bluuebunny/crossref_metadata_2025 huggingface dataset:
Nonspecular reflection of sound by elastic bodies and wave front reversal
Authors: Ljamshev Leonid M.
Published: October 2005

Results of experimental and theoretical research of nonspecular reflection of sound (NRS) by bounded thin or thick plates, rods, and shells in water are discussed. The results had been completed and published by the author during the 1950s [L. M. Ljamshev, Akust. J. 2, 189, 228 (1956); 5, 58 (1959); Dokl. Akad. Nauk 99, 719 (1954); 110, 48 (1956) and so on, see also W. Finney, J. Acoust. Soc. Am. 29, 625 (1948)]. These results are compared with the results of NRS of ultrasonic-bounded beams by thick plates in liquids. This effect was observed and described later during the 1970s and 1980s [W. Neubauer, J. Appl. Phys. 44, 48 (1973); M. Billy and L. Adler, J. Appl. Phys. 75, 393 (1984) and in the book: La Diffusion Acustique (CEDOCAR, Paris, 1987)]. It is shown that these effects of NRS are of an identical physical (linear) nature. A new type of NRS is discussed. It arises as the result of nonlinear interaction of an incident sound wave with a vibrating body when vibration frequency equals twice the frequency of incident sound wave and the phenomena of wave front reversal appears.
Graphical representations of the spherical-shell-form-function based on Wigner distribution algorithm
Authors: Yen N. , Dragonette Louis R. , Numrich Susan K.
Published: October 2005

The reflection of acoustic waves from a spherical shell is a well-defined scattering problem that can be formulated mathematically in terms of a normalized response either in the frequency or in the time domain. Studies made from computer generated graphic displays, based on Wigner distribution algorithm, provide a clear and detailed representation of the underlying physical processes which might otherwise be incomprehensible because of the complexity of the high-order mathematical functions. The sensitivity of such graphical representations with regard to the sampling size and frequency bandwidth is analyzed with some examples.
An efficient and accurate numerical technique for low-frequency scattering from elastic bodies
Authors: Schenck Harry A. , Benthien George W.
Published: October 2005

In the low-frequency regime where the dimensions of the object and the acoustic wavelength are comparable, it is possible to model the bistatic scattering from an elastic body numerically with high spatial and spectral resolution. This paper will describe a computer model that is built around a finite-element program (MARTSAM) which provides an accurate description of a general elastic body and a boundary element or integral equation method (CHIEF) by which the acoustic scattering from a body of arbitrary shape can be calculated. An additional program has been developed and implemented to couple the CHIEF and MARTSAM outputs to provide accurate solutions of the combined structural acoustic scattering problem for objects of arbitrary shape with internal elastic structure. These programs have been implemented on a range of computers including supercomputers. Examples will be given that illustrate the accuracy and resolution of the method and of the computing time as a function of frequency.
Bistatic scattering from elliptical shells
Authors: Radlinski Ronald P. , Simon Murray
Published: October 2005

The extended boundary integral method developed for elastic wave scattering from infinite cylindrical shells of arbitrary shape [M. M. Simon and R. P. Radlinski, J. Acoust. Soc. Am. 71, 273–281 (1983)] was modified to investigate bistatic scattering from thin elliptically cylindrical shells in fluid. Thin shell theory is assumed to describe the motion of the scatterer. Expansion of the normal and tangential displacements of the shells in Fourier series which are a function of arclength reduces the shell equations to algebraic form. Scattering cross sections and bistatic scattering are shown to be a function of the orientation and material of the shell. Extensional resonances are shown to have a wide bandwidth influence on the scattering patterns, but narrow-band resonance is also found with certain materials.
Low-frequency elastic excitations of large aspect ratio, solid elastic targets and their couplings to the acoustic field
Authors: Hackman Roger H. , Sammelmann Gary S. , Williams Kevin L.
Published: October 2005

There is currently a controversy concerning the nature of the elastic resonances of large aspect ratio solids and the couplings of these resonances to the acoustic field. Previously [Williams et al., J. Acoust. Soc. Am. Suppl. 1 82, S50 (1987)], both theoretical and experimental evidence was presented to support our contention that the elastic wave underlying the low-frequency resonances is essentially a “bar wave” with a phase velocity almost twice that of the shear speed. At this meeting, further evidence to this effect is presented and is carefully examined by the couplings of the elastic and acoustic fields. It is concluded that the excitation of the low-frequency axisymmetric resonances of these targets is not due to phase matching, but that the coupling can be best described in terms of “impulse excitations.” Also established is the relationship of the prominent, off-axis radiation lobes of these resonances to the surface displacements of the bar wave and it will be demonstrated that these lobes are not consistent with the Rayleigh surface wave interpretation.
Lamb waves and fluid-borne waves on water-loaded thin spherical shells
Authors: Miller Russel D. , Talmant Maryline , Castillo Jaime , Überall Herbert , Werby M. F.
Published: October 2005

A recent study of fluid-borne waves on plates with one-sided fluid loading [M. Talmant, Ph.D thesis, University of Paris VII (1987)] allows the prediction of the corresponding waves and their resonances (as well as of the Lamb-wave resonances) on thin submerged spherical shells. Similar fluid waves and the ensuing bifurcation in the dispersion curves of the first antisymmetric vibration mode (reminiscent of the repulsion of atomic levels) on cylindrical shells were previously described [Breitenbach et al., J. Acoust. Soc. Am. 74, 1267 (1983); J. V. Subrahmanyam, Ph.D. thesis, Catholic University (1983)] and observed [Talmant et al., J. Acoust. Soc. Am. (in press)]. The predicted resonances were confirmed by scattering cross section calculations using SIERRAS (R.D.M.) and T-matrix (M.F.W.) codes.
A laboratory model for scattering by a finite thin plate
Authors: Hsu Chaur-Jian , Chang S. K.
Published: October 2005

Time domain acoustic wave scattering by a thin metal plate with a finite width is studied experimentally in a water tank. The plate is illuminated by a point pressure source, and the scattered waves are detected by a hydrophone. The source and receivers are located along a line which has a projection on the plane of the plate perpendicular to the edges of the plate. Experiments are conducted at several angles between the plate and the source-receiver line. The waveforms in the space-time domain clearly show three main events: a flat surface reflection from the plate, and two edge diffractions. The reflected waves show up as a straight line segment indicating the angle between the source-receiver line and the plane of the plate. The diffracted waves in the space-time domain have curvatures that are consistent with the location of the edge diffraction predicted by ray theory. The multiple ringings of the plate scattering are also visible as secondary events in the later time.
Piecewise constant optimal controls and minimal acoustic transmission
Authors: Scharnhorst Kurt P.
Published: October 2005

The existence of completely piecewise constant optimal controls in the wide bandwidth, normal incidence, minimal acoustic transmission problem [MIN(γ2)] for layers of viscoelastic materials is examined numerically. It has been found that such solutions exist if the Hamiltonian is linear in the controls and the dynamic compressibility is defined in terms of fractional derivatives which approximate a generalized Maxwell model of the dilatation modulus. The low-frequency dynamic compressibility and the low- to high-frequency step of the compressibility in the relaxation region are assumed to be the controls. Calculations are presented that show the resulting optimal, multilayered structures. For shallow slopes of the compressibility the optimal controls are partially continuous (singular). As the slope increases, the controls become piecewise constant. Piecewise constant optimal controls are the exception. Their appearance seems to be restricted to relatively simple definitions of the control functions. Their occurrence may also be limited to the MIN(γ2) problem in the set of three basic problems defined by transmission, reflection, and absorption.
Technical achievements of the DARPA speech recognition program
Authors: Zue Victor W.
Published: October 2005

During the first 4 years, several significant milestones have been reached by participants of the DARPA strategic computing speech program in the development of recognition systems, related component technology, and common speech databases and research tools. From the systems' standpoint, speaker-dependent, -adaptive, and -independent systems that operate in several times real time with word error rates that range between 5% and 10% have been demonstrated with a 1000-word vocabulary and a language model perplexity of 60. Real-time performance of isolated-word, speaker-dependent, 100-word recognition systems that operate under noisy and stressful environments was demonstrated with word accuracy of better than 98%. Meanwhile, advances have been made in component technology including auditory modeling, acoustic phonetic recognition, lexical representation, language modeling, and the implementation of algorithms in general- and special-purpose hardware. The component technology is intended to provide robust performance in real time, and to support future expansion in task complexity. The program also resulted in the development of several large databases, and a set of common research tools. The use of these databases in conjunction with the development and implementation of standardized performance test procedures has permitted consistent documentation of progress. This talk will describe these technical areas in greater detail. [Work supported by DARPA-ISTO.]
Speech recognition in the French national project on man-machine interface
Authors: Haton Jean-Paul
Published: October 2005

Research and development activities in the field of automatic speech processing have been relatively important in France for the past 20 years. In 1981, the Ministry of Research created the GRECO-PRC on Man-Machine Communication with the role of promoting and coordinating research activities in the fields of vision, natural language, and speech. This paper gives an overview of this national project which presently groups more than 40 French laboratories. First, its role, objectives, and means of action will be briefly introduced and then a description of the ongoing projects in the different fields that have been selected will be presented: (1) design, recording, and exploitation of a database of spoken French, BDSON, to be used for fundamental studies in acoustic phonetics as well as for assessment of speech recognition and synthesis systems; (2) morphological database of speech including knowledge-based techniques; (3) man-machine dialogue using speech; and (4) use of prosody in automatic speech recognition. The results already obtained will be presented and the future of the project will be discussed.
A comparison of matched-mode and matched-field processor performance against sound-speed and sensor position mismatch
Authors: Livingston Ellen S.
Published: October 2005

Recent results due to T. C. Yang [J. Acoust. Soc. Am. 82, 1736–1745 (1987)] have shown that matched-mode processing can provide range-depth source localization with good sidelobe and array gain performance. His simulations and success with limited data suggest that the matched-mode processor is relatively robust with respect to environmental and hydrophone position mismatch. The objective of this study is to compare the performance of matched-mode versus matched-field processing under identical array geometry and environmental conditions. Assuming a range-independent deep ocean environment, sound-speed mismatch and sensor location mismatch were simulated at second convergence zone ranges. Eigenvalue truncated (reduced rank inverse) matched-mode processor performance was compared to conventional matched-field processor performance at 30 Hz on a large aperture vertical array. In the absence of mismatch, the two methods yield near identical results on an array spanning half the water column. The results in the presence of mismatch will be discussed and quantified in terms of signal gain degradation, sidelobe level, and speckle.
Speech recognition research at AT&T Bell Laboratories
Authors: Rabiner Lawrence R.
Published: October 2005

At AT&T Bell Laboratories, a broad range of systems for speech recognition, depending on the intended application area, has been studied. These systems have been extensively studied for isolated word (and phrase) recognition for command and control applications where a single word (or phrase) suffices to effect some type of control over a system or service. Typical applications include call type recognition for automated dialing and call routing within an organization. Several systems have also been implemented for connected word recognition with vocabularies ranging from the digits up to several hundred words. Typical applications for this technology include order entry, credit card entry, and digit dialing. Most recently, several studies have been begun of large vocabulary (1000–50 000 words), continuous speech recognition. The intended application for this type of system is database management and access. Our primary focus has been in studying alternative representations of subword speech units. In particular, recognition systems based on phones (acoustic-phonetic units), diphones, and acoustically defined units have been implemented and studied. Several methods of representing and accessing words from subword units including lexical access from a stored dictionary, explicit network representations of words, and derived acoustic lexicons have been studied. An investigation using a parser and a covering grammar to represent the constraints of the language and tack on the spoken input has also been shown. In this talk, the progress of each of these areas of speech recognition research is illustrated. Also discussed are the long-term research goals and the steps intended to take to reach these goals.
A method for measuring the acoustical reflectivity of coated panels at oblique angles of incidence
Authors: Dlubac J. J. , Reader W. T.
Published: October 2005

Conventional short-pulse techniques for measuring the reflectivity of acoustic panels require that the reflected and incident waves be separated in time so that their ratio can be established. This technique is limited to moderate angles of incidence, since at high angles the direct and reflected waves overlap. An interference method is investigated as a way of making reflectivity measurements at high oblique angles. The total field (incident and reflected) in the vicinity of an oblique panel is measured relative to the incident field without the panel. The total field, which depends on frequency and angle of incidence as well as the panel material, is then examined and manipulated to extract information on the reflected wave.
Measurement of Young's modulus of rubber in the high-frequency range
Authors: Kikuchi Toshiaki , Mori Taiji , Takahashi Sumio , Hasegawa Akio
Published: October 2005

There is a close relation between Young's modulus of a medium and the oscillation behavior of a cavity in the medium. Young's modulus of polyurethane rubber was measured in the high-frequency range by a new technique. A measurement of the transmission loss of a rectangular rubber plate (1×20×20 cm) with spherical cavities in a water tank was made. It was shown that when the center-to-center spacing between the cavities was much larger than the diameter of the cavity, there exists a frequency at which the plot of the transmission loss versus frequency attains a maximum. This maximum loss frequency occurs when the circumference of the spherical cavity is equal to two shear wavelengths in rubber. Thus Young's modulus for rubber from the maximum loss frequency was obtained. It was shown that Young's modulus for rubber in the high-frequency range increased with frequency in the same manner as in the low-frequency range.
Hilbert transform method for determining the reflecting surface of acoustic materials
Authors: Miki Yasushi
Published: October 2005

A new method for measuring delay time is proposed. If an observed signal is regarded as an output signal of a minimum phase system with delay, its delay time can be estimated by use of the Hilbert transform relationship between the log magnitude and the phase of the transfer function of the minimum phase system. Discussions are mainly focused on practical cases were the observed signal is bandlimited. A procedure for extrapolating the frequency characteristics of the signal is also presented to minimize the estimation error due to the bandwidth limitation. The method is applied to measure the boundary location of acoustic materials whose reflection characteristics are minimum phase. An estimation error less than ± 0.8 mm in distance is achieved when a test signal of 5-kHz bandwidth is used. As an example, the location of the reflecting surface of gravel is estimated. Using this result, the acoustic impedance of gravel is computed.
Estimation of geoacoustic models of the seabed from bottom loss measurements
Authors: Chapman N. R. , Bannister R. W. , Falconer R.
Published: October 2005

Measurements of bottom loss obtained in deep water sites in the S.W. Pacific have been used to infer geoacoustic models of the ocean bottom. The measurements were grouped into two types according to the acoustic reflectivity at frequencies below 100 Hz. The first type was characteristic of deep ocean basins and the second was typical of shallower regions of the continental margin. This classification was supported by archival seismic and nonacoustic information that indicated that the geology and the geophysical properties of the sediment were significantly different for each type. The former was composed of a layer of silty-clay material overlying the ocean crust, whereas the latter consisted mainly of sandy turbidite layers. A ray model, which was developed to simulate bottom loss in thin sediment layers, was used to estimate values of the geoacoustic properties for each bottom type by comparing model results with the measurements. The data at low frequencies were adequately described by simple geoacoustic models consisting of a layer of sediment with a constant sound-speed gradient overlying a solid substrate. For the first bottom type the substrate material was basalt, whereas for the second type the substrate was consolidated sediment. Results are presented for an example of each bottom type.
Very-low-frequency scattering by abyssal hills of the Pacific Ocean
Authors: Diachok O. , Livingston E.
Published: October 2005

Tectonic forces operating on the upper crust over its first million years (within about 20 km of the crest of the East Pacific Rise, EPR) give rise to a “horst and grabben” structure that hypothetically remains frozen as the plates advance across the Pacific Ocean. The horsts (uplifted plateaus) and grabben (valleys) striate the Pacific bottom in a nearly north-south direction. Scripps deep tow measurements at a nearly sediment-free site about 30 km from the EPR reveal that the number, height, and width of the horsts are nominally 2/km, 60 m, and 600 m, respectively. Twersky's theory of scattering from elliptically striated surfaces appears to be appropriate for estimating/investigating scattering loss and modal stripping caused by this type of bottom. Examples of calculations will be presented at infrasonic frequencies. The resultant computations constitute real world predictions provided that the acoustic wavelength exceeds the thickness of the sediments overlaying the basalt. In the majority of the Pacific, the nominal sediment thickness is less than 100 m and the corresponding frequency range is below 15 Hz.
Forward projection of a transient tomographically reconstructed pressure field
Authors: Forbes Michael J. , Letcher Stephen V. , Stepanishen Peter R.
Published: October 2005

The forward projection of transient pressure fields from ultrasonic transducers has been investigated both experimentally and theoretically. Previously developed impulse-response methods have been used to determine the space- and time-dependent pressure field of a particular plane in front of a typical ultrasonic transducer. The pressure field of this plane is then projected to another plane further from the transducer using wave-vector and time-domain methods that are implemented using FFTs. The numerical results illustrate the accuracy of the approach using the impulse-response method for purposes of verifying the method of projection. Experimentally determined transient pressure fields in a plane in front of typical medical or NDE transducers have been obtained using optical tomographic methods to measure a two-dimensional slice of the field as a function of time. The forward projection of the space- and time-dependent field via the projection algorithm yields results that are in good agreement with the optically measured field at the new plane. A comparison of analytical and experimental results will be discussed along with potential applications of the method for transducer calibration and characterization.
Electrode shaping to achieve Gaussian ultrasonic field distribution
Authors: Breazeale M. A. , Wen J. J. , Na J.
Published: October 2005

The electrode shape required to produce a Gaussian electric field distribution at the surface of a transducer was calculated. Such electrodes were made and tried with both quartz and PZT transducers with promising results. These transducers have an advantage over those made by taking advantage of electrical fringing [Breazeale et al., J. Acoust. Soc. Am. 70, 1791–1793 (1981)] in that they do not have the limitation 2 < W/T < 4. Results of measurement of the ultrasonic field produced at 4 MHz by quartz and PZT are given, and application to NDE and acoustical microscopy is considered. [Research sponsored, in part, by UTORNL Science Alliance, A State of Tennessee Center of Excellence. PZT provided by S. Fujishima, Murata Corp.]
Alleviation of frictional electrification of the DIP type IC with ultrasonic vibration
Authors: Seya Koichiro , Nakane Tomon
Published: October 2005

There are some instances in which a DIP type IC is damaged by frictional electrification in the manufacturing process. To ground the IC is one of the means of protecting it from electrostatic damage, although it is not perfect. It is believed that decreasing the contact area between the DIP type IC and other objects is effective in reducing the charging of the IC. The effective area and the time of contact are expected to decrease if the contact point between the IC and the other object is vibrated. A DIP type IC was passed through a slideway vibrating at a frequency of 22 kHz. The charged level of the IC was found to decrease. The present method of applying ultrasonic vibration to reduce the electrification of the DIP type IC could be used in the manufacturing process.
Dual-type helical acoustic antenna
Authors: Hasegawa Akin , Takahashi Sumio , Kikuchi Toshiaki , Fujita Teruaki
Published: October 2005

A dual-type helical acoustic antenna, which operates as an under-water endfire hydrophone, is described. It consists of two brass-wire helices wound in a concentric configuration and a PZT piezoelectric transducer of Langevin type. The structure is analogous to a helical antenna operating in the axial mode of radiation at microwave frequencies. The progressive wave along the helix yields the phase shift kbI due to wave propagation corresponding to the length of one turn I, while the phase shift due to the spacing between turns (the pitch) S is ku′S, where kb and kw are the wavenumbers in the brass wire and in water. For the radiation from the corresponding segments of all turns to add in phase in the axial direction, it is necessary for the helices to satisfy the conditions of kbIi = ku′Si and kbIo = ku′So, where the subscripts of i and o represent the inner and outer helices. The relationship between directional patterns and the dual helix parameters is shown in detail.
Evaluation of a procedure for the analysis of nonstationary random vibroacoustic data
Authors: Piersol Allan G. , Himelblau Harry , Wong Donald M.
Published: October 2005

Studies have been performed using space shuttle launch vibration measurements to evaluate a proposed procedure for the improved analysis of the nonstationary random vibroacoustic data acquired during space vehicle launches. The procedure involves decomposing the launch data into separate time- and frequency-dependent components by exploiting a parametric model (called the product model) that approximates the nonstationary character of the data. This decomposition permits the time dependence of the data to be analyzed over a wide frequency bandwidth, and the frequency dependence to be analzyed with a long averaging time, in both cases providing statistically stable results with a large bandwidth-averaging time (BT) product. Applications of the procedure to sample space shuttle launch vibration measurements are compared to the results of traditional analysis techniques based upon the computation of spectra over short time durations, where overall values are a maximum. The comparisons indicate that the proposed analysis procedure not only reduces the random errors in spectral estimates, but also suppresses the tendency to underestimate the maximum vibroacoustic loads during launch. [Work supported jointly by NASA and USAF.]
Self-coherent matched-field processing
Authors: Bucker H. P.
Published: October 2005

In this paper, a simple self-cohering procedure is proposed to correct for unknown environmental parameters and inaccurate model inputs. Consider matched-field processing for one or more vertical line arrays and assume that range changes are not strong, or abrupt, so that an adiabatic mode propagation model can be used. The standard (Bartlett) estimator of a sound source being at a particular location [r = (x,y,z)] in the ocean can be written as B(r) =  ∑ k  ∑ j pkj*  ∑ m φm (0,zkj)φm (r,z) exp(i  ∫ 0r wm dr), where pkj is the Fourier coefficient, in a small frequency bin, of the pressure received at sensor j of the k th vertical line array, φm is the depth response of mode m, wm is the horizontal wavenumber of mode m, and * denotes the complex conjugate. It has been shown [e.g., T. C. Yang, J. Acoust. Soc. Am. 82, 1736–1745 (1987)] that modal decomposition is an efficient means to perform MFP. Define Umk, the mode-line response function, as the correlation between the depth function of mode m with the complex signals received at vertical line k; that is, Umk* =  ∑ j pkj*φm (0,zkj). Now we can write B(r) =  ∑ k  ∑ m Umk*φm (r,z)exp(i  ∫ 0r wm dr). There will always be differences between sound propagation in environmental models and in the “real”ocean due to unknown mesoscale processes and inaccurate model input parameters. If at some location in the ocean there is a known acoustic source (which could be a surface ship), a set of mode-line correction factors Qmk, which are complex numbers that “correct” the Umk, can be calculated to give maximum correlation at the particular location of the known source. The important question is how useful are these correction factors at points (in the 5-D space of range, bearing, depth, frequency, and time) away from the point of the known source? This paper describes some preliminary calculations to determine the usefulness of this self-cohering method.
Acoustic conditions preferred by performers
Authors: Nakayama Ichiro
Published: October 2005

To obtain a guideline for the stage enclosure design in a concert hall for a performer, subjective preference judgments for synthetic music sound fields have been performed by alto-recorder soloists in relation to objective parameters. The sound fields consisted of two early reflections and their subsequent reverberation with systematically varied amplitudes, delay times, reverberation times, angles of incidence, and music used as source signals. Music motifs were characterized by their coherence of autocorrelation function (ACF). It was found that the most preferred delay and reverberation times could be determined by the ACF and the amplitude of the reflections. The results obtained here suggest the importance of a movable wall in the stage enclosure, adjusting the distance from the performer, and suggest that such a wall should be taken into consideration in the stage enclosure design.
Vibration damping in consideration of the directivity of vibration perception
Authors: Nara Matunori , Konishi Teruhiko , Okuno Hiroyuki
Published: October 2005

The difference between vertical and horizontal vibration perception was found from consideration of the character of the receptor. The equivalent sense distribution of the vibration perception was measured at frequencies ranging from 0.5–20 Hz. The experiments were aimed at developing the optimal method for vibration damping. The relation between the frequency and the vibration level was found to be described exactly by the transfer function of a second-order integral system with delay with a pair of complex poles. Under constant frequency and the maximum vibration transmissibility, the ratio of the receptor's damping factor for the horizontal vibration to that for the vertical (at 0.0075) was 1.8. According to the relation between the impulse response and the damping factor, doubling the damping factor causes approximately 13% decrease in the impulse response. Therefore, it was found that the load of vibration on the human body could be decreased by converting a vertical vibration into a horizontal one.
Narrow-band random vibration of a van der Pol oscillator
Authors: Davies Huw G. , Liu Qiang , Nandlall D.
Published: October 2005

The response of a van der Pol oscillator to random narrow-band excitation is considered when the excitation frequencies are either near to or away from the system natural frequency. The analysis is based on a multiple time scale perturbation series and stochastic averaging. Near the system frequency, the response is similar to the purely sinusoidal case. Away from resonance, the response can include an entrained response at the system frequency driven by a random parametric excitation. Time-dependent envelope equations agree well with numerical simulations. The envelope equations are used to find the response probability density functions for very small excitation bandwidths. [Work supported by NSERC Canada.]
VLF ambient noise measurements in selected ocean environments
Authors: Ali Hassan B. , Fisher C. A.
Published: October 2005

As part of its ongoing program in very-low-frequency (VLF) acoustic propagation, the Naval Ocean Research and Development Activity has conducted experiments in several diverse geoacoustic environments. Using a vertical array of hydrophones in the water column and a number of ocean bottom seismometers on the seafloor, ambient noise measurements were made in both shallow (100 m) and deep (several km) water areas. The resulting seismoacoustic ambient noise data are analyzed in terms of their spatial and temporal characteristics, depth dependence, and frequency behavior. Spatial correlations of the hydrophone outputs as a function of depth are used to assess the spatial homogeneity of the noise fields. Not unexpectedly, the characteristics of the noise fields are markedly different in the various geographic areas. The differences are attributable in part to the different sources (shipping, wind, storms, etc.) and in part to the diverse bottom types involved. The potentially significant role played by propagation conditions in determining ambient noise levels is also discussed.
Remote sensing using hydroacoustics
Authors: Patrick Paul H. , Sim B. , Hunt G.
Published: October 2005

Laboratory experiments involving hydroacoustics (sonar) were conducted in a semi-anechoic tank (15.5 m long × 15.5 m wide × 1.5 m deep) as a remote sensing technique to collect quantitative information on fish. Initial research was directed on the capability and accuracy of commercial sonar equipment in identifying various fish species and geometric shapes. Because of bandwidth limitations in the receiver, this equipment was limited in the ability to identify targets especially at the lower pulse widths (0.1 ms). Modifications to this equipment (wideband detector module) resulted in greater spatial resolution of targets (7.5 cm), and more detailed information in the various backscattering curves required for species identification.
Two approaches to tuning and pitch systems
Authors: Balzano Gerald J.
Published: October 2005

This paper attempts to sort out some distinctions important to the understanding of tuning and pitch systems. Two different approaches are distinguished, the acoustic approach and the structural approach. In the acoustic approach, intervals are defined as whole-number ratios, and a pitch system considered as a collection of ratios, some interrelated, some incommensurable. On the structural approach, the system is taken as prior to the intervals that compose it; intervals, then, become elements in a system with properties at least partially determined by the system. In the acoustic approach, what the intervals are and how they are tuned amount to the same thing, so it makes no sense to distinguish between a pitch system and a tuning scheme. But the structural approach distinguishes sharply between the pitch system as an abstract template or structure, and a tuning scheme as a strategy for mapping the template to specific frequency relations. Acoustic and structural views of tonality and microtonality will be presented, including examples of pitch systems constructed on both approaches. The present terminology will be applied to diverse instances of musical thought and practice, and implications of the two approaches for future evolution of pitch systems will be discussed.
A study of noise reduction of mufflers
Authors: Nishimura Tsuyoshi , Ebata Masanao , Okda Josuke
Published: October 2005

There are two methods of analyzing the noise reduction characteristics of mufflers. One is a method based on acoustic theory. This method is widely used, because the computation is relatively simple and the noise reduction in each component of the muffler can be easily evaluated from the solution. However, the computation is carried out under assumptions that the system is linear, isentropic, and so on, and the obtained results do not agree with those obtained in practice. Another is the method of characteristics that gives a more accurate solution although the calculation is complicated. Thus the method becomes effective and convenient for practical application if the results obtained by the acoustic theory can be corrected based on the strict solutions obtained by the method of characteristics. This report deals with the method of correction for various shapes of mufflers composed of a straight pipe and an expansion chamber.
Forward-glory scattering from a spherical shell and backscattering from a convex hemispherical shell
Authors: Kargl Steven G. , Marston Philip L.
Published: October 2005

Observations were made of the scattering of short tone bursts from hemispherical and empty spherical stainless-steel shells in water. Lambwave contributions to the backscattering from spherical shells have been previously described [S. G. Kargl and P. L. Marston, J. Acoust. Soc. Am. Suppl. 1 81, S14 (1987)] modeled by an elastic generalization of GTD [P. L. Marston, J. Acoust. Soc. Am. 83, 25–37 (1988)]. In the present research, novel focal or “glory” properties of Lamb-wave contributions to forward scattering from the sphere and to backscattering from the hemisphere are described. The observations of a strong forward glory are important since it follows from the optical theorem that Lamb waves will strongly affect the total cross section. The observed focal properties show that the scattered waves appear to emanate from virtual ringlike sources describable from geometrical considerations. The forward-glory scattering arrived earlier than the ordinary forward diffraction; here, the Lamb waves have phase velocities c1 > c for water. Typically ka ≈ 57, where the sphere's outer radius a = 19 mm and thickness = 3 mm. The backward glory from the hemisphere occurs as a consequence of single and multiple reflections of Lamb waves from the unconstrained edges of the hemisphere; the radius b1 of the virtual source is ≈ ac/c1. These echoes were preceded by a specular echo. [Work supported by ONR.]
Application of the integral equation method to two-dimensional sound diffraction problems
Authors: Horinouchi Yoshinari , Furue Yoshihiro
Published: October 2005

This report is concerned with the numerical calculation of two-dimensional sound diffraction fields around thin rigid plates by the integral equation method. Generally, the basic equation used in this method is the Helmholtz integral equation. However, if the objects investigated are infinitely thin plates, the basic equation cannot be applied. In such cases, the normal derivative form of the Helmholtz integral formula (NDF for short) is useful [T. Terai, J. Sound Vib. 69(1), 71–100 (1980)]. When the plate is flat and rigid, the equation to be solved is reduced to an integral equation of the first kind by means of NDF. And, as the fundamental solution in two dimensions is not exp(ikr)/r, but is a zero order Hankel function of the first kind, the treatment of singularities is more complicated. The numerical examples treated in this report are problems of an acoustic barrier consisting of several rigid and flat plates. It is shown that the numerical solutions agree with the analytical ones and also with the experimental results.
An experimental reach for the localized wave packet
Authors: Ziolkowski Richard W. , Lewis D. Kent , Cook Bill D.
Published: October 2005

Ziolkowski's novel solutions to the wave equation (see previous abstract) suggest that an approximation form of these wave packets can be launched from a finite size array. An experiment was designed based on ultrasonic pulses whose spectra centers around 1 MHz. At these frequencies, and with proper choice of array shape and size, the novel features of these new solutions should be observable at distances less than 1 m from the array. The one-dimensional array had many unique features. It was synthetic, consisting of only one commercial NDE ultrasonic transducer and a laser beam-photodiode detector. By reversing the role of transmitter and receiver, the problem could be modeled as a line transmitter, and a point receiver. [Work performed under the auspices of the U.S. Dept. of Energy by Lawrence Livermore National Laboratory under Contract W-7405-Eng-48.]
Calculation of turbulence effects in an upward refracting atmosphere
Authors: Gilbert Kenneth E. , Raspet Richard
Published: October 2005

In an upward refracting atmosphere, measured values of excess attenuation (50–500 Hz) seldom exceed 20 to 30 dB at a range of 1 km. Calculations of excess attenuation for a steady (nonturbulent) atmosphere predict a deep shadow zone with much higher excess attenuation. This paper investigates the contribution of atmospheric turbulence to decreasing the predicted excess attenuation. Since no convenient analytical method presently exists which can simultaneously account for turbulence, upward refraction, and a finite-impedance ground surface, a parabolic equation method is used to numerically simulate sound propagation. As an initial test, the calculation is compared to Daigle's theoretical and experimental results for a homogeneous atmosphere [J. Acoust. Soc. Am. 65, 45–49 (1979)]. For a test with upward refraction, the calculation is compared to the experimental results of Wiener and Keast [J. Acoust. Soc. Am. 31, 724–733 (1959)].
Acoustic components supporting solo singers on concert hall stages
Authors: Nakamura Shun-ichi , Kan Shinichiro
Published: October 2005

When singing on stage, soloists require their surroundings to have certain reflective properties such that they feel acoustically supported by the hall. Soloists feel this support in some cases, but not always. It has been found that, in simulated environments, subjects can receive the impression that they are in a good concert hall, even with only a single reflection. However, the subjects' perceptions of the degree of frontal reflection were greater than the actual amount of such reflection. Various acoustic components, including direct feedback of the subjects' sound, “couple” together to amplify the perceptible frontal reflection. Primary reflections from the stage enclosure seem to be not so effective by themselves, but the stage's local reverberant field is important, particularly in large halls, in that it couples with the relatively weak frontal reflection from the auditorium. It has also been found that a forward masking effect against reverberation can occur. Results from scale model experiments suggest that there are simple size effects directly related to the level and to the decay rate of reverberant sound. It was proposed that the stationary level and the early decay rate of the reverberant sound field, as well as the rate of the reverberation's frontal component, at a certain specified time should be measured in situ on the stage in order to get more information regarding acoustic engineering.
Can ensemble condition in a hall be improved and measured?
Authors: Gade A. C.
Published: October 2005

In collaboration with the Danish Broadcasting Corporation an extensive series of experiments has been carried out in The Danish Radio Concert Hall with the practical purpose of trying to improve the ensemble conditions on the platform for the resident symphony orchestra. First, a series of experiments in a 1:20 scale model indicated that among several suggested means the following would be the most effective and acceptable: (a) changing the shape of the sidewalls in the platform area in order to make them reflect sound back to the musicians more effectively; (b) lowering and redesigning of the ceiling reflectors; and (c) changing the position of the orchestra on the platform. These variables were then tested in full scale experiments in the hall including subjective evaluation by the orchestra in order to verify their effects under practical conditions. New objective parameters, which showed very high correlations with the subjective data, also made it possible to compare the improvements with conditions as recently measured in famous European Halls. Besides providing the needed results, the experiments also shed some light on how musicians change their criteria for judging acoustic quality depending on the experimental situation—a fact which had become evident from our previous research on musicians' room acoustic conditions.
Interactive problem solving with speech
Authors: Rudnicky Alexander I. , Brennan Robert A. , Polifroni Joseph H.
Published: October 2005

Until recently, systems offering high-performance speaker-independent continuous speech recognition were not available, making it difficult to understand how speech should be used in interactive systems. The advent of the SPHINX system developed al Carnegie-Mellon University [K.-F. Lee and H.-W. Hon, Proc. IEEE ICASSP-88, 123–126 (1988)] has made it practical to address the issue of designing systems that integrate speech into “real-world” tasks. This paper describes experience with several tasks that use tightly coupled speech interaction: a programmable voice calculator, a personal scheduler, and a spreadsheet. The goal of this work is to create an environment that allows for the rapid prototyping of “spoken language” systems and allows the study of human factors issues that these entail. The environment that was developed includes the ability to rapidly configure and refine recognition systems using declarative specifications, and the ability to define control structures suitable to particular tasks. [Work supported by DARPA.]
Discrimination of stop consonants using a data-driven analysis
Authors: Iwamida Hitoshi , Kimura Shinta
Published: October 2005

The characteristics of stop consonants are time-varying. In traditional running spectrum analysis, the frames are not always synchronized with the events of speech. In this presentation, a new data-driven analysis method is proposed in which the frames are synchronized with the events to extract the features of the stop consonants accurately. In this method, a feature vector is extracted from four or five frames that are equally spaced in a consonant segment. A voiced stop segment is defined as the segment between the release and the point where power exceeds a threshold. A voiceless stop segment is defined as the segment between the release and the voice onset. In these discrimination experiments, 94.0% of the voiced and 96.3% of the voiceless stops were correctly discriminated. The speech database used for these experiments was the Japanese monosyllables (/b,d,g,p,t,k/ + /a,i,u,e,o/) uttered by 20 speakers. It was confirmed that analysis synchronizing with consonant segments is effective for stop consonant discrimination.
A frame-based system for deep understanding: Toward understanding puns and metaphors
Authors: Yanagida Masuzo , Takizawa Osamu
Published: October 2005

An approach to deep understanding of speech is proposed. Deep understanding here means detection and analysis of hidden meanings like puns and metaphors. This report focuses mainly on understanding puns that involve words composed of the same number of syllables of different or similar phonemes. The analysis procedure for puns is evoked if some of the phoneme candidates of a high confidence factor in the input phoneme lattice differ from those in the phoneme sequence of legitimate words in the dictionary even if the input phoneme sequence seems understandable using low rank phonemes. The system searches possible words in the dictionary and assigns them scores according to the degree of coincidence between the phoneme sequence in the dictionary and that in the input phoneme lattice evaluating semantic conformity. In case there exist some words with high fitting scores, the system concludes that the input sentence has a hidden meaning assuming the doubly interpreted phoneme sequence to be a pivot word. [Work partly supported by Science and Technology Agency, Japan.]
A continuous speech recognizer using augmented continuous DP matching and its application
Authors: Ogawa Ichiro , Nakagawa Sei-ichi
Published: October 2005

It is important to develop a real-time continuous speech recognizer for investigating the man-machine interface through voice. From this point of view, a continuous speech recognizer that employs general digital processors and an augmented continuous DP matching (DTW) algorithm was developed. This equipment consists of an FFT processor for feature parameter extraction, 4 ImPPs (Image Pipelined Processor, NECuPD7281) for DP matching, and a host computer (NEC PC-9801E) for handling the syntactic constraints. It can be applied to an arbitrary task that is composed of 140 reference patterns and is represented by a finite state automaton with 50 states. Additionally, the method of expression of the automaton for a given task and the semantic extraction to derive the semantic information from the recognition results were designed to improve the flexibility of the user interface. The recognizer with this user interface was applied to the Japanese-LOGO (vocabulary size: 104 words). In speaker-dependent recognition tests, the recognizer achieved a sentence accuracy of about 90%.