bluuebunny/crossref_metadata_2025 huggingface dataset:
Resonance reflections from a stratified ocean bottom
Authors: Jackins P. D. , Gaunaurd G. C. , Arvelo J.
Published: October 2005

The echoes reflected by a stratified ocean bottom that is insonified by acoustic plane waves emerging from a distant source are studied. The various strata in the bottom are modeled as elastic layers bonded together and allowing longitudinal (acoustic) and transverse (shear) waves to penetrate and be propagated through, and reflected from, the overall configuration. There can be as many layers as one wishes, all of arbitrary compositions. The bottom layer is a half-space of infinite thickness upon which all other layers rest. The top layer is the fluid layer that simulates the ocean and contains the source. The reflections versus frequency at fixed angles of incidence, or versus angle of incidence at fixed frequencies are studied. Numerous resonances are present in these returns and emerge in the analysis. This investigation has the eventual, ultimate, goal of determining details about the bottom composition and stratification, from the remotely sensed reflections. Of particular interest is a five-layer ease in which the consolidation and rigidity of the sediments in the various layers increases as one penetrates deeper into the bottom.
Holographic interferometry studies of the Tambura
Authors: Janakiram V. L.
Published: October 2005

The drone instrument called the Tambura is used in essentially all the classical music concerts in both North and South India. The Tambura has four strings tuned to the notes PSSS, corresponding to G1CCC1 of the western system. The length of the Tambura of standard size is usually between 140 and 160 cm. The resonator is of semiglobular shape. The top plank of the resonator is slightly convex and has small holes forming two circles on either side of the bridge. Traditionally Jackwood and Gourds are used for the manufacture of South and North Indian Tamburas, respectively. Under certain conditions, it is found necessary to utilize a suitable substitute wood, such as Silver Oak. Holographic interferometry has been used to examine the modes of vibration of the Tambura's resonator.
Progress with the ISO and ANSI Standards for the determination of sound power levels of noise sources using sound intensity measurements
Authors: Crocker Malcolm J.
Published: October 2005

Intensity measurements offer the possibility of accurate determination of the in situ sound power of machinery noise sources in the presence of high levels of extraneous noise. Meetings of working group ISO/TC43/SCI/WG25 began in 1982. Early meetings were devoted to discussing various measurement difficulties with a draft Standard produced in late 1985. Meetings of working group ANSI S12-21 began in 1983 and produced the first draft of a standard in 1984. It is planned to submit a draft standard to ANSI for trial and study. Although there are many similarities between the ISO and ANSI drafts, there are presently some differences. The ISO draft allows both precision and engineering grade measurements and uses a number of indicators to determine the grade achieved. The ANSI draft presently only allows engineering measurements. The ISO draft concentrates on fixed point measurements on a surface, while the ANSI draft also includes the possibility of scanning.
Localization of coherent and incoherent sound sources via an optimization process
Authors: Sidki M. , Nicolas J.
Published: October 2005

The problem of the localization and characterization of sound sources is tackled, for the case of a set of monopole sources, starting with the analytic expressions describing the field in combination with a gradient method for the optimization process [M. Box et al., Nonlinear Optimization Techniques (Olivier & Boyd, London)]. To locate the sources three measurement variables are used: the 3D vectorial active intensity, the normal active intensity, and the quadratic mean pressure. Simulations for three to five emitting sources, aimed at predicting their positions and spectral densities are made. The minimum number of measurement positions that lead to convergence of the algorithm is investigated. The agreement between prediction and reality is good, even when the sources are coherent. In parallel, the impact of the choice of measurement variable on the precision of the results is studied. In the very near field of the source, it is found that identification via 3D intensimetry requires a significantly lower number of measurement positions than does that via normal intensity or quadratic mean pressure.
Using nearfield acoustical holography to analyze a source excited by multiple frequencies
Authors: Strong William Y.
Published: October 2005

Nearfield acoustical holography (NAH) is a technique that can be used to analyze the radiated sound field located to one side of a source by measuring the pressure amplitude and phase over a surface. Typically a sinusoidal source excitation is used. Additional information could be obtained from a data set if the source was excited at multiple frequencies. These frequencies were each analyzed as if they were obtained from a single frequency measurement. The frequencies would need to be chosen carefully to reduce the influence of one frequency on another caused by spectral leakage when performing a discrete Fourier transform (DFT) on the time data. In the limit, a maximum of N/2 analysis frequencies could be measured where N is the number of points in the DFT, if those frequencies were perfectly aligned with the center frequencies of the DFT frequency bins. This presentation will discuss the use of an arbitrary function generator to supply the multiple frequency signal required and will present the results of doing an NAH analysis of a rectangular plate excited by this signal.
The use of random noise for on-line transducer modeling in an adaptive active attenuation system
Authors: Eriksson L. J. , Allie M. C.
Published: October 2005

Active sound attenuation systems may be described using a system identification framework in which an adaptive filter is used to model the performance of an unknown acoustical plant. An error signal may be obtained from a location following an acoustical summing junction where the undesired noise is combined with the output of a secondary sound source. In order for the model output to properly converge to a value that will minimize the error signal, it is frequently necessary to determine the transfer function of the secondary sound source and the path to the error signal measurement. Since these transfer functions are continuously changing in a real system, it is desirable to perform continuous on-line modeling of the output transducer and error path. In this paper, the use of an auxiliary random noise generator for this modeling is described. Based on a Galois sequence, this technique is easy to implement, provides continuous on-line modeling, and has minimal effect on the final value of the error signal.
Phased array active noise controller for ducts
Authors: Richarz Werner G.
Published: October 2005

The phased array active noise controller offers a cost-effective solution to noise control in ducts at frequencies below “cutoff.” The system consists of an array of loudspeakers phased to generate a downstream propagating wave that is an inverted replicate of an incident sound wave. The signal to be controlled is sensed by a microphone or microphones upstream of the array. Each element of the array provides a fraction of the overall cancellation signal, a feature making the system a robust one. Furthermore, detrimental upstream signals arrive out of phase, and the destabilizing positive feedback paths are virtually eliminated. There are, however, installation effects, a consequence of the relative positions of the transducers in the duct. It can be shown that the bandwidth of operation as well as the maximum attenuation is governed by the relative loudspeaker-microphone separations. Analysis provides some insight and “optimum” relative positions can be found. The other installation effect is due to the radiation impedance seen by the array. Measurements of relevant impedances are presented. The overall system performance is documented for the case of a rectangular duct with and without air flow. [Work supported by the Natural Sciences and Engineering Research Council of Canada.]
A blowdown water tunnel for investigation of tones from multihole plates
Authors: Elder S. A.
Published: October 2005

A low-noise “blowdown” water tunnel has been constructed for investigation of tones generated by multihole plates. Multihole orifices are mounted in the wall of a compact resonator chamber, simulating standing-wave conditions. The test chamber, immersed in a 260-gal open tank, is supplied by 4-in. piping connected to a 55-gal reservoir mounted on a 16-ft tower. The system is able to supply up to 80 kg/s of mass flow at a speed of 10 m/s. Water level in the test tank is kept level by means of a weir near the top, which empties into a collection barrel. Flow rate is monitored by weighing the barrel over a measured time. Results of preliminary tests will be described. [Work supported by Naval Sea Systems Command General Hydromechanics Research Program, administered by the David W. Taylor Naval Ship R & D Center, Bethesda, MD.]
Utility of wind direction for predicting worst-case firing noise levels
Authors: Luz George A. , Bensch David H. , Bosle Klaus , Stokes Carl-Heinz
Published: October 2005

Several German procedures for assessing the environmental noise from firing ranges have been based on the assumption that worst-case noise levels are found when the surface wind is blowing from the firing range toward the impacted community. Although there is some evidence that this assumption works for small arms ranges (e.g., 7.62-mm rifle), its utility at ranges firing larger weapons was unclear. Tests conducted with a mid-size weapon (25-mm gun) over a period of 30 days showed that the surface wind direction was not as useful a predictor as the direction of wind at 500 m above ground. With measurement sites located between 1500 and 6000 m from the firing point, the 500-m wind proved to be a fairly reliable indicator of worst case.
Effect of laser pulse energy on optoacoustic conversion in liquids
Authors: Bass Henry E. , Cheyne Stanley A.
Published: October 2005

The effect of laser pulse energy on optoacoustic conversion has been studied experimentally. A short UV laser pulse (337 nm in wavelength and 800 ps in duration) is allowed to impinge on a test cell of propanol where it is absorbed in a short distance (∼ 0.87 ram). The absorbed energy results in a density variation in the fluid which propagates outward from the excitation zone. A second, 10-μm-diam cw He-Ne probe laser beam parallel to the excitation pulse is deflected by the passing wave. The deflection is measured with a photodiode [B. Sullivan and A. C. Tam, “Profile of laser-produced acoustic pulse in a liquid,” J. Acoust. Soc. Am. 75, 437–441 (1984)]. The time evolution as well as the amplitude of the measured signal is considered. The amplitude of the optoacoustic signal is found to vary linearly with input energy between 7.5 × 1016 W/m3 and 5.3 × 1015 W/m3. Theoretical limits on the observed linearity will be presented. [Work supported by the ONR.]
Reflection characteristics for an immersed transversely isotropic layer
Authors: Pilarski Aleksander , Rose Joseph L.
Published: October 2005

In order to obtain more information about a composite material with respect to material characterization and defect analysis, utilizing oblique ultrasonic beams or guided waves inside the structure is proposed. Measurements of ultrasonic reflectivity from composite plates immersed in a liquid have been carried out. Initial experiments have been performed on a undirectional graphite-epoxy composite. To choose the appropriate testing parameters (transducer frequency, angle of incidence, etc.) and to provide a quantitative analysis of the obtained results, model calculations were carried out for a transversely isotropic fluid-loaded layer using Thompson's recurrence formulas connecting the wave amplitudes in neighboring media. Employing long wavelengths, it was assumed that the layer was homogeneous with five effective, independent elastic coefficients. The calculations were carried out for different values of volume fractions of the voids and fibers. These data have been compared to experimental results. The experimental measurements have been performed using frequency spectrum analysis of a pulse which has passed through a composite layer in the fiber direction. Experimental and theoretical results are encouraging. [Work supported by NRL.]
Split mode excitation of traveling waves in ring resonators
Authors: Ceperley Peter H.
Published: October 2005

Traveling wave excitation in ring resonators has application to surface wave motors [H. Yamamoto et al., U.S. Patent 4,504,760 (1985)] and thermoacoustic heat engines [P. Ceperley, J. Acoust. Soc. Am. 77, 1239–1244 (1985)] as well as in other potential areas. The standard method of exciting traveling waves in ring resonators involves two wave sources spaced a quarter-wavelength apart and driven with a 90° phase difference. This paper presents a method of introducing a perturbation into the ring resonator that allows excitation of traveling waves with a single exciting source. This method also has application to similar rotating mode excitation in simply connected resonators that approximate objects of revolution.
Leaky Rayleigh wave reflection from anisotropic wedges
Authors: Billy Michel de , Nagy Peter B. , Quentin Gerard , Adler Laszlo
Published: October 2005

In order to study the interaction of surface waves at anisotropic boundaries, a single crystal wedge was used in an immersion broadband experiment. A new method was introduced which separates the reflection coefficient due to anisotropy from geometrical effects. According to this approach, the reflection coefficients of the surface wave due to the geometry of the sample are the same from both directions, while the anisotropic component of the reflection coefficients are of the same modulus but opposite in phase. The leaky Rayleigh wave attenuation coefficient's frequency dependence was also measured, and accounted for the reflection coefficient. [Work partially supported by DOE and NATO.]
An experimental study of the acoustical impedance and fluid mechanics of two orifices in series
Authors: Elger Donald F. , Adams Ronald L.
Published: October 2005

Certain ink-jet printer nozzles consist of two orifices in series. Since the pressure impinging on the nozzle is transient, a study of nozzle impedance and fluid mechanics was motivated. Air was used, and the orifices were fabricated in equal diameter plates attached to a piston driven cylinder. The space between the plates, and the outer orifice, were open to ambient. The independent variables studied were: 0.23⩽d2/d1⩽0.94; 0.4⩽t/d1⩽2.6; and 660⩽P′/ρων⩽4120, where d1 and d2 are the inner and outer orifice diameters, ω is frequency, P′ is the acoustic pressure magnitude, and t is plate spacing. Hot-wire anemometry and a standing wave resonance method were used to obtain nozzle impedance and air velocity measurements. The impedance data showed the nonlinear behavior characteristic of a single orifice. The velocity data showed that t/d1 has the largest effect on the magnitude and nonlinearity of the fluid velocity at the outer orifice.
A transfer function approach to transient propagation in media with a quadratic frequency dependence of attenuation coefficient
Authors: Guyomar Daniel , Powers John
Published: October 2005

The propagation of short acoustical pulses in lossy media is presented as a time-varying transfer function (or spatial filter) that acts on the spatial spectrum of the source excitation. The source is a planar source of arbitrary (but separable) spatial and temporal dependence mounted in a rigid baffle. The homogeneous medium obeys Stokes' equation, resulting in a quadratic dependence of the plane-wave attenuation coefficient. Using distribution theory, an integral expression for the field is derived in terms of the normal velocity of the source, the Green's function, and its normal derivative on the boundary. This expression leads to an approximate solution for a point source in the source plane, which in turn produces a spatial transform that is the propagation transfer function. This transfer function is simply related to the transfer function of lossless propagation. Computer simulations of fields as well as a discussion of some casuality problems introduced by the use of the Stokes' equation are presented.
Dynamic elastic moduli of several thermoplastic elastomers for use in acoustic systems
Authors: Bruno Anthony B. , Kasper Rolf G.
Published: October 2005

A number of dynamic mechanical measurements have been performed on the Metravib-04 viscoanalyzer for a variety of thermoplastics. The material was a thermoplastic elastomer composed of styrene ethylene butylene styrene (SEBS) as a copolymer base, plasticized with a parfinic oil, and blended with a high-density polyethylene plus a stabilizer. The tested materials differed by the amount of high-density polyethylene blended into each sample. Each material was tested in tension compression and in shear for the frequency range 5 Hz–1 kHz at room temperature. The data yielded independent estimates of complex elastic moduli, e.g., Young's modulus and shear modulus as a function of frequency. The data showed good correlation between elastic moduli and percentage of polyethylene in each sample. However, no corresponding correlation with damping was observed. The dilatational and shear wave speeds and the Poisson's ratio were also computed as a function of frequency for each sample. These data were used to evaluate the performance of various acoustic systems. A number of examples will be discussed.
Annoyance of moderate level of noise spectra with predominating pure tones
Authors: Bienvenue Gordon R. , Wood Brian , Glass Susan R. , Celmer Robert D.
Published: October 2005

Annoyance due to noise has been a concern to researchers for over 30 years. Recently, concern has centered upon the annoyance engendered by noise spectra with predominating pure tones since the effect of tones on annoyance is marked and because modern business equipment often demonstrates this type of spectrum. Though most business equipment operates at sound levels at or below 65 dB SPL, much of the annoyance research has been directed towards aircraft noise and has been conducted with noise spectra exceeding 80 dB SPL. The present research centers on the overall levels of primary concern in evaluating business equipment annoyance while maintaining a link to the established (high level) data base. Twenty-five subjects were presented with 1200 tape recorded sounds of varying tone-noise ensembles including some business machinery sounds. The subjects rated these sounds for their levels of loudness, tonal prominence, and annoyance. Results of this procedure are discussed with implications for noise control. Of particular interest was the effect of modifying the tonal components of real equipment spectra in both amplitude and frequency using a digital signal processing system. [Work supported by IBM Corporation.]
Additivity of induced loudness adaptation
Authors: Scharf Bertram , Canévet Georges , Buus Søren
Published: October 2005

The loudness of a mid-level tone can be made to go down in at least two ways: (1) by intermittently incrementing the tone, (2) by presenting an intermittent tone to the contralateral ear. Magnitude estimates given during the intervals when the ipsilateral increment or contralateral tone is absent have shown that loudness decreases approximately 50% in 2–3 min. The present experiments investigated the effect of combining the ipsilateral and contralateral intermittencies. Observers judged the loudness of a monaural steady tone (0.25, 0.5, 1, or 4 kHz at 50 dB SPL) presented: (a) alone, (b) with a 20-dB increment every 20 s for 10 s, (c) with a 50-dB tone to the contralateral ear every 20 s for 10 s, (d) under both (b) and (c). Under (a), loudness changed little over 2–3 min. Under (b), loudness decreased about 60%. Under (c), loudness decreased by 50% at 4 kHz, but less at lower frequencies. (This frequency effect occurs because contralateral intermittency appears to induce adaptation in fewer subjects at low than at high frequencies.) Under (d), loudness decreased less than predicted for independent summation of ipsilaterally and contralaterally induced loudness adaptation. [Work supported by NIH-NINCDS grant ROI-NS07270].
Pure-tone sensitivity of human infants
Authors: Olsho Lynne Werner , Koch Elizabeth G. , Carter Elizabeth A. , Halpin Christopher F. , Spetner Nancy B.
Published: October 2005

The ability of 3-, 6-, and 12-month-old infants to detect pure tones in quiet was tested at frequencies ranging from 250–8000 Hz. Stimuli were presented monaurally via headphone. Signal trials consisted of ten repetitions of a 500-ms tone burst, with 10-ms rise-fall time and 500 ms between bursts; no-signal trials were 10-s intervals of quiet. The infant's response to a tone was judged by an observer, who, blind to trial type, decided whether or not a tone had been presented on each trial, based on the infant's behavior. Comparison of infant thresholds, determined using an adaptive rule, to those of adults tested under similar conditions showed a progressive improvement in threshold from 3–12 months. Three-month-olds' threshold were relatively poorer at 250 and at 8000 Hz compared to adults. The 6- and 12-month-olds' thresholds were somewhat closer to those of adults at 4000 and at 8000 Hz than at lower frequencies. Maturation of absolute sensitivity, of frequency resolution, and of nonsensory processing may all contribute to these age-related changes. [Work supported by NIH.]
A comparison of two-, three-, and four-alternative forced-choice staircase procedures
Authors: Schlauch R. S. , Rose R. M.
Published: October 2005

Threshold estimates from multiple-interval forced-choice staircase procedures were studied using computer simulations. Two frequently cited psychometric function shapes, those for the signal known exactly and for high uncertainty, governed the hypothetical subject's responses in the simulations. Threshold estimates based on 100 trials were obtained for both psychometric functions, for three different stimulus change values, and for decision rules that estimated 70.7% and 79.4% correct levels. A total of 1000 replications in each condition were run. Each threshold estimate was calculated by averaging the stimulus levels at which a reversal in stimulus direction change occurred. For many of the simulated conditions, three-alternative forced choice was the most efficient procedure (i.e., the variability of the estimates per unit of collection time was smallest). The validity of the simulations was tested using human performance data. [Work supported by NINCDS and IBM.]
Basic sensitivity and context memory on two-consonant continua
Authors: Goldberg Rina , Macmillan Nell A. , Braida Louis D.
Published: October 2005

According to the Perceptual Anchor Model of Context Coding [Braida et al., J. Acoust. Soc. Am. 76, 722–731 (1984)], resolution tasks differ in memory coding requirements: basic sensory limitations are measured in fixed discrimination, while context memory limitations can be inferred by comparing fixed discrimination with identification. The theory was applied to two synthetic consonant continua, one varying in voicing (from /ba/ to /pa/), the other in place of articulation (/ba/-/da/-/ga/). For both continua, regions of high basic sensitivity were found between the best phonemic exemplars, while optimal context coding occurred near the exemplars. Neither result occurs for an /i/-/ɪ/-/ε/ vowel continuum [Goldberg et al., J. Acoust. Soc. Am. Suppl. 1 77, S7 (1985)]. [Work supported by NSF.]
The influence of spectral and temporal properties on stop-consonant perception for two-formant stimuli
Authors: Ohde Ralph N.
Published: October 2005

The results from a previous study on stop-consonant perception for two-formant stimuli showed that the most reliable responses were for those stimuli with substantial second-formant (F2) transitions, and that there were few consistent velar responses [R. N. Ohde and K. N. Stevens, J. Acoust. Soc. Am. Suppl. 1 75, S66 (1984)]. From these results, it was hypothesized that consistent velar responses may only be obtained by delaying the frequency increase in the first formant (F1) or by making the F 1 transition slower. The aim of this study was to test this hypothesis by systematically varying the duration of F 1 between 10 and 40 ms, and the duration of an onset of a single pulse preceding formant frequency excitation between 10 and 20 ms. Two two-formant CV stimulus continua were synthesized, each covering a range of F2 starting frequencies, for vowels corresponding roughly to [l, ɑ]. Subjects identified these syllables under both an open and a forced-choice response set. The results showed that as the duration of these temporal properties increased, velar responses increased for some stimuli. However, considerable variability was observed across subjects. The results will be discussed relative to trading relations between spectral and temporal properties, and relative to their practical and theoretical implications. [Work supported in part by a Vanderbilt University Research Council Grant.]
An auditory basis for rate normalization in stops and glides
Authors: Diehl Randy L. , Walsh Margaret A.
Published: October 2005

When cued by formant transition duration, the perceptual distinction between stops and glides (e.g., [b] vs [w]) is subject to a “rate normalization” effect: longer steady-state vowel segments shift the perceptual boundary toward longer transition durations [J. L. Miller and A. M. Liberman, Percept. Psychophys. 25, 457–465 (1979)]. However, rate normalization apparently does not occur when the stop-glide distinction is primarily signaled by rise time [P. C. Shinn, S. E. Blumstein, and A. Jongman, Percept. Psychophys. 38, 397–407 (1985)]. Both sets of findings have essentially been duplicated using nonspeech analog stimuli. In one series of experiments, the stimuli consisted of a frequency-ramped tone of varying duration, immediately followed by either a short or long steady-state sinusoid. The effect of the long steady-state segment was to shift the categorization boundary toward longer ramp durations. In another series of experiments, the stimuli consisted of a fixed-frequency sinusoid or square wave with varying rise time and either a short or long peak amplitude interval. Categorization on the basis of rise time was little affected by peak amplitude duration. Given these findings, it is suggested that the earlier speech results have a basis in general audition. [Work supported by NICHD.]
Analysis and perception of sibilant fricatives: Shona data
Authors: Clark Christopher , Bladon Anthony , Mickey Katrina
Published: October 2005

In addition to the commonly found [s,z] and [∫ȝ], the Shona language has a third pair of sibilants [s,zw w] usually described as labialized alveolar fricatives and sometimes called “whistling” fricatives. Analysis of all these sibilants revealed that the location of the main frequency peak was typically 3 Bark higher for [s,z] than for the remaining four fricatives which did not differ significantly in this respect. Durational differences were slight throughout, and seemed unlikely to contribute as place cues. A likelier possibility for cueing the place distinction, we hypothesized, was an interaction of peak location with the gradient of the high- and low-frequency slopes adjacent to the peak, together perhaps with contributions from the VCV transitions. Experimental manipulation of these variables in synthetic versions of the three voiceless sibilants yielded listener identifications suggesting that (1) a flat HF slope cued [s] overridingly, irrespective of the peak location, but (2) peak location became important when a falling HF slope is used; and (3) a steep LF slope failed to create enough of a “whistling” percept as to outweigh transitional cues to the [∫]−[sw] distinction.
Dialect differences and the perception of the fricative-affricate-stop distinction in English
Authors: Wardrip-Fruin Carolyn
Published: October 2005

The fricated portions of the naturally produced words ship, shop, shoe (as produced by a standard English speaker) were systematically reduced by computer editing. These edited syllables were randomized and presented to 20 adult listeners, 10 of whom spoke a variety of standard English and 10 of whom spoke a variety of nonstandard English which has been influenced by Spanish. Listeners judged each syllable as having initial fricative, affricate, or stop (e.g., ship, chip, or tip). Results show a cross-over in judgments from fricative to affricate (e.g., ship to chip) and from affricate to stop (e.g., chip to tip) in specific ranges of fricative durations for both groups of listeners. However, the range of durations judged ambiguous was greater for the speakers of the Spanish influenced dialect particularly between the fricative and affricate categories. This may be a reflection of the phonological status of this distinction in the nonstandard dialect.
Effects of speaking rate on the perceived internal structure of phonetic categories
Authors: Miller Joanne L. , Volaitis Lydia E.
Published: October 2005

It is well known that when listeners identify consonants as voiced or voiceless on the basis of voice onset time (VOT), they do so in a rate-dependent manner: As rate of speech decreases such that overall syllable duration increases, the category boundary moves toward a longer VOT value. We have been examining whether this alteration in boundary location is accompanied by a change in the specific stimuli judged to be the best category exemplars. For our experiment we created two /bi/-/pi/ series with overall syllable durations of 125 and 325 ms, respectively. VOT varied from 10–120 ms for the 125-ms series and from 10–320 ms for the 325-ms series. We asked listeners to judge each syllable for its goodness as a member of the /p/ category, using a scale from 1–10; the higher the number, the better the exemplar. For each series, as VOT increased there was an initial increase in rating response, followed by a subsequent decline. Thus for both the short and long syllables, the /p/ category was perceived as having internal structure, with a limited range of stimuli serving as the best exemplars. However, the entire function for the longer syllables was shifted toward longer VOT values. That is, as syllable duration increased, so too did the VOT values of the stimuli judged to be the best category members. Moreover, in a subsequent control experiment we established that the shift in rating funtion was not due to range or frequency effects, but to the alteration in syllable duration, per se. These findings strongly suggest that the adjustment for speaking rate, whatever the underlying mechanism, involves a comprehensive modification in the mapping between acoustical signal and phonetic structure.
Using P centers to assess the coherence of syllables
Authors: Hoequist Charles , Smith Mary R.
Published: October 2005

Across studies when listeners perceive a syllable to have a fricative-stop initial cluster, they report only one P center. Further, the locations of such P centers shift in a one-to-one millisecond manner with changes in pre-vocalic acoustic durations. We also know there is a lower limit on the duration of silence which allows such clusters to be perceived. In this study we ask about the upper limit on intra-syllabic silence and we ask if the number and location of P centers can be used to assess listeners' sensitivity to the segmental coherence of syllables. Using natural tokens of [s], [trip] and [strip] by two talkers, the silent interval was varied from (a) isochronous alternation to (b) overlap of [s] and [trip]. Listeners judged the number of events and the location of P centers. For (a) they perceive two separate events, each with a P center. For (b) they perceive two simultaneous events with one P center. When listeners report one event with one P center, they show sensitivity to the naturally produced silent intervals. [Work supported by Alvey.]
Hemispheric specialization for familiar voice recognition: Dichotic listening evidence
Authors: Kreiman Jody , Van Lancker Diana
Published: October 2005

Previous dichotic listening studies of voice recognition have produced ambiguous results, perhaps due to the use of unfamiliar voices as stimuli: Evidence from brain-damaged subjects [e.g., D. Van Lancker and J. Kreiman, UCLA Working Papers in Phonetics 62, 50–60 (1985)] suggested that both hemispheres are required to discriminate among unfamiliar voices. In contrast, recognition of familiar voices was observed to require an intact right hemisphere in the unilaterally brain-damaged population. To obtain data on ear advantages for dichotically presented voice stimuli, therefore, we used voices of politicians and entertainers familiar to many listeners. Subjects monitored one ear at a time and identified both the speaker and the word spoken on each trial. Although not significant, the ear advantage observed for voice recognition did differ significantly from the significant right ear advantage found for word identification. In light of current models of hemispheric specialization, we interpret this relative left ear advantage for voice recognition as a reflection of a right hemisphere superiority for familiar speaker recognition in normal subjects.
Timbre: A better musical analogy to speech
Authors: Samuel Arthur G.
Published: October 2005

A recurring claim in speech research has been that “speech is special”—different perceptual processing is involved for speech than for other acoustic stimuli. In response to this claim, several demonstrations of speech-like results have been reported for nonspeech stimuli, including musical intervals (e.g., major versus minor chords), and manner of playing a string (e.g., pluck versus bow). The research to be reported explores another, possibly more appropriate, musical analog: timbre. The notion is that families of instruments (e.g., horns, strings) may be analogous to families of phonemes (e.g., liquids, stops, fricatives). To test this notion, stimuli were synthesized to sound like various instruments. Standard speech experiments were then conducted, including forced-choice identification, ABX discrimination, duplex perception, selective adaptation, and paired contrast. The similarity of the obtained results to those for speech will be reported, and the implications of these findings for theories of speech perception will be discussed. [Work supported by AFOSR.]
Necessary conditions for superresonant interaction between monopole scatterers
Authors: Tolstoy I.
Published: October 2005

Systems of precisely spaced bubbles, balloons, or air-filled thin shells in water, in full spaces, or near elastic boundaries, insonified at frequencies close to the fundamental resonance value ω0 (“bubble frequency”) and interacting via multiple scatter may develop true resonant modes or superresonances (SRs). Under SR conditions the pressure amplification relative to the incident field will be of order (kRa)−2, as opposed to (kRa)−1 for single scatter at ω0, kR being the wavenumber in water and a the scatterer radius. For simple configurations this leads to theoretical SR amplifications between 103 and 5 × 103. It is shown here that, in order to observe the SR effect, spacings and volumes must be controlled to 12 or better. Typical pair spacings are of order 13 λacoust or (roughly) multiples thereof. When a bubble/balloon pair is near a thin plate the basic spacing is about one flexure-wave wavelength and SR amplification becomes very sensitive to the direction of flexure mode arrivals, tending to vanish entirely for angles intermediate between broadside and endfire incidence. [Work supported by ONR.]
Nonlinear distortion of the bubble pulse in a layered inhomogeneous ocean
Authors: Epstein David
Published: October 2005

Many years ago, the author analyzed a representative deep-fired long-range underwater explosion waveform [J. Acoust. Soc. Am. 35, 800 (1963)]. The most striking feature was the so-called “double shock” formation, e.g., a strong resemblance between the shock wave and bubble pulse in both amplitude and shape. The phenomenon is attributed to the combined effects of detonation depth and waveform propagation. In a previous paper the characteristics of an explosion waveform at the source, as function of shot depth was established [Proc. 11th ICA, Paris (1983), Vol. 1, pp. 345–348]. Weak shock theory was used to study nonlinear distortion of the bubble pulse as it propagates in an homogeneous, unbounded, dissipationless medium [Proc. 10th Int. Symp. Nonlinear Acoust., Kobe, Japan (1984), pp. 79–82]. It was found that the deepest fired shots suffer the greatest distortion but that, in general, extremely long range is required for shock wave formation. However, focusing due to inhomogeneous structure may substantially increase pulse strength thereby enhancing nonlinear effects. Here the effect of inhomogeneous layering on the range required for shock wave formation is investigated.
Resonance identification through bistatic scattering for elastic spheres and spheroids
Authors: Überall H. , Werby M. F. , Brown S. H. , Dickey J. W.
Published: October 2005

Free vibrations of a submerged elastic object, resonantly excited by an incident acoustic wave or wave train, occur in the characteristic fashion of one of an infinite series of allowed eigenvibrations (normal modes for objects of separable geometry) if the excitation takes place at the eigenfrequency of that resonance. While such resonances appear in the backscattering cross section plotted versus frequency, interfering with the contribution of specular echoes, only resonance frequencies and resonance widths can be determined from such a plot but not the resonance order. For targets of separable geometry it has been shown both experimentally [G. Maze and J. Ripoche, J. Acoust. Soc. Am. 73, 41 (1983)] and theoretically [M. F. Werby and H. Uberall, submitted to J. Acoust. Soc. Am.] that bistatic observations can determine the resonance order after subtraction of the specular background, and the dipole resonance of the Rayleigh wave has been theoretically identified in this way. The method is here applied to bistatic scattering from elastic spheroids, as analyzed by the T-matrix formalism and computer code. [H. Überall is supported by the David W. Taylor Naval Ship Research and Development Center, and by the Office of Naval Research.]
Wideband compression and spectral sharpening for hearing-impaired listeners
Authors: Bustamante Diane K. , Braida Louis D.
Published: October 2005

Normal-hearing listeners with hearing losses simulated by masking noise benefit from wideband amplitude compression under conditions (e.g. no interference, most-comfortable presentation level) for which listeners with sensorineural hearing impairments do not. Such findings suggest that speech intelligibility may be limited by degraded spectral and temporal resolution as well as reduced dynamic range for hearing-impaired listeners. To combat the combination of reduced dynamic range and reduced frequency selectivity, we have developed a speech-processing system which provides both widehand compression and a sharpening of short-term spectral shape. The sharpening is accomplished by an orthogonal decomposition of the critical-band spectrum (similar to a principal-component decomposition) followed by expansion of the component weights which specify the peak-valley structure of the spectrum. We compare speech intelligibility with this system to that with linear amplification and wideband compression alone for hearing-impaired listeners. Preliminary results for CVC nonsense syllables indicate a modest benefit for the compression/sharpening system for a male speaker, but little benefit for a female speaker. [Work supported by NIH.]
Comparison of four hearing aid prescriptive procedures
Authors: Sullivan Jean A. , Levitt Harry , Hwang Jain-Yih , Hennessey Ann Marie
Published: October 2005

A digital master hearing aid configured around a programmable digital filter was used to simulate hearing aids prescribed by four different procedures. The prescriptive procedures were modeled on those developed by Skinner, Byrne and Toonison, Lybarger, and Levitt et al. Real-time simulation allowed for adaptive paired-comparison testing to be used in evaluating the prescribed hearing aids. Data were obtained for three signal levels corresponding to input levels of 60 and 70 dB SPL and an output level corresponding to the subject's most comfortable loudness level. Speech discrimination scores were obtained for each simulated hearing aid at each signal level, as well as paired-comparison judgments of relative intelligibility and overall quality. No single prescriptive procedure was found to be uniformly superior to the other procedures considered. Differences between procedures were largest at low signal levels and smallest at the subject's most comfortable level. [Research supported by NIHR.]
A log-linear approach to consonant confusion analysis
Authors: Bell Theodore S. , Dirks Donald D. , Carterette Edward C.
Published: October 2005

Log-linear modeling techniques, along with the likelihood-ratio chi-square (G2) statistic, were used to assess the effects of high-frequency filtering, background noise (shaped to shift thresholds 20 dB), and presentation level on consonant confusion patterns. Specific log-linear parameters were tested to determine the significance of each of the variables and their associated interactions. Low-pass filtering changed the pattern of errors relative to unfiltered conditions as presentation level increased. This effect was evidenced by a significant interaction between stimulus, response, level, and filtering. Patterns of errors were also affected by the presence of an external noise. Signal-to-noise ratios were selected to produce equivalent performance scores relative to quiet conditions. Thus, even though the same number of errors occurred in the quiet and noise conditions, the relationship between them was significantly different. Interestingly, presentation level was also a significant variable affecting error patterns. Obviously, as presentation level increased, the number of errors decreased. However, these data indicated that the pattern of errors did not remain constant as level changed, and this effect was not similar across filtered or noise conditions. [Work supported by NINCDS.]
The binaural advantage in reverberation for stimuli with controlled predictability
Authors: Boney Stephen J. , Humes Larry E.
Published: October 2005

Monaural near-ear and binaural speech-recognition performance was assessed under two levels of reverberation (T = 0.0 and 0.8 s), at two signal-to-babble ratios (+4 and −2 dB), and at two speaker-to-listener orientations (speech at 0° and noise at 180°, and speech at 45° and noise at 225°). Speech-recognition performance was assessed using the high (PH) and low (PL) predictability sentences from the revised-SPIN test (Bilger, 1984). The PH and PL items from the SPIN were recorded through a KEMAR manikin under the above conditions and played back to normal-hearing subjects via earphones. A binaural advantage was observed across all listening conditions for both the PH and PL items. The average binaural advantage for the PH items at T = 0.0 s was about two times greater than at T = 0.8 s. The binaural advantage was approximately the same for the PL items regardless of reverberation time. Significant intersubject variability was observed for the binaural advantage across listening conditions for both types of speech stimuli. In addition, there were several interaction effects between the various acoustic parameters examined for the PH and PL items.
The NST in varying speech-to-noise ratios
Authors: Johnson C. E. , Danhauer J. L.
Published: October 2005

The effects of a speech-spectrum noise competitor presented in varying speech-to-noise ratios on ten young adult normal-hearing listeners' responses to the nonsense syllable test (NST) were evaluated. Responses were compared to the CID W-22 test presented in the same conditions. Stimuli were presented via earphones at − 10, −5, 0, + 5, and + 10 dB speech-to-noise ratios. Subjects' responses were scored and analyzed statistically. Results indicated: (1) performance decreased with speech-to-noise ratio; and (2) responses were poorer on the NST than on CID W-22. Clinical implications will be discussed.
Spectrographic analysis of the NST—Implications for assessment and error analysis
Authors: Johnson C. E. , Asp C. W. , Danhauer J. L.
Published: October 2005

Two recordings of the same word list constitute different word discrimination tests—due to talker differences and recording procedures [T. Frank and C. H. Craig, J. Speech Hear. Disorders 49, 267–271 (1984)]. Thus acoustic data of each test should be available—particularly of nonsense stimuli. A spectrographic analysis of the Edgerton-Danhauer NST was performed. Fundamental frequency F0, formant frequencies (F1-F3), relative amplitude, and duration of each phoneme (when applicable) have been documented, compared across phonetic contexts, and to the results of previous research. Implications for assessment will be discussed in terms of intra/inter-speaker variability and the use of nonsense stimuli in controlled phonetic contexts. In addition, data will be used to interpret normal-hearing and hearing-impaired listeners' errors on the NST.
A psychoacoustic assessment of central auditory processing in children who exhibit learning impairments
Authors: DeFrosse Carlton O. , DeFrosse Carlton O.
Published: October 2005

A psychoacoustic test battery consisting of the Jack Willeford Central Auditory Assessment Battery and the Jack Katz Staggered Spondaic Word (SSW) Test was administered to 45 learning-impaired children that ranged in age from 6 to 16 years. Their learning impairments occurred in one or in a combination of the following academic areas: math, reading, and writing. None of the children qualified for placement in existing learning-impaired classrooms. Subtest analysis of right- and left-ear scores was performed to determine the effectiveness of the test material for identification of auditory perceptual deficits that may contribute to the children's learning impairments. Diagnostic patterns and educational implications will be discussed and interpreted.
Planar elastic vibrators in a uniform flow field with broadband mechanical excitations
Authors: Li Dong-Jye , Stepanishen Peter R.
Published: October 2005

Flow-induced forces on structures can significantly affect their dynamic response. A new method is presented to evaluate the dynamic response of planar elastic vibrators in uniform fluid flow fields to broadband mechanical excitations. The approach is based on the use of in vacuo eigenfunction expansion to solve the fluid-loaded problem. The time-dependent coefficients of the modal expansion for the velocity are shown to satisfy a set of coupled convolution integral equations which involve mode and Mach number-dependent impulse responses. Pressures in the field are simply expressed as a summation of uncoupled convolution integrals. Numerical results are presented to illustrate the effects of Mach number on the transient response of a plate subjected to a broadband mechanical excitation. [Work supported by ONR.]