bluuebunny/crossref_metadata_2025 huggingface dataset:
Correction of energy shadowing for scan images of ultrasonic data
Authors: Kishoni Doron
Published: October 2005

In most of the scan image applications of ultrasonic measurements (C-scan etc.), parameters such as the maximum amplitude of the reflected waves are drawn as a function of an x-y location. Function of depth may be included too. Many times the only correction that is made is the attenuation compensation, i.e., the amplitude of the reflected wave is amplified according to an exponential relation as function of depth. However, an additional parameter should be considered. This is especially important in complex materials, such as composites, where several defects and delaminations along the ultrasonic wave path may be encountered. The parameter is the reduction in energy of the transmitted wave when a reflection from a defect occurs. Further reflections will be smaller, even if the size of the defects are the same. In this paper we address the problem and present a simple and fast algorithm that may he utilized in real-time measurements to compensate for this effect. [Work supported by NRC.]
Minimum spectral contrast for vowel identification by normal and hearing-impaired listeners
Authors: Leek M. R. , Dorman M. F. , Summerfield Q.
Published: October 2005

It has been suggested that the internal spectral representation of vowels in hearing-impaired listeners contains considerably less information than is available in the acoustic speech signal. It is puzzling, then, that unless the loss is quite severe, such listeners typically have little trouble identifying vowels. Apparently, faithful preservation of the vowel spectrum is not necessary for accurate identification. In this study, we sought to determine the minimum peak-to-valley differences in formant amplitude which could adequately define particular vowels for normal-hearing and hearing-impaired listeners. Vowel-like complex tones were generated by adding 30 harmonics of a 100-Hz tone in cosine phase. Amplitudes of all harmonics were equal, except for six components selected to define the formants of four vowels, which listeners were asked to identify. The selected components varied relative to the remaining component amplitudes by ±1 to +8 dB. Results indicated that the 2–3-dB peak-to-valley contrast required for identification by normal hearing listeners had to be doubled and tripled, respectively, for normal listeners in noise and for hearing-impaired listeners. While the spectral contrast information in vowels is more than sufficient for identification by the hearing-impaired, these listeners are operating closer to the limits of correct identification than their normal-hearing counterparts. This procedure offers a paradigm for a finer quantification of differences in vowel perception by these two populations than the usual accuracy measures. [Research supported by NIH.]
Acoustic and perceptual correlates of anticipatory lip rounding
Authors: Bell-Berti F. , Gelfer C. E. , Harris K. S.
Published: October 2005

Studies of coarticulation have provided evidence of lip rounding activity during consonants immediately preceding a rounded vowel, although reports differ as to the extent of this anticipatory activity. One would naturally expect to find this rounding reflected in the acoustic spectra of the consonants, and to have perceptual salience. To establish the extent of such acoustic and perceptual effects, we conducted studies using two speakers, one who uses lip gestures for alveolar consonants in both rounded and unrounded vowel environments, and another who uses them only in rounded vowel environments. In the first study, spectral analyses of 520 V1CnV2 utterances from two speakers yielded little evidence of V2 rounding gestures effecting an unfounded V1, while there is evidence that such lip rounding gestures influence fricative noise immediately preceding a rounded or unrounded V2. The results indicate that the acoustic effect is not perceptually salient for a speaker for whom lip rounding is inherent in alveolar consonant production. [This work was supported by grant NS-13617 and RR-05596.]
Perceptual learning of synthetic words and sentences
Authors: Greenspan Steven L. , Nusbaum Howard C. , Pisoni David B.
Published: October 2005

Previous research has shown that the intelligibility of synthetic speech can be improved with training [E. C. Schwab, H. C. Nusbaum, and D. B. Pisoni, Human Factors (in press)]. In the present study, we investigated the relationship between the type of training subjects receive and the pattern of perceptual learning that occurs. Three groups of subjects were given word recognition tests consisting of synthetic words and sentences generated by the Cottax Type-'n-Talk, before and after a training period. One group of subjects was trained on isolated synthetic words. A second group was trained on fluent synthetic sentences. A control group received no training at all. Recognition of isolated synthetic words improved equally for both groups trained on synthetic speech compared to the control group. However, overall word recognition in sentences only improved for subjects that were trained on synthetic sentences. These results demonstrate that performance with synthetic sentences will predict performance with isolated words, but the converse may not always hold. [Work supported by AFOSR and NIH.]
Perceptual attention in monitoring natural and synthetic speech
Authors: Nusbaum Howard C. , Greenspan Steven L. , Pisoni David B.
Published: October 2005

The role of voice distinctiveness and phonetic discriminability in perception of natural and synthetic speech was investigated. Subjects were instructed to monitor sequences of CV syllables for a specified target syllable in several conditions: (1) targets and distractors produced by the same human talker (N/N); (2) targets produced by a synthetic talker and distractors produced by a human talker (S/N); and (3) targets produced by a synthetic talker and distractors produced by both the same synthetic talker and a natural talker (S/N + S). Results indicate that highly intelligible synthetic targets are detected faster mixed with natural distractors, than are natural targets mixed with natural distractors. However, when subjects are required to discriminate between synthetic targets and synthetic distractors, performance is much worse than for natural targets and natural distractors. The distinctive mechanical sound of synthetic speech only appears to aid perception when there is just a single synthetic message among natural messages. When listeners must discriminate among synthetic messages, performance is significantly worse than when they must discriminate among natural messages. [Work supported by AFOSR and NIH.]
Use of Commodore computers in the undergraduate mechanical engineering instrumentation laboratory course
Authors: Oehmke Roger L.T. , Wepfer William J.
Published: October 2005

Integration of microcomputers into a required junior-level mechanical engineering course resulted during a recent modernization of the undergraduate program. The course presents the microcomputer as a powerful laboratory instrument (tool) and follows the approach taken by Saltsburg, Heist, and Olsen [MICRO 53, 53–55 (October 1982); 55, 59–63 (December 1982); 56, 38–43 (January 1983); 57, 89–91 (February 1983)] at the University of Rochester. The emphasis is on the applications of the computer as a flexible data acquisition and control device. After completion of a 3-h introductory laboratory the students undertake a 6-h lab exercise in which the computer is used to monitor and control the RPM of a dc motor. Remaining experiments focus on traditional mechanical measurements, many involving use of the computer. An example is the use of a digital oscilloscope peripheral to collect vibration data (acceleration versus time) from an accelerometer mounted on one of two bearing blocks that support a motor-driven rotating shaft.
Microcomputer based data acquisition and analysis systems for graduate student research in Acoustics
Authors: McDaniel Oliver H. , Lamancusa John S. , Ward William C. , Todd Kevin
Published: October 2005

The development and commercial availability of dedicated multichannel fast-Fourier transform (FFT) analyzers have resulted in a significant advancement in productivity in research in acoustics. The capability of at least two-channel FFT instrumentation is a must for any acoustics laboratory; however, the acquisition of a single instrument represents a major item in a research budget. Two personal-computer-based multichannel FFT systems will be described and demonstrated. These systems are currently being used by graduate students in theses research in internal combustion engine noise and in piping system noise. These relatively low-cost systems were assembled with commercially available data acquisition components and software. Microphone signal conditioning preamplifiers and anti-aliasing filters were designed by a senior electrical engineering student. Preamplifier gains and filter cutoff frequencies are controlled by the microcomputer.
Waveform synthesis on a personal computer
Authors: Knott George
Published: October 2005

An inexpensive music synthesis program (Concertware, by Great Wave Software) is shown to be an alternative to traditional analog methods in the teaching of waveform synthesis concepts in the laboratory. In addition, the pedagogical ramifications of using a streamlined software package such as Concertware in what can best be called the “computer laboratory” versus the traditional equipment intensive lab environment are discussed.
Fiber-matrix mode acoustic wave propagation in carbon-epoxy composites
Authors: Yost William T. , Cantrell John H.
Published: October 2005

We report the generation and detection of coupled fiber-matrix mode acoustic wave propagation along the fiber direction in eight-ply unidirectional carbon-epoxy composites using a surface generation and detection mechanism. Compressional, 0.8-to 1.2-MHz acoustic waves are transmitted and received through transducer angle blocks placed on the sample surface. The angles are adjusted such that the component of the wave propagation vectors in the angle blocks along the sample surfaces (hence, along the fiber direction) equals the component of the wave propagation vector in the composite along the fiber direction (Snell's law). The measured sound velocity of (9.29 ± 0.04) × 105 cm/s at 1 MHz using this technique does not correspond to the values obtained for wave propagation in either the fiber or the matrix alone. A model based on the law of mixtures together with the assumption of hexagonal symmetry for the unidirectional composite shows that the wave propagates as a coupled fiber-matrix mode along the C-axis direction. The model predicts a value of the sound velocity in such a mode that is consistent (to within experimental error) with that obtained from the experiments.
Analogies between nonflat ground and nonuniform meteorological profiles in outdoor sound propagation
Authors: Embleton Tony F. W.
Published: October 2005

The analogy can be made between downwardly curving ray paths (as in a temperature inversion) over flat ground and an upwardly curving ground (a rising hillside) in a neutral atmosphere. Conversely, temperature lapses and flat ground are analogous to a neutral atmosphere and falling ground. The change in sound levels relative to those at the same range either in free space or over a flat, hard ground in a neutral atmosphere can be predicted by considering the appropriate ray paths or wave components involved. In all cases the change in sound level is a function of frequency. In general, sound levels increase during a temperature inversion or up a rising hillsidem, and decrease during a temperature lapse or down a falling hillside. Some field measurements exist to check the general structure of these predictions, but not many at ranges of several kilometers, and there is a need for verification at a variety of topographical sites under various meteorological conditions.
Vibration isolation of a bus driving lane in a multipurpose structure
Authors: Hougland Dana S.
Published: October 2005

A 19 500 square foot, vibration-isolated, bus driving lane was designed and installed above grade in a multifunction structure. The isolated driving lane was designed to prevent structural fatigue and vibrational annoyance in a high-rise office tower erected on the same substructure. A criterion for bus-induced acceleration in the structure adjoining the bus driving lane was established at a maximum of −55 dB (re: 1.0grms) in the 10- to 20-Hz frequency range. The isolation system was designed in separate slab sections using removable springs with neoprene dampening elements. Edge and end conditions were treated at each slab section to allow smooth transfer between slab sections. Porting was provided beneath the slab to minimize the effects of air viscosity and the additional dynamic stiffness of entrapped air. The resonant frequencies and mode shapes of each slab section were studied using finite element analysis techniques. Preliminary test results on the vibration-isolated bus driving lane and adjoining structure are presented.
A backward-type band-split adaptive predictive coding system with dynamic bit allocation for wideband speech and sound signals
Authors: Hayashi Shinji , Honda Masaaki , Kitawaki Nobuhiko
Published: October 2005

This paper proposes a new backward-type speech coding system (ADPCM-AB) for wideband (7 kHz) speech and sound signals. In this system, a split-band adaptive predictive coding scheme with gradient PARCOR lattice filter and a dynamic bit allocation scheme are employed, where quantization bits are dynamically allocated over the subbands (frequency), the subintervals (time), and the channels (stereo) in accordance with the distribution of the residual energies. They serve to remove the redundancies due to the periodic concentration of the prediction residual energy and the nonuniform nature of the speech spectrum. The ADPCM-AB needs neither longer delay time than 4 ms nor transmission of side information parameters because the parameters for predictive coding and dynamic bit allocation are calculated with the locally decoded signals. It is clarified that the ADPCM-AB system has the best speech quality among the conventional backward-type coding systems. It is also shown that this system provides speech quality subjectively equivalent to 11-bit linear PCM (176 kb/s) at 64 kb/s.
Optimization of monitoring signal reflection patterns in recording studio control rooms
Authors: Wrightson J. M. , Brubaker Brad S.
Published: October 2005

Recent control room design practice has often included rear/side wall reflective surfaces to provide a discrete, delayed reflection of the direct monitor speaker signal. This delayed signal is typically directed to the ear contralateral from the direct sound. To assess the perceptual effects of this practice, listeners were presented with pulses, speech, and music to both ears with a delayed repetition to a single ear. The level and delay duration of the monaural signal were varied. Under some conditions, 50 additional channels of delay at low intensity were presented diotically to simulate room reflections. Listeners were asked to make judgments of detectability for the monaural signal and to describe any lateralization changes in the undelayed signal when differences were heard. These conditions were repeated in a control room environment. The results of these judgments indicate that addition of the monaural delayed signal is easily detected for most conditions and substantially alters the spatial perception of the undelayed signal.
Intelligibility of speech in a noisy environment
Authors: Sanders Bruce M. , Rossing Thomas D.
Published: October 2005

Articulation indices for various modes of speaking masked by varying levels of white and pink noise have been measured. Pink noise is found to be a more effective masker than white noise at the same level. Shouted speech is more easily masked than normal speech presented at the same level. Very little difference is found between an actor's “stage voice” and normal speech. Enhancing speech in the 2- to 4-kHz range (region of the “singers formant”) also makes little difference in intelligibility in a noisy environment.
Integral processing in the perception of phonemes within and across syllables
Authors: Tomiak Gail R. , Sawusch James R.
Published: October 2005

Analogs of speech syllables were used in a speeded classification task with subjects instructed to treat the stimuli as speech or as nonspeech. Previous research using this paradigm with noise-tone analogs of monosyllables and disyllables has shown that single syllable stimuli show integral processing of adjacent phones (speech) but separable processing of adjacent pitch and amplitude information (nonspeech). Furthermore, this integrality effect is more pronounced within syllables than across syllable boundaries. We suggested that this pattern of results reflects a speech mode of processing in which phonetic information and knowledge of the consequences of coarticulation is largely represented within syllabic units. To assess the generality of these previous findings, new analogs of monosyllabic and disyllabic speech stimuli were constructed and tested in the speeded classification task. Results will be discussed in relation to the use of coarticulatory information in speech processing. [Work supported by NINCDS.]
Speech pattern audiometric assessment of hearing-impaired children
Authors: Hazan V. , Fourcin A. J.
Published: October 2005

A microprocessor-controlled speech pattern audiometer has been applied to study the long-term speech perceptual development of a group of hearing-impaired children. This technique makes use of high-quality synthetic speech to form minimal pairs of increasing acoustic complexity. The efficiency of the labeling tests is improved through the use of an interactive procedure. Individual children's development in the ability to process a range of increasingly complex contrasts has been assessed over a 3-year period. Development of perceptual ability is found to be similar to that of normally hearing children but delayed. The order of development is explainable in terms of the acoustic complexity of the stimuli. The primary use made of specific cues, such as the first formant in the labeling of the vowel contrast, and spectral rather than temporal cues in the labeling of the voicing contrast, is highlighted. The importance of intersubject differences in the development of perceptual abilities is also stressed.
Power flow techniques in the study of machinery installations for the purpose of vibration control
Authors: White R. G.
Published: October 2005

A typical machine installation consists of a source of vibration mounted on resilient isolators attached to a flexible substructure. There are usually also other vibration transmission paths, such a pipework, shafts, etc., which act as short-circuit elements across the isolators. It can thus be seen that there are a variety of mechanisms by which vibration is transmitted from the source to the point of interest on the substructure. The power flow approach to this type of problem is a basic concept which enables the relative importance of the various transmission paths to be critically assessed. Work has been carried out on the estimation of vibrational power flows between coupled systems, using approximate mobility methods, and this has led to the establishment of design rules for machinery seatings, together with simple formulas for estimating point mobilities of structures. Experimental techniques have also been developed. One development concerns measurement of vibrational power flow through isolators so that, for one machine mounted on a set of isolators or even an array of machines, the power inputs to the substructure at each isolator connecting point can be determined. The other, most recent development is the structural intensity meter which enables one- and two-dimensional power flows to be measured in beams and plates. Intensity maps can be plotted for plate-type structures, for example. The paper outlines the power flow approach to the machinery installation problem, with the objective of vibration control, and reviews the developments noted above.
Gaussian beam synthetic seismograms
Authors: Nowack Robert
Published: October 2005

This presentation is a tutorial on the Gaussian beam method used for the asymptotic synthesis of seismic and acoustic wave fields in inhomogeneous media. The method is based on the superposition of beam solutions, each of which is an approximate solution of the wave equation along particular rays. Smoothness conditions on the medium are required for the approximate propagation of beam solutions. Within a smoothly varying medium, various choices of the beam parameters can be used. Specifying broad planar beams at the source would result in a plane-wave decomposition of the visible spectrum. The standard ray method would result by specifying narrow planar beams at the receiver. Another choice is to specify the minimum integral beam width along each ray. There are several advantages of the Gaussian beam method, including finite amplitudes at caustics, smoothing of endpoint errors, and the reduction of amplitude variability resulting from model parameterization. Several numerical examples will be given in 2-D and 3-D, and comparisons will be shown with other numerical methods.
Maslov's method in seismology
Authors: Thomson Colin
Published: October 2005

Classical ray theory is an asymptotic form of a more general wave solution for smoothly inhomogeneous media, one which is uniformly asymptotic (i.e., valid at caustics) and has the WKBJ seismogram as a special case. This method involves “Fourier integral operators,” which exploit well-known properties of wave fronts and rays. Physically interpreted, it accounts for rays that miss the receiver or, equivalently, it is a local sum of plane waves, looking like a finite Fourier transform. Phases and amplitudes for these “plane” waves are derived from adjacent ray travel times and amplitudes by Legendre transformation and canonical transformation, respectively. Time domain seismograms are obtained using the efficient algorithm of Chapman: Most work is in ray tracing, not waveform construction. Smooth boundaries are accounted for using plane-wave coefficients. This covers, e.g., the intersection of a caustic and reflector or the Fresnel transition into the shadow of a grazing ray. Here, R/T coefficient discontinuities at critical and grazing angles cause low-frequency errors, as do finite integration limits, but these are recoverable.
A history of ray-theory development at Naval Ocean Systems Center
Authors: Pedersen M. A. , Gordon D. F.
Published: October 2005

This paper presents examples of the evolution of ray theory of underwater acoustics at the Naval Ocean Systems Center and its predecessor laboratories since 1952. The motivations for this evolution were to bring ray-theory results into better agreement with experiments, to distinguish true acoustic properties of the ocean from artifacts resulting from the sound-speed model or from shortcomings in ray theory, and to develop simple controls for use in evaluating complicated propagation models. Examples of such controls are the five-parameter Epstein profile, profiles for which various ray pencils focus at a point, and closed-form solutions for profiles with range as well as depth dependence. The most significant topics presented are the development of continuous-gradient sound-speed profiles, the effect of Earth's curvature and of improvements in experimental sound speeds, the application of uniform asymptotics in the boundary layer about caustics, and the development of a method using complex parameters for the evaluation of shadow-zone fields. Various examples of ray-theory propagation losses are compared with experiment or with the results of normal-mode theory.
Foundations of rigorous ray tracing
Authors: Floyd Edward R.
Published: October 2005

Rigorous ray tracing may be derived by inserting an ansatz into the Helmholtz equation to develop an alternate Hamilton-Jacobi equation for rigorous ray tracing expressed by (∂W∂z2) + Cm−2 − C−2 = −12ω2 ∂3W/∂z3∂W/∂z + 34ω2 (∂2W/∂z2∂W/∂z)2, where W is Hamilton's characteristic function (the generator of motion), z is depth, C is the sound velocity profile, Cm is the constant of motion (vertex velocity), and ω is the radial frequency. The left side of the above equation manifests the Hamilton-Jacobi equation for classical ray tracing. The terms on the right side of the above equation, which contain the common factor ω−2, compensate for finite wavelength. This alternate Hamilton-Jacobi equation does not have any associated auxiliary equation and may be related to the original Hamilton-Jacobi equation for rigorous ray tracing [E. R. Floyd, J. Acoust. Soc. Am. 75, 803–808 (1984)]. Equations of motion for both ray paths and wave normals follow from the generator of motion W.
A coherent shallow water eigenray model
Authors: Novick Arnold W.
Published: October 2005

An efficient range-independent coherent shallow water propagation loss model based on eigenrays is described. The model uses logic which guarantees finding all ray paths. Travel time data are also accurately computed. Predictions at 64, 256, and 1024 Hz for a complex shallow water environment (140-m water depth) are compared with measured data, and predictions using a detailed geoacoustic model (data and geoacoustic data reported by D. D. Ellis and D. M. F. Chapman, Defence Research Establishment Atlantic). Propagation loss predictions out to 85 km are in good agreement with measurements at 64 and 256 Hz using this relatively simple range-independent ray tracing model with a multilayer bottom. Surprisingly, the predictions made at 1024 Hz, where ray theory may be expected to be more accurate, are not quite as good. Comparison of model predictions with other measured and theoretical data is also presented. The results generally support the validity of coherent ray-based propagation modeling in shallow water areas over a wide frequency range.
Adaptive sound control—A tutorial review
Authors: Burgess John C.
Published: October 2005

Adaptive methods are being implemented increasingly to control the response of systems to meet performance objectives in the presence of unpredictable time-varying constraints. Typical open- or closed-loop control systems often fail under such conditions because their controllers (filters) have fixed coefficients. Adaptive controllers have time-variable coefficients determined on-line to minimize the difference at a system's output between desired and actual outputs. Adaptive methods are particularly appropriate for sound control. It is well known that active fixed-coefficient methods for sound cancellation fail with time as they become detuned by time-varying constraints such as changing medium temperature and motion. The purpose of this tutorial review is to provide some of the background of adaptive control with specific application to electro-acoustic sound cancellation.
A generalized adaptive control approach to electroacoustic noise cancellation
Authors: Swanson David C. , Tichy Jiri
Published: October 2005

Electroacoustic noise cancellation is instigated using a general adaptive control system similar to those used in controlling industrial process. An optional control signal in the form of a sampled electrical input to a loudspeaker is calculated via least-squares state estimation so as to minimize the variance of the measured control system output representing the sampled electrical output of a microphone. The control system is modeled as an auxiliary input autoregressive moving-average (ARMAX) process where the characteristics of the noise as well as the transducers are included in the system identification process. The mathematical properties of ARMAX control systems have been well studied [G. Goodwin and K. Sin, Adaptive Filtering, Prediction, and Control (Prentice-Hall, Englewood Cliffs, NJ, 1984)] and the closed-loop system will be stable in practice so long as the acoustic propagation time delays are known and a lower bound on the model order can be assumed. Numerical stimulations indicate a significant improvement in noise reduction at the microphone when the more complicated ARMAX model is used over ordinary least-squares. The algorithm can be expanded for multiple inputs and outputs permitting sound to be canceled at several microphone locations by one or more loudspeaker.
Propagation of normal acoustic modes in an atmospheric boundary layer
Authors: Zorumski William E. , Willshire William L.
Published: October 2005

The Obukhov quasipotential function for the acoustic field in a boundary layer of exponential profile is used to obtain a modal description of low-frequency sound propagation. As the wind speed approaches zero, the governing equations approach the Helmholtz equation with an impedance boundary condition. The solutions for the acoustic field with a boundary layer can be given as a continuous plane-wave spectrum with variable amplitudes given by generalized hypergeometric functions. An analysis with the hypergeometric functions gives one or more acoustic modes, depending on frequency. The acoustic modes propagate as cylindrical waves, with amplitude varying inversely with the square root of distance. An estimate of the wavenumber of the fundamental mode shows that its attenuation is proportional to the product of wind speed and boundary layer displacement thickness. The propagation theory is compared to data from a wind turbine at Medicine Bow, Wyoming. A microphone array was used to measure low-frequency sound at ground level at distances from 200 to 20 00(3 m from the turbine. Atmospheric temperature and wind speed profiles were measured, as was ground impedance, so that the theory may be compared without ambiguity to the data.
Turbulence effects on sound propagation from an elevated source
Authors: Raspet Richard , Bobak Michael T. , Johnson Mark A.
Published: October 2005

In a study of sound propagation from an elevated source, the turbulence effects appeared to be much smaller than was expected from Daigle's paper on turbulence effects on sound propagation with source and receiver close to the ground [J. Acoust. Soc. Am. 65, 45–48 (1979)]. In this paper, we describe the application of an atmospheric model for turbulence developed by Johnson and Rasper [J. Acoust. Soc. Am. Suppl. 1 77, S92 (1984)] to the refractive turbulence effects theory of Clifford and Lataitas [J. Acoust. Soc. Am. 73, 1545–1550 (1983)]. This theory predicts the reduction of turbulence effects observed in the noise study for elevated sources. The implications of this study for aircraft noise measurements will be briefly discussed.
Reducing the aural detectability of a 30-kW motor-generator set
Authors: Garinther G. R.
Published: October 2005

The noise produced by motor-generator sets in Army field situations causes detectability and speech intelligibility problems. A program for computing the aural detectability distance of Army material was developed and used to guide a noise reduction effort for this generator. A field expedient solution was pursued which was developed through the testing of a number of increasingly complicated barriers and enclosures. This report describes the procedure used and shows the 1/3-octave band level, the insertion loss, and the detectability contour obtained for each configuration. The insertion loss values are compared to theory, and cooling problems associated with noise reduction of generator sets are discussed. The final enclosure configuration attenuated energy in those low frequencies which cause detection, by up to 20 dB, and reduced computed detectability from 1000 to 200 m. The use of such a program can be of assistance, for the development of Army material, and for directing noise control efforts to those frequencies which control aural detectability.
Diffraction of sound over curved ground
Authors: Daigle G. A. , Embleton T. F. W.
Published: October 2005

Theory for predicting the diffraction of sound (creeping waves) into shadow regions, resulting from propagation in either a stratified medium or above a convex curved surface, is well known and has been used mostly in underwater sound. This theory has also been extended more recently to the case of a stratified atmosphere. However, this theory generally has not been used to predict the energy diffracted over curved ground, such as rolling hills on berms. More commonly, the diffracted energy in these cases has been predicted by adapting theory valid for thick barriers or wedges. The creeping wave theory may be more appropriate in cases such as the diffraction of sound over berms, or of sound of very low frequency over rolling ground. Preliminary experiments have been performed over convex curved (radius of curvature < 100 m) ground of finite and infinite impedance in the frequency range between 250 and 8000 Hz using propagation distances up to 30 m. Controlled measurements are also planned indoors over a carefully constructed curve surface. The measurements are compared with various aspects of theory.
Theoretical predictions of turbulent boundary layer pressure fluctuations transmitted into a viscoelastic-fluid composite layer
Authors: Ko Sung H.
Published: October 2005

The objective of this paper is to develop a model for evaluating the turbulent boundary layer pressure fluctuations transmitted into a viscoelastic-fluid composite layer. The theoretical model considered here is a plane layer of viscoelastic-fluid composite structure backed by a rigid plane surface. The other side of the composite layer is exposed to turbulent flow. The transmitted flow noise received by a rectangular hydrophone embedded in the composite layer was calculated for the turbulent boundary layer wall pressure spectrum proposed by Corcos. The transmitted flow noise was characterized by the frequency spectral density expressed in decibels. The main results presented in this paper are turbulent boundary layer noise reductions, which are given relative to the noise levels evaluated in the absence of the composite layer. Effects of the viscoelastic layer thickness, the fluid layer thickness, properties of the viscoelastic layer, and flow conditions on the noise reduction are discussed.
Application of the parabolic wave equation to atmospheric sound propagation over a locally reacting ground surface
Authors: White Michael J. , Gilbert Kenneth E.
Published: October 2005

Although the parabolic wave equation (PE) is widely used in underwater acoustics, it has seen only limited use in atmospheric sound propagation. This paper assesses the usefulness of the PE for outdoor sound propagation over a flat, locally reacting ground surface. The investigation addresses three questions: (1) Can the PE accurately treat the boundary condition at a locally reacting ground surface? (2) What starting field is needed for the PE to accurately describe the sound field near the source? (3) Does the PE have sufficient phase accuracy to predict tim field far from the source? In order to answer these questions, we compare PE calculations to benchmark calculations from a fast field program (FFP) and to measure data. The PE calculations presented are for a nonturbulent atmosphere. The extension of the calculations to account for turbulence is discussed.
Performance on interaural time discrimination: Direction does make a difference
Authors: Stager Sheila V. , Langford Ted L.
Published: October 2005

The question of differences in subjects' performances on interaural time discrimination depending on which ear received the slight delay was explored. Twenty subjects with normal hearing sensitivity and no previous training on interaural time discrimination participated in the study. A “same/different” procedure was used to measure the discriminability of low-frequency noise bursts with interaural phase differences of 20 and 12 deg favoring first one ear, and then the other. From the two resulting psychometric functions, the interaural differences (in degrees) which produced a 75% correct level of performance were determined. Subjects could be classified according to their performances. Some subjects performed differently depending on ear favored. Some subjects performed equally well regardless of ear favored. Some subjects did not reach the criterion of 75% correct with the introduction of a phase difference of 20 deg regardless of ear favored. These results are considered in light of previous studies of interaural time discrimination, and of time- versus intensity-sensitive subjects (McFadden et al., 1973). [Work supported in part by NIH grant ♯NS16396.]
Binaural fusion, apparent motion, and the precedence effect
Authors: Perrott David R. , Strybel Thomas Z.
Published: October 2005

Wideband noise pulses, 50 ms in duration, were presented from pairs of speakers. Across sessions, the angular separation of the sources was varied between 6 and 160°. Within sessions, interstimulus onset intervals (ISOI) varied between 0.1 to 500 ms. Eight listeners were required to perform two concurrent discrimination tasks. The first was an objective procedure in which they were required to report the temporal order in which the sources were activated. The second task required them to assign the event to one of five descriptive categories: single stationary image; multiple stationary images; single moving image; an interrupted but moving image; or two successive images. Discrimination of the temporal order of speaker activation was only moderately affected by the angular separation of the sources (thresholds ranged between 12–20 ms). These results are well in line with other experiments in which temporal order judgments are required. However, when subjects report “apparent motion,” a relatively rare event with small ISOI's, judgments of temporal order are remarkably accurate (exceeding 80% correct) with all but the shortest ISOI's employed. The implications of these results will be discussed. [Work supported by NSF.]
The accuracy of head saccades under monaural and binaural listening conditions
Authors: Tucker J. , Perrott David R.
Published: October 2005

Literature examining head motion and auditory function has typically focused upon whether changes in position facilitate auditory spatial performance. The present experiment addresses the reverse of this question by attempting to quantify the spatial characteristics of the head saccade itself, in the context of various localization tasks. Horizontal head position was measured every 16 ms as both binaural and monaural listeners turned to face a sound source located along an arc of + or −60 deg azimuth. The signals were 0.5-, 2.0-, and 3.7-kHz tones presented at 40 dB SPL. Both single pulse and pulse train conditions were employed. Resolution was best with speakers near the subjects median plane for binaural conditions, and on the side opposite the occluded ear in monaural conditions. Precision of the final head position could be predicted by the frequency used. The current results follow known auditory spatial functions and support the 81-year-old hypothesis that while binaural localization is superior to monaural, given the opportunity to move one's head in the presence of an ongoing acoustic signal, monaural listeners may still be quite accurate. [Work supported by NIH.]
Localization of pure tones in the front-back and vertical dimension under monaural and binaural conditions
Authors: Musicant Alan D. , Tucker J. , Perrott David R.
Published: October 2005

Butler and Flannery (1980) proposed the spatial referent map theory. The present experiment was designed to test this theory under extreme conditions. Presentation of pure tones in the monaural condition should stimulate the spatial referent map directly. Here 500-ms pure tone pulses were used. Frequencies ranged from 0.5 to 9.5 kHz. Speakers, circumaural earphones, or insert earphones were utilized. The method of paired comparison with a four-alterative, forced-choice response task was employed. Subjects were requested to respond on both front-back and vertical dimensions. Without any possibility of interaural difference cues, tones did appear to come from different spatial locations. Despite a difference in the monaural condition the free field performances supported Butler's theory. In all conditions, responses on the vertical dimension showed a monotonic relationship between frequency and phenomenal vertical position. The data on the front-back dimension were relatively stable across subjects in the monaural condition, however, the between-subject variability was too great to make any kind of definitive statement.
Binaural detection and discrimination: Normal listeners
Authors: Passaro Carrin , Koehnke Janet , Colburn H. Steven
Published: October 2005

In conjunction with a study of binaural performance in impaired listeners, the performance of normal listeners in several binaural tasks were measured. Specifically, just-noticeable differences (jnd's) were measured in interaural correlation (IC), interaural time delay (ITD), and interaural intensity difference (IID) for one-third octave noise bands centered at 500 Hz and 4000 Hz. In addition, we measured NoSπ detection thresholds at the same two frequencies for a tone centered in a third-octave band of masking noise. Measurements were made at several reference values of ITD and IID, including −300, 0, and + 300 μs and −12, 0, and +12 dB. jnd's and thresholds were estimated using a relatively crude adaptive method so that a complete set of measurements could be obtained quickly for each subject. Results are discussed in terms of the dependence on reference interaural conditions and in terms of the ability to predict performance on each task from performance on the other tasks. So far, the only consistent effect of changing interaural reference parameters is larger jnd's in IID with reinforcing values of ITD and IID. Also, IC discrimination performance is directly predictable from the detection threshold and vice versa. The ability to predict both of these tasks from ITD and IID discrimination data is evaluated for various theoretical models. [Work supported by NIH.]
Perception of modulations in pitch and lateralization
Authors: Beecher Laural , Stern Richard M.
Published: October 2005

The ability to perceive low-frequency sinusoidal modulations of monaurally and dichotically created pitch was compared to the perception of modulations of the subjective lateral position of a binaural image. The stimuli used in the dichotic pitch experiments had low-pass spectra that were flat between 0 and 2000 Hz, and they were created by modulating the time delay of a multiple-phase-shift filter [F. A. Bilsen, J. Acoust. Soc. Am. 59, 467–468 (1976)]. Stimuli with similar spectra and sinusoidally modulated interaural time delays (ITD's) were used for the latralization experiments [D. W. Grantham and F. L. Wightman, J. Acoust. Soc. Am. 63, 511–523 (1978)]. Also examined was the perception of monaural frequency-modulated pure tones, and monaural low-pass stimuli with power spectra similar in shape to the “internal spectra” of the dichotic pitch stimuli. Subjects discriminated between sinusoidally modulated and unmodulated stimuli using two-cue, two-alternative 4IFC paradigms, and we determined the threshold frequency deviation or ITD as a function of modulation frequency. Results were similar in form for all experiments: threshold frequency deviation or ITD was constant for modulation frequencies up to a particular “corner” frequency, and then increased as a power function of modulation frequency up to at least 32 Hz. This corner frequency was 4 Hz or less for the binaural lateralization and dichotic pitch experiments, and approximately 10 Hz for the two monaural pitch experiments. [Work supported by NIH.]
Simultaneous and nonsimultaneous masking within natural speech
Authors: Spiegel Murray F.
Published: October 2005

Auditory masking functions are well known only for steady-state tones and noises, not for complex sounds such as speech. To investigate the contribution of simultaneous and nonsimultaneous masking in natural speech, thresholds were obtained for 15-ms probe tones placed in the closure portion of naturally spoken VCV utterances. The utterances contained a long closure (stop/d/) or a short closure (flap/P/). Two additional conditions were tested: flap closure was replaced by a short portion of stop closure, and the surrounding vowels were digitally attenuated. Critical-band filtering and spread of masking characterize the simultaneous-masking results. Some conditions reveal significant amounts of nonsimultaneous masking. These effects are less straightforward than the effects of simultaneous masking, primarily due to the time-varying nature of the natural speech maskers. Results of this line of research may help refine perceptually weighted filters used in speech coders to reduce the perceived level of quantization noise [cf. M. R. Schroeder, B. S. Atal, and J. L. Hall, J. Acoust. Soc. Am. 66, 1647–1652 (1979)].
Band importance functions for certain consonant features
Authors: Duggirala Vasanta , Studebaker Gerald A. , Pavlovic Chaslav V. , Sherbecoe Robert L.
Published: October 2005

Miller and Nicely (1955) demonstrated that there are differences in the most important frequency regions for the recognition of different linguistic features. The present study was designed to determine the relative importance of each one-third octave band between 178 and 8912 Hz for each of several consonant features. The test instrument was the diagnostic rhyme test (DRT) of Voiers, Cohen, and Mickunas (1965): The analytical methods of French and Steinberg (1947) were used to derive band-importance functions for the DRT as a whole and for each of the features, voicing, nasality, sustention, sibilation, graveness, and compactness. The one-third octave band importance weights for the DRT and the various features were found to be quite different from each other and from those reported earlier for nonsense syllables (ANSI S.3.5-1969; Kryter, 1962). Curves describing the importance per Hertz and the cumulative band importance will be presented. [Work supported by NIH grant ♯ NS 15996.]
Speech reception in noise by hearing-impaired listeners
Authors: Zurek P. M. , Delhorne L. A.
Published: October 2005

The goal of this study was to determine the extent to which the difficulty experienced by impaired listeners in understanding noisy speech may be explained merely on the basis of elevated detection thresholds. Twenty impaired ears of 14 subjects, spanning a variety of audiometric configurations with average hearing losses to 75 dB, were tested for reception of consonants in a speech-spectrum noise. Speech level, noise level, and frequency-gain characteristic were varied to generate a range of listening conditions. Results for impaired listeners are compared to those of normal-hearing listeners tested under the same conditions with extra noise added to approximate the impaired listener's thresholds. Although there are a few exceptions, the conclusion based on this sample of moderate-to-severe hearing loss is that, when compared to normals listening under similar conditions of threshold shift (or, more generally, similar values of articulation index), hearing-impaired listeners exhibit little or no handicap in speech reception. [Work supported by NIH.]