bluuebunny/crossref_metadata_2025 huggingface dataset:
Field tests of a high-temperature fiber optic lever microphone
Authors: Zuckerwar Allan J. , Cuomo Frank W.
Published: October 2005

The operating principle of a high-temperature fiber optic lever microphone (FOLM), rated at 538 °C (1000 °F) continuous service, is based on intensity modulation of light reflected from a vibrating membrane. The frequency response of the microphone, calibrated in the laboratory at ambient temperature and 538 °C by the electrostatic actuator method, is typically 60–70 kHz. Applications to high-temperature field testing include the thermal acoustic fatigue apparatus at Langley Research Center, an advanced short takeoff and vertical landing model at Lewis Research Center, the Mach 22 helium tunnel at Langley (ambient temperature only), interior mixer-ejector surfaces at GE Aircraft Engines, Cincinnati, and a high temperature acoustic impedance tube at Langley. In most of these applications the FOLM made possible high-temperature, high-frequency acoustic data to be taken for the first time. Typical test results will be discussed.
Wind turbulence spectra for heights from 2 to 32 m measured during the JAPE experiment
Authors: Hansen Scott D. , Gilbert Kenneth E.
Published: October 2005

During the Joint Acoustic Propagation Experiment conducted at the White Sands Missile Range in the summer of 1991, wind speed measurements were made on two 32-m-high towers. The vector wind speed was sampled continuously at 10 Hz for periods of up to 2 h. The power spectra for wind speed fluctuations (turbulence spectra) have been computed at heights of 2, 4, 8, 16, and 32 m. For a given sample length, the spectra are nearly the same at all heights. At all heights, however, there is significant sensitivity to the length of the sampling period. The meteorological reasons for the sensitivity are discussed. In addition, selected acoustic propagation calculations are presented to demonstrate the influence of the different spectra on acoustic propagation. [Work supported by the Army Research Laboratory.]
Panel response to jet noise under near sonic conditions
Authors: Fenno Charles C. , Bayliss Alvin , Maestrello Lucio
Published: October 2005

The problem of the response of an array of flexible aircraft-type panels excited by noise from a jet under near sonic conditions from a converging nozzle is numerically studied. The problem is computed by solving the Euler equations for the unsteady field in the jet, fully coupled to equations describing the panel motion. Computations of the far-field sound, panel response, and panel radiation are presented. The effect of nozzle geometry on the radiated sound and jet instabilities is studied. In addition, the relationship between panel location, relative to the jet exit, and panel response is determined. The computation simulates the development of large amplitude, slowly propagating instability waves in the jet which act as additional sources of sound. Thus the computation allows for direct computation of the natural sources of jet noise, as well as the propagation of the resulting jet noise. The results demonstrate large disturbances in the vicinity of the converging nozzle which act as additional sources of sound. The panel response is concentrated in progressively lower frequencies as distance from the jet exit increases.
Active control of broadband random noise using energy density control
Authors: Park Young C. , Sommerfeldt Scott D.
Published: October 2005

Previous work on active noise control in enclosures has indicated that one can often achieve improved global attenuation of deterministic signals by minimizing the acoustic energy density, rather than the squared pressure. The purpose of this work is to present numerical results that compare the global attenuation of broadband noise achieved by minimizing energy density with the attenuation achieved by minimizing squared pressure in a one-dimensional enclosure. A theoretical approach based on the frequency domain enables one to establish the basic physical limitations of active noise control systems. However, this approach cannot necessarily be applied when controlling a broadband random noise, since it often yields optimal control solutions that are noncausal in the time domain, even though such a frequency domain approach is entirely satisfactory for deterministic signals. Analyses reported here are undertaken in the time domain in a manner that yields the form of the causally constrained optimal controller. Numerical results are presented to predict the performance of the active noise control system designed to control a broadband noise, and to indicate the improved global attenuation of broadband noise that can be achieved by minimizing energy density, rather than squared pressure.
Insertion loss studies of a baffle-simple expansion chamber system
Authors: Xiao P. , Prasad M. G.
Published: October 2005

Insertion loss (IL) is the difference of sound pressure levels measured at a reference point without and with a muffler in a duct system. Thus, IL describes the acoustical performance of the whole system which includes source, muffler, and tail pipe. IL studies of a simple expansion chamber has been reported in the literature. However, there are not many IL studies of complicated geometries in a muffler from a system design point of view. This study reports both experimental and analytical work on the influence of a baffle on IL of a simple expansion chamber. The hole diameter and location of the baffler are the parameters of this study. An Altec electroacoustic driver is used as a source for experimental work conducted in an anechoic chamber. The analytical work is carried out using the boundary element method. The results indicate a significant influence (due to the interaction of source and load) on the insertion loss.
Reflection of higher-order modes at the end of rectangular ducts
Authors: Muehleisen Ralph T. , Swanson David C.
Published: October 2005

Reflection coefficients for higher-order modes impinging on the ends of finite rectangular ducts were predicted and measured. The results show coupling between propagating and evanescent modes. Expressions for the reflection coefficients in terms of modal coefficients and modal impedances have been developed. Radiation patterns were predicted from the knowledge of the coupling coefficients. It was found that higher-order modes radiate most of their energy off towards the sides and less straight ahead than the plane wave mode.
Acoustical studies of a duct system using a power flow model
Authors: Kim Woo-sung , Prasad M. G.
Published: October 2005

Any duct system from an acoustical point of view can be modeled as a source–load system. The power flow analysis in a source–load system is important for a system’s acoustical design. The present work describes the development of a power flow model utilizing the source and load impedances as parameters. Also, the source is described in terms of its power rather than the pressure or the volume velocity. The study includes both analytical and experimental work. The analytical work deals with the development of expressions for power flow in the system. The experimental work is carried out using an Altec electroacoustic driver as the source and various ducts as the loads. The system performance is evaluated using the power flow model for various loads. The study has yielded good results validating the power flow model for duct systems.
A multi-mode acoustic filter based on Bragg resonance
Authors: Hawwa Muhammad A. , Fuller Chris R. , Burdisso Ricardo A.
Published: October 2005

A unified method is proposed to suppress duct acoustic waves. According to this method, a circular cylindrical duct is designed to have a geometric periodicity at its wall in order to act as a wave filter. The filter action of the periodic rigid wall is based on the mode coupling occurring under the Bragg condition resulting in a strong stopband interaction. The problem is attacked using the perturbation method of multiple scales, leading to the derivation of the coupled-mode equations. The performance of the duct as a wave attenuator is presented in terms of the reflection coefficient or the transmission loss factor. Due to the nature of the dispersion curves of the cylindrical duct, many modes could share the same stopband. Hence, a multi-mode acoustic filter could be designed over a certain frequency band. This theoretical study might provide a basis for design concepts of acoustic silencers.
Current strategies in sonic-boom research
Authors: Darden Christine M.
Published: October 2005

During NASA’s high-speed research program, a renewed emphasis has been placed on understanding and reducing the sonic boom generated by an airplane during flight at speeds faster than the speed of sound. When the current program was initiated in 1990, a three-pronged approach was instituted: (1) human acceptabilities studies; (2) atmospheric propagation studies; and (3) airplane design and operation studies. Original goals were to use special design and operation techniques to reduce the sonic boom to levels which would be acceptable for either unrestricted overland flight or overland flight through low-populated corridors. Recent human response studies have indicated that sonic booms are much more disturbing than continuous noise sources such as engines around airports; thus any commercial overland supersonic flight is highly unlikely within the near future. These conclusions have resulted in a redirection of the sonic-boom program. This talk will discuss current efforts at sonic-boom softening (small design changes), continuing efforts to understand the impact of the atmosphere on the rise time of the sonic-boom signature, and strategies to understand what impact, if any, the sonic boom has on marine life.
Studies of sonic booms with seismic networks
Authors: Sturtevant Bradford , Cates Joseph E. , Kanamori Hiroo
Published: October 2005

Seismographs are sufficiently sensitive to detect ground motions induced by atmospheric pressure waves, so seismic networks have the potential to monitor sonic booms over large areas of the United States. They are especially well suited for the analysis of long-range sonic-boom propagation. Ground motion or displacement data provide accurate arrival times and useful estimates of wave amplitude and waveform. The instrumentation is most sensitive to the disturbance produced by the arrival of sonic booms at the measuring station, thus serving as sonic-boom event recorders, but seismographs have also detected seismic waves remotely generated by anomalous coupling of sonic boom into soil. Direct and indirect sonic booms from aircraft operations are routinely detected by the Southern California Seismic Network which consists of 250 seismic stations covering 50 000 sq km. Indirect booms from space shuttle landings has been observed at ranges of hundreds of kilometers from the flight path. Data from the network identified ‘‘mystery booms’’ heard in 1992–93 to be long-range indirect sonic booms from offshore operations. Sonic booms generated by space shuttle reentry at Mach 20 and by meteoritic entries into the atmosphere have been detected by seismic networks in the Northwest and Canada.
The theoretical and computational basis of focused sonic booms
Authors: Plotkin Kenneth J.
Published: October 2005

For the past 20 years sonic-boom signatures in focus ‘‘superboom’’ regions have been computed using a method based on a scaling law by Guiraud [J. Mech. 4, 215–267 (1965)] and a numeric solution by Gill [Ph.D. thesis, Cornell University (1974)]. The method recognizes that the smooth caustic case treated by Guiraud and Gill is by far the most common focusing situation, and that smooth caustics caused by aircraft accelerations and turns and by atmospheric gradients are topologically similar. Gill’s numeric result for a shock may thus be treated as a canonical result, and scaled to arbitrary situations by Guiraud’s similitude. The process has been incorporated into an existing sonic-boom program. When a focus is detected, the program fits the solution by (a) computing ray and caustic curvatures required by the theory, (b) determining a matching position consistent with Gill’s boundary conditions, and (c) applying the scaled Gill solution to each shock. The computational requirements of this process are reviewed, together with empirical support for the results.
Sonic boom from the Titan IV space launch vehicle launched from Vandenberg Air Force Base
Authors: Francine Jon K. , Stewart Brent S.
Published: October 2005

The sonic boom generated by a Titan IV space launch vehicle launched on 7 November 1991 was recorded at two locations on San Miguel Island, California. A Bruel & Kjaer microphone carrier system recorded the overpressure near a haul-out site for about 40 000 pinnipeds (seals and sea lions); a boom event analysis recorder (BEAR) was placed under the predicted focal region. The boom reached the pinniped site 4 min and 38 s after launch. It was neither an N nor a U wave at either site; rather, it had several distinct positive peaks, which indicates that the focal region was uprange of the recording sites. The first and fourth positive peaks were 853 ms apart. At the pinniped site, most of the boom’s energy was at low frequencies (ca<500 Hz), as indicated by the 46-dB difference between the 124-dB unweighted sound exposure level (SEL) and the 78 dB A-weighted SEL. The peak overpressure recorded near the pinnipeds was 64.1 Pa (1.34 psf; 130 dB peak SPL). The BEAR system recorded a peak overpressure of 90.5 Pa (1.89 psf; 133 dB peak SPL). Empirical tests of predictive models for focal regions and descriptions of the frequency characteristics of focused booms are still needed to evaluate the potential risks to wildlife from these booms.
Female alaryngeal speech enhancement for improved speaker identification using linear predictive synthesis
Authors: Tull Renetta G. , Rutledge Janet C. , Mahler Jerry J.
Published: October 2005

Linear predictive synthesis was applied to female alaryngeal speech as an enhancement technique to reduce gender ambiguity during speaker identification. Esophageal speech, a form of alaryngeal speech resulting from the removal of the larynx due to cancer, perturbs the theoretical acoustic models for laryngeal speech. This form of pathological speech also changes the expected spectral characteristics and affects perception. The female esophageal speaker’s low fundamental frequency often causes her to be identified as a male speaker over telephone lines. The errors made in speaker identification can be reduced by obtaining vocal tract parameters of the aperiodic, esophageal speech input from a linear predictive coding (LPC) model and resynthesizing the LPC prediction coefficients using an appropriate pitch period that is better suited for female voice. This enhancement technique was implemented using the TMS320C30 floating point digital signal processor from Texas Instruments. This implementation supports the potential of a portable processor that enhances the perceptual quality of the female alaryngeal voice and improves the input speech signal used by both human listeners and automatic speech and speaker identification systems.
Removal of phase distortion from vowel-like signals
Authors: Shipley Kenton A. , Winkel Casey R. , Vaidya P. G.
Published: October 2005

In this paper, specific types of signals are considered. To generate these signals, we begin with x(t)=A cos[wt+φ(t)], where A and w are constants, but φ(t) is a function of time generated by a random walk. The other step to generate the final signal is to pass x(t) through a nonlinear filter such that, were φ(t) to be constant the output would be of simple harmonics at w, 2w, 3w, etc. However, since x(t) is nonstationary, the output signal, say y(t), would also be nonstationary. It would be shown that human vowel signals seem to follow this nonlinear pattern. It is of interest to remove (or modify) the phase distortion due to the randomness of φ(t). This has been accomplished by using two different methods. One uses the trans-spectral coherence technique [P. G. Vaidya and M. J. Anderson, 2370–2378 (1991)] and the other uses the complex FFT components in a manner similar to a rigid body rotation in dynamics. These methods have been used to produce both distortion-free and highly modulated versions from a given human vowel signal. a)Now at Martin Marietta, Knolls Atomic Power Labs, Schenectady, NY.
Identification of vowels resynthesized from /hVd/ utterances: Effects of formant contour
Authors: Hillenbrand James
Published: October 2005

The purpose of this study was to examine the role of formant frequency movements in vowel recognition. Measurements of fundamental frequency and formant contours were taken from a database of acoustic measurements of 1668 /hVd/ utterances spoken by 45 men, 48 women, and 46 children [Hillenbrand etal., J. Acoust. Soc. Am. 97 (in press)]. A total of 300 utterances were selected from this database, representing equal numbers of 12 vowels (i,i,e,eh-,ash,ligature,a,open oh,o,u,u,inverted v,hooked backward eh and approximately equal numbers of tokens produced by men, women, and children. Listeners were asked to identify three versions of each utterance: (1) the original, naturally produced signal, (2) a formant-synthesized version that followed the measured contours of F1–F3, and (3) a formant-synthesized version with F1–F3 fixed at their steady-state values. Preliminary results indicate: (1) the original signals are identified with substantially greater accuracy than either of the synthetic versions, and (2) the synthetic signals that follow the original formant contours are identified with greater accuracy than the flat-formant signals.
Generating gestural scores from articulatory data using temporal decomposition
Authors: Collins Michael J. , Ahalt Stanley C. , Krishnamurthy Ashok K.
Published: October 2005

Through empirical investigations, the automatic generation of gestural scores for articulatory data corresponding to consonant–vowel–consonant (CVC) tokens was studied. The articulatory data consist of the movements of flesh points measured using an x-ray microbeam, and is first ‘‘warped’’ to resemble the vocal tract variables of constriction location and constriction degree. The multichannel warped data are then analyzed using temporal decomposition. The resulting target functions provide candidates for gestures, from which the best candidates are chosen statistically by examining the magnitudes of the elements of their associated reconstruction weights. Onset and durational analysis of the candidate target functions result in the gestural score. Human and Elman Recurrent Neural Network recognition tests are performed to ascertain the accuracy of the generated gestural scores. Comparisons with ‘‘correct’’ gestural scores are also performed. The results of this work should provide a stepping stone for future acoustic and articulatory based recognizers employing the same strategy.
Layered language understander: A speech recognition system using layered abduction
Authors: Lenzo Kevin A. , Josephson John , Fujimura Osamu , Bailey-Kellogg Christopher
Published: October 2005

A speech recognition system is under development in which a general-purpose inference engine for natural language processing is loaded with domain knowledge and a preprocessing periphery that allows for the integration of prosodic information. Processing takes place in well-defined layers, in which results at one level of abstraction become data to be explained by hypotheses in the next, using a layered-abduction inference mechanism that allows for integration of both top-down and bottom-up information [J. R. Josephson and S. G. Josephson, eds., AbductiveInference: Computation, Philosophy, Technology (Cambridge U. P., 1994)]. The hypothesis types and their interactions are based upon the converter/distributor model of phonetic implementation [ O. Fujimura, ‘‘Syllable Timing Computation in the C/D Model,’’ Proceedings of the Third International Conference on Spoken Language Processing (1994)]. Each hypothesis is annotated with a numerical magnitude that is used in computing a ‘‘prosodic contour’’ for the overall utterance, which aids in the generation of expectations, implications, and knowledge of which hypotheses can account for what data. Preprocessing events are explained by the features, which, in turn, are explained by hypotheses at the demisyllabic layer, the syllabic layer, and, finally, by the word-level hypotheses.
Adding articulatory features to acoustic features for automatic speech recognition
Authors: Zlokarnik Igor
Published: October 2005

A hidden-Markov-model (HMM) based speech recognition system was evaluated that makes use of simultaneously recorded acoustic and articulatory data. The articulatory measurements were gathered by means of electromagnetic articulography and describe the movement of small coils fixed to the speakers’ tongue and jaw during the production of German V1CV2 sequences [P. Hoole and S. Gfoerer, J. Acoust. Soc. Am. Suppl. 1 87, S123 (1990)]. Using the coordinates of the coil positions as an articulatory representation, acoustic and articulatory features were combined to make up an acoustic–articulatory feature vector. The discriminant power of this combined representation was evaluated for two subjects on a speaker-dependent isolated word recognition task. When the articulatory measurements were used both for training and testing the HMMs, the articulatory representation was capable of reducing the error rate of comparable acoustic-based HMMs by a relative percentage of more than 60%. In a separate experiment, the articulatory movements during the testing phase were estimated using a multilayer perceptron that performed an acoustic-to-articulatory mapping. Under these more realistic conditions, when articulatory measurements are only available during the training, the error rate could be reduced by a relative percentage of 18% to 25%.
Vocal tract normalization in speech recognition: Compensating for systematic speaker variability
Authors: Cohen Jordan , Kamm Terri , Andreou Andreas G.
Published: October 2005

The performance of speech recognition systems is often improved by accounting explicitly for sources of variability in the data. In the SWITCHBOARD corpus, studied during the 1994 CAIP workshop [Frontiers in Speech Processing Workshop II, CAIP (August 1994)], an attempt was made to compensate for the systematic variability due to different vocal tract lengths of various speakers. The method found a maximum probability parameter for each speaker which mapped an acoustic model to the mean of the models taken from a homogeneous speaker population. The underlying acoustic model was that of a straight tube, and the parameter estimation was accomplished by warping the spectrum of each speaker linearly over a 20% range (actually accomplished by digitally resampling the data), and finding the maximum aposteriori probability of the data given the warp. The technique produces statistically significant improvements in accuracy on a speech transcription task using each of four different speech recognition systems. The best parametrizations were later found to correlate well with vocal tract estimates computed manually from spectrograms.
Use of TRACTTALK for adaptive voice mimic
Authors: Lin Qiguang , Richard Gael , Zou Jingyun , Sinder Dan , Flanagan James
Published: October 2005

Various speech-processing technologies necessitate parametrization of the speech waveform. Cepstrum coefficients (including their derivatives and variants) are to date commonly used in speech and speaker recognition. This paper seeks more compact parametric description of speech information based on the adaptive voice mimic [Flanagan etal., 780–791 (1980)]. The mimic system utilizes an articulatory-based speech synthesizer to generate synthetic speech, which is adapted to arbitrary speech input. The perceptually weighted spectral difference between the input and synthesized speech is next minimized by optimizing the underlying articulatory parameters until the difference is driven below a predetermined level. The resultant representation, adapted moment by moment, provides efficient parametrization of the signal information by which the problems of speech synthesis, speech recognition, and low bit-rate speech coding are coalesced into a compact framework. In this paper, an articulatory speech synthesizer, TRACTTALK, is first described. TRACTTALK simulates the vocal tract based on principles of linear acoustics and incorporates features which include interaction between the voice source and the vocal tract. Preliminary results of adaptive mimicking using TRACTTALK are presented and discussed. [Work supported by ARPA Contract ♯DAST 63-93-C-0064.]
Testing a regional ocean model using acoustic travel time observations
Authors: Cornuelle Bruce D. , Worcester Peter F. , Chester David B. , Dushaw Brian D. , Howe Bruce M.
Published: October 2005

The Acoustic Mid-Ocean Dynamics Experiment (AMODE) measured time series of reciprocal acoustic travel time and temperature with a 700-km-diam array of six moorings south of Bermuda during 1991–1992. Time-independent inversions give a series of snapshots of the barotropic and baroclinic eddy field. Observations taken at different times can be combined, using a dynamical ocean model, to generate improved estimates of the evolving eddy field, provided that the model is accurate. The AMODE observations have been used to test a nonlinear, quasigeostrophic, limited-area ocean model by attempting to fit a long-term dataset, and by initializing the model with a ‘training’ dataset and then predicting forward in time for comparison to data not used in the initialization. Using an rms error measure of misfit between the observations and the predicted data, the model prediction has 40% less error than persistence over 12 days, and about 20% less error than persistence over 24 days. The extent to which the data misfit is sensitive to external model parameters, such as the radius of deformation or viscosity, allows the parameters to be optimized to give maximum predictability, and also gives a measure of how well the model has been tested by the dataset. [Work supported by ONR.]
Assimilating tomographic travel time data using an open ocean circulation model
Authors: Walter Chris G. , Howe Bruce M. , Mercer James A. , Miller Robert N.
Published: October 2005

Observing and predicting the ocean’s meso-scale circulation presents a formidable task for oceanographers. Ocean acoustic tomography uses travel times between acoustic sources and receivers to measure the ocean’s low wave-number sound speed (temperature) and current fields. Experiments are being performed with simulated ocean data to determine the relative effectiveness of tomographic and point measurements in constraining a numerical circulation model. The primary tool is data assimilation, i.e., an optimized combination of model output and observations. Relative weights of observation and model output are based on apriori estimates of the model and observation error covariance. Results from identical twin simulations using tomographic and point measurements are presented. In the twin experiment, two assimilation schemes are examined, one in which the estimated model error covariance is fixed (optimal interpolation) and another in which a Monte Carlo [G. Evensen, J. Geophys. Res. 99, 10 143–10 162 (1994)] method is used to estimate the model error covariance dynamically. Error statistics of the energy and flux fields are used to quantify the performance for each observation and updating scheme. The number of measurements required to achieve a target error level is estimated for each method. [Work supported by ONR and by the Strategic Environmental Research and Development Program through ARPA.]
A 4-D view of convection by assimilation of acoustic and classical measurements
Authors: Gaillard F. , Desaubies Y. , Send U. , Schott F.
Published: October 2005

Acoustic tomography has been applied to observation of winter convection in the Gulf of Lion during the THETIS-1 experiment conducted during the 1991–92 winter by a European group (IFM Kiel, IFREMER Brest, and IACM Heraklion). A global analysis of all types of temperature measurements collected during THETIS 1 will be presented. Data issued from temperature sensors, thermistor chains, and CTD have been merged with the tomographic measurements in order to recover the time evolution of the full 3-D temperature field. The method relies on inversion and Kalman filtering methods. It can be considered as a new step toward ‘‘observation systems.’’ Monitoring of the whole area during the ‘‘preconditioning’’ phase shows the influence of surface cooling on the vertical stratification. This effect is inhomogeneous and the situation evolves with time, influenced by horizontal advection. The relative importance of surface forcing and advection is responsible for the intermittent nature of convection which happens with different intensity from year to year. After convection, the behavior of the mixed water chimney and the evolution of the vertical profile toward the initial state is documented.
Ocean structure inversion by horizontal refraction tomography
Authors: Voronovich A. G. , Shang E. C.
Published: October 2005

Numerical simulations for the horizontal refraction tomography (HRT) proposed previously [A. G. Voronovich and E. C. Shang, 2851(A) (1994)] were conducted. HRT is based on measurements of horizontal refraction angles (HRA) related to different acoustic modes. HRA are measured with the help of two distant ocean interferometers (each one consists of a pair of long mode resolving vertical arrays separated about 10 km apart). The signal is produced by tonal towed acoustic source. Both model eddy and real oceanic eddy of 100 km×100 km scale measured in Canary Basin were used for simulations. The 3-D tomography inversion proceeded in two stages: (1) retrieving the values of propagation constants in the knots of a horizontal rectangular greed (linear 2-D problem) and (2) reconstruction of the sound-speed profile (SSP) in each knot (1-D nonlinear problem). Up to 81 values of propagation constants for each mode were retrieved in both cases. Then SSP were calculated in each knot with the help of a special invariant-imbedding based procedure. With error-free simulated HRA data, SSPs were exactly reconstructed. The effect of errors on tomography inversion and requirements to the accuracy of phase measurements were also estimated. [Work supported by ONR.]
Determination of the inhomogeneity position by the time structure of a scattered signal
Authors: Fokin Vladimir N. , Fokina Margarita S.
Published: October 2005

The method is based on the fact that the mean velocity of sound pulse propagation along the ray trajectory depends on the angle of the ray exit. The efficiency of the method is analyzed numerically for the signal reception at a horizontal array and a single hydrophone. It is shown that the signal of illumination may be suppressed and the scattered signal may be separated by time gating. It is also shown that the depth of the deposition of a space-localized inhomogeneity may be estimated by the width of the scattered signal autocorrelation function. The distance to the inhomogeneity may be defined by the cross-correlation function of the scattered signal at various array receivers. The influence of the depth of the submergence of a scatterer and the distance between a source and a receiver on the scattered signal structure is analyzed. [Work supported by RFFI project N 94-05-1725-a.]
A fiber optic dual-beam laser Doppler vibrometer for measurement of electrostrictive and piezoactive response of thin films
Authors: Guillot F. , Jarzynski J. , Balizer E.
Published: October 2005

A two-beam laser Doppler vibrometer (LDV) is described, designed to measure the d33 electromechanical coupling coefficient for electrostrictive and piezoactive thin films. Optical fiber couplers are used to split the light into multiple beams and optical fibers are used to simultaneously illuminate the same spot on both sides of the film. The LDV measures the normal displacement on each side and the change in thickness of the film is determined from the algebraic sum of the two LDV signals. It is found that large bending motion of the film can lead to a significant error in the measured d33 coefficient. This is illustrated with data for PVF2 films and sources of this error are identified. The application of the LDV system to measure the large electrostrictive effect in polyurethane films is discussed. [Work supported by the Office of Naval Research.]
Target visibility measurements: Three-dimensional laser vibrometer
Authors: Vignola Joseph , Zehl Otis G. , Houston Brian H.
Published: October 2005

A three-dimensional laser vibrometer has been constructed, tested, and used for simultaneous measurements of three components of induced transient motion of target surfaces. Such target surfaces need to satisfy a wider range of optical scatter requirements than those for one-dimensional motion measurement. This presentation outlines some of these requirements for high signal visibility and describes several interesting target surfaces under test, including stainless steel, paints, polymer, and retroreflecting preparations. Data presented for these surfaces include signal strength (visibility), repeatability, stability, and effects of surface roughening and a water environment. Results are interpreted in terms of simple physical models where possible to better understand potential specialized target fabrication for improving future vibrometer measurement system performance.
On local versus global parametrization of short-pulse-excited scattering and spectra: Poisson summation revisited
Authors: Felsen Leopold B.
Published: October 2005

Poisson summation has conventionally been employed for conversion of local scatterings due to individual elements in an infinite periodic array into global Bragg spectra that characterize the collective phenomena due to the entire array. In a recent generalization, time harmonic and transient local–global phenomena in finite periodic and quasiperiodic arrays have been related via finite Poisson summation, with the global outcome interpreted as radiation from equivalent sources distributed over the finite array aperture [L. B. Felsen and L. Carin, 638–649 (1994)]. This analysis is now re-examined in the context of multiple scatter scenarios under short-pulse time-domain excitation. Since short enough pulsed incident fields can time-gate individual scattered field arrivals, the early time response at the observer is necessarily parametrized locally. As multiple interaction develops, one may reparametrize any finite number of these collectively in terms of spectra associated with equivalent sources that are smoothly distributed over the corresponding multipass finite space-time aperture. This results in global algorithms based partly in the configuration domain and partly in the spectral domain. These concepts are developed and examined rigorously and asymptotically with respect to the time evolution of global spectra from highly resolved early scatters under short-pulse time-domain conditions. Corresponding statistical aspects, when the scattering hierarchy is randomly perturbed, are explored as well. [Work supported by AFOSR and ONR.]
Short pulse sound wave scattering by a fluid shell model target near a randomly distorted bottom interface: Wave oriented data processing for target-clutter discrimination
Authors: Hsu T. T. , Carin L. , Felsen L. B.
Published: October 2005

A finite difference time-domain (FDTD) code has been used to assemble a comprehensive data base of realizations for short-pulse plane wave scattering by a fluid shell model target submerged in the presence of a randomly distorted bottom interface. The entire problem is two dimensional with the scattered acoustic pressure observed along an elevated track above the target and parallel to the mean bottom boundary. Two bottom topographies are examined: (1) a smoothly deformed interface between the wated column and a fluid bottom; (2) a collection of soft randomly pitched thin flat strips. The database furnished by the ensemble of realizations for the randomly irregular bottom, in which the deterministic target remains unchanged, is subjected to wave-oriented data processing that yields space-wave number and time-frequency phase space distributions extracted via windowed transforms and refined by locally applied high resolution algorithms. The processing is applied to the target alone, the cluster alone, and the target–clutter combination. Target resonances are regarded as a major discriminant which one needs to enhance by input signal shaping and output filtering while simultaneously de-emphasizing the corrupting effect of the clutter. Statistical measures pertaining to the clutter are given consideration, as is the effect of system noise added to the data. Results are presented and discussed. [Work supported by ONR and AFOSR.]
Sound diffraction by a ground with barriers or impedance discontinuities in a refracting atmosphere
Authors: Li Y. L. , White Michael J.
Published: October 2005

The exact solution for the total field is obtained for the sound diffraction due to a time-harmonic line source above a ground in a medium with linear sound-speed profile. The ground may have barriers with arbitrary shapes or impedance discontinuities parallel to the line source. The solution is in terms of an integral equation which is numerically solved by an extension of the method of moments. The effects of scattering in the region around barriers or impedance discontinuities are studied for situations for various values of the refracting strength of the sound-speed profile.
Application of the PE method to up-slope sound propagation
Authors: Galindo Marta , Rasmussen Karsten Bo
Published: October 2005

The wide-angle PE method is applied to the prediction of sound propagation in a range-dependent environment. The finite-difference model treats the sloping ground by a staircase approximation. The method allows impedance and slope angle variations to be taken into account. The implementation of sloping interfaces in one-way PE models introduces a fundamental problem of energy conservation. This is generated at the lower boundary condition. The ground is assumed to be a locally reacting surface where the ground impedance is defined as the ratio of pressure to the normal component of particle velocity at the surface. The staircase PE method approximates the normal at the slope surface with the vertical component at the stair-step surface. A numerical correction can be included. Using a scale model facility [K. B. Rasmussen, 3617–3620 (1994)], a series of measurements is carried out. The results are compared with the PE calculations. The magnitude of the error due to the interface condition is evaluated.
The effects of talker variability across CV and VC environments
Authors: Farnsworth Lynn M. , Mullennix John W.
Published: October 2005

A consonantal speeded classification task was used to assess the effects of talker variability across CV and VC environments. In addition, conditions were blocked by syllable type (CV or VC) or consisted of mixed sets of CV and VC syllables. Talker variability was manipulated by presenting stimuli spoken by a single talker or by many talkers. The results showed that the magnitude of the effect of talker variability was approximately the same when comparing performance across talker contexts for CV’s and VC’s. In addition, the talker variability effect was stronger under conditions where syllables were blocked. The results provide further information about the underlying mechanisms involved in processing perceptual variation in the speech signal.
Speech during simultaneous communication: Temporal characteristics and perceived naturalness
Authors: Whitehead Robert L. , Schiavetti Nicholas , Whitehead Brenda H. , Metz Dale Even
Published: October 2005

Simultaneous communication is a process which combines spoken English with various manual representations of English words in the form of signs and fingerspelling. The present study investigated the durational characteristics and perceived naturalness of speech produced during simultaneous communication with increasing fingerspelling complexity. Ten normal-hearing adult females, skilled in the use of simultaneous communication, served as speakers. Experimental words of differing fingerspelling complexity were embedded in sentences which were: (a) spoken and (b) spoken and signed, except for the experimental words which were fingerspelled. Five temporal measures of speech were calculated: (a) sentence duration, (b) experimental word duration, (c) diphthong duration preceding the experimental word, (d) pause duration preceding the experimental word; and (c) pause duration following the experimental word. Speech naturalness was rated using a nine-point naturalness rating scale for all stimulus samples. Results indicated significant differences in durational measures and naturalness ratings between the speech-only and simultaneous communication conditions, and between levels of fingerspelling complexity. A multiple regression analysis was performed to predict speech naturalness from the durational measures used as independent variables. The multiple R2 was 0.698. [Work supported by U.S. Department of Education.]
The effects of feedback in auditory looming tasks
Authors: Wuestefeld A. Paige , Rosenblum Lawrence D.
Published: October 2005

Previous research [Rosenblum etal., Perception 22, 1467–1482 (1993)] demonstrates that listeners are able to make somewhat accurate anticipatory judgments regarding the time to arrival of a looming sound source and that performance feedback significantly improves these judgments. The present research examines the effects on performance of providing feedback to listeners and subsequently withdrawing it. Some visual research involving the extrapolation of movement suggests that, upon withdrawing feedback, performance will revert back to its original level. Subjects made judgments about the time to arrival of an approaching car based on various portions of the event. Each subject participated in three experimental sessions on consecutive days. Subjects were divided into three groups which received differing feedback schedules. For all groups, the first session contained no performance feedback. The withdrawal group received feedback during the second session, but not the third. One control group received no feedback for the remainder of the experiment. A second group control group received feedback for both the second and third sessions. When feedback was withdrawn, performance between the second and third sessions did not differ significantly. Thus, unlike the previous visual effects, performance did not revert back to its original level.
The relationship between the discriminability of formant transitions and certain cognitive measures
Authors: Carrell Thomas , Kraus Nina , McGee Terry , Zecker Steven
Published: October 2005

This work concerns the relationship between two measures of auditory processing and several measures of cognitive abilities. The discriminabilities of formant transitions in synthesized consonant–vowel syllables were collected from over 100 children aged 7–11. Specifically, the discriminabilities of the F3 onset frequencies were measured in a set of stimuli which ranged perceptually from /dopen aye/ from /gopen aye/. Also, the discriminabilities of the onset transition durations of F1, F2, and F3 were measured in stimuli which ranged perceptually from /bopen aye/ to /wopen aye/. A PEST adaptive tracking paradigm was employed to make these measurements. The average just noticeable difference was 101 Hz for the F3 onset frequencies and 6 ms for the F1, F2, and F3 frequency ramps. A battery of tests of cognitive ability were administered to the same children. The cognitive ability tests revealed a wide range of scores in the subjects tested. A correlational analysis of these data indicated that better onsetfrequency discriminability was significantly correlated with higher scores on the reading subtests, whereas better scores on transitionduration tasks were not significantly correlated with any of the cognitive ability measures. It was also found that better formant onset frequency discriminability did not predict higher transition duration discriminability. That is, better /dopen aye/ vs /gopen aye/ discrimination did not predict better /bopen aye/ vs /wopen aye/ discrimination. [Work supported by NI
Influence of the number of syllables and morae on reaction time in Japanese speech production
Authors: Masaki Shinobu , Honda Kiyoshi
Published: October 2005

In order to estimate the processing unit of Japanese speech production, the influence of the number of syllables and morae on reaction time (RT) in speech production was analyzed. In experiment 1, RT was measured for one- to four-mora nonwords consisting only of Japanese CV syllables [ka] or [ta]. Five subjects produced stimulus words as soon as possible after the visual onset signal was presented. RT was defined as the duration from the moment of onset signal presentation to the release of word-initial plosive consonant. Results showed a tendency for RT to increase as the number of CV syllables increases. In experiment 2, stimulus words consisted of CV syllables and vowels without a syllable-initial consonant, or CV syllables and nasal sounds [N] without a syllable-final vowel. Note that, in Japanese, a vowel and nasal sound [N] can be a mora like a CV syllable. For these stimuli, the RT was not influenced by the increase of the number of morae. These results suggests a possibility that a mora consisting of a CV syllable is the smallest segment of the processing unit in Japanese speech production. However, morae consisting of a single vowel or nasal sound cannot perform as independent processing units.
Horizontal multipaths caused by mesoscale structure and their effects on global tomography with near-axial paths
Authors: Tappert F. D. , Wolfson M. A.
Published: October 2005

The optimistic and questionable assumption that the lowest acoustic normal modes are adiabatic is used. Then, for each low mode a parabolic wave equation in the horizontal plane is derived that contains lateral variability of sound speed near the sound channel axis caused by mesoscale structure that is modeled by homogeneous and isotropic fluctuations having a single scale length, L∼100 km. In the geometrical acoustics limit, horizontal rays are found to be chaotic with the growth rate (Lyapunov exponent) given by ν∼ε2/3/L, where ε∼5×10−3 is the rms sound speed fluctuation. Horizontal multipaths begin at the focal range, rf∼ν−1∼3 Mm, and thereafter the number of horizontal multipaths per mode increases exponentially at the rate ν until diffraction effects limit this ray chaos. Full saturation is found to occur at the ‘‘log range,’’ rs∼rf ln(L2/λrf)∼14 Mm, for the acoustic wavelength λ∼30 m. Since mode coupling is neglected, this saturation range is a hard upper bound for doing global ocean acoustic tomography with near-axial paths. [Work supported by ONR and ARPA.]
Travel time effects of mesoscale structure on rays and waves at global ranges
Authors: Tappert F. D. , Tang Xin
Published: October 2005

A range-dependent ray trace model and a broadband PE model are used to model sound propagation at fixed bearing through a field of mesoscale baroclinic modes in order to study the effects of mesoscale structure on travel time at long ranges. The ray model exhibits chaos at ranges beyond a few Mm as manifested by an exponentially increasing number of eigenrays and triplications, especially in the late-time near-axial arrivals. In addition, the ray model predicts a mesoscale travel time bias, in the direction of later time, of the last axial arrival amounting to about 100–200 ms/Mm. At center frequency 75 Hz and bandwidth 50 Hz, the full-wave PE model qualitatively confirms the ray trace predictions of chaos, and shows that the later near-axial arrivals are smeared out into a continuum of unresolvable multipaths characterized by saturated (Gaussian) statistics at ranges of a few Mm, and also quantitatively confirms the mesoscale bias of 100–200 ms/Mm of the last axial arrival. The steeper, early raylike arrivals are relatively stable in the presence of mesoscale structure and may be useful for long-range tomography. [Work supported by ONR.]