bluuebunny/crossref_metadata_2025 huggingface dataset:
Quantal nature of speech production for vowels: Implications of cineradiographic and palatographic observations
Authors: Dowla Farid , Perkell Joseph S.
Published: October 2005

It has been suggested that distinctions among the classes of high, mid, and low vowels are produced by “quantally” different amounts of contact of the sides of the tongue with the hard palate and upper teeth [K. N. Stevens, J. Acoust. Soc. Am. 57, S70 (A) (1975)]. Low vowels would have no contact; mid vowels, slight contact; and high vowels, maximal contact. In a test of this hypothesis, direct palatographic and cineradiographic observations of the productions of American English vowels in the utterance /hə′bVb/ have been made for one speaker. The cineradiographic and palatographic measurements agree with one another in showing the classical height distinctions and in showing no tongue to upper tooth or palate contact for low vowels. Evidence for clear-cut distinctions in contact patterns for mid versus high vowels is obscured by differences along the front-back dimension. Front vowels do show the hypothesized mid versus high difference, but if the same difference exists for back vowels, it is manifested further back in the oral cavity than in the region normally observed by direct palatography. Further palatographic observations made to beyond the posterior limit of the maxillary alveolar ridge on additional subjects should enable us to explore the hypothesis further. [This work was supported by the National Institutes of Health grant NS04332.]
Three types of diphthongization
Authors: Collier R. , Raphael L. J.
Published: October 2005

Diphthongization of vocalic syllable nuclei is not a process of one kind. In Dutch, for instance, diphthongization can be of three different types, depending on whether the syllabic nucleus is a long vowel, a genuine diphthong, or a vowel + semivowel sequence. The perceptual and acoustic aspects of these differences are well documented in the literature. This paper complements the description with some articulatory data, based on electromyographic recordings of extrinsic tongue muscle activity in relation to the three types of diphthongization. [Work supported by NINCHDS grant NS 13870 and NFWO, Belgium.]
Vowel fundamental frequency and tongue height
Authors: Lubker James , McAllister Robert , Lindblom Björn
Published: October 2005

This research was designed to substantiate the results of Ohala [J. Acoust. Soc. Am. 60, S44(A) (1976)] which supported the “tongue-pull” hypothesis of intrinsic vowel pitch. Our goal was explanation of the observation that vowels with high tongue position have higher F0 than do vowels with low tongue position. It has been suggested that the anatomical interactions between tongue larynx put an added stress on that structure when the tongue is in an elevated position, thus raising F0 for the vowel. We replicated Ohala's use of bite blocks to force the speaker to produce articulatory “supershapes.” If the tongue-pull hypothesis is correct. a given vowel would have higher F0 with the jaw blocked in an unnaturally open position than with the jaw free. This was the case with Ohala's results, However, in our preliminary work with three subjects we have not been able to substantiate those results. One of our subjects showed no consistent bite-block effect on F0, while both of the remaining subjects used a lower F0 for vowels in the bite-block condition. Alternative solutions and criticisms of the assumptions of the present research will also be discussed.
Optical simulation of acoustic propagation in the ocean
Authors: Estes L. E. , Fain G.
Published: October 2005

We have constructed, using lasers and optical components, a simulator for acoustic waves propagating in a nonhomogeneous two-dimensional ocean. The resulting wavelength scaling provides for the contraction of tens of kilometer ocean distances to centimeter distances on the optical bench. Ocean velocity profiles which vary in range and depth are created by inducing a temperature distribution in a fluid with a high velocity-temperature gradient. This is produced by imaging a transparency (with the desired distribution) using a high-power-pulse ruby laser onto the fluid. A blue dye in the fluid selectively absorbs the ruby light. The acoustic beam is simulated by a cw blue argon-ion laser and the source and receiver patterns are generated via appropriate optics. Subwavelength-size scatters make the blue beam visible from the side providing a photographic measure of sound intensity at all ranges and depths. A simultaneous measurement of the light velocity profile is made using a Michelson interferometer. The timing of the experiment consists of a 1-msec ruby pulse immediately followed by a 1-msec shuttering of the sound wave simulating and interferometer laser beams. [Work supported by the Office of Naval Research.]
Propagation of the mutual coherence function in a bounded waveguide
Authors: DeSanto John A.
Published: October 2005

The mutual coherence function Γ is defined as the product of the pressure field p times its complex conjugate evaluated at a different point. In three dimensions this is Γ = p(r,θ,z)p*(r,θ̄,z̄), where both pressure fields are restricted to the same range plane. Integral equations for Γ in a bounded waveguide are derived assuming p satisfies the usual parabolic equation, and using a Green's function for the waveguide which satisfies a parabolic equation but with no sound-speed term. The Green's function thus depends only on the boundaries (its use is analogous to using the Laplace Green's function). We derive the equation for the ensemble average of Γ for the case of a random sound speed. An example is presented for a flat waveguide with a sound-speed field composed of refractive plus random parts, and the computability of such problems is discussed.
Numerical studies of basin bathymetry on three-dimensional ray traces
Authors: DeWitt R. N.
Published: October 2005

Many ray-tracing programs allow for a sloping or structured bottom. But when the normal to the bottom is tilted out of the vertical plane, a reflected ray is deviated from a straight-line path as viewed from above. Thus if multiple reflections occur between the sea surface and the bottom, a general curving of the ray path is expected. For low frequencies when the bottom losses are small, this ray curving can persist through many reflections. A computer program has been written to study actual ray paths taken by such propagation. By digitizing basin bathymetry and velocity profiles, three-dimensional ray traces are produced which display the effects of sea mounts, steep ridges, and curved basin walls. General characteristics of the computer program are addressed along with specific examples for certain areas in the Norwegian and Greenland Seas.
Three-dimensional ray paths in basins, troughs, and near sea mounts by use of ray invariants
Authors: Harrison C. H.
Published: October 2005

Long-range underwater sound propagation is often influenced by regions of repeated surface and bottom reflection. Under these conditions variations in water depth and bottom slope may produce a small change in ray heading after each bottom reflection, and the many resulting facets of the ray paths give the appearance of ray curvature in the horizontal plane. At low frequencies, where reflection losses are small, this curvature provides a mechanism for turning back rays from sea mounts or basin walls, and thus can introduce arbitrary increases in travel times and alterations in apparent bearing of sound sources. This paper uses ray invariants to derive analytical solutions for the horizontal projections of ray paths for many types of basins, troughs, sea mounts, or ridges. The isovelocity invariants for elevation angle and ray heading are extended to cover a basin with rotational symmetry as well as a trough of constant cross section, and the additional effects of refraction in a stratified medium are considered. Rough figures for the geometrical spreading loss and reflection loss at the edge of a basin are derived.
Asymmetry in judgments of temporal duration
Authors: Bull G. L. , Ruth R. A. , Schoeny Z. G. , McDonald W. E.
Published: October 2005

Two-sound sequences with interpulse intervals of 312 sec were presented to subjects in order to evaluate the character of judgments involved in temporal comparisons of the successive sounds. A 1000-Hz tone and wide-band noise were presented in all possible two-sound combinations. The duration of the first (referent) sound was 500 msec, and the duration of the second (comparison) sound ranged from 300 to 700 msec. The subjects' task was to state whether the second sound was shorter, of the same duration, or longer than the first sound. As the difference in the durations of the two sounds increased, accuracy of judgments improved. The effect was asymmetrical; judgments were more accurate when the second sound was shorter than the first sound and less accurate when the second sound was longer, given equivalent differences in duration. Subjective impressions were that noise bursts were longer than tone presentations of the same duration, which accords with the finding that the asymmetry in accuracy of judgments was diminished by tone-noise combinations and enhanced by noise-tone combinations.
Ultrasonic field determination by a form of computer-aided tomography applied to acousto-optic data
Authors: Cook Bill D.
Published: October 2005

The principles of reconstructing a two-dimensional field from projections can be applied to acousto-optic data to reconstruct an ultrasonic pressure field in a plane. Moreover, using the principle of wave propagation in linear acoustics, it is possible to reconstruct the total three-dimensional pressure field from the information acquired by interrogating a single plane. The “Fourier reconstruction” techniques are particularly adopted for this type of reconstruction.
Computer wavefront imaging from simulated acoustic holograms of a dipole and the bending wave in a plate
Authors: Cohen R. L.
Published: October 2005

It is commonly said that holograms have a high redundancy, i.e., any part of the hologram contains information about the entire image. This is not true in long-wavelength acoustical holography in which the number of fringes across the recording aperture is commonly less than 50. This is clearly seen when the hologram and image of a simulated acoustic dipole are examined. No matter how close together the positive and negative poles are, the image shows two sources a wavelength apart. However, using only half of the hologram at a time and making use of the phase information in the computer-generated wavefront reconstruction, the two sources can be imaged one at a time in their proper locations and their phases can be determined. Holographic images of simulated acoustic radiation from a one-dimensional bending wave in a semi-infinite plate show radiation from the entire plate when vibrating above its coincidence frequency but only from the ends of the plate when vibrating below its coincidence frequency. [Work supported by NAVSEA.]
Ultrasonic imaging system incorporating computer-aided coherent processing
Authors: Powers J. P. , O'Bryon R. T. , Patton J. W.
Published: October 2005

An ultrasonic imaging system incorporating two-dimensional coherent data processing and computer image processing has been built and tested. The system consists of two parts: data acquisition, and computer processing and display. The data acquisition portion coherently detects both the amplitude and phase of an ultrasonic diffraction pattern emitted by or reflected from an object. This diffraction pattern is then quantized and entered into the computer memory. The computer portion of the system processes the data and displays the resulting image on a high-resolution computer-controlled graphics terminal. The presence of both amplitude and phase information as the input to the processing portion of the system offers advantages in that coherent processing techniques can be used to process the data as well as incoherent techniques. This processing capability allows one to perform such operations as matched filtering, removal of the receiving transducer directivity pattern, edge enhancement, etc. Application of the system to ultrasonic imaging, acoustic transducer calibration, and studies of scalar wave propagation and phenomena will be given.
Ultrasonic field mapping using conventional schlieren images
Authors: Hughes Ronald G. , Bucaro Joseph A. , Dardy Henry D.
Published: October 2005

Conventional schlieren is a technique commonly used for visualizing the pressure distribution of a sound field parallel to its direction of propagation. Unfortunately, the phase and amplitude variations of the sound in planes transverse to its direction of propagation are integrated out in this visualization technique. However, an optical method has been proposed by Berlinghieri and Cook [J. Acoust. Soc. Am. 58, 823 (1975)] to extract these sound field characteristics lost in schlieren visualization. In this paper, a computer-based analysis method is established employing a modification of the Berlinghieri and Cook method. The analysis procedure is applied directly to a conventional schlieren image of a 1-MHz transducer. The amplitude and phase (variations) of the sound field of the transducer for a series of transverse planes are extracted from the schlieren image in order to demonstrate a method of ultrasonic field mapping.
Holographic visualization of low-frequency acoustic fields
Authors: Clark Joseph A.
Published: October 2005

A novel optical holographic method has been developed which uses conventional films for the visualization and measurement of low-frequency sound fields. The method uses two nearly coincident optic beams, instead of the single beam used in schlieren optical methods, to produce a gratinglike image plane hologram. Two holograms are superposed and then optically processed to reconstruct an image of the visualization area. The reconstructed image contains a pattern of optical interference fringes which characterize the sound field. If the acoustic pressure is constant in one direction, the interference fringes can be interpreted in a particularly simple way: as contours of equal instantaneous acoustic pressure. Sensitivities of the order of 196 dB re 1 μPa are achieved. Results obtained with the new holographic visualization method are compared with those of classical schlieren visualization methods. The new holographic visualization method is found to have better sensitivity, dynamic range, and spatial resolution than schlieren methods at frequencies of most practical interest in underwater and evironmental acoustics problems (below 50 kHz). The new method could find important application to experimental investigations of the nearfield of objects radiating or scattering sound at audio and sonar frequencies.
Acoustic camera using inverse diffraction
Authors: Sondhi M. M. , Schroeder M. R.
Published: October 2005

A method is presented for reconstructing the shape of an object from a measurement of the diffraction pattern of the object when placed in the path of a monochromatic sound wave. It is shown that measurements of amplitude and phase in a plane at a distance from the object enable reconstruction of the wavefront everywhere. It is further shown that measurements at only a discrete set of points (a square lattice of suitable dimensions) suffice. The results of computer simulation, as well as of an experiment conducted in an anechoic chamber are presented.
Visualization of ultrasonic wavefronts by double-pulsed holographic interferometry
Authors: Sanford R. J. , Chaskelis H. H.
Published: October 2005

A holographic method for visualizing ultrasonic wavefronts in liquid media has been developed. This technique is based on the use of an off-axis double-pulsed optical holographic system. Upon reconstruction, the hologram exhibits an interference pattern related to the change in the refractive index field caused by the acoustic pressure wave. In addition to qualitative value for evaluating transducers these holograms can, in concept, be quantitatively analyzed with established procedures for studying three-dimensional refractive index fields. To date this technique has been used to study the near- and farfield patterns of transducers and various reflective targets. Additional experiments have been performed using various liquids and relative atmospheric pressures in an effort to increase the sensitivity of the method. A description of the system and the results of preliminary experiments in the 400-kHz to 5-MHz range will be presented.
Contributions of the Theory of Signal Detectability (TSD) and applications in psychoacoustics to the study of vision
Authors: Cohn T. E.
Published: October 2005

TSD was first applied in psychophysics in the context of visual detection by Tanner and Swets. In early experiments a key prediction of TSD (the linearity of d′ with signal energy) failed and efforts turned to applications in audition. In 1961, Tanner suggested that results obtained for audition might be applicable in vision, that the too steep visual psychometric function might be due to imperfect memory for parameters of the signal to be detected. This paper describes subsequent experiments to test Tanner's conjecture. First, a method of analyzing nerve messages involving receiver operating characteristic analysis is used to obtain psychometric functions in cat and frog retinal ganglion cells. Second human psychophysical studies are described: Physical uncertainty (spatial, temporal, or chromatic) depresses luminance increment detectability by an amount closely matched by predictions of TSD. A luminance pedestal linearizes the psychometric function, physical uncertainty makes it nonlinear again and the nonlinearity persists in the presence of added external noise. These results support Tanner's conjecture and clarify the mechanisms underlying vision. [Work supported by NSF: BNS 76-18829 and by NIH: EY 01481.]
Recognition-detection relations for continuous and binary systems
Authors: Green David M. , Birdsall T. G.
Published: October 2005

Some theoretical results are presented contrasting the detection and recognition performance of several different multiple-channel systems. In all cases, the channels are treated as independent. We first consider a system in which the output of each channel is assumed to be continuous, then we consider systems in which the outputs are assumed to be binary. Two different decision rules are examined in the binary case. One of these leads to predicted relations between detection and recognition that are virtually indistinguishable from that obtained with a continuous system, the other yields very different predicted relations. [Work supported by NIH.]
Choice-reaction times and confidence in auditory signal detection
Authors: Carterette E. C. , Smith James Mitchell
Published: October 2005

In a yes-no detection task subjects listened for an extended number of sessions to a sinusoid in wide-band noise having a priori signal probabilities [P(S)] of 0.2, 0.5, and 0.8. Subjects gave a confidence rating following each yes or no response, to which feedback of correctness was provided. Choice-reaction times (RT's) were also recorded. Confidence ratings were lower for P(S) = 0.5 than for P(S) = 0.2 or P(S) = 0.8; were higher for signal than for noise-alone trials; and higher for yes than for no responses. Mean RT's decreased in the order P(S) = 0.5, 0.2, and 0.8; were less for signal than for noise-alone trials; were less for yes than for no responses; and became shorter as confidence increased. As P(S) increased, mean RT's of yesses decreased while those of noes increased. Mean RT's on signal trials were shorter than on noise-alone trials for yesses but the reverse was true for noes. These data (based on 3.9178×104 trials) will be related to several models of choice RT's. [Work supported by USPHS, MH-7809-14.]
Signal detection under dynamic changes in interaural phase
Authors: Pollack Irwin
Published: October 2005

A brief tonal signal (500 Hz, 20 msec) was presented against a random noise background. The interaural phase of the background noise was switched periodically. Signal thresholds are determined not only by the ongoing interaural phase of the signal and of the noise, but also by the temporal proximity of the signal to a transition in interaural phase. [Research assisted by a grant of the National Science Foundation.]
Spectrographic effects of register shifts in speech production
Authors: Buhr R. D. , Keating P. A.
Published: October 2005

Infants and children use a wide range of fundamental frequency values in their vocal production, often using more than one vocal register within a single utterance. Adult speech is also characterized by the occurrence of more than one register in the same utterance. Spectrographic analysis of utterances containing shifts between registers frequently reveals the occurrence of what appear to be multiple harmonics during these shifts, in between the harmonics of the f0. This effect may be the result of instability at extreme ranges of the modal register. For example, a combined sample for five children for whom a large corpus is available indicates these shifts occur across a range of 160–250 Hz in shifts between modal and Fry registers, and across a range of 380–800 Hz in shifts between modal and high registers. Additional evidence for this effect in adults will also be presented, along with results of waveform analysis of this phenomenon.
Experimental assessment of antidepressant drug effects by spectral voice analysis
Authors: Helfrich H. , Scherer K. R.
Published: October 2005

Using 23 healthy male student volunteers, the effects of imipramine (single dose of 100 mg orally) and lofepramine (single dose of 140 mg orally) in comparison to placebo (double blind) on F0 and energy distribution in the spectrum of the voice were examined. All subjects received the three drugs in a completely counterbalanced sequence. Subjects read a standard text at three different time points after administration of the drug. An excerpt of approximately 24 sec for each reading (a total of approximately 112 h) was subjected to digital speech analysis to obtain F0 and long-term voice spectra. Analysis of variance yielded main effects for drug and interaction effects with both time period elapsed and personality. The drugs lead to a higher proportion of energy in the region below 500 Hz and affect F0 as a function of time elapsed. Further results and acoustic-phonetic implications are discussed.
Phonological variation over time: Boston /æ/
Authors: Laferriere Martha
Published: October 2005

In the Boston dialect, the phoneme /æ/ has been observed to vary allophonically. To determine if this variation is systematic over time [Labov, 1966], five speakers, ages 26–71, read a list of approximately 40 words containing /æ/ before nasals (man), voiced obstruents (bad), voiceless obstruents (fact), and spirants (pass). First and second formant values for each occurrence of /æ/ were obtained by computer analysis, and five F1 vs F2 vowel space plots were made. For the two oldest speakers, /æ/'s are distributed in a broad (700 Hz) range along the F2 axis, and a narrow (200 Hz) range along the F1 axis. /æ/'s before nasals have the highest F2 values. For the younger speakers, /æ/'s have the same broad F2 range and narrow F1 range; but for the phonetic subclass of /æ/'s before nasals, F1 values have been lowered by about 400 Hz to overlap with F1 values for /e/. These preliminary results suggest that in this dialect, in formal speech style, variations in tongue fronting (F2) are continuous, whereas variations in tongue raising (F1) are discrete.
Performance of Harpy speech recognition system for speech input with quantization noise
Authors: Yegnanarayana B. , Reddy D. Raj
Published: October 2005

One of the major problems of a speech processing system is the degradation it suffers due to distortions in the speech input. One such distortion is caused by the quantization noise of waveform encoding schemes which have several attractive features for speech transmission. The objective of this study is to evaluate the performance of the Harpy continuous speech recognition system when the speech input to the system is corrupted by the quantization noise of an ADPCM system. The Harpy system uses segmentation of continuous speech based on Itakura metric and LPC-based parameters for template generation. The effect of quantization noise on the segmentation and the estimation of LPC-based parameters is studied for different bit rates in the range 16–48 kbs of the ADPCM system and the overall word and sentence recognition accuracies are evaluated.
Specific contributions of the ARPA SUR project to speech sciences
Authors: Lea Wayne A. , Shoup June E.
Published: October 2005

This critical review of the recent ARPA Speech Understanding Research (SUR) project emphasizes the basic speech science results (only some of which were incorporated into the final systems). ARPA SUR provided significant progress in such parameter extraction methods as pitch detection and formant tracking. Advanced methods of speech segmentation and labeling ranged from allophonelike acoustic templates to lattices of alternative phonetic units, to quasiphonemic labeling, with emphasis on manner classes rather than place of articulation. The phonological tie between small- and large-unit recognition involved tradeoffs between extensive sets of alternative lexical baseforms and large sets of phonological rules. The syllable was promoted as a more basic unit than the phoneme, but difficulties arose with the use of syllable boundaries in baseforms and rules. Prosodic features were demonstrated as useful for determining sentence structure and confirming lexical hypotheses, but were not fully tested within any system. ATN grammars, semantic networks, and breadth-first parsers were effective for speech analysis, provided they were used to constrain system choices rather than simply be good models of language. [Work supported by Tri-Services and ARPA.]
Range dependence of acoustic fluctuations
Authors: Spindel Robert C. , Porter R. P.
Published: October 2005

The fluctuations of low-frequency, continuous-wave, underwater acoustic transmissions were measured in October–November 1976 during an experiment designed to reveal the effect of increasing range on acoustic variability. A linear acoustic range was established from Bermuda to a point approximately 600 km south. Moored acoustic sources at 100 and 220 Hz, moored narrow-band receivers, a moored vertical array near Bermuda and an acoustic navigation system to track mooring motion were used to obtain data at 10- to 600-km ranges. Oceanographic measurements (XBT, CTD) were conducted simultaneously. Range dependence of the data is presented and interpreted especially in terms of a new statistic, the intensity-weighted phase rate. This statistic suppresses discontinuities in the phase rate due to amplitude fades which normally dominate the phase rate spectrum, thus allowing the underlying physical process affecting phase rate to be observed. [Work supported by ONR.]
Ear dominance effects in loudness following auditory adaptation
Authors: Davis James M. , Weiler E.
Published: October 2005

In adaptation experiments, one factor which has not been reported is the possibility of differing effects depending upon whether the dominant ear or the nondominant ear received the stimulus. The present experiment utilized a monaural measurement technique which presented a 500-Hz pure tone to the experimental ear for 7 min of adaptation exposure at a 50 dB SPL. Ten subjects were adapted in the right ear and ten in the left ear. A 10 000-Hz tone was used as a comparison tone. There was a significant difference in right- versus left-ear adaptation (12.88 vs 4.20 dB) with a t of 3.8 and p less than 0.001. An experiment using binaurally presented stimuli with the adapting stimulus at 500 Hz and 60 dB SPL, while the comparison stimulus was also 10 000 Hz (as above) but presented to the opposite ear, yielded significant ear differences. The right and left ear yielded 5.0 and 6.0 dB, respectively, with t of 2.93 and p less than 0.05. Results of both studies would suggest that the ear used for adaptation does markedly affect the magnitude of loudness adaptation measured.
Limitations on the range of intensity independence of pitch mixture of dichotic chords
Authors: Yund E. W. , Efron R. , Divenyi P. L.
Published: October 2005

The relative salience of the pitch components of a two-tone dichotic chord is invariant with respect to the relative intensity of the two tones over a wide range of interaural intensity differences [R. Efron and E. W. Yund, J. Acoust. Soc. Am. 59, 889–898 (1976)]. In a recently developed model for this phenomenon, the range of intensity independence is limited by the bone-conducted energy from the more intense tone [E. W. Yund and R. Efron, J. Acoust. Soc. Am. 62, 607–617 (1977)]. The model thus predicts that a decrease in bone conduction must increase the range of intensity independence. This prediction has been confirmed by experiments using insertion earphones. The range of intensity independence with insertion earphones was 6–17 dB greater than the value obtained for the same subjects using TDH 39 earphones in MX41/AR cushions. [Work supported by the Veterans Administration.]
Effect of unilateral bandwidth increase on ear dominance of dichotic chords
Authors: Divenyi P. L. , Yund E. W. , Efron R.
Published: October 2005

When a 1650-Hz tone is presented to one ear while a 1750-Hz tone of equal intensity is presented to the other ear, most subjects report that there is a significant difference in the relative salience of the pitch of these two tones. This has been called “ear dominance” in the perception of dichotic chords [R. Efron and E. W. Yund, J. Acoust. Soc. Am. 59, 889–898 (1976)]. The present experiments were designed to neutralize or even reverse a given subject's ear dominance by increasing the bandwidth of the tone presented to his dominant ear while continuing to present a pure tone in the nondominant ear. This increase in bandwidth was accomplished by modulating, in brief, abrupt steps, the frequency of the dominant ear tone. The duration of each of the brief steady-state portions of the FM tone was random between 1.4 and 2.0 msec; its frequency was Gaussian-distributed around the selected center frequency. To make the salience of the two pitch components of a dichotic chord equal, a larger bandwidth increase was required in subjects having a stronger ear dominance. Lowering the sound pressure level of the stimuli in both ears increased the amount of bandwidth increase required. These results can be interpreted in terms of a model of binaural pitch mixing. [Work supported by the Veterans Administration.]
Extension of the maximum-likelihood method to the estimation of mode spectra
Authors: Baggeroer Arthur B.
Published: October 2005

The presence of boundaries and inhomogeneities often requires modes to characterize signal propagation in the ocean, especially for shallow water or low frequencies at long ranges. Modes have been detected temporally by the analysis of dispersion curves and spatially by the use of arrays. The array methods are particularly useful when the modes overlap temporally. Several array processing methods have been introduced including matched filtering, maximum-likelihood filtering, and covariance matching. The Maximum Likelihood Method is an array processing technique for spectral analysis which has been used successfully in several geophysical applications involving plane waves and homogeneous media. It has higher resolution of sources and better suppression of the mutual interference that occurs in using sparse arrays. It is also closely related to the determination of optimum array gains for adaptive arrays. The extension of the MLM for analyzing mode spectra and array gains for signals propagating in an inhomogeneous media is developed. Particular attention is given to detecting shallow water sources using the noise models presented in the companion paper. The extended method, however, is applicable to all situations where modal models of the signals are useful.
Comparison of calculated and measured sound power levels of large machinery and small hand-held tools
Authors: Diehl George M.
Published: October 2005

Comparison of calculated and measured sound power levels of industrial machinery is not easy to do. This is a comparatively simple procedure in accoustical laboratories where small equipment can be sound tested in anechoic or reverberant rooms. In industry the situation is quite different. Large, high horsepower compressors, pumps, gas turbines, and similar machinery must be tested where they are installed, under less than ideal conditions, and the main problem is to obtain the “true” sound power levels, against which the measured values can be compared. Environmental conditions often interfere with controlled sound tests on smaller equipment, such as portable air compressors; and strangely enough, it turns out to be rather difficult, even in the case of small hand-held tools, to determine true sound power levels from measurements of sound pressure levels. This report compares calculated and measured sound power levels of small, medium, and large machinery and discusses the reasons for conducting the tests.
Low-frequency sound radiation plates
Authors: Brooks J. E. , Waterhouse R. V.
Published: October 2005

Theoretical results are presented for the low-frequency sound pressures arising from prescribed rigid-body motions of thin rectangular and elliptical plates. The fluid adjacent to the plate is taken as incompressible. The farfield pressure is given in terms of the plate's incompressible added mass which limits the type of plate that can be represented since this quantity is known for only a few bodies of simple shape. But the comparison of the added masses for rectangular and elliptical plates suggests an approximation for the added mass of any compact but otherwise arbitrarily shaped plate in rigid, broadside motion. The approximation is incorporated into the expressions for the farfield sound pressures.
Theoretical study of sound radiation from an inlet duct
Authors: Cho Y. C. , Lansing D. L.
Published: October 2005

The sound radiation from an inlet duct is investigated by means of the eigenfunction solutions of the wave equation in the oblate spheroidal (ζ, η, φ) and cylindrical (z, r, φ) coordinate systems. The inlet duct under consideration is composed of a uniform circular duct which opens into a configuration determined as the boundary surface of η=constant, that is the half of the hyperboloid of one sheet. The more important features in this investigation are as follows: (1) The oblate spheroidal eigenfunctions with the finite boundary surfaces have been obtained, these have not been studied previously; unlike the uniform duct modes, the shape of the oblate spheroidal wave mode depends on the frequency; the shape of the fundamental mode is plane only at very low frequencies. (2) The conversion of the duct modes to the radiation modes (the oblate spheroidal wave modes) is accurately evaluated at the inlet of the flange because the surface of ζ=0 coincides with that of z=0.
Radiation from a thin lined duct
Authors: Talkington Carl
Published: October 2005

A lumped parameter model has been developed for a thin duct associated with the paper entrance and exit path on a typewriter. Consideration of the radiation impedance of a narrow slit will be included. Data have been collected and the results will be compared with those predicted by the model.
SEM assessment of acoustically damaged cochleas
Authors: Hunter-Duvar Ivan M.
Published: October 2005

Cochleas from chinchillas exposed to a 1-kHz pure tone stimulus were examined with the scanning electron microscope (SEM). Exposures to the 1-kHz tone ranged in intensity and duration from 120 dB for 15 min to 90 dB for 3 h. Cochleas were examined at periods from immediately after exposure to nine months postexposure. Both permanent and temporary damage is demonstrated. Permanent lesions were confined to lower middle and upper basal turns. In some animals, sacrificed immediately after exposure, stereocilia of first and second row outer hair cells were found to be agglutinated and undergoing autolysis at focal points of lesion areas. In some cases stereocilia were already consumed leaving only stumps. No perforations into the fluid spaces of the organ of Corti could be found in animals examined either immediately after exposure or at later periods. The only change seen in some animals exposed to 120-dB stimulation and examined immediately after exposure was a collapse o[ patches of stereocilia of inner hair cells and first row of outer hair cells. Some animals exposed to 90-dB stimulation showed only a collapse of patches of stereocilia of the first outer row of hair cells immediately after exposure. The collapsed stereocilia of inner hair cells appeared to return to an upright position within 24 h, while those of the first row outer hair cells required periods in excess of two days.
Auditory nerve fiber activity, behavioral thresholds and cochlear histology following noise induced asymptotic threshold shift
Authors: Salvi R. J. , Henderson D. , Hamernik R. P.
Published: October 2005

Chinchillas were exposed three to five days to a 95-dB SPL octave band of noise centered at 0.5 kHz. During the exposure, the behaviorally measured thresholds reached an asymptotic level that was between 40–60 dB SPL from 0.5–16.0 kHz. During a 27–15 h period following the exposure, single auditory nerve fibers and the whole nerve action potential (AP) were studied with clicks and tone bursts. Single neuron thresholds were generally 40–60 dB above normal values and were in general agreement with the behavioral audiogram. Fiber tuning curves were broader than normal with reduced slopes. Some units had a region of increased sensitivity below CF. Click thresholds for the units were generally 40–60 dB above normal levels. The visual detection level for AP was also 40–60 dB above its normal value and consistent with the single neuron data. The cochleagrams obtained 15 h after the exposure generally showed a peak outer hair cell loss (<50%) near the apex with smaller losses (<20%) at more basal locations. [Research supported by NSE.]
Evaluation of FFP in kangaroo rat: Apical or basal?
Authors: Bledsoe S. C. , Rupert A. L. , Gidda J. S.
Published: October 2005

Davis and Hirsh [Audiology 15, 181–195 (1976)] argue that the intensity required (40 dB SL) to evoke the 500-Hz frequency-following potential (FFP) in human subjects is sufficient to stimulate the basal cochlea, driving high-frequency neurons at the tails of their tuning curves. We investigated this question by comparing the threshold of the vertex FFP with the phase-locking thresholds of low and high best frequency neurons to 500-Hz tone bursts in auditory nerve and cochlear nuclei of kangaroo rats. The effects of broadband and high-pass noise on the FFP and phase-locked neuronal responses were also compared. Only neurons with best frequencies around 500 Hz locked at FFP threshold levels. Neurons with best frequencies up to 1.5 kHz locked when the sound was increased 20 dB, and a 40-dB increase produced locking in 2.0–2.5-kHz neurons. The masking data are in agreement with the conclusions that within 10 dB of FFP threshold only a restricted region of the cochlea is activated, and a greater region is involved as intensity is increased. However, with appropriate high-pass masking, the 500-Hz region of the cochlea can be selectively evaluated. [Supported by NIH and AFOSR.]
Implications of low-cost signal-processing devices for speech science
Authors: White George M.
Published: October 2005

Speech science is a product of high technology. Our understanding of what speech is acoustically has grown up with advances in signal-processing high technolgy. In particular, bandpass filtering, Fourier analysis, and linear predictive coding and all its dericatives are the primary research tools in today's speech laboratories. However, the widespread use of these techniques is inhibited by the cost of the equipment required for real-time analysis. Recent technological advances in analog LSI signal processing devices promise to reduce the equipment costs by an order of magnitude. We shall discuss the development analog LSI chips that may soon be able to perform Fourier transforms, or bandpass filtering, or autocorrelation, or, perhaps, Itakura-type LPC analysis. Each operation may be soon performed on a single chip. It is felt that speech science will be significantly advanced by the resulting low-cost speech analysis tools.
Automatic phonemic segmentation of speech by rule
Authors: Dixon N. Rex
Published: October 2005

This paper will concern an approach toward automatic phonemic segmentation of speech which is based on reliable properties of the speech production and in-hand techniques for processing the speech signal. A system has been developed to perform this segmentation and phonemic labeling of the segments. It has stages dedicated to signal analysis, spectral classification, phonemic segmentation, phonemic (steady state) classification, phoneme boundary placement, dyadic (transitional) classification, and final phoneme string consolidation. The procedures described have been implemented and evaluated on large bodies of continuous-speech data. Performance results obtained from their application are considered “state of the art” in this area of work.
Package of computer programs for articulatory modeling of the speech process
Authors: Coker C. H.
Published: October 2005

At the poster session we will describe a package of FORTRAN. programs which accomplish articulatory synthesis of speech from phonetic input. The package of subroutines will include the following: (1) a coarticulation program which, knowing featurelike descriptions of speech segments, performs the staggered sequencing of “motor commands” necessary for proper overlapping of articulatory movements; (2) a dynamics program that converts stepwise motor commands into smooth articulatory movements; (3) an instantaneous model of the vocal tract; (4) a parametric-speed model of the pneumatic system, of vocal-cord activity and fricative noise sources; and (5) a routine for converting the above articulatory details into formants and other controls for parametric synthesis. (Of course, the package of programs can also be used with a direct time-domain acoustic tube model, as well.) The formal presentation will contain a brief description of the package of programs, and comments on several aspects of speech synthesis by articulatory modeling.
Wind-induced low-frequency ambient sea noise: Mechanism and modeling
Authors: Yen N.-C. , Perrone A. J.
Published: October 2005

A study has been made on the wind-induced ambient sea noise in the frequency range of 1–10 Hz. The noise generating mechanisms due to wind turbulence, surface wave, and wave turbulence are analyzed as the possible causes for the wind-induced ambient sea noise observed in the deep ocean. Based on the assumption of a stationary ocean surface disturbance, the noise power spectra due to the aforementioned noise-generating mechanisms are derived; furthermore, with the incorporation of propagation condition and bottom loss, a relationship between wind speed and ambient noise level in a certain water depth is formulated. The ambient noise level predicted by the theory agrees in order of magnitude with the reported experimental data obtained in Bermuda and Grand Bank areas. Analysis of noise spectrum level and wind speed variation has also been made in the determination of the dominant parameters affecting the wind induced noise. As a result of this study, a semiempirical wind noise model is suggested for the purpose of data comparison and as a guideline in wind noise level prediction. [Work supported by ONR.]
Statistics of quasistationary ambient noise due to merchant shipping in restricted spatial sectors
Authors: Heine J. C.
Published: October 2005

The received noise from a restricted spatial sector containing a small but random number of transiting merchant ships is locally (in time) stationary and Gaussian but with fluctuating time-averaged power. The statistics of these fluctuations are estimated by considering the ensemble statistics of two processes. The first is the short time averaged (minutes) power received from a fixed set of N ships in the sector, where the contribution from each ship fluctuates due to multipath interference. The second process is the long time-averaged (hours) power, which is the mean value of the first process. Fluctuations here are due to variations in the number and radiated power of the ships in the observed sector. The fluctuation statistics of the former process are shown to be well represented by a gamma density function for which the mean and variance are determined by those for the received power of the individual components. The statistics of the latter are shown also to be well represented by a gamma function for which the mean and the variance are determined by the mean and variance of the ensemble of source powers and by the average number of surface ships in the observed sector.
Information processing for complex tones in older adults
Authors: Corso John F.
Published: October 2005

The ability of three age groups to identify complex tones was measured by the method of absolute judgments. Group mean ages were nominally 20, 40, and 60 yr., with ten matched subjects per group. Complex tones from ten different musical instruments were tape recorded and spliced to yield six experimental tone types per instrument with different waveforms and durations. After practice, the test tones were presented monaurally in random order in the subject's better ear at a loudness level of approximately 60 phons. Each subject made a total of 120 judgments with the correct instrumental identification being provided after each trial. Analysis of variance showed a significant difference in the mean number of correct responses for the four factors of age, instrument, tone type, and practice. For the ten instruments, the average amount of information transmitted per group in order of increasing age was 1.68, 1.49, and 1.42 bits. It is concluded that (1) the processing of complex-tone information is adversely affected by normal aging from 20 to 60 yr and (2) maximal identification is obtained for all ages when the entire tone (onset, steady state, and offset) is presented. [Work supported by SUNY Research Foundation Grant No. 023-7185A.]
Contour, intervals, and pitch in the perception of auditory patterns
Authors: Idson Wendy L.
Published: October 2005

A series of experiments explored the information used in perception of an auditory pattern. Subjects identified which of a set of auditory patterns was presented on each trial, when each of the pattern tones was followed by an irrelevant interference tone. The structure of the patterns, the frequency relationship between the pattern and interference tones, and the time separating each pattern tone from the following interference tone were varied. Both the nature of the disruption exerted by the interference tones, and the trend of confusions among the individual patterns, provide an index of the information which is perceived. The results suggest that only the most abstract information needed to uniquely specify a pattern will be utilized. When contour was sufficient to discriminate among patterns, neither interval nor frequency information was employed. When contour alone was insufficient, successive intervals were utilized as well. The absolute frequencies of the component tones were used only when the patterns were equated for both contour and intervals. These findings suggest that perception proceeds in terms of a higher-order structure provided by the frequency relationships within a pattern. [Work supported by NIMH.]