bluuebunny/crossref_metadata_2025 huggingface dataset:
Interaction between a laminar boundary layer and an elastic layer
Authors: Pierucci Mauro , Morales Pedro
Published: October 2005

A two-dimensional viscous flow field is bounded on both sides by a purely elastic layer of finite thickness that supports both shear and longitudinal waves. The mean flow field is assumed to be a Poiseuille flow. The boundary layer equations are linearized to the Orr-Sommerfeld equation and the interaction between the fluid and the elastic layer occurs through the continuity of velocity and stress at the fluid-solid interface. The stability of the disturbances in the fluid and the elastic layer is analyzed. Kaplan [“The Stability of a Laminar Incompressible Boundary Layer in the Presence of Compliant Boundaries,” MIT Report No. ASRL-TR-116-1 (1964)] analyzed the stability characteristics of the problem by solving for the eigenvalues of the problem. The numerical solution of the eigenfunctions of the Orr—Sommerfeld equation has always been a very difficult and tricky problem. Davey [“An Automatic Orthonormalization Method for Solving Stiff Boundary Valve Problems,” J. Comput. Phys. 51, 343–356 (1983)] has developed a technique which allows for quick solution for the eigenvalues and the eigenfunction of the problem at any Reynolds number. This technique has been applied to the problem at hand and it has given solutions in a quick and efficient manner. The results for the stability of the boundary layer disturbances are presented in the form of velocity profiles (eigenfunctions) within the fluid and the solid layer.
Hydroelastic analysis of multiple circular cylinders in an inviscid incompressible flow
Authors: Lin W. H.
Published: October 2005

This paper presents a theoretical analysis and numerical results of hydroelastic oscillations of a group of parallel, circular, elastic cylinders in an inviscid incompressible flow. On the assumption that the turbulent fluid loading is negligible, the principal excitations considered are the steady hydrodynamic force due to the existence of the cylinders in the smooth flow and the self-excited hydrodynamic forces caused by the motion of the cylinders. These hydrodynamic forces are obtained by solving a Lapace's equation for a potential function with Neumann conditions on the cylinders and the finiteness condition at infinity. The cylinders are assumed to be slender-beam rods with straight axes so that the hydrodynamic forces acting on the axes do not produce any twisting and the equations of motion of the cylinders' flexure can be approximated by slender-beam theory. The hydroelastic equations describing the interaction between the fluid motion and the cylinders' motion are casted into a matrix form and solved by the method of matrix inversion with the aid of digit computers. The critical flow speeds are determined by solving a complex eigenvalue problem associated with the coefficient determinant of the matrix equation with the method of iteration. Numerical results show that the added mass coefficients and the hydroelastic coefficients are symmetrical, and that the critical flow speeds depend much on the number, configuration, and orientation of the cylinders. This work was performed during the author's tenure with Argonne National Laboratory, Argonne, IL 60439.
Reducing the flow-induced self-noise of a sonar: A successful enhancement of experimental and numerical models
Authors: Garnier Bernard , Larcher Jacqueline
Published: October 2005

The noise created by an external turbulent boundary layer in the heavy fluid contained in a sonar dome strongly depends on the dynamic properties of this shell. This work was performed to define the best design of the shell for a given shape, a given flow velocity, and a given sonar frequency range to minimize the flow-induced self-noise blinding the sonar in passive listening operational situations. Prior steps of this work were presented to the 106th Meeting of ASA [J. Larcher, J. Acoust. Soc. Am. Suppl. 1 74, S48, S78 (1983)] but they only reported simulations results. Since that time, operational prototypes have been measured out at sea; the results observed on operative sonars corroborate the whole predictive tool, which efficiently imbricates simplistic theoretical models with experimental simulations of the hydrodynamic excitation on 1/4 scaled or 1/1 sonar dome prototypes.
Current military/government applications for speech recognition
Authors: Hicks James W.
Published: October 2005

This paper presents an overview of several military/government programs in which SCI Technology, Inc. has implemented and tested its speech recognition system. Included are: (1) the Speckled Trout (U.S. Air Force), (2) LHX (Light Helicopter Experimental, U.S. Army), (3) Space Shuttle (NASA), (4) Space Station, (5) AFTI F-16, and (6) Advanced Tactical Fighter (ATF). Some programs consist of technology demonstrations, while others involve flight testing, and one, Speckled Trout, operationally installing and utilizing a system on a continual basis. In some cases, the hardware consists of a SCI Voice Control Unit (VCU-5137) and others, a Voice Development System (VDS-7001). For example, the Space Station application consisted of implementing a VDS in a power system monitoring and switching network in which power switches and loads could be controlled by verbal commands. In another application, two VCUs have been delivered to NASA for future Shuttle flights in which the remote controlled cameras on board will be manipulated (switched, positioned, and focused) using speech, freeing the astronauts' hands for more manually oriented tasks such as controlling the mechanical arm in the cargo bay. The LHX applications included interrogating various on-board systems (e.g., electrical, hydraulic, transmission) by speech commands and having the system status appear on video monitors. The Speckled Trout program is the first operational speech recognition system to be installed in an aircraft as an integral component of the aircraft's systems. Three VCUs are part of an integrated radio and navigational aides control system in which the system control can be either manual or by voice commands. However, since installation several months ago, the crew reports that radio control has been carried out almost exclusively by verbal commands. The AFTI F-16 program includes integrating a VDS as part of a flight simulator for testing and evaluation. The ATF program has only recently started and will involve integration as part of an advanced cockpit. This paper will also discuss the evolutionary process that has proven essential to the successful application of speech recognition technology into military and governmental systems of the future.
Decoding the speech code—Applications of temporal decomposition
Authors: Marcus Stephen M. , Atal Bishnu S.
Published: October 2005

Articulatory phonetics describes speech as a sequence of overlapping articulatory gestures, each of which may be associated with a characteristic ideal target spectrum. In normal speech, the idealized target gestures for each speech sound are often never attained, and the speech signal exhibits only transitions between such (implicit) targets. It has been suggested that the underlying speech sounds can only be recovered by reference to detailed knowledge of the gestures by which individual speech sounds are produced. It will be shown that it is possible to decompose the speech signal into overlapping “temporal transition functions” using techniques which make no assumptions about the phonetic structure of the signal or the articulatory constraints used in speech production. Previous work has shown that these techniques can produce a large reduction in the information rate needed to represent the spectral information in speech signals [B.S. Atal, Proc. ICASSP 83, 2.6, 81–84 (1983)]. It will be shown that these methods are able to derive speech components of low bandwiths that vary on a time scale closely related to traditional phonetic events. Implications for perception and the application of such techniques both for speech coding and as a possible front end for speech recognition will be discussed.
How practice affects speaker consistency in an automatic speech recognition setting
Authors: Roberts Linda A. , Egan Dennis E. , Wilpon Jay G. , Bakk Jean
Published: October 2005

Our primary goal was to assess whether practice would enhance speaker consistency in an automatic speech recognition (ASR) setting. A second goal was to observe whether specific directions for successful ASR interaction [see L. A. Roberts, J. G. Wilpon, D. E. Egan, and J. Bakk, J. Acoust. Soc. Am. Suppl. 1 77, S89 (1985)] would improve speaker consistency throughout extended practice. Inexperienced ASR users spoke a single list of isolated words in ten daily sessions. The main result of practice was that subjects learned to form templates efficiently. No improvement over time was found for recognition accuracies or distances between utterances and their templates, likely because no feedback was provided. When utterances were compared to first-day templates, distances scores gradually increased up to the fourth testing session, after which time no further degradation was observed. Speakers who received directions were more consistent than those who did not, throughout all practice sessions. This result suggests that the directions produced behavior changes that could not be induced by practice alone.
Lexical stress and speech recognition
Authors: Pieraccini Roberto
Published: October 2005

The characteristics of lexical stress and their possible use for automatic speech recognition are analyzed in this paper, with reference to the Italian spoken language. A theoretical analysis of the constraints imposed by the stress in a large lexicon access strategy was conducted on the basis of a 12 000 word vocabulary. Additionally, an investigation on several thousands of words spoken by different speakers was made in order to extract the statistical properties of the main stress correlates. An algorithm based on a statistical model succeeds to detect with a good reliability the stressed vowel of an isolated utterance. In the continuous speech case the same model can be used to compute a measure of the likelihood that a vowel is stressed. In the first case, the hypothesized stress position could be used as a source of constraints in a large lexicon access problem. In the second case, efficient pruning strategies based on highly reliable stress decisions can be designed for continuous speech understanding systems.
Use of articulatory signals in automatic speech recognition
Authors: Braida L. D. , Picheny M. A. , Cohen J. R. , Rabinowitz W. M. , Perkell J. S.
Published: October 2005

Automatic speech recognition systems generally attempt to determine the spoken message from analysis of the acoustic speech waveform. In this research we evaluated the performance of the IBM Speech Recognition System [F. Jelinek, Proc. IEEE 73, 1616–1624 (1986)] when the input included measurements of selected articulatory actions occurring during speech production. The system achieved significant recognition rates (for isolated words in sentences) when the acoustical signal was disabled and the input was restricted to articulatory signal similar to those sensed by users of the Tadoma method of tactile-speech recognition [e.g., Reed et al., J. Acoust. Soc. Am. 77, 247–257 (1985)]. In other tests the availability of articulatory inputs improved recognition performance when the acoustical signal was sufficiently degraded by additive white noise. Improvements were observed independent of whether the recognition system made use of the likelihood that words would appear in the vicinity of the other words in a given sentence.
Spectral edge orientation as a discriminator of fricatives
Authors: Bladon Anthony , Seitz Franz
Published: October 2005

Both “physical” and “auditorily transformed” spectra of English sibilant fricatives were investigated for evidence that might: (a) discriminate between /s/ and /∫/; (b) do so independently of vowel context; and (c) normalize the substantial speaker-sex differences found. Measurements included the frequency location of the first peak of fricative noise, the location of the low-frequency edge descending from this peak, the amplitude range of this descent, and its gradient or orientation. Analyses of variance on our data demonstrated a marked superiority for the orientation of the spectral edge, which differentiated the phonemes reliability, while remaining independent of vowel context or speaker sex. This was true. however, only of the edge as transformed in auditory space. Given the analogous ability of the eye-to-track edges, the notation appears to have some plausibility for speech perception theory and an obvious implication for automatic speech recognition. Perceptual confirmation of the importance of spectral edges in fricatives is now being sought.
Toward automatic recognition of the semivowels /r, w, l, y/: A progress report
Authors: Gengel Roy W. , Hieronymus James L.
Published: October 2005

The goal is to develop a speaker-independent, semivowel detector-classifier for use with continuous speech. Preliminary analysis of semivowels within continuous speech indicates large within-(and between) speaker variability in the respective semivowel formant frequencies, amplitudes, and durations of steady states and transitions. Deviations from textbook descriptions based on isolated words are significant. Coarticulation effects are also large. Nevertheless, through the use of energy, energy ratios, zero crossings, and signal envelopes, a logical analytical scheme was developed that classifies segments of continuous speech as: semivowel, high-front vowel, vowel, nasal, fricative, stop, or plosive. The specific characteristics of this scheme will be discussed and the computer-generated displays that were used will be presented. Results of using this scheme on a large corpus of phonetically labeled speech will be presented. [Work supported, in part, by DARPA.]
Location and classification of plosives, affricates, and fricatives in natural continuous speech
Authors: Brennan Robert A. , Chigier Benjamin , Cole Ronald A.
Published: October 2005

Algorithms have been developed to locate and classify plosive ([n], [d], [g], [p], [t], and [k]), affricate ([ch] and [jh]), and fricative ([f], [th], [s], [sh], and [z]) segments in natural continuous speech. The location algorithms use rules to locate target segments. The classification algorithms use feature measurements values and multivariate classifiers to assign a label probability to each hypothesized segment. The algorithms were tested on 400 utterances produced by 40 speakers that were not used to train the system. Performance of the location and classification algorithms was compared to human performance; i.e., segment boundaries and labels provided by experienced labelers. Location results will be reported in terms of the percentage of target segments located, the number of additional segments hypothesized, and the average left and right boundary error. Classification results will be presented in terms of rank order statistics, the average probability assigned to each segment, and confusion matrices. [Work supported by NSF and DARPA.]
Glottal inverse filtering of nasalized vowels
Authors: Milenkovic Paul , Mo Feng
Published: October 2005

Linear prediction provides a means for automatically determining an inverse filter for estimating the glottal volume velocity waveform from the acoustic speech waveform. The behavior of this type of glottographic analysis is considered for nasalized vowels. The analysis is applied to non-nasal and nasal productions that are synthesized using a transmission line model of the vocal and nasal tracts. The analysis is also applied to nasal and non-nasal vowel productions made by a male speaker. The effect of the nasalization is to introduce extraneous ripple onto the closed phase portion of the glottal volume velocity waveform without changing the observed open quotient. This ripple results from a closely spaced resonance antiresonance pair that is not compensated by the all-zero linear prediction inverse filter. The low level of the ripple is accounted for by the narrow frequency range over which the inverse filter is in error. [Work supported by NIH.]
A formant tracker based on pitch synchronous spectra and bark scaling
Authors: Hieronymus James L. , Majurski William J.
Published: October 2005

A formant tracker for continuous speech recognition which uses pitch synchronous spectra has been developed. The spectra are bark scaled to control diffuseness in the higher formants. Then the longest and strongest contiguous ridges are found beginning with regions of high intensity (vocalic nuclei). These are labeled as possible formant trajectories. In regions where the ridges are discontinuous, an attempt is made to join ridges of approximately the same amplitude and frequency. Then the formants are assigned, based on the average frequencies of the ridges in the vocalic regions. Statistically based heuristics are used when several candidate ridges occupy the region where formant frequencies overlap. Difficult areas include nasalized vowels and fronted back vowels. Results of evaluating the formant tracker on a large number of continuous utterances will be presented. [Work supported, in part, by DARPA.]
Speech analysis and synthesis using a vocal tract/cord model
Authors: Schroeter Juergen , Larar Jerry , Sondhi Man Mohan
Published: October 2005

The use of our vocal cord and tract model [Schroeter and Sondhi, J. Acoust. Soc. Am. Suppl. 1 78, S6 (1985)] for speech coding at bit rates below 4.8 kb/s is being considered. Earlier work [Flanagan et al., J. Acoust. Soc. Am. 68, 780–791 (1980)] reported on applying an unconstrained optimization method for jointly finding the best vocal tract and glottal parameters directly from the speech signal (one-stage optimization). This method has been extended by optimizing tract and glottal parameters sequentially (two-stage optimization). In a first stage, a search was made for the parameters of an inverse (tract) filter that give the best match between the inverse filtered speech and the output of the vocal cord model for a given set of glottal parameters. In a second stage, the parameters of the cord model were optimized for a given set of tract parameters so that the synthesized speech matches the original speech as closely as possible. This method has the advantage of separating the source and tract into different optimization stages. By doing so, the inherent constraints of the respective models can be easily used to constrain the search space. Compared to the one-stage optimization, the search required in each stage is over a parameter space of a smaller dimension. Different start-up procedures for the two-stage optimization are investigated. Results are compared to those of the one-stage optimization.
The art of generating meaningful results with numerical codes
Authors: Jensen Finn B.
Published: October 2005

Today numerical models are widely used within the underwater acoustics community for solving complex wave-propagation problems. While it is easy for anybody to produce “interesting” field solutions with a model, it is extremely difficult even for experts to generate numerical results that can be considered an accurate solution to a stated acoustical problem. This difficulty derives from insufficient knowledge both of the approximations introduced to formulate a solvable set of equations and of the accuracy and convergence problems associated with the numerical implementation itself. When experimental data are not available for checking a numerical solution, there are, in principle, only two ways to gain confidence in the numerical result (it is here assumed that straightforward checks of reciprocity and energy conservation have been performed): (1) Use a different model to confirm the validity of the original solution; or (2) compare the numerical result to an accepted reference solution to a similar propagation problem. Currently, the intermodel comparison is the only viable approach, since few reference solutions are available, none of which is for general range-dependent problems. Illustrative examples of both easily generated incorrect field solutions and artfully constructed correct field solutions will be given for range-independent environments using computer codes based on normal-mode and fast-field theory, and for range-dependent environments using codes based on adiabatic modes, coupled modes, and the parabolic equation.
Critical assessment of a marching method for an elliptic envelope equation in underwater sound propagation
Authors: Knightly George , St. Mary Donald F.
Published: October 2005

A scheme for computing propagation loss in the farfield of a rectangular ocean waveguide (soft or hard bottom) by marching in range using the farfield elliptic envelope equation is assessed. The method has shown some promise, despite the ill-posedness of the underlying initial value problem. A theoretical analysis of the problem is presented indicating parameter ranges where the method works. Approximate solutions obtained by the method are compared with those obtained by other methods.
Three-dimensional finite element analysis of waves in an acoustic medium with inclusion
Authors: Teng Yu-chiung
Published: October 2005

By using the pressure field as the basic unknown variable, two three-dimensional acoustic finite element algorithms computer codes, viz., ACOU3.A and ACOU3.B, have been implemented. Code ACOU3.A is designed for the uniform-grid space discretization such that the storage of the global pseudostiffness matrix is reduced to a single column vector of “27” elements, applicable to any inhomogeneous media. Likewise code ACOU3.B is designed for the irregular-grid space descretization and possesses the same feature of a reduced global pseudostiffness matrix of ACOU3.A. By using ACOU3.A, three 30 × 30 × 30 elements models were investigated. including (i) a half-space, (ii) a half-space with a 10 × 7 × 5 elements inclusion. and (iii) a half-space with a 10 × 2 × 5 elements inclusion, where the inclusion has the physical property different from the halfspace. In order to demonstrate the third dimension effects on wave propagation. the above posed three-dimensional problem to the equivalent axial-symmetric and two-dimensional approximation models was reduced. By using a full three-dimensional algorithm, the amplitude of the reflected waves carries the information of the third dimension thickness of the inclusion, which cannot be obtained by either the axial-symmetric, or the two-dimensional approximations.
Acoustic cavitation generated by an extracorporeal shockwave lithotripter
Authors: Crum Lawrence A. , Dyson Mary , Coleman Andrew J. , Saunders John E.
Published: October 2005

Evidence is presented of acoustic cavitation generated by a Dornier extracorporeal shockwave lithotripter. Using x-ray film, thin aluminum sheets, and relatively thick metal plates as targets, evidence of liquid jet impacts associated with cavitation bubble collapse was observed. The jet impact was violent enough to puncture thin foils and deform metal plates. Furthermore, numerous jet impacts were generated over a volume greater than 200 cm3. It is likely that such violent cavitation will also occur in tissue, and observed biological effects, e.g., renal calculus disintegration and tissue trauma, may be related to cavitation damage. [Work supported in part by the ONR, NIH, NSF, and the Fulbright Commission.]
Sonoluminescence produced by steady cavitation
Authors: Gaitan D. F.
Published: October 2005

Sonoluminescence (SL) is generally attributed to the radiative recombination of hydroxyl free radicals produced by the high temperatures and pressures associated with cavitation bubble collapse. Therapeutic ultrasound systems are known to produce large numbers of free radicals in water and biological fluids and thus present a possible health risk. In an examination of light produced by acoustic standing wave configurations at various frequencies ranging from 20 kHz–1 MHz, steady light emissions have been observed from the cavitation field indicative of violent bubble pulsation rather than cavity collapse. In many cases, multiple flashes occur each cycle, always maintaining a fixed phase with respect to the driving acoustic pressure. This “steady” cavitation is attributed to violent bubble pulsation, and an explanation of this phenomenon will be attempted in terms of numerical studies of bubble dynamics. [Work supported in part by the ONR and the NIH.]
Transient pulsations of cavitation bubbles
Authors: Flynn H. G. , Church Charles C.
Published: October 2005

Transient behavior of small gas bubbles in a liquid set into violent motion by ultrasonic pressure waves is of interest because of widespread use of microsecond pulses in diagnostic ultrasound. Such pulses contain only a few pressure cycles and the transient pulsations of bubbles set in motion by such pulses would determine the bubble-ultrasound interaction. A computer study has been made to obtain a global representation of the pulsation amplitudes R(t) of small gas bubbles (nulcei) in water during the first few cycles of a cw ultrasonic pressure. One objective was to obtain a better understanding of cavitation phenomena where many nuclei with initial radii from 0.1 to 20 μm are set in motion at pressures ranging from 0.5–5 bars and at frequencies from 0.5–10 MHz. Results allowed construction of surfaces showing the relative bubble amplitude R/Rn, as a function of Rn, and of the time t/TA, where TA is the acoustic period. One finding is that, in the range of peak pressures found in diagnostic pulses, transient cavities would be generated during the first pressure cycle from nuclei with initial radii as small as a few microns (μm). [Research supported by NIH.]
Direct observation of the forced radial oscillations of single cavitation bubbles
Authors: Holt R. G.
Published: October 2005

The radial response of a bubble in a liquid subject to an acoustic field has long been the object of considerable interest [see, for example, W. Lauterborn, J. Acoust. Soc. Am. 59, 283–293 (1976); B. E. Noltingk and E. A. Neppiras, Proc. Phys. Soc. London Ser. B 63, 674 (1950)]. However, the most often generated graph in these theoretical and numerical studies, the radius-time (R-T) curve, has never been experimentally observed until now. The first direct observation of the R-T response of a single bubble will be presented. The measurements were made on a bubble “levitated” [M. Strasberg, J. Acoust. Soc. Am. 33, 359 (1961)] near the antinode of an acoustic stationary wave by digitizing the output of a photodiode monitoring the intensity of laser light scattered by the bubble. Comparisons will be made with the results of some recent numerical models. [Research supported by ONR.]
Thresholds for eavitation produced in water by pulsed ultrasound
Authors: Atchley Anthony A. , Frizzell Leon A. , Apfel Robert E.
Published: October 2005

Initial results of an experimental investigation of transient cavitation produced by pulsed ultrasound are presented. Water is irradiated with a focused transducer and the subsequent cavitation detected acoustically by a second transducer. The detection technique relies upon the scattering of the irradiation field by the bubble cloud associated with the transient cavitation. A novel feature of the experimental apparatus is a system with which potential cavitation nuclei can be passed directly through the focal region of the irradiation transducer. The threshold for transient cavitation was measured for 1- and 2.25-MHz pulses having durations from a few cycles to 500 μs and repetition frequencies from 50 to 5000 Hz. The data are consistent with, but extend, those of Crum and Fowlkes [Nature 319, 52–54 (1986)] in which sonoluminescence emission was used as the cavitation criterion. The results of the experiment are discussed with regards to the predictions of present theories, in particular, that of Apfel [IEEE UFFC 33, No. 2, 139–142 (1986)]. [Work supported by NIH. AAA acknowledges the support of the F. V. Hunt Postdoctoral Fellowship.]
Resonant reciprocity calibration of an ultracompliant transducer
Authors: Garrett S. L. , Swift G. W.
Published: October 2005

A reciprocity calibration technique for application to an ultracompliant acoustic transducer in a resonator will be presented. The technique was applied to a magnetohydrodynamic (MHD) transducer in an open (pressure release) trough filled with either salt water or mercury as the wave propagating medium. The agreement between the reciprocity calibration and a calibration based on a primary standard is excellent. This result demonstrates that the sensitivity of the MHD transducer is just that given by the simple application of Faraday's law, in contrast to recent measurements indicating otherwise [P. H. Moose and R. F. Klaus, J. Acoust. Soc. Am. 74, 1066 (1983)]. This reciprocity technique should be applicable to calibration of a wide range of transducera, including hydrophones and accelerometers. [Work supported by the Office of Naval Research.]
Design model of large, uniform, conformal arrays of bender bar and flextensional transducers
Authors: Brigham G. A.
Published: October 2005

The single element design technology of flextensional and bender bar transducers is several decades old but use of either type of transducer in conformal arrays presents newer and more formidable design problems. When the array is very large the effects of edge diffraction on the outer elements can be ignored to generate a zeroth ordered waveguide design model to estimate radiation loading at any steering angle. Highly eccentric shelled flextensionals and bender bar transducers are then included in one common format. Since both types are largely flexural, they are resonant at frequencies where the interelement spacing is much smaller than a wavelength in water and only the low-frequency inertial and plane-wave volume flow components of loading need be determined. Several array geometries have been studied and the reactive loadings calculated. This paper shows both the theoretical and numerical results of various mass loadings as functions of array and element geometry. [Research supported by the Naval Underwater Systems Center, New London; the Naval Ocean Systems Center, San Diego; the Electric Boat Division of General Dynamics.]
On the design technology of the uncompensated class IV flextensional transducer
Authors: Brigham G. A.
Published: October 2005

The air-backed flextensional transducer is stress constrained both in the shell and the electromechanical bar driver. When Navy type III hard lead zirconate-titanate ceramic is used, the total amount of bar preload for depth and power cannot exceed 10 kpsi with allocation an interative process in the design. The math model consists of three radiating shell modes together with the bar end velocity at the shell-bar interface. The design starts with a stress analysis of shell and bar which yields the relative dimensions of both. These are input to (a) the shell-bar equation for mechanical resonance in water to get the shell major axis width, and (b) the resonant half-power bandwidth equation to get the shell length. All remaining parameters follow from the sizing, e.g., effective coupling, weight, and peak acoustic power. This paper shows the design sequence, the results for a uniform elliptic ring used as a baseline, and application to designing various flextensionals for several new major Navy sonar transmit array programs.
Fringe counting demodulator for fiber optic interferometric sensors
Authors: Crooker C. M. , Garrett S. L.
Published: October 2005

A demodulation scheme for high sensitivity (1–10 krad regime) fiber optic interferometric sensors which is based on fringe rate has been developed. The technique is similar to that utilized in optical shaft encoders. The output of a shaft encoder driven by a pendulum resembles the fringe pattern produced by a fiber optic interferometric sensor in response to a sinusoidal perturbation. This similarity was exploited to design and calibrate the demodulation circuit. An F-to-V chip first converts the shaft rotation rate to a proportional voltage level. Additional circuitry determines whether the voltage level should remain positive or be inverted through a comparison of the in-phase and quadrature waveforms output by the shaft encoder. A 3 × 3 coupler provides these waveforms optically [S. Sheem, J. Appl. Phys. 52, 3865 (1981)]. [Work supported by the Naval Research Laboratory and the Office of Naval Research.]
Automated in-situ measurement of hydrophone sensitivity for hull-mounted submatine sonar arrays using a reciprocity method
Authors: Kerr Gibson B. , O'Neill Patricia M. , Wilson O. B. , Baker S. R.
Published: October 2005

The United States Submarine Force currently conducts lengthy pre- and post-repair evaluation of submarine hull-mounted arrays using extensive, highly specialized external equipment and facilities. A methodology using transmission of acoustic energy between array elements employing portable test equipment controlled by a desktop computer has been developed that allows very rapid measurement of voltage and current parameters required for calculating sensitivities of acoustic transducers in situ. The method is an adaptation of one described by Van Buren [NRL Memorandum Report 5642, July 1985]. The method has been successfully applied to the determination of relative sensitivities for staves of DT-276 hydrophones in submarine hull-mounted arrays. Also, the method has been successfully adapted to detect the improper wiring of hydrophones in a stave of the array. The method is valuable since all measurements are performed from onboard the submarine. An outline of the procedures and examples of results will be presented. [Work supported by Naval Sea Systems Command.]
An evanescent wave array for wavenumber calibration
Authors: Schau H. C. , Luker L. Dwight , Petrie S. , Van Buren A. L.
Published: October 2005

A transmitting array that is designed to generate specified evanescent pressure fields near the array is described. The pressure fields are convective in nature, moving parallel to the plane of the array with a phase velocity that is slow, i.e., less than the speed of sound in water. A slow phase velocity corresponds to a spatial wavenumber k which is greater than the corresponding acoustic wavenumber in water. Such an array can be used to obtain the response of extended acoustic sensors to high-wavenumber pressure components such as exist in turbulent flow fields. Two small prototype planar arrays have been designed and constructed. The first one utilizes the piezoelectric polymer (PVDF) as its active material while the other one utilizes a Neoprene-lead titanate composite material. The positive electrodes on both arrays are segmented into independently driven areas by use of equally spaced stripes. Experimental results obtained for the prototype arrays are compared with computer model predictions. [Work partially supported by ONT.]
Planar turbulent boundary layer (TBL) pressure field emulation with a reduced degree of freedom array
Authors: Schau H. C.
Published: October 2005

Planar arrays being developed for producing one-dimensional convective pressure fields with prescribed spatial wavenumber content, slow phase velocity, and temporal frequency are described in the preceding paper. Such devices are capable of “frozen” turbulence emulation but allow no specified correlation properties such as those assumed to exist beneath Turbulent Boundary Layers (TBL). Extending the one-dimensional technique (involving N independently driven stripes) to two dimensions (with N2 independently driven rectangular areas) becomes unfeasible as the array size becomes large due to the extremely high bandwidth required. In this paper, an alternate approach is proposed and the trade-off between performance and implementation is discussed. The conclusions reached in this study include the types of TBL second order statistics which the device can produce within implementation constraints and methodologies for calibration of extended area sensors to TBL noise in a repeatable fashion.
Wind-energy farm noise impacts on nearby residences—the citizen's view
Authors: Warner John R.
Published: October 2005

Wind-energy farms, with hundreds of wind turbines, have proliferated in close proximity to residential properties, without any apparent control on the noise impact consequences. Annoying and intrusive wind turbine noise occurred as soon as the first turbines were installed. The adverse noise impact has continued and intensified as additional WECS (wind energy conversion system) sites were approved and developed with the installation of hundreds of additional turbines. The adverse noise effects consist of low-frequency noise that is not fully addressed or controlled by only A-weighted sound level standards, pure tones, and higher noise levels during higher wind-speed conditions. More stringent noise controls should have been implemented in the beginning. Riverside County has recently adopted a more stringent noise standard that also includes a pure tone restriction that will apply to future sites. However, mitigation of the mistakes made at the earlier sites remains to be resolved.
The rest of the Orange County Performing Arts Center
Authors: Paoletti Dennis A.
Published: October 2005

Considerable attention has been devoted in recent years at various technical meetings, including the 112th ASA Meeting, to the room acoustics design of the main theater of the Orange County Performing Arts Center in Costa Mesa, CA. In addition to the 3000-seat multi-use hall, there are four rehearsal spaces and numerous support facilities. This complex project has required a very large and thorough consulting effort in terms of sound isolation and mechanical-systems noise and vibration control in addition to room acoustics design. This paper will discuss the overall building complex, details of specific support facilities, and many interesting aspects of the consulting process that have made this project unique.
Critique of certain concert-hall design criteria
Authors: Benade A. H.
Published: October 2005

Current concert-hall designs are often seriously unsatisfactory for performers and musically experienced listeners, who tend to pay more attention to the music itself than to the “ambiance” of a hall. Formal experiments dovetail with practical experience to support the idea that the auditory system uses early reflections to compile information on tone, pitch, spatial, and temporal location (as well as loudness) via an extension of the precedence effect. This suggests that the “early reflection” criteria with 60 < t < 80 ms, which were originally proposed before Haas, should be based on t < 30 ms for at least a handful of early reflections to assure a good sampling of (among other things) the radiation patterns of the instruments. Data abound showing poor instrumental recognition when stimuli are recorded under anechoic conditions that give the listener only what the instrument happens to radiate in the direction of the microphone. However, source/listener motions and/or a few close-in reflectors (with or without instrumental onsets and decays) restore recognizability to recorded sounds. This is consistent with experiments that compare perception of carphone-presented stimuli with those arising from the statistical sound field of a room. Outlines of such experiments and of practical examples will illustrate and support the views described, and suggest the insecure basis of hall designs guided by undefined “listener preference” data obtained with variously processed versions of music recorded in an anechoic chamber. [Work supported by NSF.]
Speech perception studies in the first year of usage of a multiple electrode cochlear implant by prelingual patients
Authors: Busby P. A. , Tong Y. C. , Clark G. M.
Published: October 2005

Vowel and consonant confusion studies, in the electrical alone (E), visual alone (V), and electrical plus visual (EV) conditions, and three Minimal Auditory Capabilities (MAC) subtests, in the E condition, were conducted on three prelingual patients using a multiple electrode cochlear implant. For vowels, the one-dimensional solution from multidimensional scaling analysis for the E condition was interpreted as vowel length, and the one-dimensional and two-dimensional solutions for the V and EV conditions as visual parameters. The constant results from information transmission analysis for the E condition indicated poor transmission of all articulatory features, and minimal differences between transmission scores for the V and EV conditions were recorded. For the MAC subtests, two patients scored significantly above chance for male-female speaker identification, one patient for one syllable-two syllable identification, but no patients for spondee same-different discrimination. The results suggested that these prelingual patients did not fully utilize the information provided by a multiple electrode implant during the first year of use.
Cochlear neuronal degeneration precipitated by experimental blockage of the spiral lamina perforated zone
Authors: Lusted Hugh S.
Published: October 2005

Observation of patterns of neuronal degeneration in cats with cochlear implants has led to the hypothesis that blockage of the spiral lamina perforations (canaliculae perforantes of Schuknecht) by the implant produces localized dendritic degeneration and ganglion cell loss. To test this hypothesis four cats were implanted with 1 × 1 mm squares of polyethylene (PE) film placed directly on the perforated zone (four ears). Four control ears received similar PE squares placed on the scala tympani wall opposite the basilar membrane. Histological examination of the cochleae after 3 months survival time revealed almost total dendritic degeneration and 50%–75% ganglion cell loss in localized regions of the cochlea where the implant fibrous encapsulation could be seen in direct contact with the perforated zone. No neuronal degeneration was observed in the cochleas which showed no fibrous tissue in contact with the perforated zone. These results indicate that the perforated zone is a contact critical area and needs to be avoided by implanted electrodes intended for functional stimulation of the cochlea. [Work supported by NIH.]
Processor controlled hearing aid
Authors: Gilman Samuel
Published: October 2005

The amplified signal input from the hearing-aid microphone is split into one-third octave bands, each having its own processor-controlled band amplilfier. Control signals are obtained from a feedback probe microphone at the medial end of the hearing aid earmold. This signal is analyzed into one-third octaves, digitized, and compared to stored band levels in the processor (as obtained from the audiologically specified eardrum SPL and spectrum). The gain of each band amplifier is controlled by the processor so that the spectrum and SPL at the eardrum is the same as the stored spectrum and SPL regardless of variations in input signals, hearing aid characteristics, and external ear acoustics. Processor control of averaging, attack, and release times have been provided. Preliminary ANSI S3.25-1979 simulator graphic and listening tests show that band levels and averages at the eardrum are maintained within a few dB for variations of over 30 dB in the input spectrum and level.
ISPUD, an interactive environment for signal processing on a VAX computer
Authors: Peterson Patrick M. , Frisbie Joseph A.
Published: October 2005

To support our group's research in psychoacoustics and hearing aids, a useful software package, ISPUD, for computer-based signal processing has been developed. Following the work of Kopec [IEEE Trans. Acoust. Speech, Signal Process. ASSP-32, 842–851 (1984)], ISPUD uses abstract data objects to represent signals in a way that leads to a simple, interactive, and extensible environment for signal processing. In contrast to Kopec's package, which requires a LISP machine and supports one user, ISPUD runs on a VAX with the multi-user VMS operating system, a C compiler, and Tektronix-4010 compatible terminals. The user interface is a simple, LISP-like interpreter for calling defined functions or combining functions into new definitions. The package includes efficient built-in functions, written in C, for signal generation, arithmetic, composition, windowing, filtering, transforms, statistics, graphical display, and file I/O. In addition, user-written C functions can be incorporated. Examples of ISPUD usage, including strengths and weaknesses, will be presented. [Work partially supported by NIH.]
Asymptotic solution to low-frequency fluid-loaded structure problems
Authors: Crighton D. G.
Published: October 2005

This paper will review the structural and acoustical features of asymptotic solutions recently obtained for fluid-loaded structure problems. In the cases treated here, the structure is plane (a thin plate or membrane) and driven by a source of concentrated mechanical excitation. The asymptotic results, based on the smallness of the fluid-loading-at-coincidence parameter, hold generally at frequencies below coincidence, but assume particularly simple forms at much lower frequencies. Problems discussed will include semi-infinite and finite unribbed structures (with details of resonance, mode shapes, and forced response), and infinite and semi-infinite ribbed structures. An aspect subject to particular study will be the nonlocal (i.e., non-added-mass) effects of fluid loading; and generally, the aim is to expose fundamental mechanisms for structural vibration with heavy fluid loading, and to provide benchmarks for computational schemes. [Work supported by ONR.]
Fluid-loading effects upon scatter of elastic waves at discontinuities
Authors: Smith P. W.
Published: October 2005

Vibratory waves in structures constructed of many plate or shell elements joined together are scattered by the discontinuity of elastic properties at the junctions. Descriptions of that scatter (reflection, transmission, and radiation into adjacent fluid) are important in analyzing the behavior of realistically complex structures. Results of analyses that have included significant fluid loading are reviewed and their implications for structural response discussed.
Vector quantization for speech coding and recognition
Authors: Gray Robert M.
Published: October 2005

Vector quantization (VQ) is the coding or mapping of continuous or high-resolution vectors into a finite rate digital representation. The general goal is either analog-to-digital conversion or data compression with a minimum loss of fidelity. In this paper, the basic vector quantizer structures and code design techniques for fixed rate vector quantizers are surveyed and applications to speech coding and recognition are described.
Relative entropy and vector quantization
Authors: Shore John E.
Published: October 2005

Consider the problem of quantizing a vector F of measurements Fr, r = 0,…,M. In one general form of vector quantization F is quantized as the result of classification by a nearest-neighbor rule, D(F,F̂(t)) = minsεΛ D(F,F̂(s)), where D is some distortion measure and where {F(s)⋅sεΛ} is a set of predefined vectors, often called codewords. Often, the measurements Fr can be expressed as a set of expected values, ∫q† (x) fr(x)dx = Fr, and each codeword F(s) can be expressed similarly. In such cases, the relative-entropy, H(q,p) = ∫q(x)log[q(x)/p(x)]dx, can be used to define a distortion measure in (1) so that the classification is optimal in a well-defined information-theoretic sense and also computationally attractive. Furthermore, the distortion measure results in a simple method of computing codewords from training data. The method of speech coding by vector quantization is a special case of this relative-entropy method.