bluuebunny/crossref_metadata_2025 huggingface dataset:
Characterizing nonadiabatic ocean acoustic environments
Authors: Field R. L. , Ruppel T. H. , Broadhead M. K.
Published: October 2005

In range-dependent environments, the adiabatic assumption, which neglects discrete mode coupling, is often made. Up continental shelves and in shallow-water regions, abrupt changes in the environment make this assumption suspect. An issue in these environments is the degree to which the adiabatic assumption is valid. Transmission loss calculations using both a parabolic equation and coupled mode models show, that for steep slopes, much more energy reaches the shelf from deep water than predicted by the adiabatic approximation. This increase in shelf energy is shown to be due to discrete mode coupling. In this study, changes in energy flux with range are used as a criterion for when the adiabatic assumption breaks down. The flux is computed from a parabolic equation model in upslope environments as a function of range; frequency, and slope angle. It is shown that rapid flux variations are good predictors of when the adiabatic approximation breaks down and mode coupling becomes significant. [Work supported by ONR, Program Element 602435N, with technical management provided by the Naval Research Laboratory.]
Compound-matrix methods for computation of wave fields in horizontally layered fluid–solid media
Authors: Ivansson Sven
Published: October 2005

The wave field is decomposed into its frequency-wave-number components, and a compound-matrix ODE theory is formulated for multiregion fluid–solid media using multipoint boundary conditions. Forward propagation of compound entities is done to a matchingdepth, from which stabilized backward propagation to the receivers is performed. Wave-number integration as well as modal synthesis is covered. Source arrays and receivers may be arbitrarily located. An appropriate adjointproblem is defined and solved. The formula for modal excitation coefficients is generalized to cover leakymodes, which have an exponential increase with depth. Three methods for stable and automatic computation of modal depth functions, in solid as well as fluid regions, are proposed. In work by M. B. Porter and E. L. Reiss [J. Acoust. Soc. Am. 77, 1760–1767 (1985)] the mode shapes in the solid bottom were left aside, and the method by F. Schwab etal. [Bull. Seismol. Soc. Am. 74, 1555–1578 (1984)] involves experimentation with a cutoff depth for an artificial homogeneous half-space. For a medium composed of homogeneous layers, it is shown how efficient computations are obtained by writing the propagator matrices involved as products of sparse matrices, which are applied in sequence. Numerical stability can be achieved without artificially splitting thick homogeneous layers.
Chemical treatment of wood for musical instruments
Authors: Yano H. , Kajita H. , Minato K.
Published: October 2005

After a review of the structure of wood concerning its acoustic properties and a discussion of the merits and demerits of wood as a material for making soundboards of stringed instruments, the effects of three chemical treatments (a low molecular weight phenolic resin treatment, a resorcin/formaldehyde treatment, and a saligenin/formaldehyde treatment) on the acoustic properties of Sitka spruce wood are reported and compared with those of formaldehyde treatment. All three chemical treatments increased the specific dynamic Young’s modulus and greatly decreased damping (tan δ) without greatly increasing specific gravity. The range was almost equivalent to that with formaldehyde treatment whose superb effects have already been confirmed in the violin. With saligenin/formaldehyde treatment, the frequency dependence of tan δ could be controlled over a wide range by changing the saligenin concentration. In all treatments, hygroscopicity was reduced and dimensional stability was improved significantly.
Many-valued logic is reflected in formant location and spectral composition
Authors: Pollina Dean A. , Vakoch Douglas A. , Wurm Lee H.
Published: October 2005

Variation in the truth value of propositions has been shown to yield changes in cognitive event-related potentials (ERPs) [D. A. Pollina and N. K. Squires, Psychophys. 30, S52 (1993)]. Given the connection between the arousal of the central nervous system and peripheral systems, the same paradigm that was used to detect many-valued beliefs via ERPs was used to detect degree of belief by an acoustical analysis of the voice. Degree of belief was manipulated by having subjects solve a murder mystery, after which they learned facts associated with three people: (1) the person they most suspected of committing the murder, (2) the person they judged next most likely to be the murderer, and (3) the person they knew could not be the murderer. Articulatory precision of subjects’ utterances was measured by formant location and spectral composition [F. J. Tolkmitt and K. R. Scherer, J. Exp. Psychol. Human Percept. Perform. 12, 302–313 (1986)]. These measures were related to the degree to which subjects believed each statement to be true.
An efficient broadband normal-mode model for acoustoelastic ocean environments
Authors: Westwood Evan K.
Published: October 2005

A recently developed normal-mode model [E. K. Westwood etal., J. Acoust. Soc. Am. 95, 2908(A) (1994)] for acoustoelastic ocean environments has been extended to perform broadband calculations. The set of complex mode eigenvalues at the highest frequency of interest is first found. Then each eigenvalue kn is tracked as a function of frequency down to the lowest frequency of interest. Analytic derivatives with respect to frequency dkn/dω are used to extrapolate in frequency, and step sizes are adjusted adaptively. When the step size exceeds the FFT bin width, linear interpolation of eigenvalues may be used. Examples of broadband calculations are given. [Work supported by the ARL Internal Research and Development Program.]
Effect of component-relative-entropy on the discrimination of simultaneous tone complexes
Authors: Lutfi Robert A. , Doherty Karen A.
Published: October 2005

Results are reported from a number of experiments investigating the effects of stimulus uncertainty on the discrimination of simultaneous multitone complexes. With few exceptions the data were well described by a model in which average threshold for the detection of a change in one or more components varies with the components’ relative entropy (CoRE) in the complex [R. A. Lutfi, J. Acoust. Soc. Am. 94, 748–758 (1993)]. In different experiments, thresholds for the detection of a change in target level were found to be linearly related to CoRE for manipulations in the number of targets, the number of context tones, and the relative degree of level perturbation on tones from trial to trial. The slopes of the functions did not change with measures taken at different points on the psychometric function, and in most cases were close to the theoretical value of b=−1.0, as predicted by strict application of the model. Also as predicted (but for the exception of one listener) increasing the level of the target beyond that of the context had little or no effect on threshold. A correlational analysis of the trial-by-trial data revealed a tendency of listeners to give negative weight to frequencies immediately above the target, and to adjust weights in response to changes in the relative degree of level perturbation. Though inconsistent with the basic premise of the model, the variation in weights was not so great as to materially affect the predictions for mean thresholds.
Auditory brainstem response (ABR) peak amplitude variability reflects individual differences in cochlear response times
Authors: Don Manuel , Ponton Curtis W. , Eggermont Jos J. , Masuda Ann
Published: October 2005

Previously, it was shown [Don et al., J. Acoust. Soc. Am. 94, 2135–2148 (1993)] that cochlear response times are gender specific and about 13% shorter in females than in males. It is also suggested that one of the possible reasons click-evoked auditory brainstem response (ABR) waveforms recorded from females are better defined and have larger amplitudes than those of males is due to a sex difference in cochlear response times leading to better synchronization of the cochlear output across the frequency regions. Variability in cochlear response times would also lead to variability in click evoked ABR amplitudes. The high-pass noise masking derived ABR technique was used to investigate the effect of normalizing the individual temporal variability at the neural and cochlear levels. This involved adjusting for differences in neural conduction time (I–V delay) by a compression or expansion of the derived ABR waveforms and by adjusting for differences in cochlear response times by a shift of the derived ABR waveforms. A summation of the compressed and shifted ABRs results in a normalized unmasked ABR waveform that can then be compared for amplitude variability with the unprocessed unmasked ABRs. Compensation for the neutral I–V variability had little effect while compensation for cochlear response times, particularly the delay between the 5.7- and 2.8-kHz regions, greatly affected the amplitude of wave V of the compounded ABR. This work provides a better understanding of the significant relationship between cochlear response times and variability of the ABR peak amplitudes.
Effect of dynamic fluid compressibility and permeability on porous piezoelectric ceramics
Authors: Lacour Olivier , Lagier Michel , Sornette Didier
Published: October 2005

The ac piezoelectric properties of porous ceramics in the presence of compressible viscous fluids filling the pores inside the ceramics are studied. Above the Biot frequency fB, the fluid decouples from the solid and does not significantly affect the piezoelectric properties. Our study which is both experimental and theoretical focuses on the frequency domain below the Biot frequency, in which the fluid is in principle strongly coupled to the solid matrix for oscillations parallel to solid–fluid boundaries. Two regimes are found: (1) For frequencies smaller than a cross-over frequency fc∼(β/η)L−2, where L is the thickness of the sample, β the fluid bulk modulus, and η its viscosity, the piezoelectric coefficients dij are close to those of the dense nonporous ceramics weighted by the filling factor (1−φ); (2) for frequencies larger than fc, the dij tend to those of the empty porous ceramics for fluid bulk modulus significantly smaller than that of the ceramics, as is the case in our experiments. These observations are explained in terms of a frequency-dependent penetration of the acoustic fluid pressure inside the porous ceramics as a result of the interplay between fluid compressibility and viscosity.

The nature of the decoupling between solid and fluid at frequencies larger than fc but still smaller than fB is for motions, not parallel to their common boundaries (for which the Biot mechanism applies), but perpendicular to their common boundaries. This decoupling occurs due to the finite compressibility of the fluid and in details is controlled by the tube-cavity topological structure of the porous medium. The dependence of the hydrostatic piezoelectric coefficient dH=d33+d31+d32 as a function of frequency observed experimentally is compared to an effective medium theory for the piezoelectric properties coupled to a permeability model. The experimental dependence of the cross-over frequency fc on sample sizes and on the fluid viscosity are in good agreement with our theory. The study shows that it is thus possible to obtain the desirable property of a high piezoelectric coefficient dH, characteristic of light porous piezoelectric ceramics, and good mechanical properties under applied high hydrostatic pressure with the use of an internal fluid which allows the pressure to equilibrate without destroying the piezoelectric properties of the porous ceramics.
A model of a confocal ultrasonic inspection system for interfaces
Authors: Yogeswaren E. , Harris John G.
Published: October 2005

A mathematical model describing how a confocal arrangement of two focused ultrasonic transducers is used to interrogate a complex interface between two materials by scanning the focal point across the interface is outlined. A complex interface is one that has roughness and partial contact at several length scales, many of which are equal to or smaller than the compressional or shear wavelength in the material. When the focused ultrasound strikes such an interface, though the focal region be small, multiple scattering takes place among the scatterers within and adjacent to the focal region, making it unclear exactly how the interface is being sampled. To clarify this issue a specific interface model, consisting of a planar array of multiple small cracks having arbitrary lengths and spacings, is used. This interface is interrogated by a focused, antiplane shear wave. While the antiplane assumption makes it scalar, the model shows that what is measured are the multiply scattered signals averaged over the aperture of the transducer, and that the dominant contribution comes from the scattered signals that phase match to the interrogating signal. Explicit expressions relating the modeled reflected and transmitted signals to the convolution of the incident wave field with the crack-opening displacements at the focal region are given. Numerical examples are worked out for similar and contrasting materials on each side of the interface.
Shallow-water acoustic/geoacoustic experiments at the New Jersey Atlantic Generating Station site
Authors: Badiey Mohsen , Jaya Indra , Cheng Alexander H.-D.
Published: October 2005

Shallow-water acoustic experiments were conducted at the Atlantic Generating Station site on the New Jersey Continental Shelf as a part of a long-term study of the shallow-water physics in this region. A vertical hydrophone array was used to assess the dispersion characteristics of the broadband acoustic field. A geoacoustic data set including bottom impedance profiles were constructed based on previous measurements of geological parameters from sediment core data. Experimental results show that the acoustic wave-field dispersion is dependent on the azimuth angle. This dependency may be attributed to several factors relating to topography, range, and depth features.
Optimum frequency range for partial discharge detection in high-voltage transformers using ultrasonic methods
Authors: Kurusingal Joseph
Published: October 2005

Partial discharge (PD) erosion of insulation is one of the deteriorative mechanisms in transformers and high-voltage plants leading to early failure. Ultrasonic detection techniques are used regularly for location of PDs when electrical method [IEC Pub. 76/IEC 270] used in routine tests or DGA results indicate the presence of PD. The technique uses either externally or internally mounted ultrasonic transducers (J. Unsworth, J. Kurusingal, and R. E. James, Proceedings of ICPADM 94, IEEE). Unfortunately ultrasonic technique requires expert supervision to distinguish PD signal patterns from some typical transformer noise patterns. Moreover, transducers are selected based on commercial availability or for their ability to reject radio interference. A study was conducted to determine the optimum frequency range for PD detection to give maximum range sensitivity and acoustic interference rejection. This paper presents transformer noise recorded in the field as well as in the laboratory with externally and internally mounted transducers. The data records are Fourier analyzed and transducer response corrected to obtain the acoustic noise spectra. It is found that there is an upper cutoff frequency for transformer acoustic noise which is around 75 kHz and signals originating from PD have frequency components much beyond this range. Factors which influence range sensitivity are also discussed.
A dual-transducer acoustic time domain correlation technique for instantaneous blood flow velocity measurement
Authors: Kim E. J. , Yoon S. W.
Published: October 2005

A dual-transducer acoustic time domain correlation technique has been developed to measure instantaneous blood flow speed and direction. The experiment was performed with an artificial blood flow system in a water tank. The volumetric flow velocity as well as the instantaneous flow velocity at any sites in an artificial blood vessel can be measured with the dual-transducer technique. In the steady flow measurements, the volumetric flow rates estimated from the dual-transducer technique were within the error range of 8% from those of the hydrodynamic flow measurements. Even in the turbulent flow measurements, the instantaneous flow speed and direction can be determined at any sites in the vessel with the dual-transducer acoustic time domain correlation technique.
The dimensional structure of the lexicon
Authors: Wurm Lee H. , Vakoch Douglas A.
Published: October 2005

Past research indicates that the lexical access of emotion words is influenced by the dimensions of evaluation and activity. Increased reaction time to a lexical decision task using emotion words and nonwords was predicted by low loadings on evaluation and activity as determined by ratings on semantic differential scales [L. H. Wurm and D. A. Vakoch, J. Acoust. Soc. Am. 95, 2974 (1994)]. The current study examines the extent to which evaluation, activity, and potency influence lexical access of words more representative of the entire lexicon [C. E. Osgood, J. Pers. Soc. Psychol. 12, 194–199 (1969)]. Implications for a dimensional model of the structure of the lexicon are discussed.
Intelligibility training for foreign-accented speech: A preliminary study
Authors: Rogers Catherine L. , Dalby Jonathan M. , DeVane Gladys
Published: October 2005

A computer-based speech training system, the Indiana Speech Training Aid (ISTRA), has been shown to be clinically effective for improving the speech of hearing-impaired and phonologically disordered individuals [Kewley-Port etal., Clin. Ling. Phon. 5, 13–38 (1991)]. The potential value of speech recognition technology for the training of foreign-accented speech, using an overall measure of speech quality as feedback, was assessed. Phonological errors in English spoken by two native Mandarin speakers were analyzed and several training targets selected. One consonant and one vowel contrast each were selected for training using ISTRA, and by a speech-language pathologist (SLP). Pre- and posttraining tokens were rated for quality by a listener jury. Results showed significant improvement for both the ISTRA-trained consonant /l/, and the SLP-trained consonant /θ/. However, only one of the two vowel contrasts for each speaker improved significantly. ISTRA-trained sounds which showed significant improvement did so for both trained and untrained words. Two untrained phonemes, /r/ and /U/, also improved significantly, possibly indicating a generalization of improved sounds to other phonological distinctions in English. These results are promising as regards the usefulness of speech recognition technology for pronunciation training. [Work supported by NIHDCD SBIR Program.]
Training effects for identification of synthetic speech tokens for listeners with normal and impaired hearing
Authors: Horwitz Amy R. , Turner Christopher W.
Published: October 2005

Training effects for identification of speechlike stimuli having frequency, amplitude, and temporal characteristics based upon natural consonant–vowel (CV) speech tokens were measured. Two types of dynamic speech stimuli were synthesized. Specifically, in the first type formants were represented by pure tones and in the second by sweeping three ‘‘formant’’ filters across a voicing-source spectrum, to more closely approximate natural speech. Following a brief familiarization period, normal-hearing and hearing-impaired subjects listened to and identified six CV syllables in a closed-set format. The order of presentation for the type of synthesized speech tokens was counterbalanced so that initially half of the subjects were tested with the pure-tone speech syllables while the other half were tested with the broader ‘‘voiced format’’ speech. Data were collected over the course of several months. Substantial training effects were observed for both types of speech stimuli, with all subjects ultimately achieving similar identification scores for both types of syllables. [Work supported by NIDCD].
Neuroticism and openness are predicted by fundamental frequency of speech
Authors: Vakoch Douglas A. , Pollina Dean A. , Wurm Lee H.
Published: October 2005

The fundamental frequency (F0) of speech has been used for examining both transient emotional states [D. A. Vakoch, J. Acoust. Soc. Am. 95, 2974 (1994)] and more stable personality traits. For example, low perceived pitch has been associated with attributions of high emotional stability [K. R. Scherer, Eur. J. Soc. Psychol. 8, 467–487 (1978)]. In the current study, characteristics derived from F0 were related to five personality traits as measured by the NEO-PI: neuroticism, extroversion, openness, agreeableness, and conscientiousness. A high degree of openness was predicted by a high mean F0 of an utterance (p<0.025) and by a great variability in F0 within an utterance (p<0.004). The best predictor of neuroticism was peak F0 within an utterance, with subjects scoring high on this scale having a higher maximum F0 (p<0.021). The latter finding, in conjunction with previous research on attributions of emotional stability, suggests a convergence between objective acoustical measures and subjective perceptions of personality.
Effects of speaking rate, focal stress, and distinctive vowel length on syllable-internal timing in Norwegian
Authors: Behne Dawn M. , Moxness Bente
Published: October 2005

The relative timing of consonants and vowels of a syllable can be affected by factors such as speaking rate, focal stress, and distinctive vowel length. Speaking rate and focal stress tend to have a global affect on syllable-internal timing, whereas the effect of distinctive vowel length in Norwegian is generally believed to be localized to the syllable rhyme. When these factors converge on a syllable, the syllable-internal timing simultaneously reflects their concurrent effects. The goal of the present study is to examine and characterize the concurrent effects of speaking rate, focal stress, and distinctive vowel length on syllable-internal timing of C1VC2 components. Conversations were developed in which 12 Norwegian CVCs containing /i, o, a, i:, o:, a:/ were either focused or nonfocused by the discourse. The set of conversations was produced at three speaking rates by native speakers of Norwegian. Results indicate that speaking rate and focal stress have an overall affect on the timing of components in the syllable. In addition, the effects of distinctive vowel length and focal stress on components of the syllable rhyme were found to be interdependent across speaking rates. These findings are discussed in terms of syllable- and rhyme-internal timing.
Some control experiments on a model for prominence perception
Authors: Rump H. H. , Hermes Dik J.
Published: October 2005

According to the model proposed by Hermes and Rump [J. Acoust. Soc. Am. (in press)] the prominence lent by accent-lending pitch movements is proportional to the pitch-level difference between the vowel nucleus of the accented syllable and the vowel nucleus of the preceding syllable. The pitch level on the lower declination line or baseline and the pitch level on the upper declination line or topline are thus regarded as equally important. By varying the height of the pitch levels on the baselines this assumption was tested against the alternative assumption that listeners equated pitch levels on the topline of each of the stimuli when adjusting prominences to be equal. Results showed that pitch levels on the toplines were adjusted to be more or less equal, but that the pitch-level difference between the low and the high pitch level influenced the adjusted height of the high pitch level, too. It is concluded that the proposed model makes claims about the relevance of the position of the low pitch level which is too strong. The role of pitch levels in relation to the perception of prominence will be discussed.
Spectral analysis of amplitude envelopes of bandpass filtered speech
Authors: Kong King-Leung
Published: October 2005

The amplitude envelopes of rectified bandpass filtered speech have been found to provide useful cues for speech perception [K. W. Grant, L. D. Braida, and R. J. Renn, J. Acoust. Soc. Am. 95, 1065–1073 (1994)]. An analog terminal was built to yield 25 such envelopes from filters with center (carrier) frequencies from 150 to 4850 Hz. Each envelope was then subjected to another round of bandpass filtering and rectification to yield a modulation spectrum of up to nine channels with center (modulation) frequencies from half the carrier frequency to 700 Hz. The spectra were examined for cues for the identification of voicing, fundamental frequency, and consonants. Voicing was generally characterized by the concentration of formant energy at a single carrier and modulation frequency, corresponding to the formant and fundamental frequencies, respectively. The second formant of the front vowel /i/ and nasal release sometimes exhibited bimodal modulation spectra, suggesting multiple sources of modulation. Stop consonants and fricatives were characterized by elements scattered at high carrier and modulation frequencies whose occurrences might not coincide. Some consonants could be identified with elements at specific modulation frequencies: e.g., /g/ and /j/ suggested a 700-Hz source modulating carriers whose frequencies depended on the following vowel.
Cepstrogram and its application to speech analysis
Authors: Che Chiwei , Lin Qiguang
Published: October 2005

Cepstrum coefficients (including their different variants and/or derivatives) are nowadays commonly utilized in automatic speech recognition systems. Most adaptation algorithms for robust speech recognition also make use of cepstrum coefficients. It is therefore important to have a tool which can efficiently calculate and display cepstra. A computer program has been recently developed to display the cepstrogram of speech signals. The program has the following features: (1) It computes both linear prediction based and FFT based cepstra; (2) it displays cepstrum coefficients and their derivatives as a function of time and cepstrum index, as well as cepstrum coefficients at a fixed frame (section displays); (3) it generates synthesized spectra from cepstrum coefficients. In addition, the program can be used to segment and playback speech waveforms. The program has been utilized to study variations in cepstrum coefficients caused by speaker variability and channel differences. The tool is a useful supplement to conventional spectrograms for speech analysis. (The program is coded in C and runs in the X-window environment. It is available upon request.)
The PEACC method of characterization of dynamic aspects of speech
Authors: Talley Jim
Published: October 2005

In the phonetics/speech perception community, the assertion that dynamic aspects of the speech signal are employed in robust speech decoding is not particularly controversial. Though an increasing number of studies are addressing the various dynamic cues of speech, quantitative, analytical research is hampered somewhat by a lack of established methods. Methods for studying dynamic phenomena are much less well developed than those for more static properties. This paper proposes a new method for characterizing dynamic aspects of speech, the Piecewise Exponential Approximation with Continuity Constraints (PEACC) method, to help remedy this state of affairs. As its name suggests, PEACC performs piecewise fitting of exponential segments—γeαx+β, −∞<x≤0—to the sampled signal. Dynamic programming is utilized in global sequence optimization where MSE is minimized within the solution space permitted by the constraints on continuity. This method has broad applicability and produces low-distortion fits at a specifiable level of detail; however, its principle strength, from the perspective of speech science research, is that its resulting signal transition parameters have direct, intuitive interpretations. The paper concludes with a brief examination of the results of applying PEACC to a corpus of formant track data. [Work supported by NSF.]
Possible usages of LPC error for the estimation of voice characteristics
Authors: Lee Sungbok
Published: October 2005

Using the autocorrelation method, LPC error signals were computed as differences between the original waves and the predicted waves at each time instant. Analyses of synthesized and human vowel sounds show that the shapes and spectral contents of the instantaneous error signals are almost identical with those of the voice source for synthesized vowel sounds, and that the instants of glottal closure are well defined as maximally negative peaks in the error signals for both kinds of vowel sounds. Time integration of error signals was also performed. For male voices the shapes of integrated and instantaneous error signals are found to be very similar with those of EGG and differentiated EGG signals, respectively. This result supports the possibility that open and closed phases of the vocal folds within each glottal cycle can be determined from such LPC error signals. It was also found that automatic detection of valleys (or peaks) in the integrated error signal may be a robust pitch estimation method for voiced speech segments including synthetic diplophonic sounds. These and other results suggest that these simple LPC error analyses may provide significant information about the voice source.
Site selection for a long-term acoustic study of the temporal variability in a shallow-water environment
Authors: Farrell Ted , Torres Rigo , Atkinson Cola , Duckworth Greg
Published: October 2005

A site selection study was performed to determine the best location for an environmental-acoustic measurement system to be deployed for two years in shallow water. The study searched for continental United States (CONUS) locations where the environmental parameters are sufficiently variable such that, over time, the acoustic environment at these locations will be similar to the environments found at multiple littoral regions of interest outside of CONUS. Shipping densities, sound-speed profiles, wind speed, rainfall, bathymetry, and sediment characteristics were investigated using a MATLABR based geographic information system. As part of the investigation, measures of variability were defined for each environmental variable. In addition, automatic classification of sound velocity profiles was implemented on more than 44 000 measured profiles from the areas of interest. The results of the study show that several CONUS areas have enough environmental variability to satisfy the requirements of a long-term measurement program. The system developed for this effort significantly reduced the time required to analyze and visualize large amounts of environmental data. The system also provides a means to couple environmental data into various acoustic models. [Work sponsored by ONR.]
Diffraction of acoustical waves by spatially localized inhomogeneities in oceanic waveguides
Authors: Gorsky Sergey M. , Zverev Vitaly A. , Khil’ko Alexander I.
Published: October 2005

Diffraction of the acoustic fields by spatially localized inhomogeneities in oceanic waveguides has been investigated. The formula for scattering matrix of diffracted waveguide modes has been obtained using a ‘‘short-wave’’ approximation. The spatial-frequency diffraction patterns for different locations of inhomogeneities, sources and receivers, and properties both of the spatial-localized inhomogeneities and the waveguides were investigated numerically. The possibilities of diffracted fields measurement in natural waveguides were investigated under the conditions of physical modeling in the ultrasonic frequency region. The significance of physical modeling methods for investigation of diffraction phenomena in ocean was demonstrated. It was shown that the efficiency of diffracted field observations can be increased using results of calculation and measurements under model conditions. The spatial-temporal filtration by horizontal and vertical arrays and by matching processing of complex pulse illumination signals were discussed as methods for isolation of diffracted signals against the background of noise of randomly distributed oceanic inhomogeneities. Besides that, a systematic concept of the structure of perturbed signals in waveguides and an analysis of the possibilities of its observations in layered waveguides were developed.
Techniques and instruments for investigating the ultrasonic analogs of magnetooptic effects
Authors: Gudkov Vladimir V.
Published: October 2005

Magnetoacoustic phenomena involving alteration of the polarization of a transverse wave (namely, the ultrasonic analogs of the Faraday and Cotton–Mouton effects) have been found in yttrium iron garnet [H. Matthews and R. C. LeCraw, Phys. Rev. Lett. 8, 397 (1962); B. Luthi, Phys. Lett. 3, 285 (1963)]. As the ellipticity E and the angle F of rotation of the polarization plane are rather large in magneto-ordered materials, it was possible to make the first measurements using distortion of the exponential decay of the ultrasonic echo pulses in the oscilloscope screen. After discovering these effects in metals, a precise technique for E and F evaluation has been required. A technique for measuring F has been suggested by Boyd and Gavenda [Phys. Rev. 152, 645 (1966)]. It is based on measurements of amplitude versus magnetic field for two cases: field parallel and antiparallel to the wave vector. In the present paper both some original generalizations of this technique as well as a device which enables simultaneous measurements of F and E are reported. In addition, an instrument for visual observation of the ellipsoid of polarization and the direction of the major ellipsoid axis is described.
Minimization of the sound power radiated by a cubic structure using velocity constraints
Authors: Kruger Daniel H. , Mann J. Adin , Weigandt Ted
Published: October 2005

Sound power radiated by a cubic structure as a function of normal surface velocity was written using the boundary element method [K. A. Cunefare and G. H. Koopman, J. Vib. Acoust. 113, 387–394 (1991)]. This function was minimized by a gradient method with linear and second-order velocity constraints. The constraints were formulated to reflect physical limitations in the ability of constrained layer damping material to modify structural velocity. Complex velocity phase was held constant while amplitude was allowed only to decrease. Initially, simulated velocity patterns were used to test the method and software created. Sound power reductions of 3 dB were achieved while modifying the normal velocity amplitude over only 0.46% of the surface area of the cube at a single frequency. Later, velocity of a real structure was measured using a laser vibrometer. Minimum sound power and corresponding velocity distributions were calculated for nondimensional ka ranging from 0.5 to 3.0. [Work supported by NSF, Contract No. MSS-9103377.]
A numerical experiment on the coupling of structural-borne sound into an aircraft interior
Authors: Houston Brian H. , Marcus Martin H. , Williams Earl G. , Photiadis Douglas M. , Bucaro J. A.
Published: October 2005

Numerical experiments have been carried out to study the broadband structural acoustics of an aircraftlike shell under point excitation. The dominant mechanisms that lead to the enhancement of interior acoustic modes of this finite shell are examined. To this end, the shell response is studied for both the fluid-loaded and invacuo cases. A rigid shell calculation was also performed to determine the importance of the wall compliance on the interior acoustics. It was found that the low-frequency shell response is not dependent on the free-wave dispersions, and is driven by the interior volume acoustic modes. At higher frequencies, the shell and fluid are coupled at shell wave numbers determined by the free-wave dispersions. In general, the shell and fluid couple at primarily one circumferential mode.
Implementation of infinite elements in adaptive multilevel substructuring for structural acoustic calculations
Authors: DSouza Paul D. , Bennighof Jeffrey K.
Published: October 2005

A procedure is presented for incorporating infinite element modeling of the external acoustic medium in structural acoustic calculations using adaptive multilevel substructuring. The coupled structure–fluid system is modeled using finite elements for the structure, and for the acoustic fluid external to the structure and within a circumscribing prolate spheroid, and infinite elements external to the prolate spheroid, following the formulation of Burnett (1994). The adaptive multilevel substructuring approach transforms the structure model from the original finite element one in terms of nodal displacements, to a much more efficient one in terms of responses in approximate substructure vibration modes, which are selectively included in the model to optimize accuracy per degree of freedom. This approach is also used on the finite element portion of the acoustic fluid model, and results in a high level of model reduction there. Transformation of the infinite element portion of the acoustic fluid model is not practical because of its frequency dependence. The transformation process results in a linear system that is generally reduced in order, having a sparse symmetric coefficient matrix amenable to the use of a profile solver. The procedure is demonstrated on numerical examples in three dimensions. [Work supported by ONR.]
Wavelet frame expansions and wavelet response functions
Authors: Önsay Taner
Published: October 2005

Wavelet frame expansions are utilized to introduce ‘‘wavelet response functions’’ (WRF). The convolution integral is replaced by an efficient synthesis scheme by incorporating multiresolution and wavelet frame expansion concepts. Wavelet frame expansion is used to partition a wide spectral range into multiresolution frequency bands. The response from each resolution level is obtained by using a sampling interval that is matched with the bandwidth. The contributions coming from different resolution levels are combined to construct the overall system response. In the applications, the attention is focused on the synthesis of the transient time-domain vibration response of proportionally damped (Bn=η, fn, n=modes) wideband linear systems. Naturally, such systems, having impulse response durations that are inversely proportional to the modal bandwidths (Tmaxn= 2.2/Bn), require a multiresolution synthesis scheme during the construction of their transient response, which involves fast-decaying high-frequency components that are superimposed on top of slowly decaying low-frequency modal responses. The proposed construction technique averts the time-domain aliasing problem associated with the DFT schemes. The effectiveness of the wavelet frame synthesis scheme is demonstrated by considering the construction of the transient vibration response of a finite dispersive system.
Apparent anomalies in the forced vibration of a damped beam
Authors: Pierucci Mauro , Liu Wensen
Published: October 2005

An infinitely long Bernoulli beam with linear damping is acted upon by a localized force given by δ(x)u(t)eiωt. The results are then obtained in wave-number space and the inversion is carried out by using an FFT algorithm. Several interesting and not previously reported results will be presented. The solution in the transform domain is composed of four terms: two transient terms with wave numbers equal to k√η and k and two steady-state terms; one propagates energy into the far field while the other is a decaying localized disturbance. The disturbance created by this near field sloshes energy back and forth near the location of the forcing function. The apparent backward traveling wave which is present in the steady-state condition is not due to the localized continuous reflection of energy from the distributed damping but is due to the requirement that the beam vibration has to have continuity of displacement and slope. The force responsible for the continuity of the slope is the culprit for this apparent phenomena. If the forcing function is cos ωt, the steady-state solution can be obtained within 20 periods, while if the function sin ωt is used, then the steady-state solution cannot be obtained until more than 106 periods.
A Navier–Stokes solution of laryngeal flow during vocal fold oscillations
Authors: Alipour Fariborz , Titze Ingo
Published: October 2005

Dynamic modeling of vocal fold tissue movement and laryngeal airflow was combined in a computer simulations for the purpose of voice production. A finite-element model was used for the solution of tissue mechanics and a finite volume method was used in the solution of Navier–Stokes equations for the airflow. A so-called ‘‘shadow method’’ simulated the glottal constriction in the flow model to avoid the complexity of grid movement. The two-dimensional flow equations were solved in an iterative manner until the given transglottal pressure was approximated. The flow solution was then used in the estimation of the aerodynamic forces on the tissue, required in the finite element solution of tissue movement. The results indicate that glottal velocity profiles are parabolic, with maximum velocity in the exiting jet reaching 35 m/s. The time-varying Reynolds number in the glottis can reach up to 1500 at a lung pressure of 8 cm water. The jet velocity waveform is similar to that of an excised larynx and the pressure profiles are similar to those of steady flows in physical models. Also, the displacement of the inferior portion of glottis shows previously described phase differences with the superior portions. [Work supported by NIDCD Grant No. DC00831-03.]
A dynamical systems model representation of the F0 fluctuation of voice
Authors: Kakita Yuki , Okamoto Hitoshi
Published: October 2005

This paper proposes a simple model of generating time series of F0 (fundamental frequency) fluctuation of voice from the viewpoint of the dynamical systems generating the irregularity. The data examined were the F0 fluctuation for two kinds of voice, the normal and the pathological (perceived as ‘‘rough’’). The time series of F0 were obtained from the speech waveform of sustained utterances. The voice samples analyzed were for 20 normal (for five male and three female subjects) and 39 pathological (rough) (for sixteen male and one female subjects). By examining the time series and the pseudo-phase-portraits, a simple two-dimensional mathematical model generating the time series of fluctuation is proposed by combining the characteristics of the Hénon map (‘‘folding’’ operation) and the Delayed logistic map (‘‘twisting’’ operation). Using the model, basic aspects of the fluctuations as well as of the pseudo-phase-portraits are successfully described with a single control parameter. The model simulation also suggests that a resultant intermittent type of fluctuation manifests a chaotic behavior which is actually observed in a pathological rough voice. The transition of fluctuation from the normal voice to the pathological voice is simulated by this model.
Three-dimensional tongue shapes of sibilant fricatives
Authors: Narayanan Shrikanth , Alwan Abeer , Haker Kate
Published: October 2005

In this study, 3-D tongue shapes reconstructed from MR images obtained during sustained production of the fricatives /s,ℱ,z,yog/ by four speakers are analyzed. Images were collected in the coronal, axial, and sagittal planes using a GE SIGNA 1.5 T machine with image slice thickness of 3 mm and no interscan spacing. Results show that the tongue body for /s/ and /z/ is significantly lowered behind the constriction with respect to the tip and back of the tongue; the tongue body has flat or slightly convex contours in the constriction region that become significantly concave behind the constriction. The degree of concavity is speaker dependent, and decreasing concavity is observed as the posterior pharyngeal wall is approached. For /ℱ/ and /yog/, the tongue body rises slightly along its midline before it starts sloping towards its posterior end; the tongue shape behind the constriction is convex (two subjects) or flat (two subjects) and gradually turns concave towards the posterior region. The ‘‘grooving’’ effect in /s,z/ contributes to a relatively abrupt increase in the area function behind the constriction when compared to /ℱ,yog/. Similarly, raising of the tongue back in /s,z/ results in smaller areas near the velum when compared to /ℱ,yog/. The acoustic significance of these articulatory shapes are discussed. [Work supported in part by NSF.]
Determining vocal tract shape with small section length given only a few formants
Authors: Ananthapadmanabha T. V.
Published: October 2005

Vocal tract shape made up of cylindrical sections of equal length can be determined given the formant data of vowel sounds; the number of sections equals twice the number of formants. For an adult male speaker, use of four formants, results in a large section length of 2.125 cm. If section length of 0.5 cm is desired, 17 formants have to be specified which appears impractical. A method for obtaining vocal tract shape with small section lengths, given only a few lower formants (up to fourth or fifth) is presented. For vocal tract of M sections, only the lower M/2 formants are independent and upper formants are constrained by the lower formants. This constraint gives unrealistic locations for the upper formants. One method for predicting the upper formants would be to use those of a uniform tube of same length as that of the vocal tract. For a desired section length (say 0.5 cm), the required formants (say, 17) can be specified by using the given lower formants (say, four) and predicted upper formants (say, from 5 to 17). Validation of the method using x-ray data and simulated vocal tract shapes is presented.
A model of a confocal ultrasonic inspection system for interfaces
Authors: Yogeswaren E. , Harris John G.
Published: October 2005

A mathematical model describing how a confocal arrangement of two focused ultrasonic transducers is used to interrogate a complex interface between two materials by scanning the focal point across the interface is outlined. A complex interface is one that has roughness and partial contact at several length scales, many of which are equal to or smaller than the wavelength in the material. When the focused ultrasound strikes such an interface, though the focal region be small, multiple scattering takes place among the scatterers within and adjacent to the focal region, making it unclear how the interface is being sampled. To clarify this issue a specific interface model, consisting of a planar array of multiple small cracks having arbitrary lengths and spacings, is used. This interface is interrogated by a focused, antiplane shear wave. The model shows that what is measured are the multiply scattered signals averaged over the aperture of the transducer, and that the dominant contribution comes from the scattered signals that phase match to the interrogating signal. Expressions relating the modeled reflected and transmitted signals to the convolution of the incident wave field with the crack-opening displacements at the focal region are given. Numerical examples are worked out for similar and contrasting materials on each side of the interface. [Work supported by the NSF.]