bluuebunny/crossref_metadata_2025 huggingface dataset:
Analysis and prediction of vowel devocalization in isolated Japanese words
Authors: Kuwabara Hisao , Takeda Kazuya
Published: October 2005

An analysis of vowel devocalization in Japanese is presented as an example to find general rules for allophonic variations of phonemes in continuous speech, i.e., to find quantitative relations between phoneme environments and allophonic variations. A speech database consisting of about 5000 common Japanese words spoken in isolation by a professional male announcer was used. The database used is part of a large scale speech database that is now being compiled and hand transcribed at several levels of phonetic detail. Based on the acoustic-phonetic transcriptions of the speech data, factor analysis of the phonetic environment was performed for the devocalized vowels. The results reveal that the influence of the immediately following phoneme on devocalization is the greatest among the four factors examined. The second most important factor is the position of the accent nucleus. The automatic prediction of vowel devocalization is discussed using the results of this analysis.
Large-band integrated vowel representations
Authors: Mantakas Marios , Schwartz Jean-Luc , Escudier Pierre
Published: October 2005

“Gross” vowel spectrum parameters for vowel classification are of interest to many researchers. In the present study, the “effective formant” F′2, estimated by the large-band spectral integration (LBI) model [Escudier et al., Acts of the French-Swedish Seminar, Grenoble, France (1985)], in the classification of natural French front vowels /i/ vs /y/ and /e/ vs /ø/(rounding opposition), is evaluated. In parallel, formants' measures on the same corpus are provided in order to compare with and interpret the results obtained using the LBI model Classification performance is good in the case of /i/-/y/. The optimal spectral integration window is 2–2.5 Bark large (cf. with the 3–3.5 Bark “critical distance”). The model fails, however, in the /e/-/ø/ classification. Here, the second formant's frequency is by far winning. The model's weakness apparently resides in its last stage, namely peak estimation. In fact, recent results show that the LBI spectral representation can support the definition of form factors (other than peak position) for the classification of vowels.
Is there a single vibrato waveform
Authors: Maher Robert , Beauchamp James
Published: October 2005

High-fidelity singing synthesis requires careful consideration of the properties and character of natural vibrato. In many synthesis methods, a single vibrato waveform is derived and applied equally to all of the partials of the synthesized tone. The research reported here tests the validity of this linear treatment of vibrato using an analysis/synthesis procedure. The singing tone /ah/ is analyzed for bass, tenor, alto, and soprano voices, providing time-variant measurements of amplitude, frequency, and phase for each of the partials. The tones are reconstructed using an additive synthesis procedure. The importance of vibrato waveform similarities and differences between partials is evaluated by resynthesis of the singing tones from modified analysis data. The modifications include the use of the fundamental partial's vibrato waveform for all of the partials, the use of sinusoidal vibrato for all the partials, the elimination of amplitude variations for each partial, and various combinations. The analysis data and examples of the resynthesis will be presented. [Work supported, in part, by an NSF Graduate Fellowship.]
Attenuating jet aircraft noise in single-family residences near major U.S. airports
Authors: Myer Barney
Published: October 2005

Most noise attenuation of single-family residences near major U.S. airports has been achieved by acquisition and removal of homes. This noise remedy is not only extremely costly but it also disrupts social networks, undermines education and social institutions, lowers tax bases, and can generate subsequent land use problems. Acquisition as a noise remedy is limited to those areas where jet noise is the worst. The thousands of homes that cannot be purchased but endure noise peaks of near 100 dB must be addressed by other remedies. SeaTac International Airport and other airports are initiating acoustical insulation programs to lower interior noise levels to acceptable levels. An issue involved in these developing noise mitigation programs is how to set and achieve noise level goals that realistically make interior noise levels livable and are acceptable to the Federal Aviation Administration that funds up to 80% of program costs. The issue involves answering questions such as: Are interior peak noise levels (SEL) to be reduced low enough to allow uninterrupted communication (i.e., TV, telephone, conversation)? Is an average noise criterion (Ldn) sufficient? Is it possible to standardize interior noise goals when airports have widely divergent levels of activity? Noise remedy programs at SeaTac and other airports must address this basic issue to assure continued funding of their multimillion dollar efforts.
Measurement and analysis techniques for determining acoustical impacts from enroute aircraft in very quiet settings
Authors: Harrison Robin T. , Dunholter Paul H.
Published: October 2005

The National Parks of the United States have been set aside for the public enjoyment of pristine settings with minimal impacts from common urban disturbances. Ambient sound levels in these settings are generally very low, and have been measured below 20 dBA. Aircraft noise from enroute high altitude aircraft and sight-seeing aircraft have become a source of disturbance to park visitors. Under such quiet background conditions, enroute aircraft are often audible for extended durations. In the National Park setting, it has been found that even the detection of intrusive sound can result in annoyance. Numerous and continuing complaints have been generated by overflights that would not be expected to have a significant impact based upon traditional aircraft noise assessment criteria alone. This paper summarizes a National Park Service funded study that addresses various techniques for the measurement of enroute aircraft noise in these quiet settings. These techniques include A-weighted and detectability metrics for both the ambient and aircraft noise. Methods including the effects of the duration of the overflight and the number of events are also reviewed.
Frequency selectivity in focused attention as demonstrated by a multiprobe method
Authors: Dai Huanping , Scharf Bertram
Published: October 2005

A multiprobe paradigm, patterned after the probe-signal method of Greenberg and Larkin [J. Acoust. Soc. Am. 44, 1513–1523 (1968)], led observers to focus attention on a primary frequency in a detection task (2IFC). In each block of 64 trials, the signal was presented 48 times at the primary frequency and twice at each of eight probe frequencies, four higher and four lower than the primary. By presenting many probes in a block instead of only one as in most previous work, across-block variability was avoided. It also made it unlikely that observers focused on frequencies other than the primary. Results from four observers tested in separate sessions on primaries at 250, 500, 1000, and 2000 Hz revealed a consistent drop from 90% detection of the primary to chance detection of probes more than half a critical band away from the primary. At a primary of 4000 Hz, three observers showed the same kind of selectivity, but two others showed less. Generally, informed and naive observers performed much the same way. Since practice also mattered little, a sharp filterlike function could be measured in under 1000 trials. [Work supported by NIH.]
Modulation masking patterns
Authors: Bacon Sid P. , Grantham D. Wesley , Van Campen Luann E.
Published: October 2005

Modulation thresholds for sinusoidally amplitude-modulated (SAM) broadband noise were obtained for modulation frequencies from 2 to 512 Hz using a two-interval, forced-choice adaptive procedure. The noise carrier was on continuously throughout a block of trials, and was modulated for 500 ms in one of the two observation intervals. Thresholds were obtained in quiet and in the presence of a SAM broadband noise masker. In the masking conditions, the same noise carrier, presented at a spectrum level of 15 dB SPL, was used for the signal and the masker. The masker was modulated in both of the 500-ms observation intervals. The modulation frequency of the masker was 4, 16, or 64 Hz; its modulation depth (m) was 0 (no modulation), 0.5, or 1.0. For a given masker modulation frequency, the modulation masking patterns generally were bandpass, with the greatest amount of masking occurring when the signal and masker modulation frequencies were the same. With a few consistent exceptions, there was a monotonic relation between masker modulation depth and amount of masking (the greater the modulation depth the greater the amount of masking). These data indicate a tuning of the auditory system for the detection of modulation. [Work supported by NIH.]
Imaging through random media
Authors: Beran Mark J.
Published: October 2005

The distorting effects of random media on imaging are discussed. Random phase and random intensity fluctuations will be treated and the importance of isoplanicity will be emphasized. Methods of processing to overcome the distortion in conventional imaging will be considered, and the advantages of speckle interferometry and intensity interferometry will be mentioned.
Chaotic behavior of ray trajectories in a range-dependent ocean environment
Authors: Palmer D. R. , Brown M. G. , Tappert F. D. , Bezdek H. F.
Published: October 2005

It has been demonstrated that ray trajectories propagating in a range-dependent ocean environment can exhibit chaotic behavior. A particularly simple sound-speed model was considered consisting of the Munk reference sound-speed profile to which is added a small range-dependent, deterministic perturbation having an harmonic dependence on range and decreasing exponentially with depth. Chaotic ray trajectories were identified from an examination of Poincaré sections and power spectra. The sensitivity of chaotic trajectories to initial conditions and the consequent implications for predictability were investigated by considering the evolution with range of a bundle of rays that initially occupy a very small region in phase space. The largest Lyapunov exponent was determined by considering the spreading of the bundle. Since the ray equations define a nonautonomous Hamiltonian system with one degree of freedom, our results can be understood in terms of recent contributions to the study of classical chaos.
A microcomputer-based system for high-speed processing of speech materials using pitch-synchronous LPC analysis and synthesis
Authors: Eady Stephen J. , Dickson B. Craig , Snell Roy C. , Hunt Melvyn R.
Published: October 2005

This paper describes a microcomputer-based system that uses a high-speed digital signal processor (TMS-32030) for digitizing and encoding speech materials. The system uses pitch-synchronous LPC (covariance method), in conjunction with a laryngograph, to analyze previously recorded speech materials [M. J. Hunt and C. E. Harvenberg, Proceedings of the 12th International Congress on Acoustics, A4-2 (1986)]. The speech and laryngograph signals are digitized simultaneously, and speech parameters (F0, energy, formants, and bandwidths) are calculated in real time. The parameters are modified using an interactive editor, and real-time synthesis allows for auditory monitoring of the original and modified speech samples. The system is currently being used to process speech materials for a limited-vocabulary, word-concatenation synthesis system [Eady et al., Proc. ICASSP, 1473–1476 (1987)], and also in the development of a demisyllable-based text-to-speech device. [Work supported by National Research Council Canada and Science Council of British Columbia.]
A new design of a digital conference system
Authors: Gierlich Hans W.
Published: October 2005

Two main drawbacks make telephoning using a conference system troublesome: First, noise and reverberation produced at the site of the speaker are transmitted to the listener. Second, to achieve a sufficient speech volume at the listener'site, the necessary amplification for loudspeaker playback leads to instability (feedback) of the whole system. To suppress reverberation and noise, a speech controlled microphone arrangement is used. The implemented algorithm for speech detection is able to distinguish between noise and speech up to noise levels of 70 dB SPL. The noise signal may either be noiselike, impulselike, or sinusoidal. The detection algorithm needs a maximum time of 32 ms to detect a speech signal. To avoid instability, a combination of a special filter arrangement and a speech controlled amplification of microphone and loudspeaker signal is used. Two filters, having a transfer characteristic similar to a comb filter, are inserted into the transmitting and the receiving path. The transfer functions of the filters are inverse. With regard to the tone color, the filters were optimized, taking into account the special effects of signal processing used by the human ear. Hearing tests showed that speech intelligibility is not influenced by the filter arrangement.
Use of piezoelectric foil for measurement of pressure fluctuations and local shear stress in flight
Authors: Bertelrud A.
Published: October 2005

The measurement of pressure fluctuations in air flows poses several serious problems to the experimentalist. Generally a very limited frequency response is available, as the cavity above ordinary pressure transducers plays a crucial role for the sensor response. This type of transducer requires extensive work for incorporation into a surface. Recently, piezoelectric foil has become available for various applications. It offers ease of application and distributed measurements without strict frequency response limits. However, although the data acquisition is straightforward, it appears that particular caution is required for interpretation of the data as pressure fluctuations. In the present paper, emphasis is put on frequency response, true root-mean-square determination, as well as interpretation into skin friction data. Through analysis of flight data and laboratory calibrations, it is shown that the piezoelectric foil in general will respond to pressure fluctuations, and it is also shown that the piezofoil constitutes a repeatable and rugged source of information.
Bubble path in the wake of a cavitating propeller
Authors: Picrucci Mauro
Published: October 2005

Propellers that operate underwater at high rpm's cavitate at the tip. The tip cavitation creates air bubbles that are then swept downstream by the motion of the surrounding fluid. In this paper, a theory is presented to predict the local velocity and the path of the bubble. The bubble motion is assumed to be governed by a group of terms due to the acceleration of the displaced fluid, the convective term, and the drag due to the cross-sectional area of the bubble. At very low and very high Reynolds numbers, the equations have been solved in closed form. Results are presented for the bubble velocity and path for the following flow fields: (a) uniform axial flow field and (b) uniform flow field with an axially decaying swirling component. In all cases presented the bubble axial velocity component asymptotes to the free stream velocity; the manner in which it asymptotes is exponential at very low Reynolds numbers and algebraic at high Reynolds numbers. Bubble helical paths and velocity patterns are shown for different bubble sizes.
A calibrator for the measurement of the static and dynamic shear stresses along a stretched Mylar membrane
Authors: Cuomo Frank W.
Published: October 2005

Recent power spectral density measurements of the pressure fluctuations in a turbulent boundary layer obtained with a fiber-optic lever pressure sensor in a low-speed wind tunnel have compared favorably with data reported in the literature. Utilizing a similar sensor arrangement, incorporating a flush-mounted, metallized Mylar membrane, the measurement of shear stresses leading to skin friction has been contemplated. The first step toward the realization of a shear sensor has been to establish the behavior of a stretched Mylar membrane acted upon by forces parallel to the flow direction. A calibrator designed to meet the requirements normally found in a wind-tunnel environment has been constructed and tested. In this paper, its operation is described and data of Mylar samples are presented. The resolution of the device for static measurements is based on translations on the order of 1 μin. while the dynamic response covers a frequency range of dc to 18 kHz.
Lord Rayleigh and his influence on science and scientists
Authors: Crocker Malcolm J.
Published: October 2005

John William Strutt, the third Lord Rayleigh (1842–1919), was a man of considerable scientific accomplishments. He contributed to theoretical and experimental physics and was one of the great pioneers in scientific and engineering methods. He was also a man who moved in the highest social and intellectual circles of his day. He was a relative of prime ministers and received almost every honor to which a British scientist could aspire. He was a Nobel prize winner and President of the Royal Society. His book, The Theory of Sound, has remained in print for over a century. From 1880 to 1885, after J. C. Maxwell's death, Lord Rayleigh directed the famed Carendish Laboratory at Cambridge University, but after that mainly lived in Terling Place, his country house near London where he established a remarkable scientific laboratory in which he later discovered argon. During his lifetime Lord Rayleigh published over 400 scientific papers and carried on a vigorous correspondence with scientists such as Maxwell, Kelvin, Stokes and others. This paper will examine his papers and correspondence to put his scientific contributions, particularly in acoustics, into historical perspective.
Lord Rayleigh's contributions to musical acoustics
Authors: Rossing Thomas D.
Published: October 2005

Rayleigh's work touched virtually every area of acoustics in some way; musical acoustics is certainly no exception. Besides laying the groundwork for all subsequent research on musical instruments with his informative discussions of the vibrations of strings, bars, membranes, plates, air columns, and resonators, he did important theoretical and experimental work on bells and kettledrums. He identified the vibrational modes that radiated the most prominent partials in the sounds of both these instruments, and he recognized that the nominal pitch of a church bell lies one octave below the fifth partial. In the area of music perception, his most important contributions were his studies of absolute pitch and sound source localization.
Neural nets for speech recognition
Authors: Huang William Y. , Lippmann Richard P. , Nguyen Thao
Published: October 2005

Artificial neural networks are of interest for two main reasons. First, they provide architectures to implement many algorithms used in speech rccognizcrs with fine grain massive parallelism. Second, they are leading to new computational algorithms and new approaches to speech recognition inspired by biological nervous systems. Neural net approaches to the problems of speech preprocessing, pattern classification, and time alignment are reviewed. Preprocessors using auditory nerve time-synchrony models have provided improved recognition performance in noise [O. Ghitza, ICASSP 87, 2372–2375]. Highly parallel neural net architectures exist to implement many important traditional classification algorithms, such as k-nearest neighbor and Gaussian classifiers [R. Lippmann, IEEE ASSP Mag. 4(2), 4–22 (1987)]. Newer multilayer perceptron classifiers trained with back propagation can form arbitrary decision regions, are robust, and train rapidly for convex decision regions. These nets performed as well as conventional classification algorithms for digit and vowel classification tasks [W. Huang et al., 1st ICNN IV, 485 (1987)]. Neural net approaches to the problem of time alignment are reviewed, including a neural net model called a Viterbi net that implements the Viterbi decoder used successfully in continuous distribution hidden Markov models (HMMs) [R. Lippmann et al., 1st ICNN IV, 417 (1987)]. [This work was sponsored by the Defense Advanced Research Projects Agency and the Department of the Air Force.]
New findings on scattering from floating plate with lateral heterogeneities
Authors: Chamuel Jacques R. , Brooke Gary H.
Published: October 2005

Experimental ultrasonic modeling results are presented on near grazing incidence scattering from cracks, ridges, and edges of floating plates bounding a shallow water waveguide. It was observed that underwater acoustic waves propagating in a shallow water waveguide bounded by a floating plate produce a ringing wave component in the water upon scattering from a crack or a plate edge. This ringing component is associated with the water wave, travels with the water wave velocity, and its time of arrival is independent of the length of plate region along the wave path. The ringing frequency, however, is determined by the shear wave velocity and the plate thickness. The plate thickness corresponded to half a wavelength based on the shear wave velocity. The contribution of the ringing wave component to scattering from wet cracks, dry cracks, and ridges are demonstrated. The impact of these new findings on interpreting laboratory scaled studies on Arctic acoustics are discussed in view of the wide use of Plexiglas to model sea ice. Examples are given on scattering from single and multiple cracks and ridges. [Work supported by DREP and ONR.]
Acoustic fields scattered simulated three-dimensional wind-driven sea surfaces
Authors: Caruthers Jerald W. , Keiffer Richard S. , Novarini Jorge C.
Published: October 2005

Statistical characteristics of acoustic fields scattered in the forward direction from simulated three-dimensional wind-driven sea surfaces are studied for frequencies from 0.1–10 kHz, high grazing angles, and ranges in the Fresnel region. The simulated surfaces have the Pierson-Moskowitz directional spectrum for a 5-m/s wind speed. Complex pressures are calculated using Helmholtz-Kirchhoff theory with a minimum number of approximations. Statistics of the acoustic fields are calculated as ensemble averages of the complex pressures and their moments. Comparisons are presented for differences in the ensemble of surfaces, spectral forms, surface illumination patterns, and implementation of the scattering theory. The technique allows for the accurate representation of first- and higher-order moments of the acoustic field scattered from a three-dimensional surface. Such quantities are usually difficult to obtain by other techniques. [This work has been supported by NAVOCEANO and NORDA.]
The effect of finite wind fetch on the acoustic field scattered from wind-driven sea surfaces
Authors: Novarini Jorge C. , Caruthers Jerald W. , Keiffer Richard S.
Published: October 2005

The effect of a finite fetch on the acoustic field scattered by the ocean wind-driven surface for various wind speeds is analyzed through numerical simulation. The JONSWAP spectrum is adopted to describe the surface statistics of growing seas and the Pierson-Moskowitz spectrum for fully developed seas. Ensembles of three-dimensional surfaces having the desired directional spectra are numerically generated. For each surface realization, the complex pressure of the forward scattered field is calculated within the framework of the Helmholtz-Kirchhoff theory for kr > 1. Statistical moments of the scattered field are then calculated as ensemble averages. The fields scattered in the windward and cross-wind directions are evaluted for frequencies up to 1.5 kHz and for angles away from grazing incidence. Results indicate that the scattered field is sensitive to the difference between the JONSWAP and the Pierson-Moskowitz spectra. In particular, the total intensity and the coherence of the scattered field are significantly affected by the wind fetch. [Work supported by NAVOCEANO and NORDA.]
Resonator systems for in situ bubble measurements near, and at, the sea surface
Authors: Breitz Nigel D. , Medwin Herman
Published: October 2005

A parallel plate acoustical resonator has been developed to measure bubble densities and their fluctuations near the sea surface and under whitecaps. Calibration has been achieved with microbubbles whose size (radius approximately 50 μ) was evaluated by speed of rise in an aquarium. Using the “sea” generated in a 20-m-long water-wind tunnel, the resonator performance has been shown to be independent of orientation, depth, and sea surface roughness. A second device, using an upward-looking resonator, is also being developed to investigate larger bubbles at the surface. In situ acoustical devices are compared with remote sensing techniques, and early results of ocean trials are revealed. [Work supported by ONR.].
Volume scattering measurements using parametric sonar
Authors: Syck James M. , Brown Charles L.
Published: October 2005

A parametric sonar has been used to measure volume scattering strength at a number of ocean locations. The narrow beamwidth and absence of sidelobes of the parametric sonar make it an ideal tool for measuring volume scattering strength as a function of depth. The resolution of this device is such that in some circumstances individual scatterers can be identified. An example of such a situation is presented. The parametric sonar system is described. Additional sets of examples from nearshore and off-shore areas are presented for both day and night conditions.
Scattering of sound by sound from two real beams
Authors: Berntsen Jarle , Tjøtta Jacqueline Naze , Tjøtta Sigve
Published: October 2005

A theory for the nonlinear interaction between two sound beams produced by real sources in a lossless fluid was presented in a previous work [Naze Tjøtta and Tjøtta, J. Acoust. Soc. Am. 83, 487–495 (1988)]. A general solution of the governing equation in the quasilinear approximation, valid at any range, crossing angle, and frequency ratio, was obtained for prescribed boundary conditions. An asymptotic formula for the sum and difference frequency sound pressure was obtained at large distance from the sources. It relates the amplitude and directivity of the sound field in the farfield to the on-source conditions. In the present work, numerical results are presented for various types of sources (uniform piston, Gaussian source, focusing source). The influence of source geometry (separation, and intersection angles from 0° and 90°) and frequency on the beam pattern of the nonlinearly generated sound is studied. Conditions are also given to determine when scattering of sound by sound can be observed. In the special case of thin Gaussian beams intersecting at a small angle, the results are compared with that presented by Darvennes and Hamilton [J. Acoust. Soc. Am. Suppl. 1 83, S4 (1988)] using the paraxial approximation. [Work supported by the IR&D program of ARL:UT, and VISTA/STRATOIL, Norway.]
Effect of transient signal length on interaural cross-correlation functions
Authors: Yanagawa Hirofumi
Published: October 2005

The internal cross correlation is closely related to the subjective impressions of sound fields. This paper attempts to account, in a comprehensive way, for the combined effects that initial reflected sound and reverberation sound from music or other transient signals have on such impressions. To this end, cross-correlation functions of the rise and fall of the sound field from transient signals were derived from the windowing impulse responses at two points in a hall. These results were combined with the image-sources distribution patterns derived by the closely located four-point microphone method. Good agreement was found between changes with time in experimentally derived maximal cross-correlation function values and the image-sources distribution of the sound field.
Perception of timing in rhythmic patterns
Authors: Monahan Caroline B. , Hirsh Ira J.
Published: October 2005

In two studies employing an adaptive, cued-2AFC procedure, listeners discriminated between six-tone rhythmic patterns that differed only in the delay of the temporal position of one of the tones. On each trial, feedback was given and the subject's performance determined the amount of delay on the next trial. The six tones of the patterns marked off five intervals. In the first study, eight patterns comprised three “short” and two “long” intervals: thus 12121, 21121, etc., where the “long” (2) was twice the length of a “short” (1). In the second study, eight patterns comprised two “shorts” and three “longs”: thus 21212, 12212, etc. Each pattern was tested 45 times (five positions of the delayed tone × three tempos × three replications) for each of three subjects. Consistent with previous work on simple interval discrimination, absolute discrimination (Δt in ms) was poorer, the slower the tempo. Relative discrimination (Δt/t “short,” where “short” was 50, 100, or 200 ms) was better, the slower the tempo. Beyond these global results, large interactions of pattern with position of the delayed tone and tempo suggest that different models of performance are needed to explain behavior at the different tempos. [Work supported by AFOSR.]
Isolated word recognition based on the adaptive neural network
Authors: Wang J. S. , Widrow B.
Published: October 2005

For the last several years, despite many new approaches (such as application of the Markov model and VQ) in speech recognition, the classical pattern matching using dynamic programming still prevails because it yields good recognition results. However, this DP-based algorithm requires a large amount of computation. It is difficult to meet the real-time requirement for large vocabularies. An alternative pattern recognition method called the adaptive neuron in adaptive control system used by B. Widrow in the early 1960s is used here to build an isolated word recognition system. A simple and efficient learning algorithm is presented for adaptively adjusting weight vectors to fit a certain word pattern. In order to capture the variations in speech, an increasing codebook was designed during the learning phase to allow increasingly more patterns; therefore, multiple reference patterns were adopted, giving the advantages of a statistical Markov model. The performance of this recognition system has been preliminarily tested on two common vocabularies including ten digits and the English alphabet. The recognition accuracy is 100% for digits after five trainings per word and 93% for the alphabet after ten trainings per word for speaker trained. Further experiments for a complete Chinese vocabulary are under progress.
Human auditory brain-stem asymmetry in the frequency following response (FFR)
Authors: Bopanna Ballachanda B. , Moushegian George
Published: October 2005

The existence of cortical asymmetries in humans is well documented and paralleling this is the suggestion that speech sounds are processed by different central neural pathways than nonspeech sounds. This study presents data that reveal lower brain-stem asymmetry to a low-frequency sinusoidal. Frequency following responses (FFR) (500 Hz) were obtained from ten normal hearing subjects monaurally and binaurally over a range of intensities. The results provide clear evidence that the monaural FFRs have differences in onset latency and differences in latencies throughout the response waveform as well as diversities in their amplitudes. Since the FFR emanates from sites that are not rostral to the inferior colliculus, the findings support the view that the lower auditory brain stem is functionally asymmetrical.
Comparison of asymptotic and exact results for scattering from rigid spheroids
Authors: Werby Michael F. , Collins Michael D.
Published: October 2005

Results generated with the Kirchhoff approximation, which is effective for backscattering, and the diffraction integral method, which is effective for forward scattering, will be compared with the exact solution obtained with Waterman's extended boundary condition method. Calculations will be presented for kL/2 ranging from 5 to 300 and for aspect ratios ranging from 1 to 15. An additional asymptotic approximation, the method of stationary phase, is used for higher frequencies. For higher aspect ratio targets, one would expect the accuracy of high-frequency methods to be difficult to predict and to depend not only on kL/2 but also on bistatic configurations due to the large variation of Gaussian curvature on these targets. Hence, the focus of the presentation will be to illustrate how accuracy depends upon kL/2, angle of incidence, and angle of observation for high aspect ratio targets. [Work supported by ONR and NORDA.]
Two-dimensional nonlinear wave propagation in an inhomogeneous elastic solid
Authors: Kim Hyun S. , Ginsberg J. H.
Published: October 2005

This paper considers the two-dimensional problem of propagation of stress waves in an elastic half-space subjected to arbitrary excitation along a strip on the free surface. Inhomogeneity is taken to be a small-scale effect, in the form of small spatial fluctuations of the second-order elastic coefficients and density, where such variations are periodic functions of the vertical coordinate normal to the surface. General nonlinear equations of motion are derived in the Lagrangian form. Integral transform methods are used to solve the first- and second-order equations of motion, which are obtained from straightforward perturbation expansion of the displacement components. When the normal stress at the surface has a very slowly varying Gaussian distribution transversely to the strip, the asymptotic solutions are obtained in power series in terms of the small parameter for the Gaussian distribution. The longitudinal wave is shown to be the most dominant, although a shear wave does arise at the second order of approximation. It is shown that secular solutens, which are usually associated with cumulatively growing nonlinear distortion, can also be produced by the inhomogeneity, if the periodicity of the exciting stress and inhomogeneity satisfy a certain relation. [Work supported by NSF.]
Electret condenser microphones in physical acoustics research
Authors: Garrett S. L.
Published: October 2005

The simplest condenser microphone in terms of construction and use is a charged, aluminized Teflon membrane placed directly over a conducting back plate. In this configuration, the low-frequency open circuit sensitivity is determined approximately by the product of the “effective” electret bias voltage and the compressibility of the fluid trapped between the membrane and the backplate, typically 1 mV/Pa in air. This talk will describe the procedures for calibration of such a transducer in a resonator and its application to the measurement of constituents in binary or pseudobinary gas mixtures. Other aplications to low-temperature physics will be described that include acoustic measurement of temperature fluctuations with a perforated electret membrane having a detection threshold of ≈ 10 nanokelvin/ √ Hz in superfluid 4He at 1 deg above absolute zero, and the measurement of the temperature dependence of the order parameter of superfluid 3He-A and 3He-B at temperatures of 11000 of a degree above absolute zero. [Work supported by the Office of Naval Research.]
Microphone comparison with the three-port two-microphone cavity
Authors: Wong George S. K. , Embleton Tony F. W.
Published: October 2005

The performance of the three-port two-microphone cavity [Wong and Embleton, J. Acoust. Soc. Am 71, 1276–1277 (1982)] for the comparison of condenser microphones is described. The experiment involves, 1-, 12-, and 14-in. microphones from three manufacturers. With a reference microphone, the above cavity is able to measure, by comparison, the sensitivities and frequency response of various sizes of microphones under different physical conditions, such as with or without protective grids. For frequency response measurements over the frequency range from 20 Hz to several kilohertz, the experimental results are similar to those obtained with the electro-actuator method, and the cavity measurement does not require any access to the air space that is in close proximity to the microphone diaphragm. In general, the frequency response performance of the cavity improves with the smaller size microphones.
A multimicrophone system with spatial selectivity
Authors: Date Hikaru , Furuya Ken'ichi
Published: October 2005

The principle of spatial selectivity of a receiving system, which comprises several microphones distributed on a spherical boundary surface and one microphone at the center of the space enclosed by the boundary, is introduced in accordance with the theorem expressed by Kirchhoff's integral equation. The output from each of the boundary microphones are processed in the following way. Each signal is first delayed and then differentiated in both time and space in the direction normal to the spherical boundary. All of the delayed and differentiated signals are added together with appropriate weighting. The system output is finally obtained by subtracting the weighted sum from the output of the center microphone. The system output contains no components of external noise from the outside of the spherical boundary. Simulation studies were carried out for various numbers of microphones and for different ratios of signal wavelength to the radius of the spherical boundary and for both plane and spherical incident waves.
Turning the teaching of foreign pronunciations into a computer product
Authors: Thomas Raymond J.
Published: October 2005

Extending the scientific teaching of English and French pronunciations [J. Acoust. Soc. Am. Suppl. 1 76, S90 (1984)] to adults speaking other languages showed that specific variations in speech acoustics and use of muscles are related basically in the same manner in all humans, but that an added complexity arises from seemingly unrelated speech habits depending on each mother tongue. Final results of advice are thus unpredictable, but practice has shown that progress on various points could be totaled through gradual classifications of oppositions. This could be achieved with a computer program using the speech syllabic data bank (Proc. XIth ICPhS, Tallinn, 1987, Sec. 89.4) together with a much smaller bank on rhythms and stresses in the various languages. The study of fewer than 20 phonetic systems would cover all important speech habits. This would result in the capacity for the computers to “think out,” in a matter of minutes, the one among the 16 000 000 odd teaching programs it could devise (Evaluation de l'information en langues naturelles, Ms de Thèse, BU de Rouen, 1983, Vol. 3) that would be best adapted to anyone of us for learning any of the 4000 odd foreign pronunciations.
Arthur Benade—Synthesizer
Authors: Jameson George
Published: October 2005

Of the talents that led to Art Benade's considerable accomplishments in musical acoustics, one has escaped prominent mention—namely—his proclivity to synthesize. He had a strong ability to select ostensibly unrelated facts and thoughts from many disciplines, theoretical and practical, over a period of time. When these stored items began to fit together, Art was often able to discern a direction and reach the solution to an acoustical problem. The breadth of overall knowledge required to permit access to obscure relationships has often been itself obscured by the impact of Art's results. He called freely on branches of physics, chemistry, mathematics, and electronics not central to his work. Besides the sciences, he was well grounded in music, literature, and many phases of engineering. On a different level, he was a proficient machinist, mechanic, and tinkerer. Not least, he was a usable musical performer. He possessed a truly remarkable musical ear that was a major source of information. While many of the above abilities are requisite for any successful experimental physicist, Art extended his attentions further—to the user of musical instruments. Examples of the synthetic process will be cited—partly from the standpoint of the author, whose background is in manufacture and repair of instruments, and in routine professional performance and teaching of woodwinds.
Aging effects on gap detection thresholds among normal and hearing impaired subjects
Authors: Gelfand Stanley A. , Porrazzo Jody , Silman Shlomo
Published: October 2005

Age differences between normal and hearing impaired groups have clouded the effects of hearing loss on temporal selectivity in many studies. This study attempted to address this problem by comparing gap detection thresholds (GDTs) between a normal hearing group and a group with primarily high-frequency sensorineural hearing losses, each containing subjects with similar age ranges. Transformed up-down estimates of 70.7% GDTs were used in a two-alternative forced-choice paradigm with feedback. Each run involved simultaneous estimates of two gap thresholds, using interleaved up-down procedures with random switching between them. Gap detection thresholds varied from approximately 3 to 16 ms among subjects in both groups. Across all subjects, GDTs were significantly correlated with age and with hearing sensitivity at 12 000 Hz but not with auditory thresholds at any other frequencies. Regression analyses revealed that age was the only significant predictor of GDTs among these subjects. The findings suggest that temporal processing is affected by age-related auditory changes that are not necessarily reflected as shifts of absolute sensitivity.
Auditory temporal summation in normal-hearing and hearing-impaired cat
Authors: Solecki Janet M. , Gerken George M.
Published: October 2005

Auditory temporal summation functions were measured for five cats before and after they were exposed to a 2-kHz tone at 110 dBA for 48 h. Ten digitally generated stimuli with overall durations ranging from 8.32 to 275 ms were used. Stimulus frequency was 6.25 kHz. Six stimuli consisted of single tone bursts and the remaining four contained multiple tone bursts. Twelve acceptable thresholds for each stimulus were obtained from each animal before and after exposure. Threshold measurements began again no sooner than 27 days postexposure. Pre- and postexposure audiograms were also obtained and mean permanent threshold shift at 6.25 kHz was 32.6 dB. Pre- and postexposure temporal summation functions were characterized by the slope of the regression lines for threshold versus log duration. The mean slope in dB per decade of duration was − 6.6 preexposure and − 3.8 postexposure. The slope for normal-hearing cat is very similar to the slope of − 6.1 that was obtained in a companion study that used normal-hearing human subjects.
Acoustical power radiation from cylindrical shells, with or without stiffeners: Understanding of phenomena and practical aspects
Authors: Laulagnet B. , Guyader J. L. , Lesueur C.
Published: October 2005

An analytical study of radiation by finite, stiffened cylindrical shells is presented. The model of stiffener utilized allows one to treat hollow cross-section cases encountered in industry. The point driving force is on the stiffener or directly on the shell and can be radial, tangential, or longitudinal. Fluid and structure equations are solved with the modal analysis, using the in vacuo nonstiffened basis. This leads to the calculation of radiation impedances and mass and stiffness generalized terms of stiffeners. Theoretical results are presented both in air and in water, in radiated power, radial quadratic velocity, and radiation factor. Mechanical modal cross-coupling influence is shown as well as the stiffening influence on the in water vibroacoustic shell behavior. Parameters like rigidity, number and distribution of stiffeners, shell and stiffener structural damping, location, and force type are studied in air. Finally means for noise reducing in air are deduced. [Work supported by the Ministère de la Défense, Contrat D.R.E.T. No. 85065.]
Leaky Lamb wave inspection of thin plates
Authors: Nagy Peter B. , Adler Laszlo
Published: October 2005

A single-transducer immersion technique is introduced to study dispersion of leaky Lamb waves in different plates, such as fiber-reinforced composite laminates and adhesive lap joints. The plate under study is interrogated by a broadband pulse from both sides when placed between the transmitting-receiving transducer and a perpendicularly aligned mirror. The transmitted peaks in the spectrum of the received echo from this mirror correspond to double mode-converted leaky Lamb wave resonances. These peaks are measured at different angles of incidence, i.e., at given phase velocities to get velocity dispersion curves. Experimental results are presented to demonstrate the accuracy and versatility of this very simple measuring technique
Propagation loss modeling of the seabed using both shear and permeability effects
Authors: Badiey Mohsen , Yamamoto Tokuo
Published: October 2005

A complete normal mode propagation model is used to predict transmission loss coefficients for two previously reported data sets, one from the Nova Scotian Shelf and one from the Mediterranean Sea. This model employs both permeability and shear modulus of the seabed through formulations of Biot-Willis theory of acoustic waves in poro-elastic, anisotropic media. The sediments were modeled using the reported geological and core data. It was found that both shear and hydraulic (permeability) properties of the sediments contribute to the propagation loss. At low frequencies waves penetrate deep into the sediments and are effected by the higher rigidity of the lower sediment layers. Higher frequencies penetrate only into surficial sediments where the permeability is the dominant influence in the attenuation mechanism. [Work supported by ONR.]
Structural power flow analysis of coupled structures
Authors: Cuschieri J. M.
Published: October 2005

Structural power flow techniques are an effective tool for the dynamic analysis of complex coupled structures, especially those types of coupled structures that are somewhat repetitive. Using the structural power flow method, the response of a structure is analyzed by tracking the flow of the vibrational power through the coupled structures, from the location of the excitation. An advantage of the power flow technique is that, if one component of the structure is modified, the analysis need only be repeated for the modified structural component, and the reevaluation of the coupling expressions. The use of the structural power flow method is demonstrated for beamlike and platelike structures including an analysis of the influence of structural and excitation parameters on the response of the coupled structures. Power flow methods can also be used to analyze more than one type of wave motion, which is useful in cases where incident wave power is scattered into other wave types near structural discontinuities. Additionally, experimental results for structural power flow are discussed. [Work supported by NASA.]
Dominant parameters of weakly coupled systems
Authors: Igusa Takeru
Published: October 2005

A modal analysis framework was developed to identify the dominant characteristics of weakly coupled systems using perturbation analysis and mode synthesis techniques. Modal parameters were obtained to quantify four types of characteristics—tuning, coupling, nonclassical damping, and spatial interaction. These parameters were subsequently used to determine the response of the system to broadband excitation. It was found that the parameters could be used to isolate certain modes of the system with dominant effects on the response. The response characteristics were also examined in the high-frequency range by extending asymptotic modal analysis [E. H. Dowell and Y. Kubota, J. Appl. Mech. 52, 949–957 (1985)] to include the asymptotic properties of the coupled system parameters. The results were used to illustrate applications in design of isolation mounts in the low-frequency range, and in providing insight into statistical energy analysis in the high-frequency range. The work is an extension of the dynamic analysis of lumped mass secondary systems subjected to low-frequency excitation [T. Igusa and A. Der Kiureghian, J. Eng. Mech. 101, 20–41 (1985)].