bluuebunny/crossref_metadata_2025 huggingface dataset:
English vowels for Chinese speakers enrolled in English as a second language classes
Authors: Huber Therese I. , Dou Hong Wei , Weiler Ernest M. , Agnello Joseph G.
Published: October 2005

Chinese and English differ considerably in the characteristics of the vowels. The present study focuses on acoustic characteristics of Chinese production of English vowels. The seven Chinese students in the present study had all been born in mainland China, and were students at the University of Cincinnati. All were enrolled in ESL (English as a second language) classes because of difficulties with English pronunciation. Two native English speakers were used to provide a comparative analysis, by minimal pairs, for formant placement of the vowels /i/, /I/, and others. A total of five English speakers phonetically transcribed the productions and tallied the data into a confusion matrix. The acoustic analysis, performed separately by five speech science students revealed systematic differences in the production of the vowels associated with the confusion. The Chinese first formant was lower and the second was higher than found for the midwestern English speakers. Insufficient reduction in frequency between minimal pairs was the source of auditory confusion. The application to ESL speech therapy will be discussed.
Acoustic properties of primary and secondary word-level stress in Estonian
Authors: Gordon Matthew K.
Published: October 2005

Languages differ in the way they signal stress. In Estonian, the primary stressed syllable is marked by increased onset duration [Lehiste, ConsonantQuantityandPhonologicalUnitsinEstonian (Indiana Univ., Bloomington, 1966)] and by a pitch rise [Liiv, Sovetskoe Finno-Ugrovedenie 21(1), 1–13 (1985)]. The acoustic properties of secondary stress are subtle, resulting in controversy over its location [Hint, EestiKeeleSõnafonoloogia (Eesti NSV Teaduste Akadeemia, Tallinn, 1973); Eek, EPP, 20–59 (1982)]. To determine further the acoustic properties of Estonian stress, amplitude, F0, and consonant and vowel durations were measured for di-, tri-, and tetrasyllabic words. Results for primary stress conformed to those found by Lehiste and Liiv. Secondary stress was characterized by an interruption in the F0 decline (Eek, 1982) and by lengthening of the onset, but not the rhyme. Estonian is thus unusual in signaling stress by pitch and by lengthening of the onset, but not the rhyme. Unstressed open syllables were significantly longer after primary stressed syllables of both the CV [Lehiste, Language 41(3), 447–456 (1965)] and the CVV type, than after CVC, arguing against foot isochrony [Eek and Remmel, Speech Communication Seminar, 179–185 (1974)]. Implications of these phonetic measurements for proposed algorithms of stress assignment in Estonian will be considered.
Application of Skudrzyk’s ‘‘mean-value theory’’ to fluid-loaded plates
Authors: Torres Rendell R. , Sparrow Victor W. , Stuart Alan D.
Published: October 2005

The geometric-mean drive-point admittance of a complex structure can be found by examining the admittance of the corresponding infinite structure (i.e., ‘‘characteristic admittance,’’ Yc) [Skudrzyk, J. Acoust. Soc. Am. 67, 1105–1135 (1980)]. The frequency response of an infinite plate, for example, coincides with the geometric-mean response of a finite plate, i.e., the response equidistant from the resonance maxima and antiresonance minima, plotted on a logarithmic scale. Skudrzyk’s ‘‘mean-value theorem’’ was derived (and experimentally verified) without consideration of fluid coupling, which introduces a reactive effect that physically resembles a mass loading. The purpose of this research is to find whether the response of the fluid-loaded infinite plate still corresponds to the geometric-mean response of the fluid-loaded finite plate. Numerical results indicate that, in the presence of fluid loading and at low frequencies (below critical), the mean-line drive-point admittance of the finite plate still corresponds to the infinite-plate drive-point admittance that has been derived analytically [Crighton, J. Sound Vib. 54, 389–391 (1977)]. [Work supported by the Applied Research Laboratory.]
An acoustic boundary element method using analytical/numerical matching applied to a three-dimensional fluid-loaded plate
Authors: Epstein Ronald J. , Bliss Donald B.
Published: October 2005

Analytical/numerical matching (ANM) is a hybrid scheme combining a low-resolution global numerical solution with a high-resolution local analytical solution to form a composite solution. ANM is applied to a harmonically vibrating flat plate in three dimensions to calculate the radiated acoustic field and the associated fluid loading. The problem utilizes overlapping smoothed dipoles, and local corrections to calculate the dipole strength distribution on the surface of the plate. A smoothing length scale is introduced that is larger than the smallest physical scale, and smaller than the largest physical scale. The global low-resolution solution is calculated numerically using smoothed dipoles, and converges quickly. Local corrections are done with high-resolution local analytical solutions. The global numerical solution is asymptotically matched to the local analytical solutions via a matching solution. The matching solution cancels the global solution in the nearfield, and cancels the local solution in the farfield. The method is very robust, offering an insensitivity to collocation point position. ANM provides high-resolution calculations from low-resolution nu- merics with analytical corrections, while avoiding the subtlety involving singular integral equations, and their numerical implementation. [Work supported by ONR.]
Comparisons of vibration frequencies for rods and beams from one-dimensional and three-dimensional analyses
Authors: Leissa Arthur W. , So Jinyoung
Published: October 2005

A method of three-dimensional (3-D) analysis is developed for the free vibration frequencies and mode shapes of solid circular cylinders of elastic material. The method assumes for the three displacement components a Fourier series in the circumferential (θ) direction and algebraic polynomials in the radial (r) and axial (z) directions. All types of boundary conditions and arbitrary length-to-diameter ratios (L/D) may be accommodated. Extensive convergence studies show that frequencies which are exact to five or six significant figures may be obtained for cylinders with free–free ends, and that three or four figure accuracy is achievable for fixed–free ends. These accurate frequencies are compared with ones determined from known elementary and improved 1-D theories for longitudinal, torsional, and bending modes for various L/D ratios, thereby establishing the ranges of accuracy of the 1-D analyses for circular cross sections.
Prediction of signal-to-noise ratio gain for passive higher-order correlation detection of energy transients
Authors: Pflug Lisa A. , Ioup George E. , Ioup Juliette W. , Field Robert L.
Published: October 2005

In general, higher-order correlation detectors perform well in passive detection for signals of high third- and fourth-order moments. Previous studies by the authors have shown that the normalized third- and fourth-order signal moments are reliable indicators of higher-order correlation detector performance [Pflug et al. (1992b)]. For a deterministic energy transient of known moments through fourth order, it is possible to predict theoretically the amount of gain over an ordinary cross-correlation detector for a bicorrelation or tricorrelation detector applied in a noise environment of known variance. In this paper, formulas that predict detector performance for passive detection at the minimum detectable level are derived. The noise is assumed to be stationary and zero mean with Gaussian correlation central ordinate probability density functions. To test the formulas, SNR detection and gain curves are generated using hypothesis testing and Monte Carlo simulations on a set of test signals. The test signals are created by varying the time width of a pulse-like signal in a sampling window of fixed time duration, resulting in a set of test signals with varying signal moments. Good agreement is found between the simulated and theoretical results.

The effects of observation time (length of detection window) and sampling interval on detector performance are also discussed and illustrated with computer simulations. The prediction formulas indicate that decreasing the observation time or the sampling interval (assuming the signal is sufficiently sampled and the detection window contains the entire signal) improves detection performance. However, the rate of improvement is different for the three detectors. The SNR required to achieve the minimum detectable level of detection performance at a given probability of false alarm (Pfa) decreases with the fourth root of the observation time and sampling interval for the cross-correlation detector, the sixth root for the bicorrelation detector, and the eighth root for the tricorrelation detector. Relative detector performance also varies with Pfa. The probability of detection (Pd) for higher-order detectors degrades less rapidly with decreasing Pfa than the Pd for ordinary correlations. Thus higher-order correlators can be especially appropriate when a very low Pfa is required.
Residential acoustics—An historical perspective
Authors: Wetherill Ewart A.
Published: October 2005

A review of scientific and general writing on dwelling places confirms that disturbance of residents by both exterior and interior noises has been a major concern of urban living for more than 2000 years. Empirical experiments in reduction of intruding noise appear to have developed into traditional planning and building methods, such as separation of ‘‘active’’ from sleeping rooms and filling of floor joist cavities to enhance noise isolation. From a handful of references in the 19th century, the rate of publication of theoretical and experimental studies as well as nontechnical writing covering all aspects of acoustical intrusion increased dramatically in the early decades of the 20th century. These have led to international standardization of acceptability and measurement, and to regulation of acoustical performance in dwellings. Concurrent with this, however, the increase in types and locations of noise source—together with ways of reducing construction costs—have resulted in living environments in which both outdoor and neighbor noise may be worse than at any time in history.
Fundamental issues of accuracy in boundary element analysis of acoustic problems
Authors: Trevelyan J.
Published: October 2005

The boundary element method (BEM) has become well established as a primary technique for the computer simulation of acoustic fields. Its ability to provide accurate predictions of sound-pressure levels in a variety of media, including those extending to ‘‘infinity,’’ has led to a growth in the popularity of the method. Further, recent advances in acoustic diagnostic analysis, which is an extension of the traditional BEM approach, have added to the capabilities of this type of software simulation. It is important when using numerical simulation methods not to forget the potential sources of error which may be caused by improper usage of software systems. This paper discusses some of the fundamental modeling requirements of a good boundary element model, showing how the results can degrade when these requirements are violated.
Overview of musical instrument physics
Authors: Weinreich Gabriel
Published: October 2005

Most devices which can be used to ‘‘make sound,’’ in the sense of converting frequencies in the range of human gestures to those within the audible band, are nonetheless not in use as musical instruments. What is required in addition is a large, but precise, control parameter space that makes an instrument ‘‘playable,’’ that is capable of transmitting a complex musical message. Within this parameter space, the exact control of periodicity is especially important, particularly so in Western music. After first exploring how the requirements of pitch stability, acoustic output capability, and sustaining power are satisfied in various instruments that vibrate freely after an initial excitation, the discussion will continue with the more complex physics of the bowed string and the blown pipe in their various embodiments. In many cases, it will turn out that instruments in actual use are not ‘‘engineering compromises’’ but, on the contrary, based on fortuitous cooperation among otherwise unrelated factors. The lecture will conclude with a brief discussion of the computer as a musical instrument.
Internal solitons in the ocean
Authors: Apel John R. , Ostrovsky Lev A. , Stepanyants Yuri A.
Published: October 2005

Internal waves (IW) are among the important factors affecting sound propagation in the ocean. A special role may be played by solitary IWs because of their spatial localization and high magnitudes. Here, nonlinear IWs are discussed (a) from the standpoint of soliton theory and (b) from the viewpoint of experimental measurements. First, basic theoretical models for solitary IWs in the ocean are described, and various analytical solutions are treated, commencing with the well-known Korteweg–de Vries equation and its important generalizations including effects of rotation, cylindrical divergence, eddy viscosity, shear flows and instabilities, and turbulence. Experimental evidence for the existence of solitons in the upper ocean is presented both for shallow and deep sea regions. The data include radar and optical images and insitu measurements of waveforms, propagation speeds, and dispersion characteristics. It is suggested that internal solitons in the ocean are ubiquitous and are generated primarily by tidal flows over continental shelf breaks, banks, and similar bottom topographic features. [Work supported by ONR, DOD, ISF.]
Mode coupling and other effects of modeled propagation through continental-shelf internal solitary waves
Authors: Preisig James C. , Duda Timothy F.
Published: October 2005

Acoustic propagation through a continental-shelf waveguide containing internal solitary waves has been modeled with the finite-element parabolic equation (FEPE). The waveguide has two essentially mixed layers separated by a gradient layer, with a homogeneous, lossy bottom. Solitary waves (pycnocline depressions) of many sizes and shapes are included, not all of which fit the dispersion relation. Including both physical and unphysical waves aids interpretation of the forward-scattering (mode coupling) mechanism. Solitary waves of horizontal scale length 50 to 150 m cause significant mode coupling. Longer and shorter solitons give weaker coupling. For cw signals (400 Hz), the results of a sudden approximation model for a single soliton are compared to those generated with the FEPE model and provide some insight into the mode coupling behavior. The results for a broadband pulse (e.g., 400- to 525-Hz band) show a first-order difference in waveform shape after propagation through a soliton when compared to the result with no soliton. This leads to the conclusion that received signal predictability based upon perturbation of layered-only propagation is problematic in the presence of the waves. [Work supported by ONR.]
Adaptive feedforward modal space control
Authors: Clark Robert L.
Published: October 2005

An alternative method of implementing adaptive feedforward control is presented in this work based upon independent modal space control of reverberant systems. Multivariable control theory is used to derive the optimal feedforward compensators, and when the error sensors are constructed in modal coordinates, the optimal compensators reduce to a single coefficient filter per control channel (assuming the reference signal is available). If the control inputs are also implemented in modal coordinates, then the control system is termed noninteracting since each mode is orthogonal to the other. Adaptive feedforward modal space control is ideal for implementation of time-averaged gradient descent algorithms such as Newton’s method since the optimal filter coefficient is obtained from a single iteration of the algorithm, and is insensitive to changes in the resonant frequencies of the reverberant system. Formulating the control system in modal coordinates also clearly illustrates the effect of causality and nonminimum phase characteristics on the ultimate performance of any adaptive feedforward control system, whether it is implemented in physical coordinates or modal coordinates.
The effects of sinusoidal amplitude modulation on gap detection in noise
Authors: Snell Karen B.
Published: October 2005

The purpose of this study was to investigate the effects of envelope fluctuation on temporal gap detection. Temporal gap detection thresholds were measured in wideband noisebursts that had been amplitude modulated by a very low-frequency sinusoid. Detection was studied at six modulation depths and in four phase conditions which placed the onset of the silent gap at different amplitudes on the temporal envelope of the waveform. Gap thresholds were obtained in a 3IFC paradigm in four young, normal-hearing subjects. The smallest mean gap threshold, 3.5 ms, was obtained for unmodulated noisebursts. The largest mean gap threshold, 25.7 ms, was obtained at a modulation depth of 100%. Phase did not significantly affect gap detection until modulation depth was increased to 40%. At deeper depths of modulation, gap thresholds were smallest, but also most variable, when the gap onset was placed at a peak in the waveform envelope and largest when the gaps were centered on an envelope minimum or randomly placed. Results of this study are consistent with previous findings and the hypothesis that gap detection is dependent on the similarity of the gap itself to inherent fluctuations in the stimulus envelope, reflecting coherence in terms of phase and depth of envelope fluctuation.
Effects of contralateral presentation and of interaural time differences in segregating a harmonic from a vowel
Authors: Hukin R. W. , Darwin C. J.
Published: October 2005

The extent to which the 500-Hz component of a steady-state vowel contributed to its phonemic category was measured by estimating the position of the /I/–/ε/ phoneme boundary along a first formant (F1) continuum. Shifts in the phoneme boundary were calibrated against shifts produced by physical changes in the level of the 500-Hz component. When the 500-Hz component was presented contralateral to the rest of the vowel, the phoneme boundary changed by an amount equivalent to a physical reduction in the level of the component of about 5 dB (experiment 1). However, giving the 500-Hz component an interaural time difference (ITD) of −666 μs and the remainder of the vowel an ITD of +666 μs did not reduce the component’s contribution to vowel quality (experiment 2). Placing the 500-Hz component of the vowel in a sequence of 500-Hz tones substantially reduced the contribution of the 500-Hz component; the contribution was further reduced by giving both the tone sequence and the component an ITD of −666 μs and the rest of the vowel one of +666 μs (experiment 2). When these tone-sequence conditions were presented in the same experimental block as the ITD conditions without a preceding tone sequence, these latter conditions did then show an effect of grouping by ITD. The results suggest that listeners can perceptually segregate sound on the basis of different ITDs, but that this segregation is substantially enhanced if the direction has previously been cued.
An ultrasonic motor using low-frequency magnetostrictive excitation for continuous and stepwise angular rotation
Authors: Saksena T. K. , Chandra Mukesh
Published: October 2005

A new design for an ultrasonic motor operating at a frequency of about 14.5 kHz for generating continuous as well as stepwise angular rotation has been developed. The device uses a magnetostrictive ferrite transducer and a single power amplifier for its operation. A possible mechanism for the operation of the ultrasonic motor has been suggested. The variation of the speed of rotation as a function of frequency, excitation voltage, and load has been studied. The measurement of the performance characteristics reveal that the ultrasonic motor is capable of giving 100 rpm at 34.6-W electric output. The rpm of the rotating disk can be controlled by a slight adjustment of the frequency of the rf signal to the driving transducer. The load capability of the device is 1 kg. Using a tone burst of 10 ms, an angular step of 1.8 deg has been realized.
Bending vibration of cantilevered thin-walled beams subjected to time-dependent external excitations
Authors: Song Ohseop , Librescu Liviu
Published: October 2005

The bending vibration of cantilevered thin-walled beams of arbitrary closed cross section exposed to time-dependent external excitations is investigated. The beam model used in this study incorporates a number of nonclassical effects, namely, transverse shear, secondary warping, anisotropy of constituent materials, and heterogeneity of the construction. An exact methodology based on the Laplace transform technique aiming at determining the frequency-response characteristics is used, and numerical illustrations emphasizing the effects of a number of geometrical and mechanical parameters on the frequency response behavior are displayed.
A directional adaptive least-mean-square acoustic array for hearing aid enhancement
Authors: DeBrunner Victor E. , McKinney Elaine D.
Published: October 2005

This paper introduces directional microphones to adaptive array processing for hearing aid applications. Acoustic fixed arrays are designed to match a focused array gain pattern, while acoustic adaptive arrays are designed to attenuate interference noises with changing characteristics. However, as currently constructed, acoustic adaptive arrays cannot stay focused with a limited number of microphones available in a cosmetically acceptable hearing aid. In this paper, a technique is discussed that combines fixed and adaptive arrays in a system which enhances the desired signal while effectively attenuating interference speech and background noise. In particular, the design and performance of a directional adaptive least-mean-square (LMS) acoustic four-element array with a restricted geometry, where the array microphones are directional microphones, are examined. Simulations show that the directivity index of the directional adaptive array using four hypercardioid microphones is improved to between 8.6 and 11 dB. The array reduces the interference noises by 29.7 to 42.3 dB and provides a signal-to-noise ratio improvement of 11.5 to 12.2 dB over a single omnidirectional microphone. The sensitivity analysis is also discussed. It is concluded that the small size (four-element) microphone array can spatially filter interference noise effectively and so improve SNR performance significantly.
Power dissipation in a sprung mass attached to a master structure
Authors: Maidanik G.
Published: October 2005

It is shown that unlike the impedance at resonance, the power dissipation in the sprung mass appropriately approaches zero as the loss factor in this dynamic system approaches zero. Indeed, for an optimal dissipated power a finite loss factor in the sprung mass is required. It is also shown that the effective loss factor in the master structure in lieu of the power dissipated in the sprung mass is not usually at its maximum value when this dissipated power is optimal. A relevant question in designing sprung masses may thus be suggested: Should one design for maximization of this effective loss factor or opt for optimization, at least for maximization, of the dissipated power in the resonant sprung mass? The question is further compounded if the design is motivated not only by power dissipation options but also by options relating in part to the control of the response either in the master structure, in the sprung mass, or in both.
Minimum detectable level evaluation of inverse beamforming using Outpost SUNRISE data
Authors: Fabre J. Paquin , Wilson James H.
Published: October 2005

Inverse beamforming (IBF) has been very successful in detecting signals of very low levels in measured ocean data and during real-time at-sea experiments. The large performance gains for IBF are attributed to the Fourier integral method (FIM) beamforming solution [J. H. Wilson, J. Acoust. Soc. Am. 73, 1648–1656 (1983); A. Nuttall and J. H. Wilson, ibid. 90, 2004–2019 (1991)], to the inverse beamforming integral equation, to a new data thresholding technique called the eight nearest-neighbor peak picker (ENNPP), and to a postprocessing technique called the three-dimensional M of N tracker. Collectively, FIM, the ENNPP, and M of N tracker are referred to as IBF and Outpost SUNRISE data and they are used to determine the minimum detectable level (MDL) for three signal processing methods: conventional beamforming (CBF) using the LOFARGRAM display, CBF using the ENNPP and M of N tracker, and FIM using the ENNPP and M of N tracker (i.e., IBF). IBF was derived based on the environmental acoustic (EVA) properties of measured ambient noise and those EVA properties are in stark contrast to the ideal signal and noise conditions normally assumed in classical detection theory [M. Barkat, Signal Detection and Estimation (Artech House, London, 1991)]. Outpost SUNRISE noise data were chosen for the MDL evaluation because the shipping, wind, propagation, and other environmental supporting parameters were well measured during the exercise, typical of an exercise conducted by the advanced environmental acoustic support (AEAS) program. It is shown that the IBF MDLs are over 10 dB lower than those for CBF with the ENNPP and M of N tracker, and for CBF with high quality beam LOFARGRAM displays. A subsequent paper will compare IBF to other widely used high-resolution beamforming methods and the IBF theoretical gains are discussed in a companion paper [J. Wilson, J. Acoust. Soc. Am. 98, 3240–3251 (1995)] where EVA issues are illustrated by FRAZ surfaces from Outpost SUNRISE.
Vowel perception in dyslexic subjects
Authors: Bertucci Carol B.
Published: October 2005

It has been demonstrated by some investigators that poor readers may have less well-defined phonological categories than good readers. Whereas these studies have considered the perception of consonants, the current study investigated the percetion of the vowels /I, ε, æ/ by individuals with dyslexia. Perception was also compared with production and with other measures considered important for reading. A small-group case-study design was employed, in which the patterns of performance of one adult and three children with varying degrees of dyslexia were examined individually and compared with age-matched control subjects. An identification task was used, in which subjects matched synthesized stimuli, equally spaced along a continuum of F1/F2, to pictures of pit, pet, and pat. A set of intermediate stimuli was administered at a later date. The findings of this exploratory study suggest that subjects vary in their ability to perceive the vowels /I, ε, æ/ according to the severity of their reading/spelling impairment. Differences in perception were noted in the sharpness of the boundaries between phonemes and in the tendency to give abberant responses to test stimuli. Shifts in perception were also noted across days and stimulus sets in some individuals.
An integrated modal approach to waveguide reverberation
Authors: Tracey Brian H. , Schmidt Henrik
Published: October 2005

In shallow-water environments scattering from rough surfaces and volume inhomogeneities acts to degrade sonar performance. An efficient normal mode method has been developed to model the resulting reverberation. The self-consistent perturbation theory of Schmidt and Kuperman [J. Acoust. Soc. Am. 97, 2199–2210 (1995)] has been reformulated in terms of normal modes to handle scattering from rough fluid and elastic interfaces. A similar, recently developed volume scattering theory is used to model the effects of sound speed and density fluctuations in the seabed and water column. The two theories are integrated numerically and used to contrast the different scattering mechanisms. Waveguide propagation effects are seen to be extremely important. Spatial correlation and modal cross-correlation statistics are estimated and used to study decorrelation of the sound field. Excellent agreement with the wave-number integration implementation by Schmidt and Kuperman is demonstrated for the case of rough interface scattering, with the modal approach requiring significantly less computation for low-frequency shallow-water scenarios.
Reconsideration of sound attenuation in a cylindrical tube due to evaporation-condensation
Authors: Mao Yi , Sabatier James M. , Raspet Richard
Published: October 2005

Previous results of a theoretical study [J. Acoust. Soc. Am. 96, 3254(A) (1994)] of the sound propagation in a cylindrical tube indicated the sound attenuation due to the evaporation-condensation process underestimated the experimental results in porous materials. Some modifications have been introduced into our theoretical model for solving the discrepancy. First, the fundamental equations in a gas–vapor mixture are resolved including relaxation processes. Again four modes—acoustical, thermal, vorticity, and mass-diffusion modes—are obtained. Second, the temperature on the wall is no longer kept constant; the wall has a finite heat capacity and heat can conduct within the wall. The sound field propagating down the tube was determined by applying the modified boundary conditions to a superposition of the newly obtained four modes. The effects of each individual modification on the sound attenuation due to evaporation-condensation are examined. [Work supported by ONR.]
Characterization of supersonic projectile shockwaves using wavelets
Authors: Sadler Brian M. , Pham Tien
Published: October 2005

Wavelets were applied to experimentally obtained acoustic time series for detection and characterization of shockwaves generated by supersonic projectiles. Exploitation of the shockwave (N wave) for detection is useful in noisy, echo-filled environments. Wavelets generated from the derivative of a quadratic spline yield a wavelet transform (WT) with a deterministic relationship between the WT modulus-maxima and sharp transitions of the signal [S. Mallat and W. Hwang, IEEE Trans. Inf. Theory 38, 617–643 (1992)]. This WT is dyadic in scale but includes all signal shifts. Shockwave detection was accomplished by analyzing across scales. Analysis proceeds from large to small scales because large scales are more low pass and thus have fewer artifacts from high-frequency interference. With a sampling rate of 250 kHz the first seven or eight scales were found to be sufficient for detection and filtering. The N-wave produces characteristic double-peak waveforms at the various scales. Cross-scale analysis searches for these waveforms with constraints on N-wave duration. Detection of the leading and trailing edge of the shockwaves was achieved with high accuracy. The algorithm was validated using data from a variety of small (rifle) and large (tank) projectiles.
Back–front classification of English vowels using a cepstrum-to-formant model
Authors: Clermont Frantz , Broad David J.
Published: October 2005

A method is described for importing the concept of critical formant spacing into the cepstral domain, with a view toward automatic and robust classification of English vowels into back and front categories. The method is based on a cepstrum-to-formant, linear regression model reported previously [D. J. Broad and F. Clermont, J. Acoust. Soc. Am. 86, 2013–2017 (1989)], which is reformulated in order to transform low-order, linear-prediction (LP) cepstra into Bark distances between third (F3) and second (F2) formant frequencies. The back–front distinction itself is established using the critical 3.5-Bark distance as a binary decision criterion. The proposed method was evaluated through classification experiments involving (14th-order) LP cepstra and formant frequencies F2 (Barks) and F3 (Barks), which were all measured from three center frames of nine vowels spoken five times in /CVd/ context (C=/h,b,d,g,p,t,k/) by four adult, male speakers of Australian English. The accuracies obtained for correct back–front classification range between 94% and 100% both on an intra- and an interspeaker basis. These results confirm the effectiveness of relational measures of formant patterns for broad vowel distinction, but more importantly indicate the possibility of deriving certain phonetic features directly from the LP cepstrum.
Speech enhancement based on auditory perception
Authors: Yang Jun , Ribic Zlatan
Published: October 2005

It is well known that formants are more important for speech perception than other details of the spectral envelope. Therefore, for the case of speech (vowels) in noise, improving the salience of the formants by decreasing the energy in the valleys between spectral peaks would logically result in improving intelligibility. Furthermore, perceptually only the first three formants are important in determining the sound that is heard, although the higher formants are necessary to produce sounds of acceptable quality. Based on the above, this paper proposes a scheme to enhance the spectral peaks of the first three formants. In this scheme, the first-order high-pass filter with 1-kHz cutoff frequency is used to pre-emphasize the input speech. Three equalizers in parallel form are designed to enhance the amplitudes of the first three formants that are extracted by the cepstral technique and used as the corresponding resonant frequencies of these equalizers. This proposed scheme is particularly effective in the situation where clean speech can be obtained, such as theaters, concert halls, and classrooms with electrical audio-visual aids, etc. The advantages of this scheme over existing schemes will be discussed, and a demonstration tape will also be played during the presentation.
Model experiment on noise barriers having a random edge
Authors: Ho S. T. , Busch-Vishniac I. J. , Blackstock D. T.
Published: October 2005

A preliminary model experiment showed that the insertion loss for a random edge barrier exceeded that of a straight edge barrier of the same average height [J. Acoust. Soc. Am. 97, 3392(A) (1995)]. Based on this promising result, a three-level full factorial model experiment has been conducted on random edge barriers. The top of the barrier is a strip of sheet metal of fixed average height but made randomly triangular by straight line cuts. A random number generator was used to specify the height variations Δy (scaled to a prescribed maximum value) at fixed intervals Δx along the barrier. Nine different random edges provide a range of 9–37 mm in Δx and 5–18 mm in Δy. An electric spark was the sound source, a 1/8-in. B&K condenser microphone the receiver. The frequency range of the measurements is 5–21 kHz, the Fresnel number range 2–20. In most of the measurements the random edge barriers showed 3–8 dB more insertion loss than the straight barrier. For several cases, however, the straight barrier was better. [Work supported by TxDOT.]
Sound energy distribution at an intersection of underground tunnel: Small scale experiment
Authors: Imaizumi Hiroyuki , Isei Takehiro
Published: October 2005

Underground spaces generally have complex networks of tunnels which include many intersections. When a disaster such as a fire and an earthquake breaks out there, correct transmission of sound information has an important role in order for people to evacuate from underground safely [Fujimiya and Miura, J. INCE Jpn. 6(5) (1982)]. From the above point of view, sound energy distribution characteristics at an intersection of a scale model tunnel were measured under several conditions. The scale model tunnel had a square cross section, and experiments were carried out under acoustically hard and soft conditions. Impulsive sound from an electrical sound source indicated high reproductivity and omnidirectional characteristics from a preliminary experiment in the anechoic room. In a case where the scale model tunnel has an orthogonal branch, it was observed that low-frequency components were mainly propagated toward the branch, and this tendency was changed according to the angle of intersection. Furthermore, qualitative numerical simulation on sound propagation in a tunnel with an intersection was developed and the results agreed well with experimental results.
Analysis of a model structural fuzzy with long-range spatial interactions
Authors: King Melvin E.
Published: October 2005

Considerable progress has recently been made in analyzing the vibrational response of systems that enclose complex internal structures by introducing the notion of structural fuzzies. For globally distributed fuzzies that locally interact with a master structure (i.e., systems without length scales), a primary descriptor for the structural fuzzy has been identified as the density of mass per unit temporal frequency, dM/dω. Many practical structures, however, possess internals which are not local in nature, and therefore couple with the response of a master structure over various length scales. In an effort to address such problems, a prototypical model has been selected which simulates a network of pipes, ducts, and/or conduits attached to a master structure. The model consists of a flat plate (taken to be the master structure) onto which is attached a network of pipes (taken to be the spatially interacting structural fuzzy). Each pipe is attached to the plate via periodically spaced supports (assumed to be rigid and massless), and only coupled transverse motions are considered. Using this model system, basic insights into frequency wave-number interactions due to the structural fuzzy are sought and important physical parameters are identified.
Responses of bowhead whales to sonobuoy impacts
Authors: Greene Charles R. , Koski William
Published: October 2005

Air-dropped sonobuoys have been used to record sounds near bowhead whales since 1979. They have not been observed to react to sonobuoys dropped 0.5–1 km from them during numerous studies conducted during the summer; however, on a few occasions they have reacted to sonobuoys air dropped near them during their spring and fall migrations. The sound from an AN/SSQ-57A sonobuoy impact was recorded at distance 800 m in water 90 m deep in the western Beaufort Sea. Hydrophone depths were 3 and 18 m. Analysis of the impact signature and sound transmission paths revealed two reflected paths: one with a single bottom reflection and the second with two bottom reflections. A third arrival, occurring later and very weak, was also noted. The presence of a strong, low-frequency sound source nearby led us to high-pass filter the signal at 1 kHz. Analysis of the received signal amplitude, corrected for spreading loss, revealed a peak source pressure level of 211 dB re: 1 μPa/m. The positive acoustic impulse at a distance of 1 m, assuming no positive pulse spreading and scaling the pressure for spherical spreading, is estimated to be 5.55 Pa/s. The overall sound duration of the single bottom bounce arrival was 0.9 ms. [Work supported by MMS.]
Acoustics of fluid-filled pipes in fluid-filled bores
Authors: Rao V. N. Rama
Published: October 2005

The acoustic transmission characteristics of fluid-filled elastic pipes inside fluid-filled bores in an infinite elastic solid are examined here. This model is of interest in exploratory drilling to study the propagation of drilling-related noises within the borehole and into the surrounding rock. The axially symmetric solutions in the four media (fluid inside the pipe, pipe, fluid in the annulus and the surrounding solid) are coupled through the boundary conditions. The resulting characteristic determinant is 9×9. The dispersion behaviors of the various modes are analyzed. The limiting behaviors at low and high frequencies are investigated. The cases of the surrounding solid being acoustically ‘‘fast’’ or ‘‘slow’’ compared to the fluid in the annulus are also examined.
Analytical investigation of ray chaos in an underwater acoustic system
Authors: Jiang Zhong-Yue , Pitts Todd , Greenleaf James F.
Published: October 2005

It has previously been shown that acoustic ray paths in range-dependent ocean models exhibit chaotic behavior. Most of the investigations into the ray chaos phenomenon have been primarily numerical in nature. The objective of this report is to study theoretically the existence of ray chaos in a parabolic ray system with an analytically prescribed sound-speed model consisting of a double-channel profile perturbed by a typical periodic range-dependent disturbance. The perturbed Hamiltonian ray system is studied analytically via Melnikov’s method. It is shown that, under certain conditions, ray trajectories in some regions of the Poincaré section are equivalent to trajectories of the horseshoe map no matter how small the corresponding perturbation is. These conditions are sufficient for ray chaos and easily satisfied, thus explaining why double-channel propagation is very likely to exhibit chaotic behavior. [Work supported by CA43920 NIH.]
Detection and temporal integration of brief band-limited increments and decrements within a broadband noise signal
Authors: Formby C. , Heinz M. G. , Hargus S. E. , Zeiders J. W.
Published: October 2005

Detection thresholds were measured for brief band-limited increments and decrements within a broadband noise signal. The signal consisted of three temporal regions, where the first and third regions served as markers for the increment or decrement in the second region. The first and third regions had a flat magnitude spectrum from 0–6000 Hz and were presented at an average N0=40 dB/Hz. The second temporal region contained an incremented or decremented frequency band- or notch-width selected from WN=62 to 6000 Hz, with logarithmic center frequency 2500 Hz and spectrum level S0. The temporal increment or decrement was centered within the 500-ms signal. For each WN condition, detection threshold was measured as a function of increment or decrement duration (PN=10 to 480 ms) by tracking S0 adaptively. Increments were detectable for all combinations of WN and PN, but decrements were consistently detectable only for WN≳500 Hz. Increment and decrement thresholds varied inversely with WN and PN, ranging from 13 dB for small increment WN and PN to 1.5 dB for large WN and PN. Temporal integration functions were fitted to the increment detection results. The resulting estimates of integration time constants varied inversely with WN from 67 to 7 ms. [Research supported by NIH.]
Method of determining the effective scattering cross section
Authors: Ekimov Alexander E. , Lebedev Andrey V.
Published: October 2005

This paper considers the method of determining the effective acoustical scattering cross section (Seff) for an object of arbitrary form or determining the contribution of its part into Seff. The method is based on the determination of transfer coefficients by direct and reciprocal methods and calculation of Seff using the equation (without the diagram coefficient): Seff(Ω)/S∼〈Krec2(Ω)〉F×〈K dir2 (Ω)〉V, where S is the total square of the object, Ω is the angular position of the point in the external field, 〈Krec2(Ω)〉F is the root-mean-square value of the transfer coefficient linking the force F with the acoustic field, which is determined by the reciprocal method, 〈Kdir2(Ω)〉V is the root-mean-square value of the transfer coefficient linking the vibrovelocity V on the surface of the object with the acoustic field, which is determined by the direct method. The root-mean-square values of K are determined over the necessary number of points on the surface of the object. The method proposed shows high noise protection in the experimental investigations. This method allows one to use a powerful generator and vibrator to increase the signal/noise ratio. The experimental results are presented.
Acoustic radiation from a 3-D truss: Direct global stiffness matrix modeling results
Authors: Fricke J. Robert , Chiasson Leo , Bondaryk Joseph E.
Published: October 2005

This study was initiated to evaluate the normalized sound power level radiated by a driven truss. The coincidence frequency for the beam elements was sufficiently high that global truss modes are not a concern. Radiation from local beam modes dominate the field. Acoustic radiation was modeled using a combination of the direct global stiffness matrix (DGSM) method and models based on radiation efficiency of cylindrical beams. Use of a nominal structural loss factor of 10−4 for the beam material (aluminum) overestimates the measured field by an order of magnitude. Earlier studies suggest that a combination of structural loss in the joints and multipath wave-type conversion in the truss leads to a loss factor of order 10−3. For this study experimental data were inverted to estimate a frequency-dependent loss factor, which confirms the prior estimate of order 10−3. When the DGSM based model was run with the estimated frequency-dependent loss factor, the model results match the measured data closely. A power balance shows that the structural loss factor dominates the total power dissipated in the system, as one might expect for a lightly fluid-loaded structure. [Research sponsored by ARPA/ONR.] a)E-mail: fricke@mit.edu
Timing differences between prenuclear and nuclear pitch accents in Italian
Authors: D’Imperio Mariapaola
Published: October 2005

In Italian, as in English, lexically stressed syllables can receive a pitch accent, with the last accented syllable being the most prominent (i.e., ‘‘sentence stress’’). Recent analyses based on data from northern Italian varieties describe the nuclear accent of the neutral declarative contour as falling onto the stressed syllable (H+L*), while the prenuclear one, presenting a comparatively later peak, is described as qualitatively different from the other (H*). This analysis is problematic since there is no meaning contrast between H* and H+L*. An alternative analysis of the timing differences models the shape of both accents as H*, where the high in the nuclear one is retracted due to an upcoming low tone associated with the remaining of the utterance (tonal repulsion hypothesis). A similar account has been proposed for English H* by Silverman and Pierrehumbert. This analysis predicts that the timing differences will depend on such things as the number of following stressed syllables for the L-, and also that other pitch accent types (such as the L*+H of the question) will show comparable differences. The analysis was tested by measuring the location of the fundamental frequency peak relative to the beginning of the stressed vowel in various prosodic contexts.
Evaluation of source sound pressure of an ultrasonic transducer using schlieren visualization
Authors: Tuziuti Toru , Kozuka Teruyuki , Mitome Hideto
Published: October 2005

Source sound pressure has been evaluated without disturbing the sound field using an optical method of schlieren visualization. The intensity of schlieren light is described as a function of the Raman–Nath parameter ν and increases monotonously up to ν=2.405. Experimental results showed that the light intensity on the sound beam axis changed with the applied voltage to the transducer in a similar manner to the above function. Then ν can be obtained numerically by substituting normalized experimental intensity for the function. Because the light passes through inhomogeneous sound field radiated from finite aperture, ν is affected by an integral optical effect. If the measuring position is fixed, this effect can be calculated theoretically in relation to the parameter just in front of the sound source defined as ν0. Hence the source sound pressure P0 is estimated from ν0 using transducer size, wavelength of the light, and the refractive index of medium. The upper limit for P0 of this method is 1×105 Pa in the present experiment.