bluuebunny/crossref_metadata_2025 huggingface dataset:
Forced vibrations of two welded plates: A prototype for assessment of the extent of applicability of SEA concepts
Authors: Dimitriadis Emilios K. , Pierce Allan D.
Published: October 2005

Statistical Energy Analysis (SEA) has as a fundamental tenet that the power flow between two vibrating components of a system is a weighted difference of the energies of the two components, where the coefficients are intrinsic properties of the system, independent of the excitation. A prototype that may give new insight into when and to what extent such a tenet is satisfied for strongly coupled continuous systems is explored here. Two rectangular plates, welded together at right angles along a common edge and simply supported at all other edges are caused to vibrate by random forces applied on one of the plates. The displacement field of each plate is expanded in modal eigenfunctions of the uncoupled plates following a procedure analogous to that previously used for strongly coupled beams by Davies and Wahab [J. Sound Vib. 77, 311–321 (1981)]. The formally exact solution for plate energy spectral density and power flow spectral density is given and is approximated by replacing modal sums by integrals. At higher frequencies and with averages over sufficiently wide frequency bands, the theory yields approximate results consistent with SEA concepts. [Work supported by Whirlpool Corporation.] S77_4 S77 S77 4 Forced vibrations of two welded plates: A prototype for assessment of the extent of applicability of SEA concepts Emilios K. Dimitriadis Allan D. Pierce School of Mechanical Engineering, Georgia Institute of Technology, Atlanta, GA 30332 Statistical Energy Analysis (SEA) has as a fundamental tenet that the power flow between two vibrating components of a system is a weighted difference of the energies of the two components, where the coefficients are intrinsic properties of the system, independent of the excitation. A prototype that may give new insight into when and to what extent such a tenet is satisfied for strongly coupled continuous systems is explored here. Two rectangular plates, welded together at right angles along a common edge and simply supported at all other edges are caused to vibrate by random forces applied on one of the plates. The displacement field of each plate is expanded in modal eigenfunctions of the uncoupled plates following a procedure analogous to that previously used for strongly coupled beams by Davies and Wahab [J. Sound Vib. 77, 311–321 (1981)]. The formally exact solution for plate energy spectral density and power flow spectral density is given and is approximated by replacing modal sums by integrals. At higher frequencies and with averages over sufficiently wide frequency bands, the theory yields approximate results consistent with SEA concepts. [Work supported by Whirlpool Corporation.]
Estimates of power coupling two beams
Authors: Smith P. W.
Published: October 2005

Davies and Wahab have recently reported extensive precise calculations of the average power coupling the two parts of a beam held by three simple supports, averages being taken over frequency bands and an interval of the ratio of lengths of the two parts [J. Sound Vib. 77, 311 (1981)]. Those results are discussed and compared with prior studies based on one or another statistical hypothesis, including Statistical Energy Analysis. Principal results are (1) for small loss factor η (η ≲ 1/Nπ, where N is average resonant mode number), the new and old results agree very closely; this is the tightly coupled regime where half the power input to each subsystem is dissipated in the other; (2) agreement is less good for larger η; in this regime reverberation is slight since round-trip attenuation is large; coupling power varies with the source location within each subsystem; (3) for the particular structure, where the wave energy transmission coefficient of the coupling is large (0.5), there is no regime that is both loosely coupled and highly reverberant.
On reducing pitch sensitivity of LPC parameters
Authors: Singhal Sharad , Atal Bishnu S.
Published: October 2005

Mean-squared prediction error criterion used in linear predictive coding of speech has a number of inherent shortcomings. For low pitch frequencies, the LPC analysis largely ignores the pitch-related fine structure of the speech spectrum. However, the sensitivity of the LPC parameters increases rapidly as a function of the pitch frequency. This sensitivity can be directly traced to the mean-squared prediction error criterion used in LPC analysis. Moreover, in the presence of noise and errors resulting from the assumption of an all-pole model, the minimization of prediction error leads to a perceptually suboptimal solution. In this paper, the LPC analysis is considered as a short-time spectral envelope matching problem. Using the LPC-derived parameters as initial values, a search procedure is used to refine the LPC parameter estimates. The new procedure minimizes a perceptual distance metric between the spectrum based on LPC parameters and the samples of the speech spectrum at spectral peaks.
Test of an automatic syllable peak detector
Authors: Fleck M. , Liberman M. Y.
Published: October 2005

It is a venerable and commonplace intuition that syllables are “peaks of sonority.” Previous work by Mermelstein [J. Acoust. Soc. Am. 58, 880–883 (1975)] suggests that this intuition is nearly true on the acoustic surface of speech, and that relatively crude algorithms based on amplitude variation can work fairly well at counting syllables and locating their approximate extent. We have devised a simple speaker-independent algorithm for detecting sonority peaks in continuous speech, and tested its accuracy as a syllable finder on several paragraphs of casual reading by four speakers. Performance was encouraging: less than 10% misses, and less than 1% false triggers. The misses were about evenly divided among (1) vowel-vowel transition, where no sonority minimum is expected, (2) intervocalic sonorants with insufficient amplitude dip, and (3) devoiced or deleted schwas. Methods for dealing with these cases will be discussed.
The dialogue terminal
Authors: Sorace Ron
Published: October 2005

The dialogue terminal is a direct application of research in speech recognition, synthesis, and coding at Bell Laboratories. It represents a rather novel approach to state-of-the-art interactions between man and machine and has potential for a variety of applications. Current speech coding techniques permit integration of voice and data for digital tranmission and, coupled with high density memory circuits, enable storage of large amounts of speech for playback, synthesis, or recognition. Speech storage and editing and rate modification are necessary for applications such as voice-mail. Speech synthesis capability such as formant or speech-from-text techniques are available and have sufficient quality to provide a degree of intelligibility which is adequate for output of data and issuing spoken prompts. The terminal is designed to perform isolated word recognition since this method represents the state-of-the-art. However, for the more experienced user the capability to parse a continuous sequence of commands will give the terminal the ability to adapt to the experience level of the user. The resulting mode of operation consists of synthetic voice prompts and voice commands in a dialogue format.
Analyzing tones with spectrum variations due to tremolo
Authors: Nolle A. W.
Published: October 2005

Wind instrument tremolo that is produced by periodic variation of the blowing pressure involves changes in the spectrum (the relative strengths of the partials), rather than simply amplitude changes. An automatic digital waveform analysis procedure to display these changes is described. The digitized record of total length 0.5 s or more is processed piecewise, typically in samples representing four periods. Fourier analysis of each sample proceeds after (a) the frequency is determined by autocorrelation and (b) a digital representation having 2″ points period (usually 32) is constructed by interpolation. Results for the Boehm flute are shown as an example. At the low end of the register, cyclic changes as great as 10:1 can occur in the normalized amplitudes of important partials.
Keyboard-controlled digital playback of an organ-pipe tone
Authors: Nolle A. W.
Published: October 2005

The tone sample in computer memory includes the attack, ten or more periods of “steady” state, and the decay. With well-known programming techniques and digital-analog equipment, it is simple to make the sample audible. This paper considers the problem of prolonging the tone as long as the key is down without unduly increasing memory requirements. Repetition of one cycle gives the unvarying wave form of a stable electronic generator. This may be acceptable for reed pipe tones, but for flue pipe tones the absence of the small spectral fluctuations of the original source is evident. The indefinite repetition of, say, ten periods of the original tone reintroduces fluctuations, but the regular flyback may be detectable unless the original is a nearly fixed waveform. A better solution is to execute a flyback of NF periods at the end of the “steady” sample, where the integer NF is a random number that ranges from unity to the full length of the “steady” sample. Taperecorded demonstration tones will be played. A tone with tremulant can be handled by using a constant NF.
Tuning and temperament in organ flue pipes
Authors: Mulder George W. , Brzezinski James
Published: October 2005

The scaling of ranks of organ flue pipes to produce a coherent, characteristic, and tonally balanced ensemble is a problem which has confronted organ builders for many centuries, but the temperament used to tune these pipes has likewise been of concern. With an analysis of the Fantasia Chromatica by Jan Pieterszoon Sweelinck, and its contrapuntal fabric, the proposition by Simon Stevin of “finding 11 mean proportional parts between 2 and 1, learned through the 45th proposition of my French arithmetic” serves as a solution for a modification in temperament of the organ pipes at the Oude Kerk in Amsterdam where it was performed by Sweelinck.
Spatial coherence of sound scattered from a wind-driven surface: Comparison between experiment and the facet-ensemble method
Authors: Clay C. S. , Kinney W. A.
Published: October 2005

In a recent paper [W. A. Kinney, C. S. Clay, and G. A. Sandness, J. Acoust. Soc. Am., in press], the authors showed agreement between acoustic intensity values measured in a water-tank scattering experiment and values predicted using the facet-ensemble method. Medwin and Clay [J. Acoust. Soc. Am. 47, 1419–1429 (1970)] performed a laboratory experiment in which they measured the spatial correlation of underwater acoustic signals scattered from a wind-driven surface (the rms height was 0.19 cm) as a function of the separation between two receivers (0 to 40 cm). A comparison is provided here between coherence values obtained in the experiment and values predicted using the facet-ensemble method. The coherence values are given for 60-, 100-, and 160-kHz signals scattered in the specular direction at a grazing angle of 35° (the propagation distance was 420 cm). The measured coherence was dependent upon both Doppler shift and the bandwidth of the correlator, and this dependence was included in the predictions. Agreement between the predicted and measured coherence values is good in general. [Work supported by NORDA.]
Low-frequency scattering of an impulse by a cylindrically corrugated surface at grazing incidence
Authors: Jouaillec Francois , Medwin Herman
Published: October 2005

Point source radiation at grazing incidence to a rough surface generates a slightly dispersive Biot boundary wave. When the surface is a series of adjacent circular half-cylinders (radius a), the boundary wave is frequency, range, and azimuth dependent. Its characteristics have been measured at low frequencies (Ka < 1) in an anechoic chamber model to a range of 2.2 m and from 2 to 20 kHz. The results agree generally well with the theory [I. Tolstoy, I. Acoust. Soc. Am. 72, 960–972 (1982)]. [Work supported by the Office of Naval Research.]
Dynamic elastic constant determination via measurement of the acoustic field scattered by a spherical elastomer
Authors: Piquette Jean C.
Published: October 2005

Dynamic elastic constants can be determined by solving the inverse scattering problem of an infinite plane acoustic wave incident upon a spherically shaped elastomer. A least-squares analysis is used to find the values of the elastic constants that are most consistent with measured scattered pressures. The approximate values of the elastic constants required to initiate the least-squares procedure can be calculated from expressions that have been developed which constitute approximate relationships between these elastic constants and the first few coefficients of the series expansion for the scattered pressure. An experimental investigation of this technique is being developed. [Work supported by NAVSEA 63R.]
A finite difference solution for the propagation of sound in a variable area duct
Authors: Hariharan S. I. , Lester H. C.
Published: October 2005

In this paper an explicit time/space finite difference procedure is used to simulate the propagation of sound in a quasi one-dimensional duct containing high Mach number subsonic flow. Nonlinear acoustic equations are derived by perturbing the time-dependent Euler equations about a steady compressible flow. The governing difference relations are based on a fourth-order, two-step (predictor-corrector) MacCormack scheme. Difference equations for the source and termination boundary conditions are derived from the appropriate characteristic relationship. The solution algorithm functions by switching on a time harmonic source and allowing the difference equations to iterate to a steady state. A significant advantage with this approach is that the nonlinear terms can be retained and evaluated without incurring additional computer cost over and above that required for a linear model calculation. The numerical procedure is applied to a Crocco-Tsien duct with a 2.0 area reduction ratio. Throat Mach numbers up to 0.95 are computed for source levels of 135 dB. Overall spatial rms distributions are computed by temporal averaging and SPL levels of the harmonics are computed by finite Fourier transforms. Linear results are validated by comparison with the results of a Galerkin/weighted-residual finite element program. Nonlinear results are compared with experimental data.
The free-field two-microphone method for the measurement of the acoustic impedance of the ground
Authors: Glaretas Constantine
Published: October 2005

A new method for measuring the acoustic impedance of the ground has been developed. The free-field two-microphone method, as it is called, consists of measuring the sound pressure field, radiating from a loud-speaker source and propagating parallel to the ground, by two microphones located on the same vertical axis. The transfer function between these two microphone signals is shown to be, for total coherence, equal to the ratio of velocity potentials, predicted by the appropriate point-to-point propagation theory over a plane boundary. A numerical search with a digital computer provides the value of ground impedance which satisfies the transfer function relationship stated above. Two ground models have been tested: a locally reacting and an extended reacting model. From the close agreement of local and extended reaction models in predicting ground impedance, it is concluded that the grass-covered surfaces tested here can be considered as locally reacting, without significant error. The free-field two-microphone method presents certain advantages over other impedance techniques. These advantages include a reduction in possible errors resulting from uncertainties in the exact location of ground surface, and the potential to explore the acoustic properties of the ground using different ground models.
Detection of a temporal gap as a function of level and frequency
Authors: Buus Søren , Florentine Mary
Published: October 2005

The minimum detectable gap, MDG, of an octave band of noise was measured as as function of frequency and level. Seven frequencies ranging from 0.25–14 kHz were tested. For each frequency, the level of the noise varied from about 10 dB SL to a spectrum level of 47 dB SPL. The gap was produced by turning off and on the signal with a rise-fall time of 1 ms. To eliminate the effect of spectral splatter, each octave band of noise was presented with its complementary band-stop masker. The masker was continuous and its spectrum level was equal to that of the signal. The MDG was measured with a modified BUDTIF procedure and a 2I, 2AFC paradigm. Results for three normal listeners show that the MDG decreases at all frequencies with increasing level up to a spectrum level of 30–40 dB SPL. For example, the MDGs at 4 kHz were about 9 ms at 3 dB SPL (spectrum level), about 6 ms at 33 dB, and about 7.5 ms at 53 dB. The effect of frequency is much more pronounced: the MDG at the best level decreases from about 40 ms at 250 Hz to about 5 ms at 4 kHz and above. These results indicate that the auditory filter may limit performance at frequencies below 4 kHz, whereas a different mechanism seems to dominate at frequencies above 4 kHz. [Work supported by NIH grant RR07143 and Northeastern University's Research and Scholarship Development Fund.]
Intensity discrimination at high frequencies in the presence of noise
Authors: Viemeister Neal F.
Published: October 2005

Previous experiments on intensity discrimination in the presence of band-reject noise indicate that recruitment of unsaturated fibers via spread-of-excitation along the cochlear partition is not necessary for the maintenance of the large dynamic range of human hearing. The present experiment is similar to that of Viemeister [J. Acoust. Soc. Am. 56, 1594–1600 (1974)] but extends the situation to frequencies for which an intensity code based upon temporal fine structure is presumably not possible. The primary result of this experiment is that Weber's Law holds over at least a 75-dB range for intensity discrimination of 200-ms, diotic bursts of band-pass noise (6–14 kHz) in the presence of a relatively intense, complementary band-reject noise. This indicates that neither large-scale spread-of-excitation nor temporal synchrony is necessary for the auditory system to maintain its dynamic range. [Supported by NIH NS 12125.]
Boundary absorption for noise control and thermal insulation
Authors: Holger David K.
Published: October 2005

Interior industrial sound level reductions of from 3–6 dBA can often be achieved by adding acoustically absorptive materials to the space involved. When absorption is added at the boundaries, typically the ceiling and/or walls are treated, the thermal insulation provided by the boundaries may also be enhanced. Design procedures for situations involving combined sound absorption and thermal insulation are discussed. A method for estimating sound level reduction, thermal energy conservation, and economic benefits of a combined approach is presented. The method is discussed in terms of a detailed case study.
Multiple pitch induced by frequency modulation of partials of complex tones
Authors: Espinoza-Varas B. , Bregman A. S.
Published: October 2005

The stimuli were complex tones consisting of the first 12 harmonics of 100 Hz, with duration of 1.5 s. The even-numbered harmonics of the complex had a linearly falling or linearly rising spectral envelope and were frequency modulated by either a sinusoid with 90° or 270° phase or by a linear ramp. Frequency and depth of modulation were 0.66 Hz (1.0 cycle/1.5 s) and 6%,respectively. The odd-numbered harmonics had a flat spectral envelope and were frequency modulated by a 0° phase sinusoid with frequencies of 0.3, 0.66, or 2.0 Hz, and depths of 3, 6, or 10%. In agreement with McAdams [J. Acoust. Soc. Am. Suppl. 1 68, S109 (1980)], we observed that when a different modulation function is impressed upon each subset of harmonics, the 12-component complex is perceived as having two pitch values, which seem to correspond to those of the individual subsets of harmonics when presented in isolation. An absolute identification experiment was conducted in which listeners were asked to identify the spectral envelope (timbre) and pattern of frequency modulation of the even-numbered harmonics of the 12-component complex. Accuracy of identification ranged from 40%–75% over the various stimulus conditions. In a second experiment, listeners were asked to judge the strength of the periodicity pitch of two-tone complexes (2500 and 3000 Hz, or 500 and 600 Hz) in which only the higher partial was frequency modulated by a linear ramp with frequency of 1.0 cycle per signal duration. Preliminary results suggest that the periodicity pitch of the two-tone complex becomes weaker as modulation depth increases. The relation between the perceptual data and the time and frequency domain characteristics of the signals will be discussed.
Comparison of microstructures in absolute threshold and in the pitch-intensity relation—a search for a place theory of pitch perception
Authors: Hartmann W. M. , Klein M. A.
Published: October 2005

We have made fine-grained measurements of absolute threshold and of the pitch-intensity relation for sine tones as functions of frequency. For some subjects the measured microstructures have small enough error and are sufficiently stable in time that one can hope to draw conclusions from the data. For some ears, in some frequency ranges, there is little threshold microstructure. In these cases the pitch-intensity relation tends to show a frequency-independent shift. More commonly, however, there is considerable threshold microstructure, as much as 10 dB over one semitone. In those cases there is inevitably appreciable microstructure in the pitch-intensity relation; the pitch shift may change sign six or seven times per octave. We have attempted to exploit the microstructures to test models of pitch perception for sine tones. We made standard assumptions about neural driven firing rate. We also supposed that absolute threshold indicates neural sensitivity. We found that our microstructure data are inconsistent with models in which pitch is derived from either the peak locus or the centroid of an average neural firing rate pattern. This conclusion is in agreement with a previous study of pitch shifts caused by a leading tone. [W. M. Hartmann, J. Acoust. Soc. Am. Suppl. 1 68, S109 (1980)]. [Work supported by the NSF and by the NIH.]
Listeners' identification of environmental sounds
Authors: Lass Norman J. , Eastham Sandra K. , Parrish William C. , Scherbick Kathleen A. , Ralph Dawn M.
Published: October 2005

The purpose of this investigation was to determine if listeners can accurately identify various environmental sounds. A total of 40 sounds were recorded: (a) 20 animal sounds from four cows, cats, dogs, pigs, and sheep; (b) 12 inanimate sounds from four typewriters, car horns, and telephones; (c) four musical sounds from a flute, piano, guitar, and drum; and (d) four human sounds (/pʌ) from two female and two male speakers. A master tape containing the randomly arranged recordings was prepared and presented to 30 judges who were asked to identify the source of each sound and the confidence of their judgments on a seven-point rating scale. Results indicate that listeners can accurately identify environmental sounds. However, their accuracy was not equal within each sound class as well as between all four classes of sounds investigated: human sounds yielded the highest accuracy, followed by musical and inanimate sounds, and animal sounds were least accurately identified. Moreover, listeners' judgmental confidence was similar to that of their accuracy. Implications of these findings and suggestions for future research are discussed.
Fiber optic gradient hydrophone
Authors: Mills G. B. , Garrett S. L. , Carome E. F.
Published: October 2005

A laboratory study has been made of the characteristics of an interferometric type optical fiber pressure gradient hydrophone. The optical system is configured as an all fiber Mach-Zehnder interferometer excited by a single mode gallium arsenide diode laser. A pair of identical fiber coils, one in each arm of the interferometer, forms the sensing portion of the gradient hydrophone. Each coil consists of 8 m of polyethylene jacketed, single mode fiber, wound in a doughnut shaped element of mean diameter 5 cm and thickness 3 mm. The acoustic sensitivity of each coil was determined separately and then the pair was aligned coaxially and separated by 10 cm to operate as a pressure gradient device. Details of the construction of the system, calibration procedure, and hydrophone sensitivity data are presented. [Work supported by ONR.]
Articulatory synthesis from underlying dynamics
Authors: Browman Catherine P. , Goldstein Louis , Kelso J. A. S. , Rubin Philip , Saltzman Elliot
Published: October 2005

We are testing a model of articulatory coordination and control over time using an articulatory synthesizer (ASY) that converts time-varying specifications of articulator positions into speed [Rubin, Baer, and Mermelstein J. Acoust. Soc. Am. 70, 321–328 (1981)]. These articulator specifications are now generated from an underlying dynamical model that defines a particular linguistic event. Coordination among individual articulators emerges from this model, as does the time course of the event. No point-by-point temporal control is required. Synthesis will allow perceptual evaluation of different underlying dynamical models. [Work supported by NIH and ONR.]
Electromyographic and kinematic measures of articulatory coordination in a deaf speaker
Authors: McGarr Nancy S. , Kobayashi Noriko , Honda Kiyoshi
Published: October 2005

Variability in production and disruption of interarticulator timing is reported for a deaf speaker. Acoustic, electromyographic, and movement records were obtained from a deaf subject and a hearing control, producing multiple repetitions of nonsense syllables (e.g., ə bib ɑ b) with stress on either V1 or V2. Electromyographic recordings were obtained from genioglossus, geniohyoid, and anterior belly of digastricus. Temporal patterns of lip closure were recorded using an electrical conductance technique. Infrared LEDs were attached to the jaw and monitored using a modified SELSPOT optical tracking system. Tokens were sorted into perceptually correct versus error patterns according to listener judgments, and formant frequency measures obtained. The deaf speaker differed from the normal control in several ways. She was unable to consistently execute changes in stress across different syllables and did not differentiate high versus low vowels in either genioglossus or relative jaw position. Timing of geniohyoid was variable and did not accompany stress changes. While activity of digastricus paralleled jaw movement, the duration of muscle activity continued well beyond maximum displacement of the jaw, perhaps contributing to laryngeal instability. [Work supported by NINCDS.]
Remote and autogenic articulatory adaptation to jaw perturbations during speech: More on functional synergies
Authors: Bateson Eric V. , Kelso J. A. S.
Published: October 2005

The present research is a continuation of a program devoted to understanding the cooperative behavior among neuromuscular elements during speech. In an earlier study, we found functionally specific electomyographic and kinematic evidence of compensation in remote articulators (upper lip and tongue) to perturbations of the jaw [J. A. S. Kelso, B. Tuller and C. A. Fowler, J. Acoust. Soc. Am. Suppl. 1 72, S103 (1982)]. Here we extend the earlier work to include additional subjects and phonetic contrasts and to test for phase dependent perturbation effects on the extent and style of compensation. We observe in this case that compensation in perturbed productions of /bæb/ and /bæp/ is primarily autogenic, that is local to the perturbed structure itself, the lower lip-jaw complex. We also find that contrasts not involving different articulators, e.g., between final /b/ and /p/, differ only in the degree to which compensation is achieved; the paterns are kinematically similar. The phase at which the jaw cycle is perturbed does not seem to affect either the position to which the system recovers following perturbation or the style of compensation. There are, however, phase dependent acoustic differences which suggest that compensation of a largely autogenic kind is not sufficient for complete compensation. [Work supported by NINCDS and ONR.]
Acoustic backscattering at low grazing angles from the ocean bottom. I. Bottom backscattering strength
Authors: Boehme H. , Chotiros N. P. , Rolleigh L. D. , Pitt S. P. , Garcia A. L. , Goldsberry T. G. , Lamb R. A.
Published: October 2005

Acoustic backscattering measurements on a sand bottom were made at grazing angles in the range of about 2°–10° in water depth of approximately 15.5 m near San Diego, CA [T. G. Goldsberry, S. P. Pitt, and R. A. Lamb, J. Acoust. Soc. Am. Suppl. 1 72, S74 (1982)]. Data from these measurements have been analyzed to determine the mean value and standard deviation of the bottom backscattering strength per m2 as a function of grazing angle, insonified area, transmit signal type, and frequency. A curved ray path propagation model and measured sound speed profiles were used to determine grazing angle versus time. The mean value followed Lambert's law for the range of grazing angles measured and for all frequencies used. No significant differences in mean value were observed when the insonified area and transmit signal type were varied. The observed frequency dependence of the bottom backscattering strength per m2 falls in the range from f1.5 to f1.8 for this relatively flat, sandy bottom. [Work supported by NAVSEA 63R and NORDA Code 113.]
Hearing aid signal processing for noise and nonsense syllables
Authors: Preves David A.
Published: October 2005

Transfer function magnitude and phase, coherence, cross power spectrum, and cross correlation measurements for vented hearing aid fittings worn in situ near onset of acoustic feedback oscillation were obtained with linear system identification techniques utilizing a FFT spectrum analyzer. These measurements were repeated for various methods of suppressing the oscillation tendency. Time segments containing formant transitions of selected CV and VC nonsense syllables found to be difficult to perceive by hearing-impaired persons were passed through the vented hearing aid fittings worn in situ. The ability of the hearing aids to process difficult-to-perceive speech stimuli is discussed in light of interpreting the cross correlation and cross spectrum measurements.
Visual speech display for the deaf
Authors: Zahorian Stephen A. , Holland James R.
Published: October 2005

In this paper, the results of the preliminary development of a visual speech display as a speech training aid for the hearing impaired will be presented. The training aid consists of analog electronics for extracting speech parameters and a microprocessor-based system for converting the speech parameters to color parameters and for controlling the display parameters. The speech parameters consist of six spectral shape factors, similar to speech spectral principal components and a pitch signal proportional to voice fundamental frequency. The spectral shape factors are computed by appropriately combining the low-pass-filtered logarithmically scaled outputs of a bank of 16 bandpass filters. The spectral shape factors will be used to control the color of the display, and the pitch signal will be used to control “texture” in the display. The results of a vowel discrimination experiment for steady-state vowels will be presented. [Work supported by the Whitaker Foundation.]
Categorical perception of vibrotactile signals
Authors: Collins M. J. , Hurtig R. R.
Published: October 2005

Hearing-impaired children's successful use of vibrotactile devices as a substitute for audition may be dependent on the wearability of the device and the transmission of phonemically recognizable information. The present study was carried out to determine the extent to which categorical perception of voiced-voiceless cognates could be achieved with a commercially available, wearable, single-channel, vibrotactile device. Stimuli were computer-generated pairs of syllables differing only in voice onset time (ka-ga). Adult subjects with normal hearing were trained to identify tactile signals at the extreme of the continuum prior to collection of discrimination data. Peaks in discrimination functions were observed to mark voiced-voiceless boundaries for both acoustic and tactile stimulation modes. These peaks occurred at consistently longer voice onset times for tactile stimulation than for auditory stimulation. The results indicate that although categorical perception of speech sounds can be learned in the tactile mode, because of the shift in the boundary, voiced-voiceless confusions may occur with voice onset times commonly observed in real speech. Thus for effective use of the tactile modality for speech, specialized signal processing may be required.
Acoustic characterization of changes in viscoelastic properties of epoxy during cure
Authors: Winfree William P. , Parker F. Raymond
Published: October 2005

The transformation of an epoxy during cure from a viscoelastic liquid to a viscoelastic solid is characterized by the change in the shear and longitudinal ultrasonic velocities and attenuations. The measurement of these ultrasonic properties is reported for several different resins cured at room temperature. The velocities are used to calculate the bulk and shear moduli as a function of cure time. The results indicate a parallel development during cure of the shear and bulk moduli. Measurements of the longitudinal attenuation at several different frequencies indicate the viscoelastic response can be characterized by a single relaxation time which increases as the cure progresses.
Ultrasonic flaw detection in rubber composites
Authors: Covey J. F. , Corsaro R. D.
Published: October 2005

Sonar dome rubber windows are constructed much like steel-belted automobile tires, in that they are composed predominantly of rubber reinforced with wire cords. The use of convention ultrasonic flaw detection techniques to locate voids and disbonds in such a media is complicated both by the high attenuation in the rubber media and the layered construction technique used in its fabrication. This paper describes laboratory research efforts leading to the development of a simple reliable portable ultrasonic flaw detection system designed specifically for use in this complicated media. The potential benefits of using high-speed digital signal processing equipment in this application is also demonstrated and discussed.
Refractive effects on sound propagation in a nonturbulent atmosphere
Authors: Raspet Richard
Published: October 2005

Atmospheric refraction due to temperature and wind variations with altitude can produce large changes in sound level from day to day and during the course of a day. Scientific studies performed during the day in low wind conditions may not be representative of average conditions. This paper will review empirical and theoretical studies of the effects of meteorological variables on sound propagation. The results and applicability of empirical studies will be discussed with the aim of identifying data sets useful for estimating refractive effects. Theoretical solutions for specific and general sound velocity profiles will also be covered. Examples from the literature will be used to illustrate the concepts.
On the phonetics of rhythm: Evidence from Swedish
Authors: Bruce Gösta
Published: October 2005

Rhythmic alternation among sequences of unstressed syllables in Swedish utterances was examined from phonetic starting points. The results of a series of pilot experiments using both reiterant nonsense speech and original sense speech can be summarized as follows. The rhythmic alternation among unstressed syllables is reflected in the temporal domain as an alternation in syllable duration. This alternation between longer and shorter syllables can be seen as a reflex of the postulated alternation between strong and weak syllables forming rhythmic subgroups within stress groups. It appears to be quite systematic, although the size of this difference is considerably less than for a corresponding comparison of stressed versus unstressed syllables. The temporal relations also seem to be in accord with the hypothesis that the alternation is governed from the upcoming stress, so that the prestress syllable is relatively shorter, the preceding one is relatively longer, and so on by way of alternation.
Rhythmic processes in Spanish
Authors: de Manrique Ana M. B. , Massone M. I.
Published: October 2005

In a recent paper [A. M. B. de Manrique and A. M. Signorini, J. Phonet. 11, 2 (1993)], experimental evidence was obtained about the nature of Spanish rhythm. The measurement of vowel, consonant, and syllable duration showed that segments vary according to several factors. The measurement of interstress intervals showed some regularity in the occurrence of stress. From these results it was concluded that Spanish has a tendency towards stress-timed rhythm. The present work was undertaken in order to examine the processes by which regular intervals are achieved. The material consisted of two sets of three sentences. The sentences differed from each other in one word so as to increase the number of syllables of one interval, the other intervals having a fixed number. Results showed shortening and lengthening processes by which intervals, both fixed and variable, adjust the duration of their segments in accommodation to each other. It was also observed that a secondary stress becomes stronger in some variable intervals as another mechanism to attain regularity.
Acoustical dilemmas for modern violin makers
Authors: Bellamy Lance G.
Published: October 2005

This paper will discuss the dilemmas of a maker of violins with regard to three important factors. The first is the selection of wood, where density must be balanced against the resonant frequency required for a piece of a given size. The second is the choice of the degree of arching with regard to the main resonant frequencies for the back and the top. The third is the graduation (thickness as a function of position on the surface). This situation will be compared to the choices available to violin makers in (Golden Period, 1650–1750. There will be a demonstration of period and modern violins.
A new normalization algorithm for detection systems
Authors: Struzinski William A.
Published: October 2005

Normalization is a critical function in detection systems. Without proper normalization, the detection algorithm will suffer a severe performance degradation. To date, several normalization schemes have been developed. These include the Two Pass Mean, the Split Three Pass Mean, the Modified Median, and the Order Truncate Average normalizers. However, all these normalizers have a deficiency that will cause a performance degradation in detection systems (susceptibility to strong signals, excessive execution time, sensitivity to the noise probability distribution, etc.). In an effort to minimize the performance degradation, a new normalization scheme has been developed. The new approach, Split Average Exclude Average, was developed by combining the desirable features of the Two Pass Mean and the Order Truncate Average schemes. The new normalization scheme is examined in this paper, and the potential performance of this approach is demonstrated. [Work supported by NAVSEA.]
Ahead look sonar performance prediction via FFT
Authors: Matthews A. D.
Published: October 2005

It is desired to predict the probabilities of detection and false alarm for an ahead look sonar providing N echoes from a cylindrical target. The signals are detected noncoherently and the targets are assumed to be bounded by, but not restricted to, the four Swerling cases. The clutter is assumed to be Rayleigh distributed. A computer algorithm is developed whose inputs are the number of echoes, the number of degrees of freedom, the multiple correlation of the echoes, and the average single-ping signal-to-noise ratio. The outputs will be graphic representations of clutter and target probability density functions and a log-log representation of their corresponding false alarm and detection probability distributions. The process is accomplished by calculating the characteristic function reversing and conjugating, and performing an FFT to obtain an estimate of the probability density functions.
Loudness discrimination in the sound field of a room
Authors: Lindevald I. M. , Benade A. H.
Published: October 2005

Subjects moving freely in the reverberant field of a room (T≃1 s), were presented with alternating pairs of 1-s soft-switched signals from loudspeakers differing in input level by |z|<6 dB. The fraction of subjects perceiving pairs as equally loud were fitted to the curve P(z) = A/[1 + (z/J)C].A is the population fraction expected to perceive equal-level sources as equally loud, J is the 50% jnd, while δ(=4J/C) measures the width of the transition region. Similar experiments using headphones give very small values of J(≃0.5 dB) and δ(≃0.1 dB). This precision for signals specified at the ear predicts chance performance in the Rayleigh-distributed sound field typical of rooms. The experiment was done with 400-Hz signals. (1) sinusoid: A = 0.44, J = 3.57 dB, δ = 3.45 dB: (2) bandlimited tone with first five harmonics strong; A = 0.58, J = 2.79 dB, δ = 2.91 dB: (3) tone with five equal amplitude harmonics; A = 0.60, J = 2.97 dB, δ = 3.45 dB: and (4) a related “tone” having five nearly harmonic equal-amplitude sinusoids; A = 0.61, J = 2.87 dB, δ = 4.10 dB. The observed (A, J, δ) sets lie between the values obtained in headphone experiments and those predicted via the Rayleigh distribution. Increasing the number of components reduces J, as it does the SD of the overall physical signal in the room. The values of A are nearly equal for all room signals except (1). The transition region δ is smallest for signal (2) which is most similar to musical tones and largest for the least musical signal (4).
Reported outdoor speech interference and event LAX
Authors: Hall Fred L. , Birnie Susan , Taylor S. Martin
Published: October 2005

One plausible interpretation of annoyance at noise is that the annoyance arises primarily because of speech and sleep interference caused by the noise. If this is true, then it is sensible to look for relationships between activity interference and indicators of the noise from specific noisy events. Such an analysis could provide the first step toward a better understanding of the relationship between noise and annoyance. This paper describes the results of such an analysis for reported speech interference due to aircraft noise in residential areas. The event noise levels are derived from data collected during the summer of 1983 by Transport Canada's automatic monitoring system at ten locations around Toronto International Airport. For each location, this data set provided the maximum LAX and average LAX by aircraft type. The analysis relates the outdoor speech interference reported by respondents at each site to the worst case and average LAX for aircraft operations at their location, by means of a nonlinear regression. The results are compared with a similar analysis for road traffic data, to see if the same equation can serve for both types of noise source.
Impulse noise: Comparison of dose calculated by 5-dB rule and 3-dB rule
Authors: Erdreich John
Published: October 2005

In the past several years, there have been many proposals concerning incorporation of impulse noise into total worker exposure. This is a problem of particular concern in the United States as a consequence of the 5-dB trading rule mandated by regulation. The alternative approaches to integrating impulse noise are to use a direct integration with a 5-dB trading rule, to use a 3-dB rule SLOW integration according to ISO 1999 and the proposed ANSI standard for high level signals above 120 dBA or to use a 5-dB SLOW integration as performed by the ANSI Standard (S1.25) personal noise dosimeter. Simulation of each of these approaches with industrial noise data was performed. The effect of integrating high level transients with a 3-dB rule (p2) integration compared with a 5-dB rule (p1.2) integration is that the calculated dose is 3 to 15 times greater than the 5-dB rule dose. We would infer from this that substantially greater numbers of workers would be protected (or more sites would be in noncompliance) by using a 3-dB rule for impulse noise. Because of this, it is critical that a rigorous scientific basis be established for impulse noise dose prior to defining procedures for such dose determination.
Application of a nonlinear layered model to hyperthermia
Authors: Haran Michael E. , Lewin Peter A.
Published: October 2005

The use of ultrasound as a heating modality in cancer therapy gives rise to the need for delivering prescribed exposures of high-intensity ultrasound to deep tissue volumes. It is known that the attenuation coefficient of nonlinear media depends upon the local acoustic pressure amplitude for a given fundamental frequency. It is also known that such anomalous effects occur at biomedical frequencies and intensities [e.g., E. L. Carstensen et al., Acustica 51, 116–123 (1982) and F. Dunn et al., Br. J. Cancer 45, 55–58 (1982)]. In this presentation a series algorithm [M. E. Haran and B. D. Cook, J. Acoust. Soc. Am. 73, 774–779 (1983)] is applied to study the propagation of plane finite amplitude waves through multiple layered media. It is shown that in some cases harmonic distortion and the resulting anomalous attenuation are sufficient to reduce the heat generation to a large degree. It is hoped that this elementary model will aid in the development of more effective acoustic sources and exposure protocols.
The role of hyperthermia and cavitation in production of hind limb paralysis in ultrasonically irradiated mouse neonates
Authors: Borelli M. J. , Frizzell L. A. , Dunn F.
Published: October 2005

The third lumbar region of the neonatal mouse spinal cord was irradiated with 1-MHz ultrasound using intensities between 46 and 289 W/cm2 at hydrostatic pressures of 1 and 16 atm. It was previously reported that the threshold exposure duration necessary for production of hind limb paralysis of the neonates increased with hydrostatic pressure at 289 W/cm2 but not at the lower intensities, although subharmonic and harmonic signals were observed at all intensities [Frizzell et al., J. Acoust. Soc. Am. 74, 1062–1065 (1983)], suggesting that acoustic cavitation is involved at 289 W/cm2 Examination with electron and light microscopy reveals that permanent paralysis is always accompanied by altered spinal cord morphology and that the minimum tissue damage is altered synaptic morphology similar to that observed in the cat brain [Borrelli et al., J. Acoust. Soc. Am. 69, 1514–1516 (1981)]. Tissue damage was greatest at the perimeter of the spinal cord, possibly as a result of higher temperatures resulting from heat conduction from the more highly absorbing spinal column. This suggests that hyperthermia plays a role in development of tissue damage even when another mechanism is involved. [This work was supported in part by grants from the NSF and the NIH.]