bluuebunny/crossref_metadata_2025 huggingface dataset:
Right-left asymmetries in sound field localization in normal and anterior temporal lobectomized subjects
Authors: Nichols D. , Yund E. W. , Efron R.
Published: October 2005

Normal right-handed subjects (15 male and 15 female) localized 5-s complex sounds without head movements from one of seven small loud speakers (separated by 30° on the azimuth) under two conditions. In the first, the “target” sound was presented in isolation (the speakers in the unused positions being silent). In the second condition the same target was used but six discriminably different sounds were also present—each emanating from a different loudspeaker. Under the first condition there was no significant right-left asymmetry in localization accuracy in normal or temporal lobectomized subjects (33). In the second condition, a right auditory field superiority in localization accuracy was observed for the 30 normal subjects (p < 0.0003). The right anterior temporal lobectomized subjects also exhibited the (normal) right field superiority (p <0.001), while left anterior temporal resections appeared to obliterate this normal asymmetry. These effects may derive from the larger auditory association cortex (planum temporale) on the left side which may be involved in localizing sounds in a complex auditory field. [Work supported by Research Service, Veteran Administration.]
Comparison of three pattern recognition methods using broadband sonar echoes
Authors: Martin Douglas W. , Au Whitlow W. L.
Published: October 2005

Three target recognition methods were tested using broadband ultrasonic echoes from nine cylinders differing in material, size, and internal structure. Echoes were obtained from targets ensonified with simulated dolphin echolocation pulses, and were processed by the human auditory system, by the constant-Q filter-bank model of Chestnut et al. [J. Acoust. Soc. Am. 66, 140–147 (1979)], and by a time-domain feature model derived from descriptions of features used by humans. Humans appear to discriminate between these targets by using features related by differences in time separation and differences in relative amplitudes of each echo highlights. The filter-bank model uses a nearest-neighbor algorithm, described by Chestnut, to evaluate frequency-dependent reflection characteristics, and the time-domain model calculates distances between test and reference feature lists, where features contain time separation and amplitude information of the highlights. All systems peformed above 90% correct under ideal conditions. The filter model was adversely affected by slight changes in the configuration of the echo collection system which had no effect on the other models. Data comparing performance of the three recognition methods in noise will be discussed.
A projected equation for estimating horizontal coherence in the ocean
Authors: Beran Mark J. , McCoy John J.
Published: October 2005

The horizontal and vertical coherence that result from scattering of acoustic waves in the ocean cannot in general be treated separately. The governing equation couples the two effects. If, however, there are two distinct length scales present then an approximate decoupling is possible. Here we shall show what conditions are necessary for decoupling and using a two scale embedding procedure derive an equation governing the horizontal coherence in terms of effective horizontal scattering functions.
Fluctuations in shallow water acoustic transmission loss: Comparisons with environmental variability and theoretical predictions
Authors: Ali Hassan B.
Published: October 2005

An acoustic signal propagating in the sea is generally degraded not only by interactions with the bottom and surface boundaries, but also by volume inhomogeneities caused by nonuniformities in temperature, density, and salinity distributions. The degradations in the acoustic signal are manifested by fluctuations in its amplitude and phase and by an accompanying loss in its coherence properties. The results of experiments conducted in several shallow water areas are used to establish correlations between fluctuations in acoustic transmission loss and variability in the environmental parameters. The physical processes responsible for the observed fluctuations are identified primarily with diurnal and semi-diurnal tides and inertial effects. Using a modified version of SNAP (SACLANTCEN Normal Mode Acoustic Propagation Model) comparisons are made between measured and predicted acoustic transmission loss.
Scintillation behind non-Gaussian fractal phase screens
Authors: Berman David H.
Published: October 2005

Recent evidence indicates that ocean bottom height statistics are non-Gaussian (in fact K-distributed) and fractal with dimension D < 3/2. This paper investigates consequences of this fact for underwater acoustic scattering by examining the statistics of an acoustic field far behind a K-distributed fractal phase screen. In the case of Gaussian phase screens it is known that far from the screen, the scintillation index of the field saturates at its Rayleigh value of 1. In the case of K-distributed phase screens, the scintillation index again saturates to the Rayleigh value. However, for extremely non-Gaussian screens with fractal dimension D < 3/2, the rate of approach to the asymptotic limit is considerably slower than for Gaussian screens. In addition, in this last case the scintillation index approaches 1 from below rather than from above as in the Gaussian case. This means that signal detection might be easier than in the case of a Gaussian phase screen because intensity fluctuations are more constrained.
Reciprocal acoustic transmissions for monitoring ocean currents: Results from a mesoscale experiment
Authors: Worcester Peter F. , Spindel Robert C. , Howe Bruce M.
Published: October 2005

By simultaneously transmitting acoustic pulses in opposite directions between two points in midocean, one can separate current effects on acoustic propagation from the effects of sound-speed structure. Reciprocal acoustic transmissions can therefore be used to measure ocean currents. A short test conducted during summer 1983 to the west of Bermuda yielded approximately 3 weeks of reciprocal acoustic data from transceiver moorings separated by 300 km. Daily average arrival patterns at the two moorings are very similar, so that there is no difficulty identifying corresponding arrivals; pulse shapes and amplitudes of corresponding arrivals can differ significantly for a single reception, however. The differential travel times that are measured have reasonable magnitudes; a low-frequency trend is present that is probably the result of mesoscale fluctuations in the ocean.
Shallow water waveguide characterization using Hankel transforms—Preliminary experimental results
Authors: Frisk G. V. , Lynch J. F. , Doutt J. A.
Published: October 2005

Theoretical and computer simulation results recently obtained by Frisk and Lynch [G. V. Frisk and J. F. Lynch, J. Acoust. Soc. Am. 76, 205–216 (1984)] indicate that one can perform a relatively simple cw experiment which would, after processing of the data using the Hankel transform, yield a depth-dependent Green's function which contains information about the nature of the discrete and continuous modal spectra, as well as the plane wave reflection coefficients of the waveguide boundaries. Specifically, the method consists of measuring the magnitude and phase versus range of the pressure field due to a cw point source and then numerically Hankel transforming these data to obtain the Green's function versus horizontal wavenumber. In order to test this technique in shallow water, a series of experiments was performed in May 1984 at selected sites in Buzzards Bay and Nantucket Sound. The data collected at these sites appear to be of extremely high quality, and preliminary analysis using both forward modeling of the pressure field and the inverse Hankel transform (to obtain the Green's function and thus the modal structure) indicates that the method has considerable promise. Modal interference patterns in the pressure field and modal peaks in the Green's function are clearly observed. Of particular interest is the observation of a so-called “virtual” mode or “pseudoresonance” in the modal continuum, which we believe to be the first clear measurement of continuum structure in ocean acoustics. [Work supported by ONR.]
Inverse analysis for determination of the complex shear modulus of bottom sediments from surface water wave-induced bottom motion
Authors: Yamamoto Tokuo
Published: October 2005

An inverse procedure for complex shear modulus profile was developed based on the Yamamoto theory of poro-elastic bed response to surface water waves. Laboratory wave tank experiments were conducted to measure the wave-induced bed motion of sand beds and clay beds. Seismometers were used to measure the small motion of sand beds while movies were used to measure the large motion of clay beds. The complex shear modulus (rigidity and frame loss) profiles of beds were determined from the bed motion data using the inverse theory. Good agreements were observed between the inverted complex shear moduli and the complex shear moduli determined directly from the torsional resonance column tests. It is concluded that the proposed inverse method is powerful and useful for determining the bottom sediment properties in relation to geoacoustic modeling. [Work supported by ONR.]
Scattering from fluid-loaded, layered elastic objects
Authors: Feuillade C. , Werby M. F. , Gaunaurd G.
Published: October 2005

We examine acoustical scattering from submerged elastic shells with an elastic inclusion, assuming welded boundary conditions at the solid-solid interface. Backscattered form functions are examined over a large frequency range for different combinations of materials and shell thicknesses. We observe the following results. Case 1: The impedance of the inner inclusion is equal to or greater than that of the outer shell: (1) For very thin shells, the object scatters rigidly over a broad frequency range. (2) For shells of intermediate thickness the behavior is that of an elastic solid characteristic of the outer material. (3) Thick outer shells are dominated by internal reflections for suitably large frequencies. Case 2: The impedance of the outer shell is much greater than that of the inner inclusion. The backscattered form function is dominated by an interference effect between surface reflections and inner inclusion reflections. Explanations of these phenomena are presented.
Elastic acoustic scattering as a probe of microgeometry in porous media
Authors: Johnson David L. , Roberts James N.
Published: October 2005

Elastic scattering of sound by the inhomogeneities in a disordered system is studied within the weak scattering limit; this is analogous to “small-angle” scattering using neutrons or x rays. A suitable sample would be an epoxy double pore east prepared from a sedimentary rock; the correlation lengths are approximately 10—100 μm which are too large to be studied with x rays or neutrons. The scattered intensity is simply related to the autocorrelation function of the sample, which we have deduced from direct observations using SEM pictures. Thus, for the first time, one is able to predict the scattering behavior from direct observations. The relationship of the scattering to the specific area and to the topological propeties of the system are discussed.
Resonance scattering of elastic waves by an elastic sphere in an elastic medium
Authors: Brill D. , Gaunaurd G.
Published: October 2005

We study elastic wave scattering by an elastic sphere within an elastic matrix under the light of the Resonance Scattering Theory (RST). We consider shear (s) and compressional (p) plane wave incidence on a sphere of radius a, and the four cases of elastic wave scattering that they engender (i.e., pp, ps, sp, and ss). In all cases we isolate the resulting modal resonances by suitable background subtraction and generate the resonance spectrogram of the target in a wide frequency band that includes the resonance region where λ < a. We exhibit the SEM poles that appear in this general case in the complex-frequency plane. We study their connection with the surface waves that circumnavigate the spherical inclusion, both inside and outside it. This work contains four simple but important cases that we have extensively studied in the past [viz., (i) (visco) elastic sphere in a fluid, (ii) fluid-filled cavity, (iii) fluid sphere in a dissimilar fluid, and (iv) impenetrable (rigid/soft) inclusions in a (visco) elastic matrix] and they all emerge from it as particular cases. We show the specific form of the RST in the general case and the pertinent SEM poles with their interpretation and relationship to surface waves. We determine the phase velocities and attenuations from the pole positions, and also show the response surface, cross sections, modal resonances, etc. in wide frequency bands. This work considerably extends the 1960 analysis of Truell, Einspruch, and their associates on this classical problem.
The state-of-the-art parabolic equation approximation as applied to underwater acoustic propagation with discussions on intensive computations
Authors: Lee Ding
Published: October 2005

The Parabolic Equation (PE) has applications in many different scientific fields such as electromagnetics, optics theory, quantum mechanics, plasma physics, seismology, underwater acoustics, etc. This presentation centers on a discussion of the parabolic equation approximation as applied to underwater acoustic wave propagation—past, present, and future directions. A review will be given of past contributions. Recent developments are highlighted. Looking ahead we discuss what the parabolic equation method can do in order to stimulate future research and development, as well as applications. Intensive computations with respect to the parabolic equation implementation will also be discussed.
Modeling anomalous vocal tracts for investigating vowel production
Authors: Landahl Karen L. , Gould Herbert J.
Published: October 2005

Apert and Crouzon syndromes involve head and neck malformations affecting vocal-tract architecture through midface hypoplasia, malocclusion, reduction in pharyngeal depth, anomalous velar, and palatal configurations. Acoustic data on vowels from Apert and Crouzon speech are correlated with results from computer simulations of vowel production [Henke, Hosein] which convert information on physical measures from the speakers' articulations into values for a series of transfer functions. These investigations help determine consequences to vowel production resulting from these malformations. We focus first on [u]: (1) acoustic analysis reveals [u] to have the poorest relationship to reported normal formant values, (2) normal [u] articulation requires expansion of oral and pharyngeal cavities—difficult for these subjects. Physical measures were determined from midsaggital [u] and frontal cephalograms, dental casts and patient photographs. Successive alterations were made to the model to achieve convergence with known acoustic output. Current findings indicate: (1) a significantly compressed vowel space for these speakers, (2) little relationship between oral cavity volume and [u] formant structure, (3) strong relationship between pharyngeal volume, lip aperture, and oral/pharyngeal constriction for the first two formants.
Cross-linguistic issues in the relationship between semivowels and vowels
Authors: Maddieson Ian , Emmorey Karen
Published: October 2005

Do semivowels vary in quality from language to language in the way that vowels do? Is such variation related to cross-language differences in the cognate vowels? Are semivowels simply nonsyllabic variants of these vowels? To address these questions, nonsense words containing the semivowels /j/ and /w/ in the context of the cognate high vowels /i/ and /u/ as well as the low vowel /a/ were recorded from speakers of three different languages—Amharic, Zuni, and Yoruba. Formant frequencies in the semivowels and vowels were measured. Results show that semivowels do differ across these languages and that the differences are related in part to differences in vowels. Semivowels also differ from vowels in ways that suggest they are more “consonantal” in their production. Finally, these languages have distinct patterns of coarticulation between semivowels and adjacent low vowels. [Work supported by NSF.]
A two-dimensional model of jaw movement during speech
Authors: Edwards Jan , Harris Katherine S.
Published: October 2005

It has been known for some time that jaw movement during speech includes both translation and rotation [C. H. Gibbs and T. Messerman, ASHA Rep. No. 7 (1972)]. However, present models of articulation treat jaw movement either as pure rotation [P. Mermelstein, J. Acoust. Soc. Am. 53 (1973)] or as pure vertical translation [Y. Kakita and O. Fujimura, J. Acoust. Soc. Am. Suppl. 1 62, S15 (1977)], primarily because the data needed for a more accurate model have not been available. In this experiment, the jaw movement of three subjects, recorded during the production of VCV disyllables, was separated into rotation and translation, using an anatomically based model. We analyzed the relationship between these two components as a function of changes in the segmental and the suprasegmental context. Preliminary results for two subjects suggest that the additional degree of freedom of jaw movement is related to vowel front/back characteristics. [Work supported by NINCDS grant NS-13617 to Haskins Laboratories.]
A new split-step Fourier algorithm for solving the parabolic wave equation with rough surface scattering
Authors: Tappert Fred , Nghiem-Phu Lan
Published: October 2005

A novel technique based on a generalized method of images has been developed to solve the parabolic wave equation with irregular ocean surface treated as a pressure release boundary. This exact technique exploits the fact that full fast Fourier transforms performed on both real and image oceans have extra degrees of freedom not available when fast sine transforms are used, and makes possible a split-step Fourier algorithm that ensures that the pressure field vanishes at the prescribed surface elevation as a function of range. The new algorithm is nearly as fast (per range step) as the standard flat surface split-step Fourier algorithm and retains the other PE model advantages of being fully diffractive and fully range dependent. Examples obtained using the special purpose compact microcomputer system PESOGEN (PE solution generator) will be presented.
The dependence on low-frequency bottom backscattering on the parameters of the measurement systrem
Authors: Wyber R. J. , Muir T. G.
Published: October 2005

When using a parametric sonar it is possible to ensonify areas of the bottom with dimensions in the order of a wavelength. When this occurs the assumption of random independent scatterers which may be valid for conventional systems no longer applies and the structure of the scattering system becomes important. In this situation the variation of the scattering strength with pulse shape, length, and beamwidth need no longer follow the classical behavior. To measure these parameters a series of experiments was carried out off Oahu, Hawaii using a parametric sonar with a bandwidth of 0.5–5 kHz in a side-scan mode to transmit an impulsive signal. In the backscattered signal it was possible to identify individual biological scatterers and separate their effect from the bottom backscattering. The accurate position information in the measurements then allowed the data to be processed to determine the dependence of the backscattering on the ensonified area. [Work supported by the Office of Naval Research.]
Radiation of sound induced by laser beams incident on water surfaces: A tutorial survey of the world literature
Authors: Pierce Allan D. , Hsieh Hsiao-an
Published: October 2005

One of several topics associated with laser-induced sound that has received great interest in recent years is the generation of underwater sound by a laser above a water surface. Although preceded by many noteworthy contributions (dating back to nineteenth century papers by Alexander Graham Bell on the “photophone”), the first experiments explicitly exhibiting the effect were reported in JETP Letters in 1971 by F. V. Bunkin and his colleagues at the P. N. Lebedev Physics Institute. Since that time approximately 100 papers have appeared on the subject; extensive contributions have been made by V. M. Komissarov, A. I. Bozhkov, O. V. Rudenko, K. A. Naugol'nykh, L. M. Lyamshev, S. G. Kasoev, L. V. Sedov, A. M. Prokhorov, S. V. Egerev, Al. A. Kolomenskii, A. A. Karabutov, and many others. Several reviews have been recently published by Lyamshev, Sedov, and Naugol'nykh. In addition to calling attention to this research on a new and exciting area of acoustics, the present paper is offered with the hope that an independent overview may be helpful to those desiring a greater familiarity with the subject. [Work supported by ONR, Code 425-UA.]
Nondestructive determination of ultrasonic velocity and absorption in thin dielectrics by laser-induced pressure pulses
Authors: West J. E. , Sessler G. M. , Gerhard-Multhaupt R.
Published: October 2005

The laser-induced presure pulse (LIPP) generation of acoustic impulses in the 0.5-ns range allows the measurement of ultrasonic velocity and absorption in thin dielectrics in the 0.2- to 1-GHz range. This optoacoustic method generates ultrasonic pulses by thermoelastic effects and ablation due to absorption of 70-ps laser-light pulses in an opaque coupling layer deposited on one of the sample surfaces. Detection of the acoustic pulses is possible by using electrically charged or piezoelectric samples which yield electrode signals upon transit of the pulse. The sound velocity and absorption is determined from the relative delays and spectra, respectively, as the pulse is reflected from the sample surfaces. Multiple LIPP by splitting the laser beam into two time (or more) components and delaying one component relative to the other by a time T results in a boost of the spectrum at the frequency f = 1/T. Measurements on several polymers with this technique gives sound speed close to the values determined for the bulk material and attenuation coefficients that rise almost linearly with frequency in the range 0.2–1 GHz.
Optoacoustic effect in SF6
Authors: Ali Manaf , Bass Henry E.
Published: October 2005

Short pulses (∼1 μs) of 10.6-μ radiation from a CO2 laser have been used to generate optoacoustic pulses in gaseous SF6. The optical path length for 10.6-μ radiation in SF6 is very short at intermediate pressures (∼1 Torr) so a strong acoustic pulse is generated which travels in the direction of the incident laser beam (in addition to cylindrical expansion). As the gas pressure is decreased, the optical path length increases resulting in an acoustic pulse with a shape dominated by collisional transfer in SF6. As the pressure is lowered further (∼200 mTorr), thermal conduction to the test cell walls and spontaneous radiation by the excited gas become important. As these mechanisms begin to compete with slow collisional energy transfer processes, the optoacoustic signal changes in sign. A microscopic description of the behavior will be offered. [Work supported by ONR.]
Achievement of substantially higher source levels for airborne-laser-induced underwater sound
Authors: Pierce Allan D. , Hsieh Hsiao-an
Published: October 2005

A properly designed airborne laser system can generate a spatially periodic heating pattern (of length L and periodicity L/n) that moves with supersonic speed V over the water surface; such creates an n-cycle tone-burst radiating downward at an angle θ = cos−1(c/V) with the horizontal, the sound having wavelength λ = (L/n)cos θ and frequency f = c/λ. Analysis is developed for when the laser deposits power P over time T, such that the heating pattern moves a net distance VT that is much larger than L. Calculations of the farfield radiation [r > (cT)2/λ] can yield a peak value of acoustic pressure times radial distance that is approximately fT/n times larger than what results for circumstances studied by Muir, Culbertson, and Clynch [J. Acoust. Soc. Am. 59, 735–743 (1976)]. Thus, if f is 10 kHz, T is 0.1 s, and n is 10, the source level will be 40 dB higher. The analysis assumes the laser heating penetration depth is somewhat larger than λ/sin 2θ, but this should ordinarily be well satisfied, especially if blue-green light is used. At a range of 10 km, one is typically still within the Fresnel regime; the theory estimates the peak pressure to be of the order of 0.2 Pa when the power P is 100 kW. It is argued that, even with the limitations of existing laser technology, it should be feasible to develop an airborne laser system that will create detectable sound at locations of will at ranges up to 10 km or greater. [Work supported by ONR.]
Programmable artificial ear incorporating desk-top scientific computer
Authors: Kennedy William A. , Egolf David P.
Published: October 2005

Sound-pressure recordings made in real-ear simulators are more realistic than those made in 2-cc couplers. Unfortunately, most real-ear simulators (1) require factory calibration and (2) were designed to simulate only the mean acoustic impedance of normal adult ears. Due to variability of ear-canal geometry and eardrum impedance among subjects, the statistical possibility of any one subject exhibiting such “mean” characteristics—especially if that subject has a conductive pathology—is very remote. Reported herein is the development of a programmable artificial ear (PAE) which (1) requires no factory calibration and (2) can be programmed to account for individual variations in source impedance, ear-canal geometry, and eardrum impedance. The PAE consists of a 2-cc coupler, microphone, amplifier, and desk-top scientific computer, equipment which is included in most laboratory/clinic inventories. Sound pressures recorded in the 2-cc coupler are adjusted via the computer to what they would have been had the recordings been made in a particular subject's ear. Good agreement has been obtained between PAE data and real-ear simulator measurements.
Speech perception in noise by fluent, non-native listeners
Authors: Florentine Mary
Published: October 2005

This experiment measured the ability of fluent, non-native listeners to understand American English in the presence of babble noise. The Speech Perception in Noise (SPIN) test was administered at 75 dB SPL to 11 native and 14 non-native young, normal listeners. The non-native listeners were university students and teachers who had lived more than four years in the United States. Six noise levels were chosen in 2-dB steps to encompass the range from 20%–80% correct performance. At each noise level, 50 new sentences were presented and the percent correct for 25 low- and 25 high-predictability sentences was measured. After each sentence the listener wrote the last word in the sentence and repeated it to the experimenter, who also wrote it. Results show (1) the native listeners could obtain 50% correct performance at significantly higher noise levels (about 3 dB) than the non-native listeners, (2) both groups performed significantly better on the high-predictability sentences than on the low-predictability sentences, and (3) the difference in tolerable noise levels (levels yielding 50% correct) between the high- and low-predictability sentences was the same for the native and non-native groups. [Work supported by the Research and Scholarship Development Fund of Northeastern University.]
Feature analysis of normal-hearing and hearing-impaired childrens' errors on the NST
Authors: Danhauer J. , Abdula C. , Johnson C. , Asp C.
Published: October 2005

The nonsense syllable test (NST) was presented to normal-hearing and hearing-impaired children whose verbal responses were phonetically transcribed and converted to confusion matrices. These data were also submitted to multidimensional scaling analyses to determine perceptual features used by the subjects. Features for each group are presented.
Response of hearing-impaired adults to a structured program of auditory training
Authors: Rubinstein Adrienne , Boothroyd Arthur
Published: October 2005

Twenty adults with mild to moderate sensorineural hearing impairments were given three tests of auditory perception: The CUNY Nonsense Syllable Test (NST), the low predictability items of the Speech in Noise (SPIN) test, and the high predictability items of the SPIN test. The tests were administered at a signal-to-noise ratio giving a score of approximately 50%, as determined with an adaptive procedure at the beginning of the study. They were tested on four occasions: (1) at the beginning of the study, (2) after one month of “no treatment,” (3) after a month of intensive auditory training, and (4) after a further month of “no treatment.” During the treatment period, ten of the subjects spent all of the time on global tasks (i.e., activities involving sentence perception and perceptual strategy) while the other 10 spent part of the time on analytic tasks (i.e., activities involving consonant recognition). The results showed: (a) that performance improved only during the treatment period, and (b) that the improvement occurred only in the high predictability items of the SPIN test. The effect of the training method was not significant, either as a main effect or in interaction with other factors. The findings suggest that the benefits of auditory training were found in an increased use of sentence context as an aid to word recognition and that the crucial component of training was “focused listening.”
Assessing the perception of intonation contours by children with severe and profound hearing loss
Authors: Most Tova , Boothroyd Arthur
Published: October 2005

Thirty hearing-impaired children were given two tests of intonation contour perception. The first involved a receptive task in which subjects had to identify the “odd man out” in three stimuli. The second involved an imitative task in which subjects had to imitate an intonation contour. The imitations in the second task were tape recorded for subsequent rating by five trained listeners on a ten-point scale. The same stimuli were used for both tests, these being computer-generated synthetic vowels with rising, flat, and falling contours. Among the older subjects (= >5 years), all the severely deaf (3 frequency average loss <90 dB HL) and about 50% of the profoundly deaf (3 frequency average loss = >90 dB HL) passed both the receptive and the imitative task. Among the younger children (<4 years), the proportions passing the imitative task were the same as in the older group. Only one severely deaf child, however, was able to pass the receptive task. The fact that the receptive task was too difficult for the younger age group was confirmed using normally heating children. (Both hearing and deaf children, however, were able to perform a visual-spatial analog of the task.) The findings suggest that the evaluation of intonation perception in young hearing-impaired children should be based on imitative tasks, rather than receptive tasks. They further indicate that all severely deaf, and approximately 50% of profoundly deaf children should have sufficient auditory capacity for the perception of intonation contours.
The perception of intonation in speech produced by the hearing impaired
Authors: Rubin-Spitz Judith , McGarr Nancy S.
Published: October 2005

Eight severely-to-profoundly hearing-impaired children produced each of 14 phrase-length utterances. F0 measures were obtained at several locations within each utterance (starting F0, final F0, and F0 peaks). The relative timing of each F0 measure (ms from onset of the utterance) was also determined. In addition, several difference measures were derived. Listeners experienced with the speech of the deaf served as subjects for the perceptual experiment and were asked to judge whether they heard a terminal fall or a flat final intonation contour. A multiple linear regression analysis was used to determine if any combination of the acoustic measures could predict listeners' responses. Results suggest that the only variable which made a significant contribution to the regression function was the temporal interval between the terminal peak F0 and the final F0. The regression equation accounted for a statistically significant but not a large proportion of the total variance. The implications of these results will be discussed. [Work supported by PHS grant NS17764 to the Graduate Center, CUNY.]
Computer simulation of the fitting of hearing aids based on prediction of the discrimination ability of speech sounds
Authors: Hiki Shizuo , Fukuda Yumiko
Published: October 2005

The earlier version of the computer program for simulating the process of fitting hearing aids [S. Hiki and Y. Fukuda, “Prediction of the ability of discrimination of speech sounds by hearing impaired children,” J. Acoust. Soc. Am. Suppl. 1 68, S5 (1980)] has been modified so that correction for individual speaker differences can be made. The latest version of the program consists of two parts. In the first part, by inputting various characteristics of the hearing loss of an impaired child and the candidate hearing aid, parameters of the hearing aid are controlled. The goal of the adjustment is to match the sound pressure versus frequency range, audible for each impaired child with the aided characteristics, to that of typical speech sounds. In the second part, the type of identifiable phones and probable phonetic confusions among the phones are estimated by analyzing the relationship between the aided characteristics and the acoustical properties of the speech sounds. The corrections for individual speaker characteristics involve the average amount of shift of formant frequencies for all vowels due to the differences of shape of the vocal tract, the deviation of average value of the fundamental frequency, and the deviation of intensity of the unvoiced sounds relative to the vowels.
Tactile aid usage with prelinguistic, profoundly deaf children
Authors: Proctor Adele , Goldstein Moise H.
Published: October 2005

Since 1977, we have followed a total of 12 prelinguistic, profoundly deaf children who used different amounts of tactile stimulation at various stages of language acquisition. Subjects ranged in age from 6 weeks to 5 years old at the onset of tactile aid usage. Longitudinal data measuring speech perception, speech production, and general linguistic behavior suggest that our nine successful tactile aid users followed a sequence of language acquisition similar to that outlined for normal hearing children [A. Peters, The Units of Language Acquisition (Cambridge U. P., Cambridge, England, 1983)] and that Peters' work provides a theoretical basis for incorporating tactile aids into language training programs for young deaf children. Language learning difficulties and socioenvironmental factors, not device design and construction, account for why three subjects did not become frequent tactile aid users. The cumulative effect of our data for all subjects suggests that there is a need for a gradation of devices providing increasingly complex tactile stimulation as the child matures linguistically and physically. Videotapes will be used to illustrate tactile aid usage. [Work supported by The Deafness Research Foundation and grant no. RR07143, Department of Health and Human Services.]
Medium band waveform coding of speech signals based on short-time phase equalization
Authors: Honda Masaaki , Moriya Takehiro
Published: October 2005

A new speech coding method is presented utilizing perceptual redundancy for slow-varying, short-time phase characteristics. First, a speech signal model is introduced based on the all pole filter and all pass filter, which represent the power spectrum and the short-time phase of speech signals respectively. The phase equalization technique for canceling the short-time phase is then presented based on the time domain matched filtering method. Using this technique, the prediction residual of the speech signal is converted into a nearly zero-phase signal, or ideally into a pitch pulse sequence, which results in the temporal concentration of residual energies. The phase-equalized speech signal is efficiently encoded using variable-rate tree coding with temporal bit allocation. Experimental results show that quality of the phase equalized speech is almost indistinguishable from the original speech, and that the coded speech quality is equivalent to that of the 6.6 bit Log-PCM at a bit rate of 1 bit/sample. It is also confirmed that this system has the ability to make a speech quality change from natural to vocoder sounds by controlling the information bits for the main pulses and its residuals of the phase-equalized residual signal.
Synthesizing an intelligible /h/
Authors: Nartey Jonas N. A.
Published: October 2005

This paper presents follow-up work on earlier data which suggested that /h/ is a voiceless vowel, and not a fricative or an approximant, as others have claimed [e.g., Strevens, 1960 for the former, and Ladefoged, 1975 for the latter]. Acoustic measurements of English /h/ spoken in various phonetic environments were compared to those of fricatives and approximants in similar environments. Results indicate that co-articulation effects of /h/ and its phonetic environments are similar to those of neither fricatives nor approximants. In another experiment, /h/ was synthesized in English utterances using identical parameters with (a) fricatives, (b) approximants, and (c) vowels. The utterances were presented to native American English speakers for both “naturalness” and intelligibility judgments. Results favored the [h] with the characteristics of a “voiceless vowel.”
Text-independent speaker recognition experiments using codebooks in vector quantization
Authors: Shikano Kiyohiro
Published: October 2005

A text-independent speaker clustering approach to speaker-indepencent speaker recognition through vector quantization (VQ) was investigated, where the distortion value was used as a clustering measure. To show the possibility of the text-independent speaker clustering, speaker recognition experiments were carried out using the Harvard sentence database. Nine male speakers uttered ten different Harvard sentences each. Codebooks were generated from the first five sentences for each speaker using Weighted Likelihood Ratio measure (WLR) through LPC analysis. Using 128 vectors in each codebook, a speaker recognition rate of 98% was attained on the latter five Harvard sentences. Effects of codebook size and input length are also discussed. The above approach based on framewise VQ only utilizes the static distribution of LPC spectra. VQ for multiframe codebooks was used to represent the coarticulation units. The results of speaker recognition experiments based on multi-frame codebooks will be compared with fixed length VQ approaches.
Speaker-independent recognition of isolated digits using a weighted cepstral distance
Authors: Tohkura Yoh'ichi
Published: October 2005

The cepstral distance has been one of the most efficient spectral distance measures in speech and speaker recognition [S. Furui, IEEE Trans. Acoust. Speech Signal Process. ASSP-29, 254–272 (1981)]. A new weighted cepstral distance measure using LPC derived cepstrum coefficient variability was tested in a speaker-independent English digit recognition system using standard DTW alignment techniques [L. R. Rubiner, S. E. Levinson, A. E. Rosenberg, and J. G. Wilpon, IEEE Trans. Acoust. Speech Signal Process. ASSP-27, 134–141 (1979)]. The results show a recognition accuracy of > 99% for the digits [K. L. Shipley. A. E. Rosenberg, and D. E. Bock, J. Acoust. Soc. Am. Suppl. 1 72, S80 (1982)]. Recognition results using the same data base and the log likelihood LPC distance are about 97.4%. Hence there is a large improvement in performance with the new weighted cepstral distance.
A single-boarded isolated word recognizer using LPC cepstrum
Authors: Hamada M. , Bessho Y. , Norimatsu T. , Yamada A.
Published: October 2005

Minicomputer simulation was carried out to design an optimum system based on currently available signal processing LSI's. First, finite-word-length effects of Levinson-Durbin (LD) algorithm and Le-Roux (LR) algorithm [J. Le-Roux et al., IEEE Trans. Acoust. Speech Signal Process. ASSP-25, 257–259 (1977)] for extracting PARCOR coefficients were investigated regarding (1) the PARCOR/AR/cepstrum coefficient error, (2) the difference in LPC cepstrum distance between the top two candidates, and (3) recognition rate. LR was found to be almost always better than LD by each of above measures. Second, the effects order of analysis, number of template bits, and the template normalization method were examined to minimize memory size. It was shown that the number of template bits of each cepstrum coefficient can be reduced to four with little decrease of recognition rate as compared to the system with floating point number templates. A single-boarded recognizer using TMS320 for LPC analysis and MN 1263 for DP matching was implemented. The overall recognition rate of on-line test in speaker-dependent mode was 99.4% for a ten-word vocabulary (total of 1000 tokens of ten speakers). Multiple template speaker-independent mode achieved a recognition rate of 97.0% for an eight-word vocabulary.
The use of allophonic variations of /a/ in automatic continuous speech recognition of French
Authors: Vaissière Jacqueline
Published: October 2005

The acoustic characteristics of the vowels in continuous speech are affected by their duration, the stress condition, and the surrounding phonemes. The purpose of this study is to investigate the extent of the allophonic variations of the most frequent and most variable vowel in French, /a/, and the integration of such variations in an automatic speech recognition system. The preliminary corpus consists of 200 vowels /a/ extracted from 128 sentences uttered by two speakers. Using the SPIRE facilities, the sentences were digitized and segmented [Zue and Leung, J. Acoust. Soc. Am. Suppl. 1 75, S59 (1984)]. The segments labeled as SONORANT corresponding to the occurrence of /a/ were extracted. The values of F1 and F2 at the onset, offset, and middle of the segments were estimated using automatic peak tracking from the LPC spectra. Our results suggest that: (1) the MANNER of articulation of the preceding and following consonant plays a negligible role on F1 and F2 onset and offset (in contrast with other data published for English and Swedish); (2) if the PLACE of articulation of the consonants is divided into FRONT, MID, and BACK, then the difference (F2-F1) at /a/ onset allows the unique specification of the place of articulation of the preceding consonant in 86% and 57% of the cases, for SP1 and SP2, respectively, and there is no overlap between FRONT and BACK consonants; (3) (F2-F1) at /a/ offset indicates that F2 of the following vowel has to be taken into account before interpretation; (4) determination of the thresholds (for MID and BACK consonants) can be done on a few well selected words, where the effect of the context is known to be maximal. Detailed descriptions of our findings extended to the study of more speakers, and of their consequences for the coding of the permissible variations of the vowel /a/ for each word of the vocabulary will be presented. [Work supported in part by the Office of Naval Research under contract N00014-82-K-0727.]
Shallow water acoustic modeling over a sloping bottom
Authors: Tindle C. T. , Deane O. B.
Published: October 2005

Ray theory with beam displacement gives an approximate method of finding the acoustic field in shallow water. It was shown to be quite accurate for a horizontally stratified two fluid (Pekeris) model by comparison with normal mode results. [C. T. Tindle, J. Acoust. Soc. Am. 73, 1581–1586 (1983)]. The method is extended to the sloping bottom situation by simple geometric arguments and without further approximation. Results show that even small bottom slopes have a dramatic effect on the sound field. The results are compared with those for the adiabatic normal mode approximation and agreement is good for higher frequencies. At lower frequencies differences are attributed to normal modes passing through cutoff, a process which is ignored in simple adiabatic mode theory.
Magnetic biasing of a rare earth transducer using permanent magnets
Authors: Howarth Thomas R.
Published: October 2005

Magnetic (dc) biasing of rare earth transducers is a necessary operation in order to eliminate frequency doubling. The uniform biasing of rare earth materials is very difficult due to its low permeability, while other magnetostriction materials like Nickel and Permalloy are easily biased since they have a high permeability. The present method of biasing consists of superimposing the ac field onto a dc power supply with the appropriate passive blocking components. An alternate method of biasing is to place permanent magnets in close proximity to the rare earth material in such a way as to provide a uniform flux distribution throughout the rare earth element. In order to provide the uniform flux distribution an extensive computer modeling effort was performed to choose the optimum magnetic path configuration and to determine the proper magnet type and shape. Results comparing these two biasing techniques on a Terfenol D (Fe1.90Dy0.73Tb0.27) driven low-frequency ring transducer are presented.
OSHA policy and court decisions on the noise standard
Authors: Macaluso Judith
Published: October 2005

OSHA's hearing conservation amendment to the noise standard was promulgated 16 January 1981. After a 7-month period of scrutiny and debate, most of the amendment's provisions became effective in August of 1981. Some modifications were made to the remaining portions of the amendment, and virtually the entire amendment, as modified, became effective in March of 1983. The Forging Industry Association filed suit in the Fourth Circuit Court of Appeals to vacate the amendment. The Court heard arguments by OSHA and the Forging Industry in April of 1984 and in November a two-member majority voted to vacate the amendment. At this time, the amendment is not enforceable in those parts of the nation where state standards do not apply. This discussion will cover the details of the Fourth Circuit Court's decision and OSHA policy with respect to appealing the decision and enforcing the noise standard.
Distribution based definition of impulse noise
Authors: Erdreich John
Published: October 2005

Existing definitions of impulse noise are based on estimates of either the decay of the noise envelope, the repetition rate of the peaks, or a comparison of single peak level to rms level. Because each of these measures fails to fully characterize at least one of the relevant characteristics of an impulse, many exceptions to these classifications can be found. An improved alternative classification should be independent of absolute pressure magnitude, impulse duration, and repetition rate; yet, it should incorporate all these on a relative basis. The kurtosis (b2) of the amplitude distribution of a sampled impulse is such a descriptor. Examination of distribution kurtosis over samples of 1, 5, 11, and 42 s suggests that the “impulsiveness” of an exposure may be characterized by this statistic. The 11-s window also relates the statistic to click-induced TTS processes in the ear. Normally distributed data exhibit a kurtosis of zero. For comparison, average values of b2 calculated for the 11-s samples are: b2 = − 1.5 for a sinusoid, b2 = 65.4 for barrel recycling, b2 = 3.0 for coin blanking, b2 = 44.0 for forging, and b2 = 24.6 for stamping. Analysis of kutosis and crest factor shows that they are related measures of the distribution. In limited cases, therefore, crest factor may have utility as an interim indicator of impulsiveness.