bluuebunny/crossref_metadata_2025 huggingface dataset:
Influence of national languages on learning English
Authors: Jianping Pan
Published: October 2005

Every language can be described in terms of a set of phonemes, including, in part, vowels, diphthongs, semivowels, and consonants. In China, Mandarin is the standard spoken Chinese, i.e., the national language. There are many additional dialects, each with its own set of phonemes. Fluent speakers in Mandarin and another dialect can therefore pronounce two sets of phonemes. The phonemes are produced mainly by operations of the vocal tract. The connections between vocal tract movements and phonemes are established in childhood, first for the dialect and then for Mandarin. This led us to compare the ability to learn to speak English among Chinese speakers using different dialects and therefore different sets of phonemes. It was observed that the greater the similarity of the dialect to English, the more intelligible and natural were the initial attempts to speak English. This was in spite of similar fluency in the national language, Mandarin. It was concluded that early learning of English was important, independent of the national language, and it was hypothesized that similar results would be obtained in other non-English-speaking countries.
Glottal source estimation using a sum of exponentials model
Authors: Krishnamurthy A. K.
Published: October 2005

This paper describes an algorithm for simultaneously estimating the parameters of a model for the glottal source and the vocal tract filter. The glottal source signal is described by t.he four-parameter LF model [Fant et al., Speech Transmis. Lab. Q. Prog. Stat. Rep. (1984)]. The vocal tract is modeled as an all-pole filter; however, to allow for the effects of source-tract interaction, a separate model is used in the closed and open glottal phases. Under these assumptions, it is shown that the output speech signal can be modeled as a sum of complex exponential signals. The parameters of the speech signal model are estimated using the algorithms described in Kumaresan and Tufts [IEEE Trans. Acoust. Speech Signal Process. ASSP-30, 833–840 (1982)]. The electroglottograph signal is used to obtain a preliminary estimate of the location of the closed and open phases in each pitch period. The analysis of steady vowel sounds from several speakers indicates that this method provides a very accurate estimate of the glottal source and the vocal tract formants. [Work supported by the OSU Seed Grant program.]
The effect of macroprososdic context on consonant perturbations of fundamental frequency
Authors: Kingston John
Published: October 2005

Kohler argues, on the basis of German data, that differences in F0 before or after voiced and voiceless stops (the microprosodies) may be obliterated by the larger scale effects on F0 of the surrounding intonation contour (the macroprosodies) [Kohler, Phonetica 39, 199–218 (1982)]. In a similar fashion, Silverman claims, on the basis of English data, that the direction of F0 change after stops differing in voicing depends on the surrounding macroprosodic contour [Silverman, Phonetica 43, 76–91 (1986)]; The macroprosodic contour in which word-initial consonants [p, t, b, d, sp, n] were embedded at two levels was manipulated: (1) The consonant occurred either immediately before a stressed vowel or before an unstressed vowel, e.g., palace versus police; and (2) the words occurred either in focus or with focus on the preceding word, e.g., I saw a few churches on Monday, but many PALACES on Wednesday versus I saw only a few on Monday but MANY palaces on Wednesday. These manipulations were entirely orthogonal and allow us to examine both the effects of word internal as well as syntactically determined macroprosodies on the extent and direction of consonantal effects on F0 in English.
Physiological mechanisms of masking and intensity discrimination
Authors: Delgutte Bertrand
Published: October 2005

Recent electrophysiological studies in which stimulus paradigms were designed to mimic those of psychophysics suggest that a broad range of psychophysical data on masking and intensity discrimination can be simply explained by assuming that signal detection is primarily based on average discharge rate informal ion in auditory-nerve fibers, provided that information from high-threshold fibers is taken into account. These studies further suggest that tone-on-tone masking is due both to spread of the excitation produced by the masker to the signal place along the cochlea, and to suppression of the signal by the masker, with the relative importance of the two mechanisms depending on masker level and frequency separation between signal and masket. In particular, two-tone rate suppression is largely responsible for the upward spread of masking at high masker levels and for differences between simultaneous and nonsimultaneous masking techniques. New results will also be presented on unmasking due to suppression of the masker by the signal, improvements in signal detectability by off-frequency listening, and the role of olivocochlear efferents in signal detection. [Work supported by NIH Grant NS13126.]
An equation for acoustic propagation in an inhomogeneous medium with relaxation loss
Authors: Nachman Adrian I. , Smith James F. , Waag Robert C.
Published: October 2005

An equation for acoustic propagation in an inhomogeneous medium with relaxation loss is systematically derived from the classical dynamic equations together with an equation of state for relaxation. The derivation assumes small acoustic perturbations but accommodates arbitrary spatial inhomogeneities in material compressibility, density, and parameters of relaxation. The linearized wave equation obtained for n relaxation mechanisms has order n + 2, is causal, and yields the expected dependence of attenuation on frequency. Exact analytic expressions valid at all frequencies are given for the spatially varying attenuation coefficient, as well as phase velocity. A Green's function is calculated for the equation. The results may be used to model scattering for image reconstruction and the determination of statistical properties, such as average differential scattering cross section.
Extraction of information from low-frequency sounds generated within the human body
Authors: Mast T. Douglas , Pierce Allan D.
Published: October 2005

The analysis of sounds generated within the body has been a staple of medical diagnostics since Laennec's invention of the stethoscope (c. 1804). Also, interest in muscle sounds dates back to Grimaldi's treatise on optics published in 1665. More recent work by Oster and Jaffe (1980), Barry (1987), and Frangioni et al. (1987) has led to quantitative insight on the physical processes that generate such sound. A feasibility analysis is presented here for the problem of source characterization using information taken outside the human body, with emphasis on sources of frequencies less than 250 Hz. The available information is the displacement time history of all points on the body's surface, although one would want to measure displacements at a restricted number of points. The body is modeled as a dissipative fluid-filled sphere or a finite cylinder. It is argued that sources of interest can be idealized as compact, rather than distributed, combinations of monopoles, dipoles, and quadrupoles. Thus the unknowns to be determined in regard to any frequency component are the source coordinates and the strengths and relative phasing of the probable multipoles. Results reported include measurement accuracies necessary for satisfactory source characterization. [Work supported by the William E. Leonhard endowment to Pennsylvania State University.]
Mixture composition determination from measurements of the acoustic nonlinearity parameter
Authors: Everbach E. C. , Apfel R. E.
Published: October 2005

An important inverse problem in biomedical acoustics is the determination of the composition of a tissue based upon measurements of its acoustic properties. Under many circumstances, the tissue can be modeled as an ideal mixture of components, each with known or assumed properties. The measured properties of the mixture as a whole are then related to those of the components through a set of mixture laws. Once formulated, the problem can be inverted to yield the volume or mass fractions of the components present in the tissue. In this laboratory, the use of the acoustic nonlinearity parameter B/A to help infer the composition of mixtures of bioligical materials has been studied. Various mixture models have been investigated, and a modified version of Apfel's model [J. Acoust. Soc. Am. 79, 148–152 (1986)] has yielded the most useful results. This modified mixture model will be presented, along with precise data that have been obtained on biological mixtures using a novel acoustic interferometry technique reported earlier [J. Acoust. Soc. Am. Suppl. 1 80, S4 (1986)]. [Work supported by the U. S. National Institutes of Health through Grant R004GM300 19].
Bubble clouds as sources and scatterers of underwater sound
Authors: Prosperetti A. , Lu N. Q. , Lezzi A.
Published: October 2005

A model of a bubble cloud based on a set of averaged equations is used to study the acoustic emission and scattering properties of these objects. From a study of the normal modes of the cloud, it is inferred that the lowest modes have a frequency much lower than that of the individual bubbles. This indicates that clouds of bubbles can radiate at such low frequencies if these modes are excited. The results of some scattering calculations at different wavelengths are also shown as a function of the incidence and scattering angles. [Work supported by ONR.]
Streaming generated by aggregations in a rotating ultrasonic waveguide
Authors: Whitworth Glenn
Published: October 2005

A chamber cavity, which has a square cross section and pressure-release walls, is used to produce a well-defined, 160-kHz standing ultrasonic field. A suspension of latex microspheres in aqueous metrizamide fills the chamber. The chamber rotates about a horizontal axis producing the centripetal force necessary to contain the buoyant spheres in the axial region. At low particle concentrations, aggregations of microspheres form at half-wavelength intervals near the axial positions of acoustic pressure amplitude (PA) minima, as expected because of rotational and acoustic radiation forces. At higher concentrations, additional particle distributions are often seen that suggest the presence of flow. When high concentrations of larger particles are used, small aggregations also form at axial positions of PA maxima. To account for the flow, theory for acoustic streaming in a rotating fluid was developed and applied to flow near the larger aggregations. Reasonable agreement with observations was obtained when a term was added arising from drag on the aggregations. [Work supported by NIH Grant CA42947.]
Experimental studies of a quasi two-dimensional nonpropagating hydrodynamic soliton
Authors: Winkler Erik M. , Wu Junru
Published: October 2005

A nonpropagating hydrodynamic soliton was first observed by Wu et al. [Phys. Rev. Lett. 52, 1421 (1984)] in a rectangular water trough that was parametrically driven at a frequency below twice that of the linear cutoff frequency of (0, 1) mode. A soliton of the same type was recently studied that forms when the trough is driven at a frequency below twice that of the linear cutoff frequency of (0,2) mode. The profile of the soliton in length direction as well as in width direction is determined. A video tape of the experiment will be shown during the presentation. [Work supported by NSF and Vermont Epscor.]
Measurements of transmission loss to ultrasound through two-dimensional ensembles of trapped bubbles versus the amplitude of the incident wave
Authors: Wu Junru , Nyborg Wesley L.
Published: October 2005

In continuation of work reported previously, the frequency spectra of transmission coefficients to ultrasound through sheets of gas-filled micropores at incident amplitudes up to 3 × 104 Pa have been measured. It is found that as the amplitude of the incident wave is increased, the peak frequency of transmission loss through a bubble ensemble shifts to lower values by as much as 47%. Results of the experiments and the possible theoretical explanation will be presented and discussed. [Works of JW are supported by NSF and Vermont Epscor. Works of WLN are supported by NIH CA42947.]
Acoustic radiation forces used to control the motion of a glass microneedle
Authors: Wu Junru , Work Steven S. , Warshaw David M.
Published: October 2005

A method has been developed using a glass microneedle to tug on protein filaments and to measure their tensile strength (the force necessary to break the bond between individual protein molecules) [A. Kishino and T. Yanagida, Nature 334, 74–76 (1988)]. To precisely control the motion of such a glass microneedle, a technique that uses acoustic radiation force at megahertz frequencies has been developed. Results, from experiments using a rectangular resonant chamber and a video detection system have indicated that forces of 5 nN exerted on a glass microneedle and displacements of 4 μm can be resolved and were reproducible. A video tape of the experiment will be shown. [Works of JW supported by NSF and Vermont Epscor; DMW supported by NIH AM34872 and AHA.]
Room acoustics for THX® motion-picture theaters
Authors: Holman Tomlinson
Published: October 2005

The THX® Sound System is installed in over 300 premium motion-picture theaters and dubbing facilities worldwide. An important part of the program is specified room acoustics, including both local acoustical environment of loudspeakers and global room acoustics issues including background and intrusive noise control and reverberation time criteria. Acoustical standards established for this program will be examined, and the methods of measurement will be explained. Time-honored techniques such as using appropriate measurement microphones for the sound fields encountered, as well as time and space averaging, are applied in new, more portable, forms.
Localization schemes for beam-type sources
Authors: Lu I. T.
Published: October 2005

A time-harmonic isotropic source in stratified waveguides can be localized by performing matched-field processing in “mode space.” Because the range and depth information of the source are contained only in the phase and magnitude, respectively, of complex model amplitudes, the range and depth data can be processed independently. This is not true for beam-type sources. When propagating in a waveguide, an initially collimated beam undergoes diffusion after successive reflections and refractions and is converted eventually into the oscillatory pattern of one or more guided modes. However, the upgoing and downgoing plane wave constituents of a given mode do not have the same excitation strength as in the case of isotropic sources. Here, localization schemes of a Gaussian beam source that is modeled via the complex source point technique are considered. The beam direction, beam width parameter, and waist location are determined in “mode space.” The procedure is greatly simplified if the source is a well-collimated beam. [Work supported by NSF.]
Source localization by inversion of the parabolic equation method
Authors: Bates Susan M. , Bates Bruce J.
Published: October 2005

The inverse split-step parabolic equation method is derived. Spatial localization of single and multiple low-frequency harmonic point sources in a deep ocean is demonstrated by using simulated pressure field measurements from a vertical array. In addition, the method is applied to a source extended in range and depth. This method is solved in a single iteration, unlike some inverse techniques that vary parameters.
An equivalent circuit appreciation of Warren P. Mason
Authors: Ballato Arthur
Published: October 2005

Warren Mason was a man of many parts and prolific in them all. One area in which he exercised a decisive influence was that of equivalent network representations of electromechanical systems. This paper salutes Mason's accomplishments in helping to bridge the disciplines of acoustics and electronics. It begins with a brief discussion of analogs and describes the Butterworth-VanDyke circuit of a piezoelectric vibrator. This is where Mason started his productive work in the subject, introducing acoustic transmission lines, mechanical ports, and piezoelectric transformers. Today, the Mason equivalent circuit is universally used for bulk and surface acoustic wave device characterization. It has also given rise to a variety of alternative formulations such as analog networks, the KLM equivalent circuit, and systems models, which are discussed.
Acoustical filters in plane-wave fields
Authors: Cook Richard K.
Published: October 2005

Lumped-parameter acoustical filters were thoroughly described by Warren P. Mason in his book on electromechanical transducers and wave filters. Today, the filtering action of open-ended circular (standing-wave) tubes and of plane-parallel waveguides is described. The description is most simply in terms of the spatial wave fields of open-ended tubes and slots. These were first analyzed accurately by L. A. Weinstein (Vaynshteyn) in the 1940s–1950s for both acoustical and electromagnetic waves. The basic mathematical technique was detailed examination of the diffracted field as a wave emerged from the open end after propagation inside the waveguide. The solutions to the diffraction problems are applied, in acoustical engineering, to the accurate analysis of open resonators, as well as to the analysis of probe tubes used for measurement of sound fields. Earlier investigators had been able to do no more than arrive at rough approximations in the form of “end corrections” to the length of a resonating circular tube, fianged at its open end.
Acoustic cavitation 42 years after the Briggs, Johnson, and Mason paper
Authors: Apfel Robert E.
Published: October 2005

Warren Mason was one of the early pioneers of high-power ultrasonics who studied its effects on liquids [Briggs et al., J. Acoust. Soc. Am 19, 664–677 (1947)]. One of these effects is acoustically induced bubble activity. Such activity can be desirable, as in ultrasonic cleaning, or undesirable, as with cavitation on sonar transducers. This talk will review the basic physics underlying the onset of cavitation, the dynamics of bubble motion, and the effects caused by cavitation, with emphasis on how to optimize or minimize these effects, depending on the application. [Much of this work has been supported by the Office of Naval Research and by the National Institutes of Health through Grant 1R01CA39374.]
Interior idle noise in current model passenger cars
Authors: Foulkes Timothy J.
Published: October 2005

The author, a consultant in architectural acoustics, became interesting in the vast differences in interior noise levels between passenger cars in today's market. Octave-band noise levels were measured in 20 different automobile models. In order to reduce the number of variables, the measurements were limited to the driver's position with the car stopped and engine at idle. In the 31-Hz band, noise levels varied 20 dB between two different models by the same manufacturer. The range of A-weighted levels spanned only 8 dB. While the A-weighted sound levels as published by most consumer magazines provide some indication of relative loudness, examination of octave-band levels shows radical differences between models. Some observations and conclusions are drawn by an “enlightened” consumer.
Cancellation of noise in the presence of signals of short duration
Authors: Raman Vijay , Mohammed Osama A. , Yen Kang , Ramdin Kurt
Published: October 2005

This work focuses on the active cancellation of acoustic noise in industrial environments such as factories, airports, etc., by the use of earphones with active components. An important consideration in this problem is that undesired components (noise) should be reduced without eliminating certain desired signals. More specifically, the cancellation technique must incorporate the significant constraint that warning sounds and brief verbal exchanges must not be canceled. Achieving this goal is particularly difficult when the characteristics of the desired signals resemble that of the noise. In the course of this work, these researchers have used the fact that these sounds are of relatively short duration, whereas the ambient noise environment is persistent, to formulate and evaluate appropriate signal detection and adaptive noise-canceling schemes. The filtering algorithm adopted is a variation of the well-known adaptive least-mean-squares algorithm. [Work supported by the Florida High Technology and Industry Council.]
Results of a computer simulation scheme for active noise reduction
Authors: Yen Kang , Raman Vijay , Mohammed Osama A. , Ramdin Kurt
Published: October 2005

This paper presents the results of a computer simulation scheme for active noise minimization in a random noise-canceling headphone. The noise-canceling algorithm was developed in a separate paper by these authors. The performance of the noise-canceling algorithm has been tested via digital simulation of the actual noise field distributions within the ear canal. Farfield conditions have been assumed by the modeling environmental noise field to be due to sources far from the cancellation headset. The ear is initially modeled as an axisymmetric pipe, and the resulting field is calculated as a function of the location of noise canceling sources. The results proved helpful in determining the appropriate locations of the noice-canceling sources in order to achieve maximum noise energy reduction at a predetermined point in the pipe. [Project supported by the Florida High Technology and Industry Council.]
Use of the Articulation Index to assess “noise reduction” hearing aids
Authors: Fabry David A. , Van Tasell Dianne J.
Published: October 2005

The Articulation Index (AI) was used to evaluate a commercially available “noise reduction” hearing aid. First, a transfer function relating calculated AI to rated speech intelligibility was derived empirically for a group of normal-hearing subjects. This function was used to predict speech recognition by 12 hearing-impaired subjects wearing either the noise reduction hearing aid or conventional amplification. Subsequently, subjects rated the intelligibility of connected discourse presented under both aided conditions at five signal-to-noise (S/N) ratios in speech-weighted noise. The AI transfer function predicted speech intelligibility accurately for half the subjects. For the remaining subjects, predicted scores were typically higher than observed scores. If the effects of upward spread of masking were included in the AI calculations, these differences were minimized. For all subjects, speech intelligibility increased monotonically with AI. AI scores and speech intelligibility ratings were similar between the two hearing-aid conditions, suggesting that S/N ratio was not increased by the noise reduction hearing aid relative to conventional amplification. [Work supported by NINCDS NS 12125.]
Implementation of principal components spectral analysis of speech in a wearable tactile aid
Authors: Rahrer T. J. , Kwasniewski T. A. , Tansley B. W.
Published: October 2005

One common configuration of a vibrotactile sensory aid for the hearing impaired is the vocoder scheme employing a bank of bandpass filters that model human audition. These systems result in 16 or more spectral energy outputs, each of which drives a vibrator. Although successful in laboratory and classroom settings, the large number of vibrators required makes the implementation of wearable versions difficult. By performing a principal components analysis of the filter bank outputs, the spectral information can be recorded in as few as four parameters. A compact principal components analysis system has been implemented using digital signal-processing techniques. The system consists of a 1/3-oct bandpass filter bank, high-speed energy detectors for each filter, logarithmic scaling of spectral energies, and a principal component calculator, all in software running on a commercial DSP microchip. In this paper, the principal components system is compared to the traditional vocoder method, as well as to the pitch and formant extraction schemes.
Real-time cepstral analysis, speech feature extraction, for the cochlear prosthesis
Authors: Gedeon Ibrahim J. , Kwasniewski Tad A. , Rahrer Timothy J.
Published: October 2005

The most common speech-processing strategies for use in the cochlear implant employ multifeature extraction. For the proposed application, the first three formant frequencies and amplitudes along with the pitch period are calculated for each speech frame processed. This paper presents real-time cepstral analysis as a complete speech analysis strategy for the cochlear implant. Cepstral analysis has long been known as an accurate method for pitch extraction and formant tracking. However, it has not been previously considered for real-time signal-processing applications, due to the complexity of the calculations involved. Taking advantage of today's microprocessor technology, cepstral analysis has been implemented using the Texas Instruments TMS320C25 digital signal processor. Results obtained with a 10-pole LPC analysis using the same processor are compared with those of the proposed method. The results are also compared to a commercial software package (ILS from STI) for validation.
Vowel length and closure duration in word-medial VC sequences
Authors: Davis Stuart , Van Summers W.
Published: October 2005

Previous research has shown that English vowel length varies depending on the voicing characteristic of the following consonant. For stop consonants, closure durations also vary as a function of consonantal voicing. Generally, vowel-stop sequences containing voiced consonants show longer vowel durations and shorter closure durations than similar sequences containing voiceless consonants. These previous studies have focused on stressed vowels in monosyllabic or bisyllabic words. Very little research has examined the effects of postvocalic voicing on stressless vowels. In the present study, the influence of postvocalic voicing on vowel and closure durations in VCV́ and V́CV sequences is studied. Subjects produced sentence pairs containing target words contrasting in intervocalic consonantal voicing (e.g., adopt-atop, tabbing-tapping). Both stressed and unstressed vowels tended to lengthen before voiced consonants. However, the vowel-lengthening effect was not as consistent for stressless vowels as for stressed vowels. Closure durations were longer for voiceless stops than voiced stops after a stressed vowel. However, voicing effects on closure duration were inconsistent after stressless vowels. The results have implications concerning perceptual cues for intervocalic voicing and for issues concerning syllable-internal structure. [Work supported by NIH.]
An articulatory study of consonant-vowel overlap
Authors: de Jong Kenneth
Published: October 2005

Fowler [Phonetica (1981); J. Exp. Psychol. (1983)] has proposed that speech production is a series of global articulatory movements associated with the vowels of an utterance, upon which smaller, consonantal movements are superimposed. A possible articulatory test of this theory models consonants and vowels as alternations between a relatively closed and open vocal tract. Using the Wisconsin x-ray microbeam system, articulatory records were obtained of English phrases with varied accent placement and alveolar consonant sequences of varied duration. Analysis of one speaker's productions is now complete. When the data are pooled over all conditions, the total duration of the bisyllabic sequence and the time difference between sonority peaks is strongly related to the length of the medial consonants; there is no evidence for articulatory overlap—i.e., vowel shortening in the neighborhood of long consonants or shifting of sonority peaks toward longer consonants. However, correlations within accent types uncover a strong shortening effect for accented syllables preceding long consonants, along with a reduced displacement in the following vowel. Thus any comprehensive overlapping consonant-vowel theory of speech production must take into account the general prosodic structure of the utterance. [Work supported by the National Science Foundation under grant No. IRI-8617852 to Mary Beckman.]
The timing of lip rounding and tongue backing for /u/
Authors: Lee Gina M.
Published: October 2005

A small corpus of x-ray microbeam data was examined to test the predictions made by two well-known views of anticipatory coarticulation: time locking and feature spreading. Lip rounding and tongue backing associated with English /u/ were investigated. The token types were VCn V sequences where V = /i, u/ and C = /s, t/, and consonants are assumed to be neutral with respect to both features under study. The speaker showed no evidence of time locking or feature spreading of the onset of lip rounding in /iCn u/-type tokens. Within the same token types, there was an absence of tongue backing activity during the production of the consonants. However, other articulatory phenomena are temporally fixed. The maximum degree of rounding was fixed relative to the acoustic onset of /u/ [as stated in G. M. Lee, J. Acoust. Soc. Am. Suppl. 1 81, S37 (1987)]. This is consistent with past findings involving EMG peaks [Hirose et al., Ann. Bull. RILP (1968, 1969)]. The timing of the minimum degree of backing was also fixed. These findings may provide evidence for the notion of a (static) target in speech production.
Stretch-activated ion channels as acoustical transducers
Authors: Sachs Frederick
Published: October 2005

Ion channels sensitive to membrane distortion have been demonstrated in nearly all cells, from plants to animals, in specialized sensory transducers, such as the crayfish stretch-receptor and, in traditional motor outputs, such as skeletal muscle. The channels respond to membrane tension by opening with a probability that (for low tensions) follows an exponential in the square of tension. The data can be explained with a simple model in which tension does Hookean work on the channel raising its energy, which leads towards transitions to the open state. Even in skeletal muscle, the channels are sufficiently sensitive to tension to serve as productive models of hair cell transduction. If auditory transduction is due to stetch sensitive ion channels in ciliary membranes, a very simple traditional model of force coupling between stereocilia leads to the following predictions [F. Sachs, “Biophysics of mechanoreception,” Membrane Biochemistry 6 (2), 173–195 (1986)]: (1) tufts are tapered in order to provide a region of strain that is first order in the angle of deflection; (2) channels may be located over the cilia, but will only be turned on at the distal tips in the strained region ofmembrane; (3) for maximum sensitivity, the optimum number of channels is on the order of one/tip; (4) given a constant channel density, to maintain optimum sensitivity at different mean pressures requires some form of mechanical relaxation to the optimal tension; (5) specialized tip links connected to channels are not required since strain is applied in the plane of the membrane; and (6) the channels are sufficiently fast to account for the response time of the generator current and the transduction system is sufficiently sensitive to account for the observed psychophysical “thresholds.” [Work supported by NIADDK-13195 and USARO 22560-LS.]
A review of air-moving device noise: Mechanisms and measurement methods
Authors: Yeager David M.
Published: October 2005

The predominant acoustic noise sources of electronic systems are very often due to convection cooling devices. A wide range of capacities of air-moving devices (AMDs) are used in computers and business equipment, typically varying in total flow rate from 0.01 to 1.0 m3/s, and in static pressure drop from 10 to 1000 Pa. This review paper will cover several aspects of air-moving device noise: (1) basic mechanisms of aerodynamic noise generation, especially in the context of air-moving device noise, (2) measurement methods of characterizing aeroacoustic noise sources on airfoils immersed in turbulence, (3) important parameters in assessing aeroacoustic performance, (4) standardized measurement methods for determining noise emission levels of AMDs; (5) use of sound power levels of AMDS for predicting overall system noise levels; and (6) nonstandardized measurement methods for quantifying air-moving devices based on sound intensity techniques.
A model for mechanotransduction in a Pacinian corpuscle
Authors: Holmes Mark H. , Bell Jon
Published: October 2005

A Pacinian corpuscle (PC) is a rapidly adapting mechanoreceptor that resides in the skin and other tissues. The transduction process in the PC consists of a mechanical filtering by the capsule that surrounds the dendrite, along with a receptor membrane component. A model is described that accounts for these. The layer structure of the capsule is taken as formed from concentric elastic membranes with the interlamellar spaces filled with a viscous incompressible fluid. From this the hoop strain in the receptor membrane is found and this serves as the input to a receptor model, which has a strain-sensitive activation mechanism for the transducer conductance. The transient behavior of the receptor potential due to various compression stimuli to the capsule is investigated. Of particular interest is how the viscoelastic properties of the capsule contribute to the adaptive behavior of the PC. To this end the dependence of the hoop strain on the rate of compression and the imposed strain are discussed along with the response of the PC when the loading is removed.
Modal studies in a double second Caribbean steel drum
Authors: Hansen Uwe J. , Rossing Thomas D.
Published: October 2005

Caribbean steel drums characteristically contain several sections of separately tuned areas imbedded in one quasihemispherically shaped drum head that is formed in the end of a 55-gallon steel drum. The double second pan under study for this report includes three A♭ sections tuned in octaves. Pitch and spectral content for each section depend on section geometry, mass distribution, and elastic properties. Section geometry is determined by hammer-embossed boundary delineation while mass and tension are modified by selective hammering in the section interior, which enables tuning of several lower partials harmonically. Spectral nearfield studies illustrate harmonic content as well as departures from harmonicity in some of the notes under study. Time average holographic interferometry shows strong amplitude dependence of spectral content of notes and illustrates coupling between sections. Complications in using impact excited modal analysis due to noncoherent input contributions are discussed, and computer-animated modes are illustrated.
Physiological “release from masking”
Authors: Mott J. B. , McDonald L. P. , Sinex D. G.
Published: October 2005

Psychophysical “release from masking” [S. Buus, J. Acoust. Soc. Am. 78, 1958–1965 (1985)] is observed when the masking exerted by stimuli with equal power but different envelopes is compared. Depending on the masker-to-probe frequency ratio, thresholds for flat-envelope maskers are higher than those for corresponding fluctuating-envelope maskers. A similar phenomenon was previously reported for masked thresholds based on rate responses of chinchilla auditory-nerve fibers [J. B. Mott et al., Absts. A.R.O. 12 (1989)]. Those results were interpreted as suggesting that neural activity during periods of low masker energy was important for detection. The present study examined whether synchronized responses of auditory-nerve fibers also contribute to “release from masking.” Synchronized rates were measured in short-time windows, centered on periods of high and low energy in the maskers' envelopes. Synchronized rate at the probe frequency grew more rapidly as a function of level during low-energy windows than high-energy windows. Statistical thresholds based on synchronized behavior will also be reported. [Work supported by NINCDS NS23242.]
The amplitude-latency trade-off demonstrates a temporal basis for fine echo delay acuity in bat sonar
Authors: Simmons J. A. , Ferragamo M. , Moss C. F.
Published: October 2005

Echolocating bats (Eptesicus fuscus) can perceive a change in echo delay smaller than 1 μs in two-choice, echo-jitter discrimination tasks. Although the bats emit sounds short enough to avoid overlap of stimulus echoes with incidental echoes, and although the transfer function of the delay electronics reveals no amplitude or spectral cues correlated with delay, it is difficult to believe that such fine temporal is not based on spectral cues. If the delay of echoes is encoded by neural discharge timing rather than from spectral information, perceived delay should shift according to the neural amplitude-latency trading function, which is − 13 to − 18 μs/dB for N1-N4 responses in Eptesicus. The apparent delay of echoes in the jitter task shifts by − 16 to − 17 μs/dB, demonstrating that the bats perceive the delay of jittered echoes from their arrival time rather than from some spectral consequence of the particular value of echo delay. The corresponding shift of the entire jitter psychometric function demonstrates that bats may perceive the phase of echoes relative to emissions from information encoded in the timing of neural discharges, as previously proposed from coincidence of the jitter psychometric function with the cross-correlation function of echoes.
Profile analysis by parallel distributed networks
Authors: Howard James H. , Harpster Jeffrey L. , Miller Maxwell H.
Published: October 2005

Several profile analysis (spectral shape discrimination) phenomena were simulated using simple, three-layer, feed-forward, parallel networks. As in previous studies with human listeners, pairs of multitonal sounds were used—one with all components at a uniform, background level and the other with a tonal signal added in-phase to the central component. Networks were trained to select the signal alternative in a 2AFC paradigm using the generalized delta rule with error propagation and were then tested under various conditions. Background levels were varied randomly either between or within trials. In the former case, the network learned to compare energy at the signal frequency across the two sounds (intensity discrimination), whereas, in the latter case, it learned to compare the distribution of spectral energy within sounds (profile analysis). These results mimic previous findings with human listeners reported by Green and his colleagues [D. M. Green, Profile Analysis (Oxford U. P., New York, 1988)]. In follow-up simulations, improved profile analysis have been demonstrated when the frequency range of tonal components is expanded and as the duration of the complex increases to 100 ms. [Work supported by ONR.]
Spectral pattern discrimination in children
Authors: Allen Prudence , Wightman Frederic
Published: October 2005

This experiment used a video-game-based adaptive forced-choice paradigm to assess the effect of noise on the ability of children to discriminate among stimuli with different spectral shape. The stimuli were synthesized vowels and consonants, and tonal complexes that had sinusoidally rippled spectra. Preschool children, school-aged children, and adults served as subjects. In order to encourage the subjects to base their discriminations on spectral shape rather than on local energy concentrations, the overall level of the stimuli was randomized over a 20-dB range for each presentation. In general, the results suggest that spectral pattern discrimination ability for all stimulus types improves with age. The age at which the children's performance reaches adult levels varies with stimulus type. Moreover, noise has a particularly deleterious effect on the spectral pattern discrimination ability of the younger subjects. [Work supported by DRF and the Waisman Center Rowan Trust.]
Sound sources of high-speed jet impingement
Authors: Powell Alan
Published: October 2005

For high-speed flows the source region is often not compact and the low-speed arguments are invalid: Then the motion of the flow boundary may provide insight as to the nature of the sound sources. (i) When a “two-dimensional” jet impinges on a large flat plate, the sinuous jet puffs from side to side at the plate, approximated by simple sources separated by the high impedance jet. (ii) The high-speed edge tone is similar. (iii) For small plates sound will be generated by the axially symmetric modulation of the strength of the unstable stand-off shock wave. Then, (a) changes in flow yield the total annular simple source strength or (b) the nearly conical boundary of the flow enveloping the plate appears to vibrate sympathetically providing an alternative source model. Some results of an alternate approach are given, in which the boundary is considered to be generating the evanescent waves, partially scattered at impingement or radiating at initiation.
Modeling of pressure-gradient effects on flow noise
Authors: Christoph George H.
Published: October 2005

A methodology to predict the effects of pressure gradients on flow noise is presented. The method was developed from the pressure-gradient data of Schloemer [J. Acoust. Soc. Am. 42, 93–113 (1966)] and of Burton [MIT Acoustics and Vibration Lab. Rep. 70208-9 (1973)] and implemented with the Corcos wall-pressure model. Predicted boundary-layer parameters yield a wall-pressure frequency spectrum for a particular flow condition, thus allowing one to predict conditions where data do not exist. For increasing pressure flows, rms wall pressures were predicted to be unaffected for Burton's test cases but increased for Schloemer's experiments. This apparent contradiction is in agreement with the experiments. Also, it is shown that erroneous conclusions result from a zero-pressure-gradient wall-pressure model. Computations of wavenumber spectra in adverse gradients gave higher spectral levels except in the vicinity of convective peaks where levels are noticeably lower, especially at the high frequencies. For Burton's test cases, frequency-spectral-density calculations showed increases at low frequencies and reductions at high frequencies, although not as dramatic as Burton's data indicated for the most severe adverse-pressure gradients. Wall-pressure levels were predicted to increase at all frequencies for Schloemer's data. The model clearly shows that these differences are due to the way that the adverse-pressure gradients are formed.
Computer simulation model useable for testing underwater communication techniques
Authors: Estes Lee E. , Fain Gilbert , Caron Paul R.
Published: October 2005

A model, and its software implementation, that is designed to simulate the transmitted and ocean-degraded received signal of an acoustic underwater transmission is presented. The approach taken for the model is the development of a stochastic impulse response characterized by its reverberation statistics and decay time and shape. A correlation time for the evolution of the nonstationary character of the impulse response is also defined. Signal transmission and reception is then modeled by processing signals through two statistically independent realizations of the impulse response. The two realizations are stationary during a correlation interval and are linearly summed after weighing with leap-frogged raised cosines. New realizations are generated for each correlation interval. Results appropriate to transmission in a shallow-water long horizontal channel are presented and compared with experimental results. [Work supported by the Massachusetts Centers of Excellence Corporation.]
A normal-mode approach to simulation of very low-frequency reverberation in an Arctic environment
Authors: Hayward Thomas J.
Published: October 2005

A normal-mode-based approach to simulation of very low-frequency underice reverberation in an Arctic environment is presented. Acoustic backscattering from the ice layer and the ocean bottom is simulated using a normal-mode-based model, with underice and bottom backscattering represented by coupling of forward- and backward-traveling waves associated with the normal modes. Results of simulations performed to test the feasibility of associated methods for estimating backscattering strengths from reverberation data are also presented.
A study of two standard speech intelligibility measures
Authors: Greenspan Steven L. , Bennett Raymond W. , Syrdal Ann K.
Published: October 2005

In the diagnostic rhyme test (DRT) participants identify each test word by chosing one of two response alternatives that differ only in their initial consonants and only by a single binary distinctive feature. Although the DRT is an accepted industry and military standard for measuring initial consonant intelligibility, other intelligibility measures may be more appropriate when stimuli produce near maximum DRT scores. Moreover, the DRT incorporates assumptions that may be valid for natural speech, but are untested for high-quality, low-bit rate coded speech. In particular, the DRT implicitly assumes that segmental intelligibility can be adequately measured by examining only single feature confusions. To examine this assumption, subjects were asked to identify the initial consonants of consonant-vowel syllables. Multifeature confusions were far more common with coded speech than with natural speech. Moreover, the consonant identification procedure reliably discriminated between speech coding devices that had near maximum, statistically indiscriminable DRT scores. These results suggest that open-response identification procedures may be more suitable than the DRT for evaluating high-quality, low-bit rate coders.
Improved excitation prediction and quantization in optimal amplitude multipulse coders
Authors: Lin Daniel
Published: October 2005

Improvements to the excitation prediction and quantization procedure in optimal amplitude multipulse coders are described. The new procedure successively reoptimizes the pulse amplitudes and predictor gain of the long-delay correlation filter as each new pulse is found. Furthermore, the amplitudes and gain are requantized at each step so that the new pulse amplitudes and locations can correct for the quantization errors in the existing excitation. To perform the joint amplitude-predictor reoptimization, nonrecursive pitch prediction structure is used in our analysis. The predictor is implemented as an adaptive vector quantizer whose codebook is populated with past excitation sequences. This structure allows for a computationally efficient closed-loop analysis procedure. The results showed that at 9.6 kbps, the new technique achieved an average segmental SNR of over 18.5 dB. Informal subjective tests indicated that the reconstructed speech is toll quality.
Review of the effects of noise on performance
Authors: Suter Alice H.
Published: October 2005

The effects of noise on job performance are not as easily discerned and predictable as other effects, such as those on hearing or speech communication. The extent to which noise affects performance depends on numerous nonacoustical factors, such as the subject's biological and psychological state, as well as certain external factors. Despite these and other difficulties involved in comparing research results, a recent review of the noise and performance literature provides sufficient evidence to indicate adverse effects under certain circumstances. The probability of performance decrements increases with increased (1) noise level, (2) intermittency, (3) aperiodicity, (4) lack of controllability. (5) task complexity, (6) task duration, and (7) the addition of certain other stressors. The review was sponsored by the U.S. Army Human Engineering Laboratory, and performed under the auspices of Gallaudet University.