bluuebunny/crossref_metadata_2025 huggingface dataset:
The acoustic scattering by a submerged, elastic spherical shell: The transition from thin to thick shells
Authors: Sammelmann Gary S. , Hackman Roger H.
Published: October 2005

A fundamentally oriented analysis of the pole structure of the acoustic scattering matrix in the low- to high-frequency region (0⩽ka⩽1000) as a function of mode number, density, and sound speed for several shell thicknesses has been previously presented [J. Acoust. Soc. Am. Suppl. 1 83, S94 (1988); Suppl. 1 84, S185 (1988); Suppl. 1 85, S95 (1989)]. The three most interesting results of this analysis were: (1) Due to fluid loading, the vacuum antisymmetric Lamb mode a0 and the rigid Franz modes “switch tails” to produce a subsonic a0_ mode that resembles the vacuum antisymmetric Lamb mode at low ka and an a0+ mode that resembles a0 at high ka; (2) the mid-frequency enhancement of a thin shell is associated with this bifurcation of the antisymmetric Lamb wave [J. Acoust. Soc. Am. 85, 114–124 (1989)]; and (3) the existence of strong thickness resonances associated with the existence of regions of negative group velocity for the third antisymmetric Lamb wave on a thin spherical shell. In this presentation, the trajectories of the poles of the acoustic S matrix as a function of the thickness of the shell are followed. A unified picture of the resonance structure of a spherical shell as a function of shell thickness is presented, and the importance of tail switching of the elastic and diffractive degrees of freedom in the transition from a thick to a thin shell is illustrated.
High-frequency acoustic scattering from a doubly periodic ellipsoidal surface: Neumann boundary conditions
Authors: Bishop Garner C. , Smith Judy
Published: October 2005

An exact solution is obtained for the scatter of an acoustic plane wave from an infinite surface constructed from a doubly periodic array of infinite and parallel elliptical semicylinders. Neumann boundary conditions are imposed and the Helmholtz-Kirchhoff integral used to calculate the scattered pressure field. Fredholm integral equations of the first and second kind are used to calculate the surface field. Numerical calculations are performed to determine the effects of geometric parameters on the scattered pressure field and the dependence of the surface field on the type of Fredholm integral equation used. Results are also compared with those for scattering from a sinusoidal surface.
The ASTM E-756 Damping Standard—The good, the dangerous, and the pitfalls
Authors: Drake Michael L.
Published: October 2005

The resonant cantilever beam test procedure is a reliable test method from which complex modulus data can be determined. The ASTM E-756-83 Standard was developed around this test procedure. This test method has been used since Oberst started work in the early 1950s. Ross, Kerwin, and Ungar expanded the usefulness of the test through the development of sandwich equations. Although the test procedure is reliable, it was developed when the fundamental interest in complex modulus data was focused on the transition region. As a result, when users begin to stretch the viscoelastic properties testing further into the rubbery and glassy regions problems developed. This paper will detail the idiosyncrasies of this test method and the analytical equations used to calculate the complex modulus data from the raw test data. It will be demonstrated that apparently good, self-consistent data can be very inaccurate. The effects of test specimen configuration and the modulus of the viscoelastic material on the accuracy of the complex modulus data and the utility of the various test specimens will be discussed. The commonality of the problems in the best test with other popular test methods will also be discussed and suggested test parameters will be given to enhance complex modulus data accuracy.
Characterizing viscoelastic materials using the free volume microprobe
Authors: Mayo Bret A. , Pfau James P. , Mangaraj Duryodhan
Published: October 2005

A great deal of effort has been directed toward the development of new polymer blends and interpenetrating polymer networks in recent years. This approach has emerged as an important means of developing new polymeric materials and improving properties such as sound damping. The thermodynamic models that best describe polymer-polymer interactions have an important free volume contribution. Until recently, however, there was no technique that could make a direct, nondestructive measurement of the molecular free volume. Using the free volume microprobe (FVM), it is now possible to characterize both the average free volume site size and the relative number of free volume sites. Thus it is possible to make some assessment of the total free volume as well as the free volume distribution. The basic theory of the FVM technique will be presented as well as some supporting data from a series of miscible, immiscible, and partially miscible polymer blends.
Boundary integral method for acoustic propagation in laterally inhomogenous medium
Authors: Seong Woojae , Schmidt Henrik
Published: October 2005

A hybrid analytical-numerical method that combines boundary element and full-wave solution technique is presented for two-dimensional acoustic propagation in laterally inhomogenous and simultaneously vertically stratified medium. First, the range-dependent ocean is divided into range-independent sectors and the field within each sector is expressed by a boundary integral over the vertical sector boundaries in terms of a set of unknown boundary displacements. Proper choice of a set of distribution functions (here it is found to be Legendre polynomials) for these boundary displacements leads to efficient evaluation of the kernel of the boundary integral by means of global matrix approach, as implemented in the SAFARI code [H. Schmidt and F. B. Jensen, J. Acoust. Soc. Am. 77, 813–825 (1985)]. This leads to a matrix relation between the displacement and stress amplitudes along the vertical boundaries, similar to the stiffness matrix of finite element method. Once solved for the displacement amplitudes at the sector boundaries using a Galerkin approach, the boundary integral formulation directly yields the total field within each sector including both the forward propagating and reverberated field. A few examples of the canonical problems will be given to explore the feasibility and efficiency of the proposed new method for solving various range-dependent ocean acoustic propagation problems.
Analysis of finite element and finite difference methods for the “parabolic” equation
Authors: Akrivis Georgios D. , Dougalis Vassilios A.
Published: October 2005

Some recent results of the authors on the construction and analysis of finite element and finite difference methods for the Schrödinger equation (viewed as one-dimensional “parabolic” approximation to the Helmholtz equation in underwater acoustics) will be surveyed. The results include the construction, analysis, and implementation of efficient, conservative finite element methods of high order of accuracy based on spline approximations in the depth variable and suitable Padé-type techniques in the range. In addition, optimal error estimates for classical second-order accurate finite difference techniques of Crank-Nicolson type for the “parabolic” equation with interfaces and variable bottom will be presented.
A marching method for some elliptic models of wave propagation
Authors: Knightly George H. , St. Mary Donald F.
Published: October 2005

Parabolic equation (PE) methods for wave propagation in the farfield are frequently obtained from elliptic equations valid in the farfield. Here, properties of the elliptic models are discussed directly. The equations are discretized and a marching method is obtained. Although the underlying initial value problems are not well posed, in certain parameter ranges, the marching method is stable when the step sizes are restricted in a suitable way. Under such restrictions, propagation loss curves are obtained that show good agreement with exact solutions of some test problems. Both fluid and elastic wave problems are considered. [Work supported in part by ONR.]
Teleconference room acoustics
Authors: Herbert R. Kring
Published: October 2005

In dedicated teleconference rooms, there is an acoustical conflict between finishes for optimum microphone pick-up and those for optimum speech intelligibility within the space. This paper looks at the implications of this for teleconference room finishes and geometries as well as other acoustically critical factors such as sound isolation and ambient noise control. Recommendations are given for optimum performance.
Room acoustic and sound system design for full duplex teleconference facilities using ceiling-mounted transducers
Authors: Bray Wade R.
Published: October 2005

Room acoustics and the effects of differing conditions on performance of a full-duplex audio teleconference system, which nulls direct-path coupling, leaving only coupling by reflected energy, were studied. Room impulse responses and signal-to-noise ratios at talker locations and at transducer locations were taken for rooms with differing acoustic conditions, as were binaural recordings of the impulse response tests and of sample teleconferences in the same situations, for subjective comparison and simultaneous analysis. Long working distances from talkers to microphones offer the advantage to users of freedom of placement and movement, but at the cost of strict limits on ambient noise loudness and, more particularly, noise structure (annoyance components). Early reflections within the psychoacoustic fusion time limit described by the Haas effect were found beneficial, while reflections beyond that time proved detrimental both to intelligibility and to circuit function. Discrete energetic late reflections are especially undesirable. Optimization for various room configurations was studied, and criteria presented for obtaining subjectively high-quality teleconferences between rooms and from rooms to handsets.
Lips caustics in light backscattered from an acoustically levitated spheroidal water drop
Authors: Simpson Harry J. , Marston Philip L.
Published: October 2005

Cusp diffraction catastrophes that open up roughly transverse to the propagation direction are known to exist for both light and sound [P. L. Marston, J. Acoust. Soc. Am. 81, 226–232 (1987)]. They may be produced by reflecting high-frequency sound from curved surfaces. In the present research, a closely related caustic in which the cusp curves join to form a pair of lips is studied. This caustic was predicted to exist [J. F. Nye, Nature 312, 531–532 (1984)] in light backscattered from horizontally illuminated oblate drops of water, provided the axis ratio q = D/H was within certain ranges. The associated rays are diffracted at the drop's surface and have only one internal reflection. As q is increased above the critical value q4≈1.31 associated with a hyperbolic umbilic focal section, the cusp points at the ends of the lips caustic were predicted to merge in the backward direction at a “lips event” when q = qL1 ≈ 1.42. For qL1 < q < qL2 ≈ 1.58, no caustics for this class of ray are expected while a second lips event occurs at qL2. Observations of farfield scattering from levitated drops support Nye's analysis and illustrate a mechanism for producing lips caustics. The observed backscattering is weak for q between qL1 and qL2. [Work supported by ONR.]
Calibration of underwater acoustic transducers at NRL/USRD
Authors: Van Buren A. L. , Blue J. E.
Published: October 2005

At the beginning of World War II, the United States recognized the need for establishing systematic methods for calibrating and evaluating sonar transducers and established the Underwater Sound Reference Laboratory. The Underwater Sound Reference Detachment of the Naval Research Laboratory evolved from this beginning and now serves as the focus for underwater acoustic metrology in the United States. This paper covers the measurement facilities and methods for carrying out the metrology function. Evaluation of sonar transducers and some of the ongoing research in calibration measurement methodology are also discussed. Finally, some comments are made about calibration system requirements for implementing new measurement techniques.
Precision electroacoustical measurements at the NIST (formerly the NBS): Selected examples
Authors: Nedzelnitsky Victor
Published: October 2005

In addition to primary calibrations determining the pressure and free-field sensitivity levels of laboratory standard microphones by reciprocity techniques, a broad variety of electroacoustical measurements are performed at the NIST. These include the calibration and characterization of measuring microphone systems, audiometers and audiometric earphones, and other instruments at very low sound-pressure levels, as well as the investigation of the linearity of these instruments over relatively large dynamic ranges. Techniques and apparatus used for these purposes include free-field measurements in the NIST anechoic chambers and sound-pressure measurements in acoustic couplers. Separation of signals from noise is achieved by means including analog filters, synchronous detectors (lock-in amplifiers), and dynamic signal analyzers. These incorporate a variety of excitation source types (e.g., sine-wave, chirp, bandlimited-impulse, periodic noise, and random noise signals) and signal-processing techniques (e.g., FFT, extraction of the fundamental component of a sine-wave signal from a digitized time record). Specific examples from the NIST research and measurement services are discussed.
Acoustical instrument calibration and standards
Authors: Wong George S. K.
Published: October 2005

In the field of acoustical metrology, one often encounters questions such as: Why do we need to calibrate instruments from time to time and at what intervals? How does one select an acceptable calibration procedure to implement traceability with a specific uncertainty? In order to have confidence in measurements, the overall measuring system should be calibrated in accordance with the requirements of a standard or an accepted procedure. In general, “type testing” refers to tests that are essential to verify the design integrity of the instruments; “calibration” refers to tests that are designed to monitor operational performances which may change with the passage of time; and “fieldcheck” is a relatively simple procedure, which can be performed in situ, to give some indication of whether the instrument is operational and functioning properly. To implement traceability for a specific uncertainty of a calibration procedure is a completely different matter. Examples and factors that determine the calibration schedule are discussed.
Experimental evidence of localized wave phenomena
Authors: Lewis D. Kent , Benson Steven E. , Kirk Floyd C. , Cook Bill D.
Published: October 2005

Recent interest in creating beams that exhibit more localized behavior on the propagation axis than traditional solutions has led to many formalisms for creating and describing energy transport. One set of these solutions, the localized linear wave solution developed by Ziolkowski, has been under investigation through acoustic experiments for some time. Experiments using linear superposition to simulate two-dimensional acoustic arrays are discussed. These experiments use two methods, acousto-optics, to simulate an array of line sources and a point detector, as well as the traditional two-transducer arrangement, to simulate square arrays of point sources. Preliminary experiments with an actual two-dimensional acoustic array are also described. The technique of signal preprocessing to overcome damping problems and interelement interactions is discussed. [Work performed by the Lawrence Livermore National Laboratory under the auspices of the U.S. Department of Energy under Contract No. W-7405-ENG-48.]
Propagation of localized pulse trains in a circular acoustic waveguide
Authors: Besieris I. M. , Shaarawi A. M. , Ziolkowski R. W.
Published: October 2005

A brief overview of a novel approach to the synthesis of wave signals [cf. I. M. Besieris, A. M. Shaarawi, and R. W. Ziolkowski, J. Math. Phys. 30, 1254 (1989)] will be presented. This approach, referred to as the bidirectional method, was originally introduced in order to understand the salient features of Brittingham-like solutions. Its scope is broader, however, and encompasses classes of problems altogether different from wave propagation in an unbounded homogeneous domain. The efficacy of the bidirectional method in geometrics involving boundaries has already been demonstrated [cf. A. M. Shaarawi, I. M. Besieris, and R. W. Ziolkowski, J. Appl. Phys. 65, 805 (1989)]. In this presentation, the propagation of localized pulse trains in an infinitely long, circular, acoustic waveguide will be examined in detail. The farfields radiated out of a semi-infinite, circular, acoustic waveguide, excited by a localized initial pulse, will also be studied. These approximate solutions, which are computed using Kirchhoff's integral formula with a retarded Green's function, are causal, have finite energy, and exhibit a slow energy decay.
An experimental study of the field profile of a Bessel beam ultrasonic transducer
Authors: Hsu D. K. , Margetan F. J. , Thompson D. O.
Published: October 2005

The first experimental investigation of ultrasonic beams produced by a Bessel transducer is reported. Bessel sources are of particular interest in attempts to achieve diffractionless beams; Durnin et al. [Phys. Rev. Lett. 58, 1499 (1987)] showed that a Bessel beam of light was diffraction-free. Using a novel technique of nonuniform poling, the polarization strength of a piezoelectric ceramic disk was made to follow the pattern of an axially symmetric, truncated Bessel function J0 in both amplitude and phase. Field profiles and propagation behavior of the Bessel ultrasonic beam were measured experimentally in a water immersion tank. The measured profiles agreed well with calculated results using a Gauss-Hermite beam model. Effects of the number of lobes, frequency, and beam width on diffraction behavior were investigated. [Work supported by Basic Energy Sciences, U.S. Department of Energy.]
Modification of the NPE computer code to describe the propagation of axisymmetric sound beams in infinite media
Authors: Too Gee-Pinn James , Ginsberg Jerry H.
Published: October 2005

NPE is a nonlinear progressive wave equation computer code developed by McDonald and Kuperman [Comp. Math. Appl. 11, 843–851 (1985)] to evaluate transient propagation in acoustic waveguides. It is suitable for two-dimensional phenomena, as well as for radial propagation of azimuthally symmetric waves. The present study describes the modifications required to employ NPE for the evaluation of axially propagation axisymmetric waves, particularly those associated with sound beams radiated by a baffled piston. In addition to implementing a formulation in which the propagation is essentially parallel to the axis of a set of cylindrical coordinates, it is necessary to account for the transverse spreading of the beam into an infinite medium. Another issue is the manner in which the initial waveform input to NPE is obtained. The predictions of NPE for a linear sound beam when the input is obtained from the King integral, which is an exact solution in quadrature form, are compared to the results obtained for a simple input based on assumption of planar wave behavior in the vicinity of the transducer. The results in both cases are also compared to the analytical solution for the farfield radiation pattern.
Computational aspects of synthetic seismograms for layered media
Authors: Gerstoft P. , Vilmann O. , Krenk S.
Published: October 2005

A modified direct global matrix method for computation of synthetic seismograms in a layered viscoelastic media has been developed by combining information obtained from continuous and discrete formulations. The resulting code is stable for vertical seismic profiling as well as normal horizontal profiling and has proven efficient in the generation of full-scale seismograms. The wave field is first computed in the wave slowness domain. A proper integration back to the frequency domain is a critical part of a forward modeling code. By considering the discrete wavenumber formulation it is shown that numerical integration with equally spaced integration points corresponds to a fictitious cylindrical reflective surface with diameter inversely proportional to the sampling interval. It is demonstrated that for large ranges Filon integration is a natural extension. It is demonstrated that a trapezoidal Filon integration is more accurate than higher-order Filon schemes, which generate earlier spurious arrivals. In order to yield seismograms without spurious arrivals, the wavenumber or slowness sampling must be sufficiently dense. However, in some regions, the reflectivity function is quite regular. In these regions, linear interpolation is adequate and the number of sampling points is reduced by an adaptive scheme. Finally, in the slowness domain as opposed to the wave-number domain, the position of the peaks in the reflectivity function is nearly independent of frequency. This enables an efficient vectorization over frequency.
Binaural technology: Fundamentals and applications
Authors: Blauert Jens P.
Published: October 2005

Binaural sound systems have been used in science for quite some time, for example, to provide authentic reproduction and simulation of auditory events for sound-quality evaluation. However, it is only recently that an industry started to evolve that successfully applies binaural methods to a variety of practical tasks, and supplies the necessary equipment for doing so: the binaural-technology industry. In this lecture, the fundamentals of binaural technology will be reviewed and a few current applications discussed, such as dummy-head stereophony, binaural mixing consoles, binaural room simulation, binaural noise diagnosis, virtual auditory environment, and telepresence. Special emphasis will be put on the possibilities of incorporating binaural technology into design processes in architectural acoustics.
Acoustic feature development during unsupervised learning by a neural net
Authors: Seebach Bradley S. , Intrator Nathan
Published: October 2005

A biologically plausible neural network model that employs unsupervised learning was applied to various sets of CV syllables. This network has been shown to develop recognition of input signals on the basis of distinctive signal features rather than overall signal shape [N. Intrator and B. Seebach, Int. Neural Network Soc. Abstr. 1, Suppl. 1, 299 (1988)]. Syllables pronounced in isolation by male and female speakers were digitized and sampled in short (8–32 ms) overlapping time windows, then filtered into overlapping critical bandwidths [E. Zwicker, J. Acoust. Soc. Am. 33, 248 (1961)] to produce three-dimensional energy surfaces in time and frequency. A portion of these syllabic tokens was used as a training set for the net. Those remaining were used to test generalization of network solutions both within a single speaker's utterances and across speakers. For example, when trained on a single speaker's tokens, and tested for classification of place of articulation in stop consonants, the network might correctly identify approximately 80% of similar tokens from a different speaker.
HMM-based word recognition using word category prediction neural network
Authors: Maruyama Katsuteru , Nakamura Masami , Kawabata Takeshi , Shikano Kiyohiro
Published: October 2005

Word category prediction is used to implement an accurate word recognition system. Traditional statistical approaches require considerable training data to estimate the probabilities of word sequences and many parameters to memorize probabilities. It is also difficult to predict unseen data. To solve this problem, NETgram (Neural Network for word category prediction) has been proposed. The performance of the NETgram was comparable to that of the statistical trigram model [M. Nakamura et al.L, Proc. IEEE ICASSP '89, p. 731, s13.10 (1989.5)]. In order to evaluate the ability of the NETgram, the NETgram is applied to an English word recognition experiment based on HMM. English phone models for HMM were trained using the forward-backward algorithm according to the pronunciation symbols of training speech. The NETgram is efficiently applied to improve word recognition results using a dynamic programming technique. The word recognition rate using HMM is 81.0%. The NETgram (trigram model) trained by 512 sentences in the Brown Corpus Text Database improves the word recognition rate to 86.3%. The statistical trigram model improves the word recognition rate to 85.4%. The NETgram performs better than the statistical trigram model although the training data are insufficient to estimate the probabilities of the trigram word category sequence. This experiment shows the NETgram can learn general characteristics from limited training data, that is to say, the NETgram can interpolate sparse training data to a certain degree.
Acoustical images of a submerged aluminum sphere
Authors: Gaumond Charles F. , Houston Brian H. , Woo Sheila
Published: October 2005

Acoustical images using reflection tomography [P. B. Abraham and C. F. Gaumond, J. Acoust. Soc. Am. 82, 1303–1314 (1987)] were made from 3-D acoustical scattering data. The data were taken from a 7.6-cm (3.0-in.) solid aluminum sphere using a semicircular hoop array with radius 0.85 m, which rotated around the sphere. The element spacing in the hoop was 2°. The hoop was rotated in 2° increments from the backscattered direction to the forward direction. The frequency range of the data was 2.7–25 kHz. These acoustical images were compared with analytically produced acoustical images.
Prediction of bending resonances on elastic spheroids due to oblique incident plane waves from exact beam theory and comparison to T-matrix calculations
Authors: Werby M. F. , Gaunaurd G. C.
Published: October 2005

Exact beam theory predicts the presence of “bending” or flexural resonances from oblique incident plane waves. In an earlier paper, Timoshenko theory, which assumes a constant moment of inertia along the beam, was employed to derive an approximate theory to predict flexural resonances. Good agreement was found (for the lower modes) with the exact T-matrix calculations based on exact 3-D elastodynamics. It is possible to solve the exact beam equation using phase matching techniques to predict “bending” resonance locations. Such calculations are performed for spheroids for aspect ratios from 1.5 to 10, with extremely good agreement between beam theory and the exact T-matrix calculations for all modes.
Diffraction tomographic interferometry, a new imaging technique
Authors: Gan Woon S.
Published: October 2005

A new imaging technique, diffraction tomographic interferometry (DTI), is proposed. Double-exposure DTI and time-averaged DTI are considered. The Born approximation is used to obtain the scattered wave and the filtered-backpropagation algorithm in the reconstruction. For double-exposure DTI, summation of the two tomograms gives the diffraction tomographic interferogram. Reconstruction gives a cross-sectional image covered with interference fringes. For time-averaged DTI, the intensity of the reconstructed image is expressed in terms of the characteristic function. This gives a cross-sectional contoured image of the object modulated by a system of interference fringes. For the proposed experimental setup for the double-exposure DTI, the interferogram of the object, a turbine blade in its unstressed state and stressed state, is recorded with the object illuminated from various directions. For the proposed experimental setup for the time-averaged DTI, a diffraction tomogram is exposed for a period of time by rotating the illuminating source around the object several times, during which the object executes a motion.
Echolocation signals of odontocetes in open waters
Authors: Au Whitlow W. L.
Published: October 2005

The characteristics of echolocation signals used by the Atlantic bottlenose dolphin (Tursiops truncatus) and by the beluga or white whale (Delphinapterus leucas) in target detection and discrimination experiments in open waters will be discussed. Echolocation signals measured in open waters have been found to have considerably different properties than those measured in tanks. The primary reason for the differences in echolocation signals may be attributed to the use of very-low-intensity signals by animals in tanks and the use of higher-intensity (> 40 dB) signals in open waters. Signals are projected in the forward direction along a narrow beam and echoes are detected by a receiving beam that overlaps the transmission beam. Acoustic properties of these signals, such as source sound-pressure levels, signal waveform, frequency spectrum, peak frequency, bandwidth, and click intervals, will be considered. The acoustic characteristics of echolocation signals propagating from the animal's head will also be discussed. The structure of the acoustic field directly forward of the animal, the transition region between the near and far acoustic fields, the directivity patterns in the vertical and horizontal planes, and off-axis distortion of signals will be addressed.
Ineffectiveness of ultrasonic flea collars
Authors: Dryden Michael W. , Long Glenis R. , Gaafar Sayed M.
Published: October 2005

Ultrasonic flea collars from two companies were evaluated for their ability to reduce flea numbers on cats with experimentally induced flea infestations. Sounds produced by the collars were also evaluated both before and after use to ensure that the flea collars were functional. Each brand of ultrasonic flea collar was evaluated on five cats with each cat being infested with 50 cat fleas (Ctenocephalides felis) for a 7-day treatment and control period. Cats were declawed and fitted with Elizabethan collars to reduce normal grooming activity. Collars generated 40-kHz pulses (0.8-ms exponential rise and fall) every 50 or 20 ms (depending on the device). The level of the stimuli was between 92 and 80 dB SPL at 10 cm and would thus be audible to both cats and dogs. An average of 99.6% and 97.4% of the fleas were still on the cats after treatment and control periods, respectively. The ultrasonic flea collars were totally ineffective at reducing flea numbers on these cats.
Effects of sodium salicylate and intense noise on thresholds in the chinchilla
Authors: Bancroft B. R. , Boettcher F. A. , Salvi R. J.
Published: October 2005

Salicylate administration exacerbates the temporary threshold shift (TTS) caused by noise exposure in humans [D. McFadden and H. S. Plattsmier, Hear. Res. 9, 295–316 (1983)]. However, results from animal experiments are conflicting as to whether salicylate administration potentiates noise-induced TTS. Because of the controversy in the literature and the lack of data on the potential permanent interaction of noise and salicylates, chinchillas were exposed to noise (105 dB SPL, 0.5-kHz octave band, 15 days), salicylate (300 mg/kg, daily i.p. injection), or both, concurrently. Salicylate administration resulted in a TTS of 4 to 8 dB, depending on frequency, and no permanent threshold shift (PTS). Noise exposure caused 46 to 68 dB of TTS and 10 to 26 dB of PTS. The group exposed to both agents received 57 to 70 dB of TTS and 8 to 30 dB of PTS. There was little difference between the interaction and noise-alone groups for both TTS and PTS measures. In conclusion, salicylate did not exacerbate the temporary or permanent threshold shifts caused by intense noise exposure. [Work supported by NIH 5RO1-NS23894-03.]
Wave localization in anisotropic random media
Authors: Sheng Ping
Published: October 2005

As a phenomenon generic to waves in random media, localization has many general features common to both quantum particles, e.g., electrons, and classical waves, e.g., electromagnetic and elastic waves. One such aspect is that all waves localize in two- or one-dimensional systems with an arbitrary amount of randomness, and that in three dimensions a wave localizes only in certain energy regimes that are separated from the delocalized regimes by so-called mobility edges. In this paper, the results of recent work on the anisotropic dimensional crossover behavior for wave localization are described. Starting from a randomly layered medium, some scattering centers were introduced, i.e., inhomogeneities, into each of the layers in a controlled manner. These inhomogeneities cause the wave to scatter out of the layering direction. When the strength and the density of the scattering inhomogeneities become equal to the randomness encountered in propagating from layer to layer, then the system becomes an isotropic, 3-D random medium. Therefore, in increasing the scattering strength and density of the scatterers there is essentially a “cross over” from one-dimensional randomness, where all waves are localized, to isotropic three-dimensional randomness, where there can be mobility edges. What is found is that there is a critical anisotropy below which the system behaves as 1 D and above which the system behaves as 3 D. In other words, the transition is achieved in a discontinuous manner. The talk will emphasize the underlying physics of the localization and its anisotropic critical transition.
Surface acoustic waves on nonlinear substrates
Authors: Maradudin Alexei A.
Published: October 2005

The nonlinearity of the substrate on which a surface acoustic wave propagates causes acoustical rectification of the surface wave, harmonic generation and nonlinear mixing of surface waves, and the acoustoelastic effect. It gives rise to weakly nonlinear surface acoustic waves, and to associated surface acoustic solitary waves. The spatial dispersion required for the existence of the latter is introduced by coating the substrate by a thin film with different material properties. Explicit conditions for the formation of envelope solitons are presented, as well as for self-focusing of the surface acoustic waves. The nonlinearity can also cause Love waves to become leaky with an amplitude-dependent damping constant. Finally, a periodic corrugation of the surface of a nonlinear elastic substrate is used to introduce spatial dispersion into the system. An amplitude dependence of the stop bands for the propagation of weakly nonlinear surface acoustic waves across the resulting periodic structure, and an amplitude-dependent attenuation in the case of sagittal polarization, are found. [Work supported by NSF Grant No. DMR 88-15866.]
Homovocalic patterns in two- and three-syllable words: Implications for automatic speech recognition
Authors: Smith Mary Regina
Published: October 2005

In 11 067 two-syllable words and 9640 three-syllable words of the MRC Psycholinguistic Database, 5% of the former class and 15% of the latter have the same vowel quality in two consecutive syllables. Of the 20 allowed vocalic nuclei, 95% occur homovocalically in two-syllable words. For the first two syllables of the three-syllable words, 75% of the vowel types occur homovocalically, while only 45% occur in the final two syllables of three-syllable words. Which vowels occur in these sequences is significantly affected by stress pattern, as is their relative frequency in such sequences. The likelihood that a vowel type will occur in a homovocalic sequence can be expressed as a function of quality, stress pattern, and combinatory range. The resulting patterns reduce the search space and promise improved probabilities in some automatic speech recognition strategies. [Work supported by Alvey (U.K.).]
Acoustic impedance measurements in the gerbil ear
Authors: Ravicz Michael E. , Rosowski John J. , Voight Herbert F.
Published: October 2005

The external-ear radiation impedance ZE and the middle-ear input impedance ZM were measured in the Mongolian gerbil from 10 Hz to 18 kHz. The frequency range of measurement was spanned by two acoustic sources of different characteristics. Here, ZE was measured looking out from the tympanic ring through the external ear; ZM was calculated at the tympanic membrane from measurements at the lateral surface of the skull. The magnitude of both ZE and ZM is higher in gerbil than in chinchilla, cat, or human, consistent with the smaller ear dimensions in gerbil. Also, ZM is stiffness dominated below 1 kHz and has an equivalent volume of 140 μ/, approximately two-thirds of the volume of the middle-ear air spaces. This result and measurements of ZM with the middle-ear air spaces opened suggest that these spaces dominate the impedance in this animal at low frequencies. The external ear can be modeled roughly by a rigid-walled tube of 2.5-mm diameter and 10.6-mm length. Comparisons of ZE and ZM suggest that roughly half of the sound power available at the tympanic membrane at frequencies above 2 kHz enters the middle ear and that a much smaller proportion is transmitted at lower frequencies.
Some temporal effects on the sensitivity of the CAP to changes in sound intensity and frequency
Authors: Chatterjee Monita , Smith Robert L. , Relkin Evan M.
Published: October 2005

The sensitivity of the compound action potential (CAP) of the auditory nerve to changes in sound intensity and frequency was obtained in anesthetized Mongolian gerbils. Detection of the change in N1 peak magnitude caused by a probe stimulus was determined using a two-interval tracking procedure [E. M. Relkin and R. L. Smith, J. Acoust. Soc. Am. Suppl. 1 83, S98 (1988)], a technique that depends on both the N1 magnitude and its variability. The probe was added to a background sound consisting of a train of short tone bursts, and the time delay from background onset to probe onset was varied. Adaptation, i.e., the number of prior tone bursts, reduced the magnitude and variability of the CAP in response to successive background tone bursts, but did not appear to directly affect detection of the probe tone. However, detection was found to be a sensitive function of the latency between the onset of the probe and the corresponding background tone burst, particularly for probes that differed in frequency from the background. This dependency apparently resulted from variations in neural synchrony caused by cochlear travel time. [Research sponsored by NSF and NIH.]
A description of articulatory movements in an auditory-perceptual domain
Authors: Lee Sungbok , McCutcheon Martin J.
Published: October 2005

A longstanding tradition in speech research is to correlate articulatory events with acoustic outputs. In this paper, articulatory movements were studied in a two-dimensional modified auditory-perceptual space (MAPS) based on work by Miller [J. Acoust. Soc. Am. 85, 2114–2134 (1989)]. The goal of MAPS was to eliminate z' degeneracy of the original APS. Tongue shapes (n = 1660) were generated by simulating the front- and back-raising gestures described by Ladefoged et al. [J. Acoust. Soc. Am. 64, 1027–1035 (1978)]. Two articulatory parameters (location and degree of constriction) were extracted from the derived tongue shapes and lip opening was varied in 20 steps as another parameter. The F1−F3 frequencies were calculated using a 37-section tube model for the resulting 33 200 vocal tract shapes. The modes of distribution of points in MAPS and formant frequency spaces were compared as a function of articulatory parameters. Differences and similarities between two spaces, and the possibility of describing a direct quantitative relationship between oral gestures and vowel perception, will be discussed.
Speech enhancement by selective spectral filtering
Authors: O'Shaughnessy D.
Published: October 2005

Raising the quality of noisy speech signals has met with limited success when only one signal is available for processing. Resynthesis techniques (e.g., LPC) eliminate noise but significantly distort the speech, whereas standard spectral subtraction methods often leave the enhanced speech still noisy and assume stationary additive noise. To enhance a noise speech signal, rather than subtracting a spectral estimate of the noise, complete elimination of the signal energy in frequency ranges where speech energy is weakest is suggested. Thus signal energy is only preserved at frequencies where the signal-to-noise ratio (SNR) is strongest. No modification is performed at frequencies of strong energy (unlike the spectral subtraction method), on the assumption that noise at such frequencies is masked by speech energy. Selection of which frequency ranges to pass is done on a basis of relative energy, spectral continuity (e.g., energy at a given frequency may change abruptly only at phoneme boundaries), and expected formant spacing (e.g., one formant every kilohertz). Informal listening tests have shown improved speech with input SNR as low as 0 dB. [Work supported by NSERC.]
Simulation of vocalic gestures using an articulatory model driven by a sequential neural network
Authors: Bailly Gérard , Jordan Michael I. , Mantakas Marios , Schwartz Jean-Luc , Bach Morten , Olsen Motten
Published: October 2005

The “sequential neural network” proposed by Jordan (MIT COINS Technical Report 88-27, 1988) enables learning of motor skill problems involving excess degrees of freedom. This structure was used with (i) the articulatory model developed by Meada [J. Acoust. Soc. Am. Suppl. 1 65, S22 (1979)] as an internal “forward” model relating a given set of articulatory commands (five articulators, e.g., lips, jaws, tongue body, tongue dorsum, and tongue tip) to their acoustic consequences, (ii) vocalic targets specified in the formant (F1, F2, F3) space, and (iii) smoothness constraints leading to coarticulation phenomena. A first set of results about learning of such vocalic gestures will be presented and the following will be discussed: (i) the pattern of coarticulation phenomena observed in this structure, such as between-articulator relationships for achieving a given formant target and temporal organization of the trajectories in the articulatory space, (ii) the role of gesture duration, and (iii) the compensation ability of the network in relation to perturbations such as those observed in the bite-block experiments by Lindblom et al. [J. Phonet. 7, 147–162 (1979)].
Phone recognition in continuous speech (Dutch)
Authors: van Alphen Paul
Published: October 2005

The use of hidden Markov models is a powerful tool in building automatic speech recognition systems for continuous speech. Instead of just using frame-by-frame information, dynamic information is also introduced by taking the time derivative of static parameter vectors. The static and dynamic parameters are processed in separate codebooks and these observations are then integrated at a probabilistic level [Gupta et al., Proc. ICASSP, 697–700 (1987)]. The single-user, connected-word recognition system is developed for a subvocabulary of the Dutch language (appropriate for banking transactions). The phone models (used as subword units) are made more robust by using the co-occurrence smoothing algorithm [K. F. Lee and H. W. Hon, IEEE Trans. Acoust. Speech Signal Process. ASSP-37(11), 1641–1648 (1989)], which enables accurate recognition, even with limited training data. Results will be presented at the meeting.
Speaker normalization using second-order connectionist networks
Authors: Watrous Raymond
Published: October 2005

A method for speaker-adaptive classification of vowels using connectionist networks is developed. A speaker-specific linear transformation of observations of the speech signal is computed using second-order connectionist network units. Vowel classification is accomplished by a multilayer network, which operates on the transformed speech data. The network may be adapted for a new talker by modifying the transformation parameters while leaving the classifier network fixed. This is accomplished by back-propagating classification error through the classifier to the transformation parameters. A model of this type was evaluated for 75 speakers using the first two formant values of the Peterson/Barney data (ten vowels). When the speaker-dependent transformation and nonlinear classifier were simultaneously optimized, a vowel recognition accuracy of 96.8% was obtained. However, when adapted from average speaker parameters, the network was shown to be very sensitive to initial values. When the linear transformation was optimized separately by minimizing the mean-squared error between vowels across talkers, and the classifier optimized for this normalized data, better results were obtained. Recognition accuracy using the normalized data was 91.7%; when adapted to each speaker, the accuracy improved to 96.6%.
A study of LSF parameters for speaker recognition
Authors: Paliwal K. K.
Published: October 2005

Linear prediction (LP) analysis of speech can lead to a number of parametric representations (such as cepstral coefficients, reflection coefficients, etc.), each of which provides equivalent information about the spectral envelope. The line spectral-pair frequency (LSF) representation is an alternate LP parametric representation. For speech coding and recognition applications, this representation has been recently found to be more useful than the other LP parametric representations. For speaker recognition application, the cepstral coefficient representation is traditionally used as an LP parametric representation. In this paper, the LSF representation for speaker identification is studied and its performance is compared with that of the cepstral coefficient representation. This representation results in speaker identificaton accuracy of 94% for the diagonal covariance matrix case and 95% for the full covariance matrix case; the corresponding results for the cepstral coefficient representation are 91.5% and 94.2%. Thus the LSF representation is better suited for speaker identification than the cepstral coefficient representation.
Degradation of speaker identification for LPC formant-coded speech
Authors: Eggen J. H. , Vogten L. L. M.
Published: October 2005

Degradation of speaker characteristics for LPC formant-coded speech was measured with a speaker identification experiment. Spontaneous speech spoken by 14 speakers was analyzed and synthesized with two LPC speech-coding schemes using 2.4 and 12 kbit/s, respectively. PCM-coded speech (120 kbit/s) served as reference speech. Here 20 subjects had to identify the speakers by listening to the utterances. Listeners' familiarity with speakers was scaled before the experiment. The percentage that correctly identified speakers differs signficantly for the three speech types and depends on familiarity. The degradation of speaker identification can also be expressed by the index of loss in voice recognition. This index is defined as the ratio of the percentage correctly identified LPC voices and the percentage correctly identified PCM voices. The index is a function of familiarity only for 2.4 kbit/s LPC speech. If familiarity is controlled, the speaker identification test provides a suitable method for evaluating speech-coding techniques with respect to voice recognition. [Work supported by the Dutch government as part of the national SPIN program “Analysis and synthesis of speech.”]