October 2005

Theoretical and experimental results are presented to illustrate the propagation of leaky Rayleigh waves on lossless, layered half-spaces in contact with a fluid, and their influence on reflected acoustic beams incident at the Rayleigh angle. The presence of the layer gives rise to dispersive effects in both the Rayleigh wave velocity and the rate of its energy leakage. Good agreement between the measured reflected beam profile and the prediction of an extended analytical model is demonstrated under most conditions. For layers softer than the half-space (loading case), an infinite number of propagating surface modes exist. In the stiffening case, however, only a single mode exists, whose Rayleigh wavespeed is bounded by the Rayleigh and transverse wavespeeds of the substrate in a narrow band of the layer thickness over acoustic wavelength, d/λ. At the substrate transverse wavespeed, the mode ceases to propagate. We find that except for the lowest-order mode for the loaded half-space, the imaginary part of the wavenumber of all other propagating modes approaches zero at the respective wave cutoffs. Comparisons of the reflected beam profile predicted by the extended analytical mode close to a cutoff are shown to deviate from results of direct numerical integration of the reflected field integral, in qualitative agreement with experimental observations. [A. H. N. supported by NSF Grant CME 8021054 and AF Contract F33615-80-C-5015.]
Inverse scattering in a stratified medium
Authors: Stickler D. C.
Published: October 2005

A point harmonic source at frequencies ω1 and ω2 is located in the region z < 0, where both the sound-speed and density profiles are known. These two profiles are recovered from a measurement of the pressure field for all r at some fixed depth z < 0 at the two frequencies. Trace formula methods are used. The following assumptions are needed. First, c(z) and ρ(z) approach c1, ρ1 as z→ − ∞ and c2, ρ2 as z→ + ∞, and although c2, ρ2 are not known, it must be known that c2⩾c1. Second, the angular frequencies ω1 and ω2 must excite no proper modes, i.e., only the continuous spectrum is excited. Third, the sound speed c(z) may be complex, but c1 and c2 must be real. If c(z) also depends on frequency and the form of the dependence is known, say c(z) = cR(z) + iωλ(z), cl(z), then cR(z), cl(z), λ (z) can be recovered along with the density. If the density is known and c(z) is real then only one frequency is required. Two numerical examples are given, using parameters associated with ocean bottom sediments.
The effect of noisy reflection data on an inverse method for determining the structure of a layered ocean bottom
Authors: Thomson David J. , Herstein Peter D.
Published: October 2005

A basic inverse problem in underwater acoustics is the determination of the structure of the seabed from a limited knowledge of the reflection coefficient. For many applications, an adequate model for studying the acoustic interaction is provided by the scattering of plane waves by a layered liquid medium. In contrast to formally exact solutions to this inverse scattering problem, Candel et al. [J. Sound Vib. 68, 571–595 (1980)] develop and approximate scheme which is readily implemented numerically. Under the assumption of nonabsorptive media, the reconstruction of both density and sound-speed profiles is effected by the numerical integration Of a system of four first-order differential equations requiring impulse response data for two distinct grazing angles. In this paper, the effect of additive noise on the inversion algorithm is examined for synthetic reflection data generated for a geoacoustic model of deep-sea sediments representative of the Hatteras Abyssal Plain.
Acoustic tomography in a randomly fluctuating ocean
Authors: Kendall Elizabeth A.
Published: October 2005

Ocean tomography [W. Munk and C. Wunsch, Deep Sea Res. 26, 123–61 (1979)] introduces several complications beyond those of conventional inverse theory. The ocean environment represents a significant challenge to the problems of time keeping and data handling. Further, the ocean is a randomly fluctuating medium that changes rapidly in time. These random effects must be either utilized or filtered out by estimation procedures before inverse problems in underwater acoustics can be solved. The random effects are most significant when amplitude ocean acoustic tomography is attempted because micromultipath interference can cause large amplitude discrepancies. A previous attempt to invert for sound-speed structure in a randomly fluctuating ocean failed [M. Brown, U. Calif. San Diego, Ph.D. thesis (1992)]. This paper outlines a successful method for inversion in a randomly fluctuating medium. Simple random inverse problems, such as two-dimensional travel time and one-dimensional waves, are studied. Criteria for stability, uniqueness and model resolution are established. The size of the fluctuations and the extent of the randomness are varied, and an iterative scheme for higher order terms is discussed. The paper provides an analysis of the stochastic forward or direct problem. Parametric and nonparametric estimation procedures are used to solve the inverse problem, and estimator and filter performance are analyzed. Computer solutions to simulated random inverse problems are given. This work is ongoing and will culminate in a successful solution to the random ocean inverse problem.
Hale J. Sabine—The Fiberglas years, 1961–1974
Authors: Huggins Donald , Sundberg Donald
Published: October 2005

Hale Sabine moved from the Chicago area to Granville, Ohio in November of 1961 to become the manager of the Sound Laboratory at the Owens-Corning Technical Center where he worked until his retirement in 1974. The sound laboratory had been in operation since May 1960 conducting acoustical testing in the areas of product noise control, sound transmission through walls and between rooms through suspended ceiling paths, as well as sound absorption by reverberation room methods. Under Hale's management this work continued and expanded so that by early 1962 Owens-Corning published the “Speech Privacy Design Analyzer” containing both noise reduction through walls, ceilings, air handling systems, and doors as well as masking sounds produced by HVAC terminal devices. Additional field and laboratory evaluations were published in 1963 as “Solutions to noise control problems in construction.” While speech privacy analysis did not gain acceptance for closed room acoustical design, the increase in open-offices after 1970 reinforced the need to evaluate acoustical design in terms of both speech level reduction and masking background. Hale's published papers on the Acoustics of Open-Plan Offices were among his best work from the Fiberglas Years.
History of sound absorption testing
Authors: Northwood Thomas D.
Published: October 2005

Although people must have been aware of the acoustic pecularities of rooms since the first buildings were built, the first systematic studies of room acoustics were those begun by W. C. Sabine around 1895. Sabine discovered the relation between reverberation in a room and its sound absorption. He found that, to an adequate approximation, he could calculate the room absorption by adding up the areas of the various room surfaces multiplied by appropriate sound absorption coefficients for the respective materials involved. Thus the way was open for the design and prediction of the reverberation properties of rooms. Sabine's instrumentation consisted of a stopwatch, a set of organ pipes, and his two ears, and although this sufficed to solve his immediate problems, it was recognized as a pretty primitive approach. Both the theoretical bases and the measuring technology have been evolving ever since. The paper traces these processes from the viewpoint of their effects on acoustical design.
Velocity thresholds for simulated moving auditory signals
Authors: Grantham D. Wesley
Published: October 2005

An experiment was conducted to determine the auditory system's sensitivity to simulated movement of acoustic events in the horizontal plane. Movement was simulated by dynamically varying both the interaural time and intensity differences of waveforms presented dichotically over headphones. Stimuli were pure tones of 250, 500, 1000, or 2000 Hz. Durations varied from 100–500 ms; overall level was 83 dB SPL. In a two-interval forced-choice procedure subjects selected which of two tones was the “moving” tone. One of the two intervals contained a stationary signal. The other interval contained a single, left-to-right sweep of the same duration as the stationary signal; its velocity was varied in an adaptive procedure to track threshold. For 500-ms stimuli thresholds were 10–15 deg/s for the three lower frequencies tested and about 40 deg/s for the 2000-Hz signal. As stimulus duration decreased, thresholds increased, but not in such a way that “distance traveled” remained constant. At all durations tested, performance remained significantly worse at 2000 Hz than at the lower three frequencies. The best thresholds in this experiment were about two orders of magnitude greater than velocity thresholds determined for the visual system. [Work supported by NSF.]
Detection of interaural differences of intensity in trains of filtered clicks
Authors: Dye R. H.
Published: October 2005

Threshold values of interaural differences of intensity (Δ IDI) were measured as a function of the number of clicks(n) and interclick interval (ICI = 1,2,5,10 ms) for trains of clicks that were bandpass filtered at either 4000 or 2000 Hz. Data were also gathered for trains whose clicks alternated between 4000 and 2000 Hz. The log-log plots of Δ IDI (in dB) vs, n are quite similar for the single-band and alternating-band conditions, all showing shallower slopes at shorter ICIs. Furthermore, the slopes at a given ICI are quite similar across filtering conditions. These data are discussed in terms of possible mechanisms of binaural integration across bands of frequency.
An ear asymmetry in learning gap detection
Authors: Efron R. , Yund E. W. , Nichols-Mello D.
Published: October 2005

Gap detection performance was tested in each ear of 56 normal hearing subjects using 300 ms wide-band (200–4000 Hz) unfrozen noise bursts. No ear asymmetry was observed in the gap detection thresholds. In subsequent studies of individual subjects in which the same noise burst was used in all trial (frozen noise) a progressive improvement in gap detection threshold was observed over a 20-day period for both ears. Over the 20-day period, however, a highly significant ear difference was observed in the learning curves of the two ears despite the fact that the right and left ears were tested on alternate trials. The ear asymmetry in gap detection was not present in the first few days for two subjects but became progressively more obvious as testing proceeded. The third subject exhibited an ear asymmetry on the first day's block of trials which increased only slightly over the 20-day period. The right-left asymmetry in learning gap detection in frozen noise is difficult to explain without postulating some (asymmetrical) central processing mechanism.
Cued dichotic listening
Authors: Carney E. J. , Carney A. E. , Chmiel R. A. , Fitzsimmons L. K.
Published: October 2005

A new dichotic paradigm, called cued dichotic listening, was developed in which subjects receive the monotic presentation of one member of a dichotic pair of syllables prior to the dichotic presentation of the stimulus pair. In this procedure, subjects receive maximum information about the stimuli in one ear, called the cued ear. During dichotic presentation of a pair, the subject must identify only the stimulus in the noncued ear. Stimuli were synthetic syllables composed of the stop consanants /p,t,k,b,d,g/ plus the vowel /ɑ/, arranged into 30 dichotic pairs. Subjects received cues in both left and right ears on successive runs. Right ear advantage (REA) was calculated by subtracting the left-ear score (cued right ear) from the right-ear score (cued left ear). Subjects also received a standard dichotic condition with no cueing. Results indicated that the REA was increased in the cued condition compared to the standard condition. Cueing the left ear improved right-ear performance; cueing the right ear did not affect left-ear performance compared to the standard condition. These results suggest changes in phonetic processing, rather than auditory or attentional factors, contribute more to the REA.