bluuebunny/crossref_metadata_2025 huggingface dataset:
Sidelobe suppression in correlated multipath estimates
Authors: Mignerey Peter C.
Published: October 2005

Some environments cause a propagating signal to split along several different paths. When such multipath propagation occurs, the covariance among signals traveling along rays emanating from a common source is expected to be larger than the covariance between signals generated by independent sources. An estimate of the covariance between signals arriving from two different directions is shown to be a bilinear form. The ability of the bilinear form to distinguish a correlated arrival from an independent source is studied using a simulated acoustic field. Peaks associated with the correlated pair of arrivals are 24 dB above an uncorrelated background, while no similar peaks occur for the independent source. The 8-dB sidelobes are found to be a problem. An adaptive set of filter vectors, obtained from the classical minimum variance problem, produces a bilinear form which is found to minimize sidelobe interference to 2 dB. However, the peaks indicating a correlated arrival are reduced to 18-dB output above the uncorrelated background. The constrained minimum variance estimate of the covariance between two arrivals of a multipath is apparently effective in suppressing sidelobe interference at the cost of a lower main peak. [Work supported by ONR.]
Analysis of multilayered wideband transducers
Authors: Fujita Hajime , Koyano Kiyoshi
Published: October 2005

Single piezoelectric material used as a transducer element usually gives a high Q resonance characteristic due to high acoustic impedance relative to that of water. Epoxy resins or similar materials are used as impedance matching layers in order to obtain a low Q transducer. Although analysis of a composite structure element is traditionally made with an electromechanical equivalent circuit, numerical analysis is increasingly in use. In this paper, a simple and useful design method for a multilayered transducer utilizing both an equivalent circuit and a finite element method is described. Transducers with a half-wavelength piezoelectric material with one or two quarter wavelength matching layers are analyzed for transient response with pulsed sine wave input as well as for steady-state response, and vibration, stress, and sound radiation characteristics are obtained. The results indicate that the best matching impedance is different from the value usually suggested [B. V. Smith and B. K. Gazey, IEE Proc. 113, 285–297 (1984)].
Ultrasonic microprobe hydrophones
Authors: Moffett Mark B. , Powers James M. , Clay William L.
Published: October 2005

Miniature, bilaminar ultrasonic hydrophones have been constructed with elements of piezoelectric lead titanate (PbTiO3), lead metaniobate (PbNb2O6), and poly(vinylidene fluoride) (PVDF). The overall shape of the bilaminar element was cubical, with dimensions of about 0.8 mm for the PbTiO3 and PbNb2O6 probes and 1.3 mm for the PVDF probe. When used with a 19-dB-gain, 100-pF-input capacitance preamplifier, the sensitivities ranged from − 229 dB//IV/μPa for the PbTiO3 to − 236 dB//1V/μPa for the PVDF probe. The resonance frequency of the PbTiO3 and PbNb2O6 probes, about 0.9 MHz, was lower than expected; this is probably due to the relatively high percentage of epoxy used in constructing the elements. If one defines the useful frequency range as that for which the response is flat within ± 1 dB and the beam pattern is omnidirectional within ± 1 dB, the useful upper frequency limit extends to 100 kHz for the (larger) PVDF probes and to 200 kHz for the PbTiO3 and PbNb3O6 microprobes. These upper frequency limits correspond to element dimensions about one-tenth wavelength in water. [Work supported by NUSC's Independent Exploratory Development Program (funded by DNL) and by the Transducer Technology Project (managed by NRL/USRD and funded by ONT).]
Analysis of a magnetostrictive Tonpilz transducer
Authors: Claeyssen F. , Boucher D. , Pohlenz C.
Published: October 2005

A rare-earth iron Tonpilz transducer has been developed in a compact size, low-frequency, high-power design. The main reason for this first transducer choice is that its behavior is rather well understood, the rods being used on their longitudinal mode. The active elements of the Tonpilz are four rods of random oriented TbO.3DyO.7 Fel.8 alloy. They are magnetically coupled through a laminated flux return path. Force-cooled coils provide the dynamic magnetic field as well as the polarizing bias. Data on electrical impedance, transmitting current response, transmitting voltage response, effective coupling coefficient, efficiency, and linearity versus the bias are presented. The in-air and in-water measurements are compared with the results predicted by a plane wave model taking into account leakage flux, reluctance of the return path, eddy current losses in the rods and in the return path, and mechanical damping. The material constants of the magnetostrictive alloy used in the model are measured independently with a low-frequency, two-rods device under high-power conditions. New developments in modeling and technology design will be presented in a subsequent paper.
Hydrophonic probe for underwater intensity measurements
Authors: Pauzin S. , Biron D.
Published: October 2005

Two hydrophones are mounted side by side with a separation distance of 20 mm for the frequency range 1–10 kHz. In an anechoic chamber, the probe is validated with a microphonic intensity probe for acoustic power measurements of a reference sound source. The results are compared and agree very well. Directional characteristics of the probe are carried out. Then, an experimental setup is developed in a small cavity (2 × 1 × 0.8 m3) filled with water. A hydrophone as projector is put in the middle of the tank. The characterization of this test facility is done in terms of the time response between the different boundaries and the receiver. The sound power measurements obtained by an hydrophone associated with a gating system to simulate free-field and those obtained by the hydrophonic probe are compared. Sound source localizations are also investigated. The results are in good agreement in spite of the small bounded water tank. This technique seems to be very interesting to study material characteristics like transmission or absorption and source localization.
Statistical properties of auditory threshold selection in guinea pigs
Authors: Hood Linda J. , Barlow Elizabeth K. , Berlin Charles I.
Published: October 2005

Measurement of electrophysiological thresholds for specific frequencies in animals has been limited due to spectral problems when using brief broadband pulse stimuli necessary to elicit synchronous neural discharge. Frequency specific stimuli usually fail to yield responses any closer than 10–20 dB to actual behavioral thresholds. When a compound action potential obtained from a pulse or tone-burst stimulus is subtracted from a potential obtained with the original stiumuls plus a low-intensity continuous pure-tone masker, estimates of thresholds are near behavioral thresholds than measurements obtained using short duration stimuli alone. The effects of presentation rate (27.7 to 77.7/s), number of sweeps (625 to 10 000), and level of the continuous tone stimulus (−10 to 50 dB SPL) in a series of guinea pigs have been examined. Optimal recording conditions yielded thresholds to sine waves that are within 5 dB of behavioral data presented by Heffner et al. [J. Acoust. Soc. Am. 49, 1888–1895 (1971)], and somewhat less sensitive than data presented by Prosen et al. [J. Acoust. Soc. Am. 63, 559–566 (1978)]. [Work supported by DRF.]
Individual differences in auditory-evoked potentials: Variability of middle-latency responses, including comparisons with brain-stem AEPs
Authors: Lauter Judith L. , Karzon Roanne G.
Published: October 2005

The variability of auditory-evoked potentials has been studied based on a repeated-measures design where individuals are tested under the same conditions in a series of eight weekly sessions. In previous reports to this Society, it has been shown that for auditory brain-stem responses (ABRs): (1) within-subject variability of waveform parameters such as peak latency and amplitude is smaller than between-subject variability; (2) individual subjects show characteristic variability profiles which can, in turn, classify individuals into groups; and (3) variability measures reveal differences in auditory-system response to right, left, and binaural stimulus presentations. In this paper, these measures are extended to middle-latency responses (MLRs), represented by three vertex-negative and two vertex-positive peaks. As expected, variability profiles for MLRs show much higher degrees of variability than for ABRs, even in within-individual comparisons; however, sensitivity to individual differences and ear of input are still apparent. In addition, our variability measure provides yet another means of distinguishing between the reputed brain-stem component of the MLR and later MLR peaks. [Work supported in part by AFOSR.]
Fiber optic sensor for measurement of pressure fluctuations in a turbulent boundary layer. Part II: Preliminary test results
Authors: Zuckerwar Allan J. , Watson Ralph D. , Cuomo Frank W.
Published: October 2005

The fiber-optic sensor described by Cuomo et al. (Part I) was used to measure the pressure fluctuations of a turbulent boundary layer in the 7- × 11-in. low-speed wind tunnel at Langley Research Center. The sensor was flush mounted in the bottom plate of the test section at a distance of 30 in. from a trip wire. The free-stream flow speed was varied from 0–140 ft/s in five intervals. At each flow condition the static pressure in a plenum chamber surrounding the test section was adjusted to match the pressure inside to eliminate inflow or outflow. At the same time, the position of one wall was adjusted to provide negligible longitudinal pressure gradients due to the growth of the boundary layer at the tunnel wall. The nondimensionalized power spectral density compares favorably with other measurements reported in the literature. [Work supported by NASA.]
Learning phonetic features using connectionist networks
Authors: Watrous Raymond L. , Shastri Lokendra
Published: October 2005

A method for learning phonetic features from speech data using connectionist networks is described. A temporal flow model is introduced in which sampled speech data flow through a parallel network from input to output units. The network uses hidden units with recurrent links to capture spectral/temporal characteristics of phonetic features. A supervised learning algorithm is presented which performs gradient descent in weight space using a course approximation of the desired output as an evaluation function. A simple connectionist network with recurrent links was trained on a single instance of the work pair “no” and “go,” and successfully learned a discriminatory mechanism. The trained network also correctly discriminated 98% of 25 other tokens of each word by the same speaker. The discriminatory feature was formed without segmentation of the input, and without a direct comparison of the two items. The network formed an internal representation of a single, integrated spectral feature which has a theoretical basis in human acoustic-phonetic perception.
Text-independent speaker verification using linear predictive hidden Markov models
Authors: Tishby Naftali
Published: October 2005

The application of hidden Markov models to text-independent speaker verification was studied. Linear predictive hidden Markov models have been proved to be an efficient way for statistical modeling of speech signals in all the forms of speech recognition. As was already suggested by Alan Poritz, such models can be used for statistical characterization of the talker himself. In our case, ergodic Markov models with four-seven states, are shown to discriminate between speakers, where the spectral density of the states is characterized by eight order LPC coefficients. A large data base of 100 talkers with about 20 000 isolated digits was used. The states, as well as the optimal model dimensionality, tend to supply enough information to verify the talker with less than 3% error rate, on sufficiently large testing data. The results show, however, that most of the speaker-dependent information is contained in the spectral densities and very little is left in the transition matrix. This may justify the vector quantization approach to the problem.
On optimizing the innovation codebook of stochastic coder
Authors: Lin Daniel
Published: October 2005

In stochasticoding, the speech waveform is represented as a nonstationary Gaussian random process and is reconstructed by filtering an i.i.d. Gaussian source (innovation) with a time-varying linear filter [M. R. Schroeder and B. S. Atal, Proc. ICASSP 85]. The design of optimum block codes for the white Gaussian innovation using a “polar” product code representation is considered. Each codeword in the product codebook consists of a gain and a vector's “orientation” (specified by a set of vectors on the unit hypersphere). The amplitude factors of the codewords are encoded with a Max quantizer. The codeword “orientations” are iteratively generated from an ensemble of random spherical vectors so as to maximize the minimum spherical angle between the orientation vectors. The product codes are jointly searched to determine the optimum Gaussian innovation. This approach is compared with probabilistically generated codes as well as iteratively generated (cluster) codes for performance. Preliminary simulation results have shown an improvement of nearly 1 dB in the average SNR.
Reduction of the noise of a model counterrotation propeller at cruise by reducing the aft propeller diameter
Authors: Dittmar James H. , Stang David B.
Published: October 2005

The forward propeller of a model counterrotation propeller was tested with its original aft propeller and with a reduced diameter aft propeller. Noise reductions with the reduced diameter aft propeller were measured at simulated cruise conditions. Reductions of up to 7.5 dB were measured for the aft propeller blade passing tone, and reductions in the harmonics were also measured. Reductions in the interaction tones were observed as the result of the reduced diameter aft blades no longer interacting with the forward propeller tip vortex. Significant reductions in the total noise at each harmonic were observed. The chief noise reduction at each harmonic came from reduced aft propeller-alone noise, with the interaction tones contributing little to the totals at cruise. Total cruise noise reductions, as much as 3 dB at given angles for the blade passing tone, and as high as 10 dB at some of the harmonies were observed. These reductions would improve the fuselage interior noise levels and represent a cruise noise benefit for using reduced diameter aft propeller blades.
Reduction of wind noise using correlation in outdoor acoustical measurements
Authors: Brunner John , Raspet Richard , Schomer Paul
Published: October 2005

Accurate outdoor acoustical measurements are difficult under high wind conditions. Use of two microphones and windscreens may significantly reduce wind noise. The basic technique uses two facts. First, and most important, is that as the separation of the two microphones increases, the turbulence of the wind becomes largely uncorrelated. Second, the turbulence created by placing the microphone in a steady wind is uncorrelated. A cross-multiplication algorithm on the data from the two microphones will show the effectiveness of this process as compared to using just a windscreen.
Experimental investigation of the diffraction of sound by a curved surface of finite impedance
Authors: Berthelot Yves H. , Kearns James A. , Pierce Allan D. , Main Geoffrey L.
Published: October 2005

Laboratory scale experiments have been conducted to study the diffraction of sound by a single curved ridge on top of a flat table. The sound source is an electric spark that produces an N-wave type transient of about 40-μs duration. Two 14-in. microphones capture the signal at a reference position (before the ridge), and at a field point, respectively. The signals are gated to eliminate undesirable reflections, digitized (250 kHz/channel), low-pass filtered (50-kHz cutoff frequency), and processed by an IMB PC which computes the Fourier transforms of both reference and field pressure waveforms. Regardless of the spark variability, the ratio R of the Fourier transforms is constant to within 0.5 dB for the frequency range of interest (5–40 kHz). Consequently, R represents the insertion loss of the ridge as a function of frequency. Experimental results include, for two different surface impedances (1) the insertion loss on the curved surface, and (2) the diffraction pattern in the transition region between the illuminated and shadow zones (penumbra), measured at a horizontal distance of 0, 0.71, and 2.09 m from the apex of the ridge. Data will be compared with theoretical results presented earlier. [Work supported by NASA.]
A general-purpose parabolic equation model for atmospheric and ocean acoustics
Authors: White Michael J. , Gilbert Kenneth E.
Published: October 2005

A wide-angle parabolic equation model has been developed that is applicable to a variety of propagation problems in the atmosphere and ocean. Linear finite elements are used to discretize the vertical dependence of the acoustic field. This method of discretizatiun enables the model to efficiently handle both small-scale and large-scale vertical variations in complex wavenumber and density. For example, the model can be applied to a turbulent atmosphere and to a randomly layered ocean sediment. The computational grid is terminated at the top and bottom with a complex impedance condition so that a locally reacting surface for atmospheric problems and a pressure-release surface for underwater problems can easily be incorporated. Results from selected applications will be presented to demonstrate the capabilities of the model.
Elastic wave propagation on a multilayer interface between two solids
Authors: Rokhlin S. I.
Published: October 2005

The propagation of waves on a multilayered interface is investigated by matrix method. The solution is obtained as a series expansion with respect to the thickness of the interface layers. In the first approximation the solution is determined by a simple characteristic equation for the velocity of the interface wave, where the interface layer is described by effective elastic properties, depending mainly on shear moduli. The results are compared with the case of thin surface layers when the effective properties depend on both elastic constants. The theoretical results obtained can be used for calculation of the elastic properties of interface layers based on experimental data on surface and interface wave velocity. Such applicability is demonstrated by several experimental examples.
The finite element method applied to ocean acoustic propagation
Authors: Murphy Joseph E. , Chin-Bing Stanley A.
Published: October 2005

The finite element method offers the possibility of a full-wave solution for range-dependent acoustic propagation. It allows an accurate treatment of the basic physics of the problem in realistic range-dependent environments, and thus, for those problems not amenable to exact analytic solutions, it could serve as the benchmark against which adiabatic mode, coupled mode, parabolic equation, and other models might be compared. The two major problems to be solved in implementing a finite element model for ocean acoustic propagation are (i) the huge systems of equations which must be solved, and (ii) accurate treatment of the radiation condition at the computational boundary of the model. Solutions for both of these problems will be proposed.
Composite slotted cylinder
Authors: Henriquez T. A. , Siders Clementina R. , Montgomery Robert E.
Published: October 2005

The composite slotted cylinder is a candidate for a low-frequency, high-powered, compact, underwater acoustic source that is being considered by the Naval Air Development Center. The geometry and boundary conditions of the slotted cylinder do not lend themselves to a complete closed-form solution. Therefore, a finite element model was employed to predict the behavior of a particular slotted cylinder configuration with and without radiation loading. Experimental modal analysis was used to confirm the assumptions used in the models of the unloaded transducer. The optimization of the acoustic performance was achieved using the finite element models developed. High-drive effects were also modeled. Comparisons are shown between predictive acoustic performance and the actual measurements. [Work supported by ONT.]
Numerical models used in design and analysis of volumetric transducer arrays
Authors: Hudson Scott A.
Published: October 2005

Analysis of transducer element and array performance has been aided by development of computer-based numerical models that include computation of mutual coupling effects. The farfield pressure distribution and array source level calculations necessitate accurate modeling of the individual transducers (in free field) as well as the varying radiation impedances due to the acoustic coupling between elements. Electromechanical equivalent circuits are employed to determine the complex drive voltages required to achieve velocity control within the array. Optimization methods that result in increased array source level and efficiency without forsaking velocity control have been investigated. This paper discusses numerical computations of farfield pressure, acoustic power, and mutual radiation impedances within volumetric projector arrays. Methods of optimization are also discussed. Predicted characteristics such as beam patterns and source level curves for various arrays of low-frequency flexural disk and flextensional transducers will be presented and compared to measurements.
Fiber-optic acoustic sensor using the Fabry-Perot interferometer
Authors: Fujimura Katsunori , Matsumoto Michio , Hattori Katsuji , Naono Hiroyuki
Published: October 2005

A highly sensitive and compact butt-coupled-type fiber-optic acoustic sensor has been developed based on the Fabry-Perot interferometer (FPI) for use in remote sensing or sensing in narrow spaces. The system consists of a sensor head, a controller, and an optical fiber transmission line. The sensor head is based on the FPI, and is formed from the high-reflectance-coated confronting surfaces of an optical fiber and a diaphragm. The sensitivity of the sensor head is related to the finesse of the FPI. As the finesse increases, so does the sensitivity. To ensure system stability and increase the dynamic range, the controller is composed of a DFB laser diode with a wavelength of 1300 nm, double optical isolators, an automatic temperature controller, and an automatic power controller.
Age variation in the upper limit of hearing
Authors: Takeda Shintaro , Morioka Ikuharu , Yoshida Yoshiaki , Matsumoto Kenji
Published: October 2005

The equipment for measuring the upper limit of hearing was produced to estimate hearing changes with age. The upper limits of hearing were individually measured with 4293 audiometrically, otologically, and neurologically normal ears ranging in age from 5–85 years. The results were as follows. In each age group of both sexes, the upper limits of hearing showed an approximately normal distribution if a logarithmic scale was used for the age axis. The median values of the distributions shifted to a lower frequency, and their deviation became larger, with increasing age. It is of much interest that from early childhood the deterioration of the upper limit of hearing is already in progress. Standard upper limit age curves were established by calculating 10, 25, 50, 75, and 90 percentiles for each age group.
Flux lattices in the high-Tc superconductors
Authors: Bishop D. J.
Published: October 2005

Two types of experiments that provide information about the statics and dynamics of flux lattices in YBa2Cu3O7 will be discussed. The first are magnetic decoration experiments in which the flux lines are decorated with magnetic particles. From these experiments, it has been learned that the flux lattice is triangular with the normal value for the flux quantum. In ultrasound and high-Q mechanical oscillator experiments flux lattice melting has been examined. These experiments show that the flux lattice melts well below Hc2 into a vortex liquid state. These results suggest that the vortex solid and its melting is essentially two dimensional in character.
Anomalies in the elastic properties of the high-temperature superconductors
Authors: Almond D. P. , Saunders G. A. , Lambson E. F. , Al-Kheffaji A. , Cankartaran M.
Published: October 2005

Measurements will be presented of the ultrasonic wave velocity and attenuation in high-density polycrystalline samples of YBa2Cu3O7 at temperatures between 4.2 and 300 K. Both longitudinal and shear wave data were found to be characterized by thermal hysteresis and run-to-run variations. At temperatures below Tc, the hysteresis disappeared, the elastic constants increased, and the attenuation fell. The higher temperature measurements were affected by sample annealing and thermal history and exhibited recovery effects. The pressure dependence of the elastic constants was found to be very large and also to exhibit hysteresis effects. These phenomena are consistent with the suggestion that these ceramics are pseudoplastic solids in which the plasticity is attributed to the mobility of twin boundaries. Recent measurements of Bi-Ca-Sr-Cu-O superconductors will also be presented.
Evaluation of English pronunciation based on the static and dynamic spectral characteristics of words spoken by Japanese
Authors: Hamada Hiroshi , Nakatsu Ryohei
Published: October 2005

To develop an English pronunciation training system, a method is proposed for evaluating the pronunciation ability of Japanese speakers. The quality of English pronunciation is assumed to be determined by the static characteristics of phonetic spectra, the dynamic structure of spectrum sequences, and prosodic characteristics of utterances. Since it is difficult to evaluate these factors directly, evaluation is achieved by comparing English words uttered by a Japanese speaker with those uttered by a native speaker using speech recognition techniques. The static characteristics of phonemes are evaluated by measuring the stability of mapping functions that adapt phonetic spectra of Japanese speakers to those of native speakers. The mapping functions are obtained by speaker adaptation through vector quantization. Evaluation values for the dynamic spectral structure are defined by the DTW matching distance between words spoken by Japanese and those spoken by native speakers. Although prosodic characteristics are not considered, preliminary experiments show that the evaluation results obtained by the proposed method have a good correspondence with human judgments of pronunciation quality.
Tip vortex sound
Authors: Gershfeld Jonathan L. , Skiko Edward J.
Published: October 2005

The aerodynamic sound generated by the convection of a tip vortex past a trailing edge is examined. For angles of attack greater than ± 5 deg, a vortex is generated that produces a high-frequency fluctuating pressure field several hundred hertz in bandwidth, and as much as 5 dB above the boundary layer wall pressure field, centered at roughly twice the frequency of vortex shedding due to Helmholtz wake instabilities. Sound is generated as the vortex-induced pressure field is scattered by the trailing edge. The increase in the acoustic source level, quantified by fluctuating boundary layer surface pressure statistics both near the trailing edge of the foil and under the tip vortex, corresponds in level and frequency to the increase in measured radiated noise. The wall pressure field due to tip vortex flow is superimposed on the boundary layer pressure field and apparently does not alter the wall pressure field at other frequencies. The directivity of the radiated tip vortex sound shows a reduction in the sound in the plane of the foil around the tip. This suggests that a quarter plane Green's function is required to describe the radiated noise. [Work supported by O.N.R.]
Backscattering measurements in the presence of bubbles
Authors: Nützel Bernd , Herwig Heinz
Published: October 2005

The results from a recent acoustic scattering experiment, which was conducted in the North Sea, are presented with supporting environmental information. The acoustic data were obtained utilizing a high-resolution (narrow beamwidth) pulsed parametric sonar transmitter and conventional receiver. The measurements were done in the frequency regime from 20 to 80 kHz at grazing angles of 30° and 90° (normal incidence). The results show that the contribution of bubble clouds to total backscattering from the near-surface layer is negligible at normal incidence. However, bubbles can acoustically screen the surface so that backscattered energy from the surface can no longer be used to compute backscattering strength because of increased attenuation within bubble clouds. At a grazing angle of 30°, the backscattered energy is significantly influenced by bubble clouds when breaking waves occur.
Statistical model-based speech enhancement systems
Authors: Ephraim Yariv
Published: October 2005

This paper deals with the problem of enhancing speech signals that have been degraded by statistically independent quasistationary noise. The estimation of the clean speech waveform, and of the parameters of autoregressive (AR) models for the clean speech, given the noisy speech, is considered. The two problems are demonstrated to be closely related in the sense that a good solution to one of them can be used for achieving a satisfactory solution for the other. The difficulties in solving these estimation problems are mainly due to the lack of explicit knowledge of the statistics of the clean speech signal and of the noise process. Maximum likelihood estimation solutions that are based upon the E–M algorithm and its derivatives are proposed. For estimating the speech waveform, the statistics of the clean speech signal and of the noise process are first estimated by training a pair of Gaussian AR hidden Markov models, one for the clean speech and the other for the noise, using long training sequences from the two sources. Then, the speech waveform is reestimated by applying the E–M algorithm to the estimated statistics. An approximation to the E–M algorithm is interpreted as being an iterative procedure in which Wiener filtering and AR modeling are alternatively applied. The different algorithms considered here will be compared and demonstrated.
The cognitive framework for musical scales in various cultures
Authors: Dowling W. Jay
Published: October 2005

Musical scales in different cultures throughout the world share certain design features. This suggests that some general constraints influence scale structure. Some features of scales reflect constraints apparently grounded in the physical stimulus and perhaps the auditory system; for example, octave equivalence. Other features appear to derive from limitations on human information processing abilities; for example, the use of a limited number of discrete pitch levels. The latter seems to arise from the advantages of discrete categorization of elements in noisy communication channels, and cognitive limitations on the number of categories humans can handle at a time. Combining these two types of constraints further specifies the form that scales can take; octave equivalence combined with a limited number of pitch categories leads to scales having up to seven or so pitches per octave, with the pitch categories recycled through several octaves in a logarithmic pattern. Subtler constraints are discussed, and the resulting design features are illustrated with examples from several cultures.
Recent applications of acoustoelasticity
Authors: Delsanto P. P. , Quarati P. , Boschetti F. , Chaskelis H. H. , Mignogna R. B.
Published: October 2005

A perturbation treatment of the propagation of Rayleigh waves on the surface of a homogeneous initially deformed material plate has been recently proposed [P. P. Delsanto and A. V. Clark, J. Acoust. Soc. Am. 81, 952–960 (1987)]. Although an “exact” (numerical) solution is also possible, the perturbation approach has the distinctive advantage of allowing the solution of the “inverse problem” of determining texture and/or stress from the results of Rayleigh waves time-of-flight measurements at different propagation angles. Based on this perturbation approach, nondestructive techniques for the evaluation of the elastic constants, texture, and stress [Delsanto et al., submitted to J. Acoust. Soc. Am.] have been proposed for the special case of polycrystalline aggregates, which can be assumed to consist of a slightly orthotropic distribution of cubic crystallites. Typical examples are aluminum alloys. An analysis of the experimental results and a comparison with other NDE techniques confirm the reliability of this method.
Impulse sound propagation above a curved surface
Authors: Daigle G. A. , Raspet Richard
Published: October 2005

Theory for the propagation of sound in an upward refracting atmosphere or, by analogy, above a curved surface, includes a residue series solution. Controlled experiments made indoors above a carefully constructed curved surface using pure tones have shown that the diffraction of sound energy deep within the shadow can be accurately calculated from the residue series solution [A. Berry and G. A. Daigle, J. Acoust. Soc. Am. Suppl. 1 81, S97 (1987)]. However, when the theory is used to predict impulse sound propagation in a refractive atmosphere, serious discrepancies are observed [C. G. Don and A. I. Crammond, J. Acoust. Soc. Am. 80, 302–305 (1986)]. The calculated waveforms show considerably more attenuation and dispersion than the measurements. Recently, a more accurate calculation of the residue series has improved the prediction, but the discrepancy is still apparent [R. Raspet and S. J. Franke, J. Acoust. Soc. Am. 83, 1964–1967 (1988)]. In order to elucidate the discrepancy, impulse sound measurements made indoors under controlled conditions above a curved surface will be described. The measurements will be compared with the predicted waveforms. Measurements are also planned outdoors under realistic conditions.
A model of sound propagation in a temperature inversion near the ground
Authors: Van Moorhem W. K.
Published: October 2005

A model has been developed for three-dimensional sound propagation from a point source in a realistic vertical sound-speed profile corresponding to a temperature inversion near the ground surface. The ground itself is modeled as a finite normal impedance surface. The model is developed by Hankel transforming the three-dimensional wave equation with a variable vertical sound speed. An approximate analytic solution of the transformed equation is then obtained, valid for wavelengths small compared to the scale of the temperature variation. The form of this approximate solution depends on whether the source is above or below the receiver and whether rays can pass directly from the source to receiver, are reflected from the ground or are refracted by the temperature gradient. The Hankel transformed solution is returned to physical space by a numerically calculated inverse transformation. The solutions obtained show the expected interference pattern and sound-pressure level decay near the source, but no significant change in the rate of decay is observed from that of the isothermal atmosphere. [This work was supported by the NASA Langley Research Center under Grant NAG-1-283.]
Speaker-independent recognition of unvoiced plosives using a convex time pattern of a posteriori probability
Authors: Miwa Jouji
Published: October 2005

This paper describes a method of speaker-independent recognition of unvoiced plosives. In this method, the convex characteristics of the time pattern of a posteriori probability is adopted for eliminating effects of speaker difference and coarticulation. The time pattern of a posteriori probability is more suitable than that of the distance pattern. In the first stage of the method, a posteriori probabilities for four categories (/p/, /t/, /k/, and silence) are calculated frame by frame from a five-channel spectrum of five time frames using the Bayes theorem. In the next stage, the convex part of the time pattern of a posteriori probability is decided as an unvoiced plosive. The decision using the dynamic characteristics is more suitable for speaker-independent recognition than that using static threshold. The recognition experiments are conducted for about 1400 samples of unvoiced plosives in 166 Japanese city words uttered by five male speakers. These experiments are carried out under the condition of automatic phoneme detection and without the knowledge of the following vowel. The recognition rate of 81% is obtained for the speaker-dependent case and 59% for the speaker-independent case. [Work supported by Grant-in-Aid for Scientific Research on Priority Areas, The Ministry of Education, Science and Culture of Japan.]
A method for selecting an optimum phoneme sequence using a posteriori probabilities of phonemes
Authors: Makino Shozo , Moriai Satoshi , Kido Ken'iti
Published: October 2005

A phoneme recognition system using phoneme discriminant filters was proposed, where segmentation and phoneme recognition were carried out simultaneously. However, the method had many phoneme insertion errors and often produced phoneme sequences that were inconsistent with Japanese phoneme connection rules. This paper proposes methods for reducing the number of phoneme insertion errors using inhibitory functions and for selecting an optimum phoneme sequence using dynamic programming based on the a posteriori probabilities of the phonemes. The definition of inhibitory functions is based on the Japanese phoneme connection rules and the phoneme duration information. This method was tested for the recognition of phonemes in a 212-work vocabulary uttered by ten male speakers. The inhibitory functions reduced the total number of phoneme insertion errors to 1/5 and, especially, reduced the number of consonant insertion errors. The a posteriori probabilities of phonemes were computed based on distributions of the output of the phoneme discriminant filters.
Automatic speech recognition of small vocabularies within the context of unconstrained input
Authors: Bossemeyer R. W. , Wilpon J. G. , Lee C. H. , Rabiner L. R.
Published: October 2005

Speaker-independent recognition of small vocabularies, spoken over the long-distance telephone network has been demonstrated to be a viable technology. However, the algorithms tested and the tasks evaluated both assumed that user input would be restricted to only a small set of defined vocabulary words. Recently, a large scale trial of speaker-independent, isolated-word, speech-recognition technology was carried out in Hayward, California. The task chosen required that users speak, in isolation, one of five defined vocabulary words (collect, calling-card, person, third-number, and operator). Observations of customer responses during this trial, indicated that about 20% of the utterances had the desired vocabulary item along with extraneous input that ranged from nonspeech sounds to groups of words (e.g., “I want to make a collect call please”). Our current recognition algorithms have not been designed to handle this type of input. As such, a modification of the recognition algorithms had to be made to handle words embedded in speech (i.e., a form of key-word spotting). In this paper, two recognition algorithms are presented, one based on templates and the other based on hidden Markov models. Both algorithms are designed to recognize vocabulary words in the context of unconstrained speech. Currently, recognition rates of 99% for strictly isolated input (i.e., with no extraneous speech or gross artifacts) and 90% for vocabulary words spoken in unconstrained speech are being achieved.
A speech understanding system based on a topic-oriented language model
Authors: Hori Masahiro , Tsujino Katsuhiko , Mizoguchi Riichiro , Kakusho Osamu
Published: October 2005

Higher-level knowledge plays an important role in the speech understanding problem. A topic-oriented language model is used in SPURT-I (a SPeech Understanding System with Rule-based and Topic-directed architecture). SPURT-I consists of three subsystems: a rule-based acoustic analyzer SPREX (a SPeech Recognition EXpert), a word hypothesizer, and a topic-directed parser ASP (an ASsociated-based Parser). SPURT-I accepts Japanese speech divided into “bunsetus” (Japanese syntactic units), and attempts to output a corresponding word sequence. The basic assumption of this approach is that acoustically close phoneme sequences rarely correspond to semantically close words. Based on this assumption, SPREX recognizes a group of consonants such as /b,d,g,z/, /p,t,k/, and so on. As a result of this simplification, more candidate words are generated by the word hypothesizer than in the case of strict recognition. Then ASP reduces the effective number of candidates by associating the topic of utterance and the candidate words. Currently, SPURT-I has a 1000-word vocabulary for simple scenic descriptions. Experiments with actual utterances showed that nine sentences out of ten were recognized successfully and the “bunsetu” recognition rate was 97%. [Work partly supported by a Grant-in-Aid for Scientific Research, Ministry of Education.
Relationships between phoneme recognition accuracy and sentence recognition accuracy
Authors: Ohguro Yoshihisa , Nakagawa Seiichi
Published: October 2005

The relationships between phoneme recognition accuracy and sentence recognition accuracy for continuous speech recognition using a simulated phoneme recognizer are described. A left-to-right and top-down parser based on Earley's algorithm was used as a syntactic parser. The beam search technique was embedded in this parser to reduce the search space. The syntactic constraints were represented by a context-free grammar. The word lattice for an utterance was generated by a word spotting algorithm from an ambiguous phoneme sequence. The input to the parser was the word lattice of spotted words that consisted of a group of four terms: word name, beginning point, ending point. and matched score. Three tasks, “connected word (digits or city names),” “computer network (about 250 words),” and “inquiries on UNIX (about 500 words)” were used for the evaluation. From experiments, the conclusions are as follows. (1) It is difficult to recognize continuous speech without syntactic knowledge, (2) The search space ought to be large for managing an ambiguous input. (3) The sentence recognition rate is not so sensitive to vocabulary size.
Word-boundary identification in automatic continuous speech recognition
Authors: Harrington Jonathan , Watson Gordon , Cooper Maggie
Published: October 2005

Phoneme sequence constraints, such as /b l g/, that occur across word boundaries (considerable growth) but not word internally were used for word-boundary identification from continuous speech transcriptions. All three-phoneme, word-boundary sequences were derived by pairing P♯ (all word-final phonemes) with ♯PP (all word-initial, two-phoneme sequences) and PP♯ with ♯P in a 23 000-word lexicon including 70 000 reduced phonological forms. The resulting P♯PP and PP♯P sequences were matched against the same lexicon to determine which of them were excluded word internally. All boundary sequences excluded word internally were compiled into a tree structure that was matched against three sets of phonemic transcriptions corresponding to “fast,” “normal,” and “citation” speech production styles of 145 utterances. The results showed a word-boundary detection rate of between 39% and 50% in the three transcription sets. The incorporation of phoneme sequence constraints also reduces the number of word parsings compared with an earlier left-to-right chart-parsing approach for word-boundary identification [J. M. Harrington and A. Johnstone, Comput. Speech Lang. (1988), in press].
Morphemic and syntactic analyses in a Japanese dictation system
Authors: Ito Akinori , Ogawa Yoiti , Okada Michio , Makino Shozo , Kido Ken'iti
Published: October 2005

A Japanese dictation system is being developed for task-independent continuous speech recognition. To achieve task-independency, syntactic and semantic constraints arc used in the morphemic and syntactic analysis subsystems. The structure of the Japanese sentence is effectively described by a two-level grammar that consists of an intra-Bunsetsu part and an inter-Bunsetsu part. The Bunsetsu structure consists of one content word followed by several function words. Because most of the function words are short, they cannot be spotted individually with high accuracy. In the morphemic analysis subsystem, a syntax-driven continuous DP matching algorithm spots Bunsetsu-like units and generates a Bunsetsu lattice. In the spotting process, function words following a content word are predicted using the intra-Bunsetsu grammar. In the syntactic analysis subsystem, a Bunsetsu lattice is parsed by an algorithm based on the CYK algorithm. Each Bunsetsu candidate has syntactic markers. The inter-Bunsetsu grammar is expressed as the correspondence between a FEATURE marker in the modifier-Bunsetsu and a SLOT-FILLER marker in the head-Bunsetsu. A recognition rate of 81.5% was obtained from simulator-generated phoneme sequences with an 85% phoneme recognition score.
A generalized wedge assemblage method for surface scattering calculations
Authors: Keiffer R. S. , Novarini J. C.
Published: October 2005

The facet ensemble or wedge assemblage (WA) method, which was originally developed for and successfully applied to long crested 3-D surfaces [J. C. Novarini and H. Medwin, J. Acoust. Soc. Am. 64, 260–268 (1978); H. Medwin and J. C. Novarini, J. Acoust. Soc. Am. 69, 108–111 (1981); Kinney et al., J. Acoust. Soc. Am. 73, 183–194 (1983)], has been extended to anisotropic 3-D surfaces. This allows for the modeling of scattering from more realistic surfaces and may give insight into the relative roles of diffractions and reflections from 3-D rough surfaces. In this new approach, the surfaces are decomposed into elemental finite wedges made up of triangular facets. In the model, each of these wedges has associated with it a diffracted response and, depending upon its orientation relative to the source and receiver, a reflection. The diffracted response of each finite wedge is calculated by appropriately scaling the response of an equivalent infinite wedge and windowing it by the time interval associated with the finite wedge. Reflections are handled separately and are determined by geometry and the relative sizes of the Frensel zone to the reflecting area. The impulse response for the entire surface is then the accumulation of the responses from the assemblage of finite wedges that make up the rough surface. As a preliminary application, the method is used to model scattering from 3-D wind-driven sea surfaces. Results are compared with those obtained from a Helmholtz-Kirchhoff based model.