bluuebunny/crossref_metadata_2025 huggingface dataset:
Influence of frequency range in the perceptual recognition of fricatives
Authors: Feijóo Sergio , Fernández Santiago , Balsa Ramón
Published: July 2002

The objective of this paper is to study the importance of various frequency bands for the identification of fricatives. Tokens were CV syllables formed by the combination of the Galician fricatives /θ,f,s,∫/ and the vowels /a,e,i,o,u/ which were pronounced in Hyperspeech form by a man and a woman. Tokens were sampled at 32 kHz and low-pass filtered with cutoff frequencies of 11, 8, 5.5, 4, and 3 kHz. Thus, the total number of tokens was 240=4 fricatives× 5 vowels× 2 sexes× 6 frequencies. Thirty-seven listeners carried out the perceptual experiments in two conditions: (1) whole fricative noise plus 100 ms of the following vowel, and (2) whole fricative noise. The results of the perceptual experiments show that as the cutoff frequency is lowered, (a) /s/ tends to be recognized as /θ/ in both conditions; (b) the fricative noise of /θ/ tends to be recognized as /f/, and (c) recognition of /f/ and /∫/ is affected to a lesser extent. Results suggest the importance of low-frequency energy in the characterization of /f/ and /∫/. Acoustic analysis is being carried out to determine if the identification of the filtered fricatives can be explained by relations among the spectral bands.
Taiwanese prenasalized stops and poststopped nasals
Authors: Pan Ho-hsien
Published: July 2002

Prenasalized stops and poststopped nasals were classified into a category of ‘‘nasal+stop’’ segments in Chinese [Chan (1987)]. In Taiwanese, which is a member of the Chinese language family, initial prenasalized stops b,g [m○b, ŋg] would change into homorganic nasals [m,ŋ] when preceded by nasals. Moreover,strong energy attacks can be observed at the release of oral closures for b,g [m,ŋ] [Pan(1994)]. This raises the issue of whether Taiwanese prenasalized stops alternate with post stopped nasals. This study investigated the alternation between prenasalized stops and post stopped nasals in Taiwanese by using nasal airflow, oral airflow, and acoustical data. Preliminary results showed that when prenasalized stops changed into nasals, there was energy damping before the release of oral closure, and rise of acoustical energy at the release of oral closure. Though not environmentally conditioned, post stopped nasals [m○b, ŋg] were free variations of homorganic prenasalized stops [mp,ŋk]. [Work supported by NSC, Taiwan.]
Babinet’s principle for elastic waves: A numerical test
Authors: Carcione José M. , Gangi Anthony F.
Published: July 2002

Babinet’s principle states that the diffracted fields from complementary screens are the negative of each other. In electromagnetics, Babinet’s principle for infinitely thin perfectly conducting complementary screens implies that the sum, beyond the screen plane, of the electric and the magnetic fields (adjusting physical dimensions) equals the incident (unscreened) electric field. A test of the principle for the elastodynamic case was made using numerical calculations, and the results demonstrate that Babinet’s principle holds quite well for complementary plane screens with contrasting boundary conditions; that is, the complementary screen of a stress-free screen is a rigid screen with openings where the original stress-free screen existed, and vice versa. The results are exact in an anisotropic SH case; for the P–SV case, the diffracted waves, PdP, SdS, PdS, and SdP satisfy the principle exactly, while the refracted waves, PdPrSc and SdPrSc, do not satisfy the principle at all (these waves are generally much smaller than the PdS and SdP waves). Diffracted surface waves also do not satisfy the principle. The numerical method is based on a domain-decomposition technique that assigns a different mesh to each side of the screen plane. The effects of the screens on wave propagation are modeled through the boundary conditions, requiring a special boundary treatment based on characteristic variables. The algorithm solves the velocity/stress wave equations and is based on a Fourier/Chebyshev differential operator.
Kronecker autoregressive and spherical harmonic ambient noise models
Authors: Barthelemy Alain C. , Willett Peter K.
Published: July 2002

This paper develops two new models for ambient noise processes which exist in the deep ocean. The first model, termed the Kronecker autoregressive model, extends existing ideas regarding the use of linear models on uniform line arrays. The paper shows that the novel parametrization accurately models physical noise processes for uniformly sampled volumetric arrays. A new proof is presented which shows that for the case of purely sinusoidal signals, the Kronecker form provides an exact model, suggesting its use for colored noise sources. The second model uses a form of the spherical harmonic expansion to develop an original linear matricial model. Additionally, this paper presents a new algorithm needed for computing the weighting coefficients involved with this model.
Power estimation of sound sources on low-speed electric trains using a deconvolution approach
Authors: To W. M. , Yung S. M.
Published: July 2002

The traditional delay and sum beamforming method has been widely used over the past two decades to locate and characterize sound sources on moving trains. However, when the total number of major and minor sources is uncertain, this method tends to overestimate the sound powers for multiple sources. The overestimation of sound powers becomes significant when the separation of the sources is very “small” for a microphone array with fixed configuration for avoiding spatial aliasing at high frequencies for broadband noise. The separation of the sources is “small” because the resulting resolution of the delay and sum beamforming method is “low” at low frequencies. For train noise characterization, the contribution from some other parts of the train would be masked due to this low resolution power at low frequencies. This paper describes a deconvolution approach which is developed to give estimates of the sound powers on a moving line source. Simulation results illustrate that sound powers can be accurately estimated by using the proposed deconvolution method at both low- and high-frequency regions. Both the delay and sum beamforming method and the deconvolution method were applied to the measured data using an array of 15 microphones.
Modeling of scattering by partially buried elastic cylinders
Authors: Fawcett J. A.
Published: July 2002

The work of a previous paper on the modeling of scattering by a partially buried cylinder is extended to allow the cylinder to have full elasticity (and not just be a fluid structure as in the previous work). The results of computations showing the effects of increasing burial on the backscattered field, both in the spectral and temporal domains, are given. A shelled and a solid aluminum cylinder are considered for a grazing angle of incidence which is subcritical.
A low-frequency feedback-controlled audio system
Authors: Mullins Joe H.
Published: July 2002

A novel closed-loop low-frequency speaker system has been designed, built, and tested which provides the control signal through the use of a single pressure-sensitive sensor internal to the speaker enclosure. This technique allows the use of one or more passive radiators in the same radiating system while ensuring that the control signal accurately reflects the total radiation from all elements. The nature of the sensor used also tends to correct for nonlinear distortion caused by finite cabinet volume, and the use of a passive radiator allows the production of a power level well beyond that produced by the same speaker in a sealed-box alignment. Feedback factors of 30 to 50 dB have been achieved using conventional speaker elements and active analog correcting networks. The effects of acoustical delay, placement of speaker and sensor, and direct acoustical radiation on the sensor are discussed, and results described. For purposes of subjective evaluation, conventional mid-range and high-frequency elements have been added to the system, and a stereo pair has been constructed and measured acoustically.
Further evaluation of a model of loudness perception applied to cochlear hearing loss
Authors: Moore Brian C. J. , Glasberg Brian R. , Vickers Deborah A.
Published: July 2002

This paper describes further tests of a model for loudness perception in people with cochlear hearing loss. It is assumed that the hearing loss (the elevation in absolute threshold) at each audiometric frequency can be partitioned into a loss due to damage to outer hair cells (OHCs) and a loss due to damage to inner hair cells (IHCs) and/or neurons. The former affects primarily the active mechanism that amplifies the basilar membrane (BM) response to weak sounds. It is modeled by increasing the excitation level required for threshold, which results in a steeper growth of specific loudness with increasing excitation level. Loss of frequency selectivity, which results in broader excitation patterns, is also assumed to be directly related to the OHC loss. IHC damage is modeled by an attenuation of the calculated excitation level at each frequency. The model also allows for the possibility of complete loss of IHCs or functional neurons at certain places within the cochlea (“dead” regions). The parameters of the model (OHC loss at each audiometric frequency, plus frequency limits of the dead regions) were determined for three subjects with unilateral cochlear hearing loss, using data on loudness matches between sinusoids presented alternately to their two ears. Further experiments used bands of noise that were either 1-equivalent rectangular bandwidth (ERB) wide or 6-ERBs wide, centered at 1 kHz. Subjects made loudness matches for these bands of noise both within ears and across ears. The model was reasonably accurate in predicting the results of these matches without any further adjustment of the parameters.
Three apparently paradoxical results of sound radiation theory
Authors: Junger Miguel C.
Published: July 2002

Three important generally accepted theoretical results challenge physical interpretation either by being apparently paradoxical or by embodying a surprising coincidence. The first two results bear on the radiation loading of effectively infinite force- or line-excited elastic plates; the third result bears on the radiation loading of an unbaffled rigid piston.
The effects of differences in burst duration and rise/decay time on auditory apparent motion in the free field
Authors: Span Sherry A. , Strybel Thomas Z.
Published: July 2002

This study examined the effects of differences in burst duration and rise/decay time between the lead and lag stimuli on auditory apparent motion (AAM: the illusion of motion produced by the proper timing between two stimuli). Three standard burst durations (25, 100, and 400 ms) with 0-ms rise/decay times were each employed with three stimulus onset asynchronies (SOAs) that produce continuous motion for lead and lag stimuli of equal durations. During a trial block, five variable burst durations (25, 50, 100, 200, and 400 ms) and two rise/decay proportions (.00 and .15) were tested. On half the trials the standard burst was presented first. The signals were high-pass noise, and they were presented from two loudspeakers at ±20° azimuth. AAM was heard most often when the duration of the lead and lag stimuli were equivalent. The SOAs that produced AAM depended on the duration of the lead source only. No significant effects of rise/decay proportion were observed.
Visual and acoustic surveys of whales: A Monte Carlo model
Authors: Vigness Kathleen J.
Published: July 2002

There has been much interest lately in decreasing the high variability of cetacean abundance estimates by coupling passive acoustics with visual surveys. To estimate the advantages and disadvantages of a dual-mode survey, a Monte Carlo model was created. The model can reflect the species and the methodology of a particular survey. The whale population is defined by the parameters of group size and encounter rate, swimming speed and direction, surfacing rate and pattern, and source level and rate of vocalization. The survey process is constrained by the additional parameters of vessel speed, sighting probability, hydrophone array characteristics, and sound propagation. With results from the model, researchers can compare the abundance estimates and variability from a visual survey only, a passive acoustic survey only, or a dual-mode survey. They can also determine which parameters are important for a species, region, and methodology, and design their surveys accordingly. Sample results from a hypothetical minke whale population demonstrate that adding passive acoustics significantly increased the number of groups detected and decreased the time the groups were in the survey area before being detected.
Vocal discrimination of two species of manatees (Trichechus inunguis) and (T. manatus manatus) in Brazil
Authors: de Sousa Lima R. S. , Paglia A. P. , Fonseca G. A. B.
Published: July 2002

Localizations from 15 West Indian manatees (T.m. manatus) and 14 Amazonian manatees (T. inunguis) were recorded and selected for spectral analysis. Several physical parameters of these vocalizations were used to describe and compare the vocal repertoires of these two species. Five of those variables were used in a principal components analysis (PCA) in order to verify if the vocal parameters could be grouped by species. Both approaches (repertoire comparison and PCA) discriminated these two species. The main differences were in the limits of the fundamental frequency (range) of each species, 1.07 to 8 kHz for (T. inunguis) and 1.07 to 4.98 kHz for (T.m. manatus), and the number of notes per vocalization. Only the Amazonian manatee presented more than one note (1 to 4) per vocalization. The first axis of the PCA explained almost 70% of the data variation. The most important variables in this axis were related to the fundamental frequency. Therefore, the characteristics of the fundamental frequency are responsible for the discrimination of the vocal patterns of (T. inunguis) and (T.m. manatus). Such patterns are specie-specific and could be used as a taxonomic feature. [Work supported by FBPN, MacArthur Foundation, Conservation International, USFWS, and CNPq.]
A two-stage process for automatic recognition of harbor seal vocalizations
Authors: Mellinger David K.
Published: July 2002

Male harbor seals (Phoca vitulina) were continuously recorded for a year from an array of hydrophones in shallow water off the coast of central California. A two-stage automatic recognition system was used to extract sounds of interest. The first stage, fast but crude, processed the entire sound archive. It operated by: (1) making a spectrogram; (2) normalizing the spectrogram in several ways to remove some background noises and interfering sounds; and (3) detecting sounds in the 100–1000 Hz range with a minimum duration of 1 s. The second stage, slow but accurate, operated on the sounds extracted by the first stage and classified them as being either harbor seal roars or not. Classification was done by measuring a variety of acoustic characteristics—duration, frequency span, amplitude variation, etc.—in several frequency bands, and applying statistical pattern recognition techniques to the resulting feature vectors. Training data consisted of 1011 roar examples and 850 nonroar sounds. Recognition accuracy greater than 95% was achieved, with the principal errors occurring because of close resemblance between seal roars and breakingwaves. These results show that acoustic monitoring combined with automatic recognition can be a viable method for continuous monitoring of populations of wild animals. a)Currently at PMEL, 2115 S. E. OSU Dr., Newport, OR 97365.
Acoustic detection of pressure in sealed drums
Authors: Costley R. Daniel , Henderson Mark
Published: July 2002

At many waste sites, transuranic (TRU), low-level, and mixed wastes are stored in 55-gallon drums. Many of these drums contain hazardous, organic wastes as well. Radiolysis or other physical or chemical processes may result in gaseous emissions inside these drums. When this happens the pressure within the drum will increase, sometimes to unacceptable levels. In more drastic cases, these emissions may produce flammable or explosive atmospheres (e.g., hydrogen from radiolysis). Currently regulatory procedures requires that each drum be individually opened and inspected for the presence of hazardous organic waste. This situation will be dangerous for workers. A nonintrusive technique has been developed which will detect an increase in pressure over ambient levels and alert workers of potential danger and greatly increase safety. When a drum lidis tapped, it vibrates at specific frequencies. It turns out that the natural frequencies of vibration of the drum lid increase as the pressure inside the drum increases. Thus the pressure within the drum can be determined by measuring the frequency at which the drum lid vibrates. Experimental results and plans to incorporate this into a simple handheld device will be discussed. [Work supported by DOE.]
Vibroacoustical nondestructive evaluation of fatigue crack size
Authors: Bouraou Nadejda I. , Tyapchenco Alexander N.
Published: July 2002

For the vibroacoustical nondestructive evaluation of the fatigue crack’s relative size the feature set, components of which are ratios of the testing object free or forced oscillations spectral amplitudes of higher and basic harmonics, is used. The features being used are functions of the relative crack size and do not depend on initial excitation intensity. The evaluation of the above-mentioned fault is produced by using the generalized likelihood ratio method. The maximum likelihood equations for using the feature set are solved. New expressions of the relative crack size estimation are obtained. The estimation of each feature is represented as the estimation set component. On the basis of the estimation set the statistics (mean, standard deviation) are computed.
Calculation of second harmonic beam patterns for ultrasonic sources of arbitrary shape
Authors: Landsberger Brian
Published: July 2002

Ultrasonic transducers with nonaxisymmetric radiating surfaces are common in both industrial and medical applications. This presentation describes a numerical technique for calculating the second harmonic field produced in the field radiated by a transducer of arbitrary shape. The angular spectrum of the source is calculated with a two-dimensional Fourier transform, a weighted integral over the source spectrum is evaluated, and the inverse transform of the result yields the second harmonic pressure in any plane parallel to the source plane. Arbitrary absorption and dispersion are taken into account. The method is similar to that used by Alais and Hennion [Acustica 43, 1–11 (1979)] to evaluate the field of a parametric array. In the present theory, however, we do not make use of the parabolic approximation. Beam steering and focusing are introduced by appropriate phasing of the source function. Calculations of radiation patterns produced by rectangular and other sources are presented.
Dispersion relations of surface waves in semi-infinite periodic scattering arrays
Authors: Caballero D. , Sánchez-Dehesa J.
Published: July 2002

Lord Rayleigh was the first in proposing the existence of surface waves in elastic systems at the end of the 19th century. Since then, people have observed elastic surface waves in very different contexts: from the huge wavelengthed seismic movements to the ultrasonic waves traveling along electronic devices. The aim of this work is to study the propagation of elastic surface waves through structures with periodically distributed inhomogeneities with cylinder symmetry. The theoretical study is based on variational tools developed by the authors. The periodicity led us to characterize the system by a band structure. In principle, those results can be used to design devices to forbid the transmission or to enhance and drive the propagation of elastic surface waves of selected frequencies. [Work supported by CICyT of Spain.]
Origins of the IWAIF model
Authors: Feth Lawrence
Published: July 2002

The intensity weighted average of instantaneous frequency (IWAIF) model evolved from early work on the processing of frequency-modulated (FM) tones suggested by Bob Bilger [Feth et al., J. Acoust. Soc. Am. 45, 1430–1437 (1969)]. This early work on FM signals led to the incorporation of Voelker’s Unified Theory of Modulation [Proc. IEEE 54, 340–353 (1966)] into the original envelope-weighted average of instantaneous frequency (EWAIF) model [L. L. Feth, Percept. Psychophys. 15, 375–379 (1974)]. This talk will describe the development of the IWAIF model and describe its application in auditory signal processing such as formant tracking in speech processing algorithms and the perception of Doppler-frequency shifts produced by moving sound sources.
Selected aspects of across-frequency processing in binaural hearing
Authors: Trahiotis Constantine , Bernstein Leslie R. , Stern Richard M.
Published: July 2002

While considering the many contributions of Dr. R. C. Bilger to knowledge concerning auditory processing, it seemed both fitting and appropriate to honor him by discussing research in binaural hearing that is consistent with key aspects of what we perceive to be his style. Accordingly, the presentation will highlight the integration of theory, measurement, and empirical observation in selected aspects of across-frequency processing in binaural hearing. The topics addressed will be: (1) lateralization as a function of bandwidth, center frequency, and interaural time/phase disparities; (2) binaural interference; (3) the incorporation of peripheral compression, rectification, and low-pass filtering in an index that accounts for binaural detection across frequency.
Determination of the complex shear modulus from torsional waves in viscoelastic bars
Authors: Jarzynski Jacek , Doane John W.
Published: July 2002

The torsional wave method for measurements of shear modulus, described by Garrett [J. Acoust. Soc. Am. 88, 210–221 (1990)], is modified by the addition of metal end pieces to the sample. This allows measurements of the shear modulus at low frequencies (100–1000 Hz) for small samples. Measurements can be made with the sample either air-loaded or water-loaded. The system is calibrated using a laser vibrometer. Data will be presented for neoprene rubber and compared with measurements of Young’s modulus using a resonant rod technique.
Maximum-likelihood estimation of wave components for structural vibrations
Authors: Halliday Peter J. , Grosh Karl
Published: July 2002

In the analysis of structural acoustic systems, it is desirable to have a robust method for estimating the wave components (complex wave numbers and the corresponding wave amplitudes) of the structural response. A technique for evaluating steady-state wave propagation on single and multiply connected beam-like structures from noisy data is developed using previous modifications of Prony’s method [K. Grosh and E. G. Williams, J. Acoust. Soc. Am. 93, 836–848 (1993)] and a new maximum-likelihood scheme. An overdetermined exponential solution is fit to evenly spaced data points from which propagating and evanescent wave numbers and their associated amplitudes are determined for each frequency of interest. In situ estimation of the elastic modulus may be accomplished by using a least-squares fit of the analytic solution of the Timoshenko beam theory dispersion relationship to the dispersion curve estimated from experimental data. Structural intensity is evaluated from the wave component and material property information. Numerical experiments have shown the effectiveness of these methods even for relatively low signal-to-noise ratios, on the order of 30 dB.
Estimation of articulatory movement and its application to speech synthesis
Authors: Huang Jun , Levinson Stephen
Published: July 2002

In this paper, two methods are investigated to estimate the movement of the vocal tract modeled by a three-dimensional articulator described by seven physical parameters. First, we use the cubic spline method to interpolate the articulatory parameters between consecutive phonemes. Second, the articulatory parameters are piecewise linearly interpolated and then passed by a cubic spline smoothing filter. The cubic spline smoothing filter is a low-pass filter with the maximum flatness property. The filter bandwidth can be easily adjusted by a control parameter. A graphical animation is built to visualize the articulatory movement of phonetic strings and to compare the results from the two different schemes. Some English phonemes are synthesized based on the articulatory model and the governing Webster equation in frequency domain considering viscous losses. An overlap-and-add method is used to convert the frequency domain spectrum to time domain speech signal. Finally, some example of the synthesized speech phonemes will be demonstrated. [We want to thank the National Science Foundation (NSF) for the support of our work.]
Detecting filled pauses in spontaneous speech
Authors: O’Shaughnessy Douglas
Published: July 2002

Practical speech recognizers must accept normal conversational voice input (including hesitations). However, most automatic speech recognition work has involved read speech, whose acoustic aspects differ significantly from speech found in actual dialogues. Hesitations, filled pauses, and restarts (after aborted utterances) are common in natural speech, yet few recognition systems handle such disfluencies with any degree of success. Among other problems, filled pauses (e.g., ‘‘uhh,’’ ‘‘umm’’), unlike silences, resemble phones as part of words in continuous speech. The work reported here further develops techniques to allow identification of filled pauses. A distinction is made between disfluencies in actual dialogs (e.g., in the Switchboard database of natural telephone conversations, which have poor recognition rates so far) and simulated ones (e.g., the ATIS Wizard-of-Oz-style database of airline travel inquiries). It appears that speaking with actual people influences disfluencies, e.g., filled pauses tend to be shorter and more variable in pitch patterns, although unfilled pauses adjacent to filled ones remain important in both styles. While most automatic recognition methods rely entirely on spectral envelope (e.g., low-order cepstral coefficients), identifying hesitation phenomena seems to require use of fundamental frequency and duration in addition to such spectral parameters.
Speaker verification performance comparison based on traditional and electromagnetic sensor pitch extraction
Authors: Gable T. J. , Ng L. C. , Burnett G. C. , Holzrichter J. F.
Published: July 2002

This work compares the speaker verification performance between a traditional acoustic-only pitch extraction to a new electromagnetic (EM) sensor based pitch approach system. The pitch estimation approach was developed at the Lawrence Livermore National Laboratory (LLNL) utilizing Glottal Electromagnetic Micropower Sensors (GEMS, also see http://speech.llnl.gov/). This work expands previous pitch detection work by Burnett et al. [IEEE Trans. Speech and Audio Processing (to be published)] to the specific application of speaker verification using dynamic time warping. Clearly, a distinct advantage of GEMS is its insensitivity to acoustic ambient noise. This work demonstrates the clear advantage of the GEMS pitch extraction to improve speaker verification error rates. Cases with added white noise and other speech noise were also examined to show the strengths of the GEMS sensor in these conditions. The EM sensor speaker verification process operated without change over signal-to-noise (SNR) conditions ranging from −20 to −2.5 dB; the acoustic algorithms became unusable at SNR exceeding −10 dB. [Work supported by NSF and DOE.]
Backward propagation algorithms for image reconstruction: Signal processing, algorithm architecture, and applications
Authors: Lee Hua
Published: July 2002

Traditionally, image reconstruction algorithms were developed, following the design and configurations of the data-acquisition systems. So, the algorithms are typically special-purpose and system-specific. Consequently, there is a lack of consistency in terms of algorithm architecture, organization, and performance. In this paper, we present a unified framework for algorithm design and development. This allows us to implement image reconstruction for various data-acquisition configurations including active or passive sensing, linear or circular receiving apertures, CW or wideband illumination, and monostatic or bistatic formats, based on a single theoretical framework in an organized manner. The computation schemes for both linear and circular apertures will be discussed in detail. The layered backward propagation technique, as the main processing modality, provides the flexibility for dynamic updating for changes of propagation parameters. In addition, we illustrate parallel processing and recognition as integrated components for the algorithm structure. The presentation includes the theoretical background on signal processing for image formation, and overview of algorithm architecture for various configurations, a discussion on computation complexity and commonality, and several applications.
Development of hearing-aid near-field EMI testing
Authors: Candiago Marco
Published: July 2002

With the introduction of digital cellular telephone technology, cellular telephone interference to hearing aids is now a noticeable problem for both hearing-impaired digital cellular phone users and bystanders. Previously, bystander hearing-aid immunity levels (far field) were defined by the IEC committee and are contained in the current IEC 118-13 hearing-aid electromagnetic immunity standard. With the realization that an up-scaled far-field test would be insufficient to provide an accurate correlation of digital cellular telephone usability, the ANSI C63.19 working group developed a near-field test procedure using calibrated dipole antennas. This paper discusses the development of near-field testing, broken down in the following topics: (1) digital cellular phone interference and spectral properties; (2) rf interference coupling into the hearing aid; (3) far-field testing and rationalization behind the use of 1-kHz 80% AM modulation to quantitatively determine hearing-aid immunity characteristics; (4) near-field simulation using dipoles and antenna calibration; (5) near-field repeatability testing; and (6) preliminary results of near-field hearing-aid interference measurements. The result of these developments has led directly to the formulation of the near-field test procedure described in Sec. 5 of the currently developing ANSI C63.19 standard.
Demonstration of Doppler effects and sound-speed measurement using an analog frequency-to-voltage (FM) detection circuit
Authors: Huegel Angela K. , Naff Michael S. , Korman Murray S.
Published: July 2002

An audio frequency FM detector has been designed and built to measure very small Doppler shifts for classroom demonstrations. This detector uses a limiter, a series L−C phase-shift network, two sine-to-square wave circuits, a multiplier and low-pass filter. One can easily measure a deviation of 1/20 Hz in a 4000-Hz carrier. The apparatus consists of a stationary piezo-tweeter radiating a tone near a microphone that translates with periodic motion, using a motorized disk and linkage mechanism, which drives a translation stage. The output of the mic goes to the FM detector which outputs to a digital scope or meter for display. Calibration is extremely linear and is performed using the digital frequency readout on the generator. From a measure of the peak to peak Doppler shift Δf (from the periodic Doppler shift signal versus time), the source frequency f, the radius R of the disk, and the period T, one can predict the sound speed to be C=(f/Δf)(4πR/T). Here, an approximation involving the linear velocity has been made. Results for C are typically within 1% of the expected value for air at room temperature.
Physical modeling of the piano: First results for a simple monochord
Authors: Giordano N.
Published: July 2002

Initial results are presented for a project aimed at constructing a physical model of the piano. Newton’s laws are used to calculate the motion of the hammer, strings, soundboard, and room air, and thereby compute the sound produced by the instrument from ‘‘first principles.’’ In this talk we focus on a prototype instrument consisting of a monochord with a small soundboard. Modeling results for this geometry are described, and compared with experimental results for a laboratory version of the instrument. If possible, calculated tones will be presented. [Work supported by NSF Grant PHY-9722031.]
Sound production by a vibrating piano soundboard: Theory
Authors: Jiang Minghui , Giordano N.
Published: July 2002

We describe a theoretical study of the sound generated by a piano soundboard. The soundboard model, which is similar to that developed recently by our group, is driven harmonically at a point on one of the bridges, and the resulting vibrational motion obtained by numerical solution. This soundboard is situated in a numerical room, and the resulting room pressure is calculated with two different methods. In one approach the pressure throughout the room is obtained by solution of the appropriate coupled equations for the air pressure and velocity. In the other treatment every point on the soundboard is treated as a simple source, each of which contributes to the pressure at a specified listening location. The results of these two approaches are compared to each other, and to recent experiments by our group. [Work supported by NSF Grant PHY-9722031.]
Acoustics of flute sound in laboratory experimental conditions
Authors: Cossette Isabelle , Thorpe C. William
Published: July 2002

Experimental studies of musical performance are often performed in laboratories with sometimes invasive apparatus. A concern often raised by musicians is that recordings obtained in such laboratory conditions may not reflect what would occur in a real performance. This study aims to provide an assessment of the extent to which data acquired in an invasive experiment, involving the physiology of flute performance, represents what would occur in actual performance. Five professional flutists, playing on the same flute, performed four well-known pieces of the flute repertoire in invasive and noninvasive conditions. The invasive condition consisted of apparatus on the flute head joint to measure the jet velocity and lip aperture, and trans-nasal catheters to measure respiratory pressures. The noninvasive condition occurred before the apparatus had been attached, but with other conditions kept as consistent as possible. The acoustic output of the flute was recorded and characterized by measuring the fundamental frequency and relative magnitudes of the harmonics. The results did not show any significant change in the acoustic parameters overall, although several subjects exhibited some differences (<6 dB) in harmonic amplitudes between the conditions. We therefore conclude that, in professional players at least, performance is not unduely compromised by experimental constraints.
Analysis of cymbal vibrations: Lyapunov spectrum and route to chaos
Authors: Touzé Cyril , Chaigne Antoine
Published: July 2002

A new step in the analysis of cymbal vibrations is presented. The goal of this analysis is to gain a better understanding of the underlying mechanisms that govern the sound of these instruments. In a previous work, geometrical invariants (correlation dimension and percentage of false nearest neighbors) were extracted from the reconstruction of the cymbal vibration in phase space [Touzé et al., Proceedings of ISMA’98, Leavenworth (1998)]. Here, the Lyapunov exponents of the system are computed in order to assess the presence of chaos and to quantify it. The method used consists of approximating the tangent flow of the trajectory in the reconstructed phase space. An algorithm which uses a higher-order polynomial approximation has been developed. This algorithm is validated on a number of well-known theoretical time series (Henon map, Duffing equation, etc.). The Lyapunov spectrum of the cymbal shows, in particular, the convergence of the greatest exponent, which is a strong indication for the presence of chaos. A closer analysis reveals a kind of Ruelle–Takens scenario for the transition from linear to chaotic motion: Power spectra and Lyapunov exponents show that mode-locking and quasi-periodicity are present in the vibrations just before the chaotic regime.
Musical acoustics in the Twentieth Century
Authors: Weinreich Gabriel
Published: July 2002

Of humankind’s material creations that developed without benefit of logical understanding, musical instruments may well be the most complex and sophisticated. Although some of the greatest theoretical minds of the nineteenth century (such as Helmholtz and Rayleigh) applied notable efforts to the study of musical instruments, that period lacked the experimental tools to match the virtuosity of its theoreticians; so that it was not until the twentieth century, and more particularly, the period that overlaps the life of the Acoustical Society of America, that the field of Musical Acoustics was truly able to blossom forth. Specifically, ASA’s first quarter century saw the development of sophisticated vacuum-tube technology (powerfully stimulated by World War II); the second produced solid-state electronics; and the third brought personal computers to every scientist’s desk and made digital methods part of everyday life. But, unlike the situation in other fields, modern electronic methods not only made the physics of musical instruments more accessible, they themselves invaded their subject matter in the form of electronic music, giving rise to a totally new kind of symbiotic relationship. This talk will draw upon the annals of JASA to summarize that fascinating period of history.
Propagation of a monochromatic sound wave in a turbulent atmosphere near the ground: Theory and laboratory experiment
Authors: Ostashev Vladimir E. , Wasier Julie , Blanc-Benon Philippe , Juvé Danile
Published: July 2002

The theory of sound propagation in a turbulent atmosphere near the impedance ground is presented. It is assumed that the monochromatic source and receiver are located above the ground, and there is no temperature or wind velocity stratification in the atmosphere. A formula for the mean square sound pressure at the receiver is obtained and studied analytically and numerically as a function of different parameters of a problem: the variance and integral length scale of atmospheric inhomogeneities, the horizontal distance between the source and receiver, the value of impedance of the ground, etc. Theoretical results obtained are also compared with data from laboratory experiment modeling sound propagation in an atmosphere with temperature fluctuations near the ground. The experiment was carried out at Ecole Centrale de Lyon. The agreement between theoretical and experimental results is generally good. [This material is based upon work supported in part by the U.S. Army Research Office under Contract No. DAAG55-98-1-0463, and NRC-ETL Research Associateship.] a)On leave from New Mexico State Univ.
Computational methods for an inverse problem in acoustics
Authors: DeLillo Thomas , Isakov Victor , Valdivia Nicolas , Wang Lianju
Published: July 2002

The problem of computing normal velocities on the boundary of a region from pressure measurements on an interior surface is considered. The pressure satisfies the Helmholtz equation and is represented by a single layer potential. Once the density function is found, the normal velocities can be easily computed by applying a second kind of integral operator. The problem of solving for the density function from the interior pressure measurements is ill-posed. Regularization methods using the singular value decompostion and iterative methods, such as the conjugate gradient method for the normal equations, are compared for pressure data with noise. [Work supported by NSF and Cessna Aircraft Company.]
Speaking fundamental frequency of young adult Arabic men during oral reading and spontaneous speaking in both Arabic and English languages
Authors: Abu-Al-Makarem Ali , Petrosino Linda
Published: July 2002

Currently, there is a paucity of normative data on speech and voice characteristics of different linguistic and ethnic groups. In particular, there is no available published data on the speaking fundamental frequency (SFF) characteristics of the voice of the Arabic population. The purpose of this study was to obtain preliminary data on the SFF of a group of normal speaking, young adult Arabic males. Fifteen, native Arabic, adult men served as subjects and received the identical experimental treatment. Four speech samples were collected from each subject (Arabic reading, Arabic spontaneous speech, English reading, and English spontaneous speech). Results showed the mean and SD of SFF are ([146,9, 15.4], [145.8, 13.8], [149.1, 12.6], and [145.5, 12.0]), respectively. No significant differences were found in the mean SFF between language and type of speech, nor between languages. A significant difference in the mean SFF was found between the type of speech (F1,14=5.51, p=0.03). Reading was significantly higher than speech. Also, Arabic men in this study had higher SFF values than previously reported for young adult males of other ethnic groups [H. Hollien and B. Jackson, J. Phonetics 117–120 (1973); A. Hudson and A. Holbrook, J. Speech Hear. Res. 197–201 (1992)]. For Speech Communication Best Student Paper Award.
Speaking rate, fluency, and accentedness in monolingual English and bilingual Czech, French, and Japanese speakers
Authors: Burgess Cliff S.
Published: July 2002

Speaking rates (SRs) and articulation rates (ARs) were calculated for monolingual English speakers and advanced ESL learners with Czech, French, and Japanese L1s; 64 speakers in total. Data were based on two reading tasks, a cartoon description, and a personal anecdote. Bilingual speakers did these tasks in both English and their respective L1s. Short excerpts were then taken from each English language sample (264 tokens plus 64 repeats) and played to 10 native English speakers who made scalar judgments on fluency and degree of accentedness. Judgments showed a wide range of correlation with SRs and ARs that varied among language group and task type. The influence of pause type, length, and frequency was also examined. The results suggest that cues to accent and fluency perception are multidetermined and that listeners weight these cues differentially. [Work supported by SSHRC.]
Focus on Japanese-accented English interrogatives and declaratives
Authors: Knudsen Laura G.
Published: July 2002

This paper presents the results of a study of focus realization in Japanese English intonation, offering a modified and expanded replication of an earlier study [M. Ueyama and S.-A. Jun, UCLA Phonet. Lab Wkg. Paper No. 94 (1996)]. Fourteen subjects and three controls read two sets of test interrogatives and declaratives, one with fixed utterance length but focus in varying locations, and one with fixed focus location but varying utterance length after focus. F0 and absolute time measurements were taken at four or more points in each utterance, yielding measurements of focus contour, high- or low-plateau length, and utterance-final contour. Each test item was also analyzed qualitatively and coded phonologically using the ToBI transcription system. The current study includes twelve female and male EFL subjects in four levels, plus two more advanced ESL subjects in a fifth level. Results showed first of all that Japanese acquirers of English had large gaps in their inventory of intonational types. More advanced speakers did show evidence of having acquired aspects of the English intonational contours; however, not employing them as do native English speakers.
Reconstruction of Formula 1 engine instantaneous speed by acoustic emission analysis
Authors: Rizzoni Giorgio
Published: July 2002

This talk presents some results of a method aimed at extracting instantaneous engine speed information from acoustic emission measurements obtained from Formula 1 (F1) vehicles during a race, using joint time-frequency analysis methods. The analysis method used in this work is applied to acoustic emission data recorded by the microphone of the in-car cameras mounted on F1 vehicles. The data analyzed were acquired during the 1998 Grand Prix of San Marino (Imola), and pertain to the performance of the Ferrari and McLaren–Mercedes vehicles. The analysis presented in the paper is based on data acquired in three different sections of the Imola circuit: the starting straightaway and two curves, to highlight the capabilities of the method. The result of the analysis demonstrates that it is possible to estimate a number of useful variables from sound measurements using joint time-frequency analysis methods. These estimates are related to engine performance (e.g., engine speed and its acceleration, top engine speed), to engine architecture (e.g., gear ratios), to driving strategy (e.g., shifting strategy), and to vehicle performance (tire adhesion, aerodynamic behavior). The analysis includes validation against engine speed data obtained via telemetry measurements.
Backscattering enhancement mechanisms for thin tilted circular plates in water
Authors: Hefner Brian T. , Marston Philip L.
Published: July 2002

A thin tilted circular glass plate in water was illuminated with high-frequency sound. When the angle of incidence along a diameter of the plate corresponds to the extensional and flexural wave coupling angles, there is an enhancement in the backscattered response. At each of these angles, the largest response arises from a leaky wave which travels along the diameter of the tilted plate and reflects from the plate edge. A similar enhancement was studied previously for the scattering of sound from the flat end of a tilted solid cylinder and, in that case, the reflection of the leaky Rayleigh wave was modeled using methods in ray theory [K. Gipson and P. L. Marston, J. Acoust. Soc. Am. 104, 1754 (1998)]. For the circular plate, in addition to the scattering associated with this central ray, further enhancement was found near the extensional wave coupling angle associated with rays traveling along off-center paths. The mechanism for this response is believed to depend upon multiple reflections around the rim due to the geometry of the scatterer and may have a significant effect on the overall response of the scatterer. [This work is supported by the Office of Naval Research.]
Further applications of time-frequency methods in acoustics
Authors: Gaunaurd G. C. , Strifors H. C.
Published: July 2002

Time-frequency (t-f) methods have been repeatedly established as effective tools to successfully analyze a variety of signatures from many types of sources or scatterers. Many such signatures have been acoustic in nature, and we have studied these most extensively in the past. We have presented several of these examples before [viz., G. C. Gaunaurd and H. C. Strifors, J. Acoust. Soc. Am. 104, 1746, 136th ASA Mtg. (1998)]. Here, we will cover acoustical applications that we have analyzed, but which were not adequately covered in 1998. We will also outline additional ones—more extensively discussed in Appl. Mech. Rev. 50, 131–149 (1997)—and a recent radar application [i.e., ibid. IEEE Trans. Antennas Propag. 46, 1252–1262 (1998)]. In this case, the present t-f processing was shown capable of successfully identifying an air target that had been covered with a dielectric RCS-reducing layer. We believe that this general approach, jointly with the way we have used it to physically interpret broadband sonar/radar signatures, can lead to straightforward implementations for many practical target-classification problems associated with many types of sensors. [Work partially supported by the ILIR Program of the authors’ Institutions and the ONR.]