bluuebunny/crossref_metadata_2025 huggingface dataset:
GTD synthesis of resonance amplitudes in the backscattering from an elastic spherical shell
Authors: Kargl Steven G. , Marston Philip L.
Published: October 2005

An elastic generalization of the geometrical theory of diffraction [P. L. Marston, J. Acoust. Soc. Am. 83, 25–37 (1988)] was used to describe the backscattering of short tone bursts from an elastic spherical shell in water [S. G. Kargl and P. L. Marston, J. Acoust. Soc. Am. 85, 1014–1028 (1989)]. In the present research, steady-state backscattering amplitudes are synthesized. The GTD model contains explicit terms for the specular reflection and individual Lamb surface wave contributions in a Fabry-Perot form. Numerical calculations compare the exact partial-wave series result with the GTD synthesis for the frequency range 7 < kA < 100. These computations correspond to a stainless steel shell with an inner-to-outer radius ratio b/a = 0.838. An important feature of the GTD model is that only two parameters are needed to fully describe the individual Lamb wave resonance amplitudes. These parameters are the ratio of the Lamb wave phase velocity to the velocity of sound in water (c1/c) and the radiation damping β1. These should depend on local properties of the shell-water system. This simple parametrization of the direct scattering problem should facilitate novel approaches to the inverse problem. Identification of a scatterer may be achieved through an examination of the Lamb wave resonances via a numerical fitting of these parameters. [Work supported by ONR.]
Examination of finite amplitude ultrasonic pulses using light diffraction
Authors: Wolf J. W.
Published: October 2005

Finite amplitude pulses are examined acousto-optically with a light diffraction apparatus. Using an empirical model of transducer frequency response, ultrasonic pulse Fourier spectra are derived for input to a light diffraction model. To then simulate harmonic distortion, the pulse spectra are used for input to a computational model based on the Burgers' equation for propagation of finite amplitude acoustic waves in a nonlinear medium. Several examples are presented that illustrate good agreement between experimental light diffraction patterns and those predicted by the pulse spectrum, propagation, and light diffraction theories. [Work supported by ONR.]
Higher-order time-domain ultrasonic scattering
Authors: Cavicchi Thomas J.
Published: October 2005

To improve the reliability of diagnosis of tissue pathologies (e.g., cancers) based on ultrasonic imaging, quantitative images of physiological parameters, such as speed of sound and/or absorption coefficient, can, in theory, be obtained by inverse scattering procedures. Previous studies [T. J. Cavicchi, S. A. Johnson, and W. D. O'Brien, Jr., IEEE Trans. UFFC-35, 22–33 (1988)] employed a moment method matrix formulation under a monofrequency assumption. Periodicities of the field phasors caused a nonuniqueness problem in the inverse scattering solution for scatterers with phase shift magnitudes greater than π, in practical medical imaging, the phase shift would be hundreds of π. A reformulation of the method in the time domain appears more promising because the phase shifts are now merely time delays. The time-domain moment method formulation is presented for forward scattering. A preliminary numerical study has been undertaken using the moment method equations and the internal field calculated by the inverse discrete Fourier transform of the well-known monofrequency solution solved at many frequencies. Using the exact field within the integral produces a resulting good approximation of the exact field. Current studies are aimed at inversion for the forward-scattered field; future work will address inverse scattering.
Low-frequency noise: Wave turbulence interaction and bubble cloud oscillations
Authors: Carey W. M. , Browning D. G. , Fitzgerald J. W.
Published: October 2005

Low-frequency ocean ambient noise data, when not dominated by shipping noise, show evidence for wind-dependent noise at frequencies less than 500 Hz. Vertical directionality measurements have a horizontal component with a broad frequency characteristic. This effect is partly due to the coupling of wind-generated noise into the sound channel by either a shallowing sound channel or a down-slope conversion process due to basin boundaries and sea mounts. Omnidirectional measurements made below the sound channel critical depth, in sparsely shipped basins, and at high sea states indicate two distinct regions divided by the occurrence of breaking waves. Prior to wave breaking, a possible sound generation mechanism is the interaction of surface waves and turbulence in the near-surface layer [I. Z. V. Gonchavov, Atmos. Oceanic Phys. 6(11) (1970); Yen and Pertone NUSC TR5833 (1979)]. Wave breaking produces bubble clouds. The dynamic evolution of these bubble clouds is a mechanism for the production of sound of frequencies less than 500 Hz. These clouds of micron-size bubbles are regions of low sonic velocity described by Wood's volume fraction equations. These regions can be treated as a compressible body with a composite mixture speed and density that can exhibit a collective resonant oscillation and radiate as monopole and dipole sources. However, due to the proximity of the sea surface, only the monopole and its image, an effective dipole, would be of importance. When driven by the wave breaking vorticity and turbulence, these regions are shown to result in sufficient radiated sound to produce noise levels comparable to those observed and are also pronounced scatterers of low-frequency sound.
Monitoring oceanic precipitation using ambient sound—An assessment
Authors: Nystuen Jeffrey A.
Published: October 2005

Several years ago, the idea that rainfall could be measured by monitoring the unique underwater sound generated by the rain was introduced. Two mechanisms by which raindrops can produce sound underwater have been identified. Data will be presented from current laboratory experiments exploring the relative importance of these two mechanisms for different sized drops at terminal fall velocities. The influence of wind on these mechanisms is not understood and currently prevents quantification of rainfall rate from ambient sound; however, oceanic data from the Canadian Atlantic Storms Program will be presented that demonstrate that precipitation can be detected acoustically even in conditions of high winds (20 m/s) and large waves (4–6 m).
Underwater noise due to precipitation
Authors: Crum Lawrence A. , Pumphrey Hugh C. , Prosperetti Andrea , Bjorno Leif
Published: October 2005

In 1959, G. Franz published a thorough investigation of the underwater sound produced by liquid drop impacts [G. Franz, J. Acoust. Soc. Am. 31, 1080 (1959)]. He discovered that, under certain conditions, a gas bubble was entrained by the impacting droplet, and the subsequent oscillation of this bubble resulted in a large amount of radiated sound. Recently, Scrimger has measured the underwater sound produced by rainfall and has discovered that a well-defined spectral peak exists near 15 kHz [J. A. Scrimger, Nature 318, 647 (1985)]. The sound produced by the impact of water droplets on a water surface, both for individual and for multiple events such as those produced by artificial and natural rainfall, has been examined. The studies indicate that the major contribution to the underwater noise produced by both rain and snow is that associated with the oscillations of gas bubbles introduced into the water by the impact. Both experimental and theoretical evidence for these conclusions will be presented, including numerical studies of the drop impact process. [Work supported by the ONR.]
Measured wind speed dependence of the underwater acoustic ambient vertical directional spectra at high frequencies
Authors: Kennedy Robert M. , Szlyk Thomas K. , Wentworth Susan M.
Published: October 2005

A measurement of the acoustic ambient arriving from a horizontal direction along with total acoustic intensity spectra allows one to infer both the total directional spectra and some physical characteristics of the sources of “sea surface sound.” A long-term measurement of these two quantities was made at high frequency, i.e., 8 to 64 kHz, in the Tongue of the Ocean, The Bahamas. The horizontally directed ambient was measured using vertically oriented line arrays and was observed for wind speeds ranging from 1 to 30 kn. The resulting database was used to estimate the statistics of anisotropic “noise gain” relative to the isotropic noise gain. The resulting normalized array performance was found to be a continuous function of total acoustic intensity and a discontinuous function of wind speed. Differences in the functional dependence and residual statistics were found for two cases: whitecaps present or not present. The relation of these results to the total directional spectra and a model of the near-surface distribution of acoustic sources are discussed.
Attenuation of Lamb waves in a plate loaded by a liquid under pressure
Authors: Diodati Paolo , Severcan Mete
Published: October 2005

Attenuation of Lamb waves in a steel plate loaded on one side by a fluid was measured as a function of the fluid pressure. A monobloc, rectangular, steel cavity was constructed and a 1-cm-thick steel plate was used to shut pressureproof the cavity which contained castor oil, C11H10O6, in it. Acoustic pulses at 1.7 MHz were launched on the plate using a plastic wedge and a second wedge was used to detect the received pulses. The attenuation for different Lamb modes was observed to increase with pressure for all the modes except for the low-order modes, S0, a0, s1, a1, and a2, for which a decrease in attenuation was observed. Over the range of pressure used, considering the change in the wave velocities in the liquid and the steel plate, normally an increase in attenuation would be expected. The unexpected decrease in attenuation for low-order modes can be explained by the decrease in the power lost in the liquid near the plate boundary by the leaky waves, due to the shifting away from the phase-matching condition with the change in pressure.
Acoustic emission from the Gran Sasso Mountain (AEGS)
Authors: Diodati P. , Paparo G. , Scarpa R.
Published: October 2005

Rectangular parallelepiped samples extracted from a tunnel 10.2 km long and 1000 m above sea level and from the top of the highest mountain in the Appennines underwent laboratory studies for their acoustic emission (AE). The different responses are reported and explained. These are the preliminary results of the AEGS project which will involve collecting data from sensors located in different places of the mountain. In the center of the tunnel, where a laboratory of the National Institute of Nuclear Physics (INFN) exists, was placed a seismometer (to tape record seismic events with a frequency up to 100 Hz), a geophone 3d (for oscillations up to 30 kHz), and four transducers (broadband, 50 kHz–2 MHz), for AE measurements. In a perforation made on the top and 1.5 km deep, at 1000 m from the surface, a seismometer, a geophone, and two transducers for AE were placed. In addition, on the top a geophone and two transducers for AE were placed. This acquisition system of seismic data will make it possible, by means of a computer network, to send all the data to a unique integrated system to analyze the physical parameters.
Depth dependence of high-frequency ambient ocean noise
Authors: Caron Albert B.
Published: October 2005

Little or no published data exist on the depth dependence of ambient ocean noise at frequencies above 9 kHz. In this higher frequency region, absorption becomes a major effect in reducing surface-generated noise as receiver depth increases. To investigate this effect, two experiments were conducted. The first, conducted in the Tongue of the Ocean, off Andros Island in the Bahamas, utilized bottom-mounted hydrophones at a depth of 1500 m to observe ambient noise levels covering sea states 1–3 over a frequency span of 13 to 37 kHz. Near surface noise measurements were also made for comparison. The second experiment, conducted in Hawaii, covered a 9- to 25-kHz span with bottom-mounted hydrophones at depths of 751, 1173, 1769, 2481,3981, and 4505 m. Results in both cases show a reduction in ambient noise levels with increasing depth and frequency, as predicted from models developed by Urick and others. Closest agreement was observed with the assumption of a dipole surface intensity radiation pattern at the higher sea states.
The reflection phase grating diffusor: A five-year progress report
Authors: D'Antonio Peter
Published: October 2005

The reflection phase grating diffusor, RPG, which is based on mathematical number theory sequences suggested by Manfred Schroeder, has brought to architectural acoustics a missing design ingredient—broadbandwidth wide-angle sound diffusion—and opened up many possibilities for acoustical consultants and architects. The RPG diffusor system was first introducted into the recording industry in 1983 and in the short span of 5 years has become a standard ingredient in hundreds of forefront recording/broadcast facilities, worship spaces, corporate conference/teleconference and A/V facilities, performing arts facilities, and home listening rooms. A few installations include The Oak Ridge Boys, Peter Gabriel, Whitney Houston, CBS Records, Telarc, DMP, Polygram, CBS-TV, NBC, CBC, BBC, WQXR, WFMT, Houston Grand Opera, Chrysler Hall, Kentucky Center for the Performing Arts, Word of Faith Ministries, Crenshaw Christian Church, Clear Lake United Methodist Church, Duke University, SUNY, and Lakeland College. This presentation will comprehensively review the theoretical and experimental performance characteristics of the RPG and examine a broad spectrum of applications. Many case study photos and TEF time and frequency response measurements will be presented.
Assessing electroacoustic systems with auditorium acoustics measures
Authors: Bradley J. S. , Halliwell R. E.
Published: October 2005

Modern auditorium acoustics measures have been successfully used to objectively assess changes to auditoria. This paper reports the use of the same techniques to assess the acoustical changes introduced in halls by three types of electroacoustic systems. These were: a film sound system and two different room acoustics modification systems. The RAMSoft room acoustics measurement program was used with broadband exponential pulses as the source signal. The program calculates 12 different quantities in each of six octave bands and permits an evaluation in terms of the needs for both speech and music. While one of the room modification systems was not found to be very effective, the other produced considerable increases in reverberation along with some related effects. Repositioning of the film sound system loudspeakers was shown to improve the performance of the system. The results clearly demonstrate the value of objective evaluations of electroacoustic systems in rooms.
Matched field processing for a moving source
Authors: Ozard John M. , Brooke Gary H. , Tinis Scott
Published: October 2005

The detection or localization of a sound source in a multipath or multimode environment is improved by taking the multipaths into account through matched field processing (MFP). Frequency domain MFP (FDMFP) is very effective for a stationary source. However, there is an implicit averaging over time in FDMFP through the use of the Fourier transform and a further averaging if the covariance matrix is formed. Consequently, if the sound source is moving, the averaging of the time dependent signal will lead to an increase in ambiguity of the source position and a loss in array gain. When the source is moving, the received signal contains information about both source position and source motion. To take full advantage of the information in the time sequence, MFP in the time domain (TDMFP) was implemented. TDMFP is equivalent to obtaining the narrow-band gain and MFP gain in one step through the use of a Fourier transform modified by the propagation. Simulated results confirm the improved localization and array gain of TDMFP compared to FDMFP for a moving source.
A symmetry renormalization method for matched-mode sidelobe reduction
Authors: Smith George B. , Frichter George M.
Published: October 2005

Current matched-field research at NORDA is centered on techniques that attempt to match measured and predicted modal amplitudes for improved detection and localization of acoustic sources in shallow-water waveguides. Ambiguity functions generated by these modal estimators display a sidelobe structure that is symmetric about the true source peak. This symmetry represents additional information about the signal location, which can be used to further enhance detection. Here, a simple correlation algorithm is presented which enhances the signal peak and suppresses sidelobes by renormalizing each point of the ambiguity function in accordance with the symmetry around that point. Since a renormalized ambiguity function retains the range symmetry of the original, the technique can (within limits) be applied iteratively. Computer simulations of a shallow-water Pekeris waveguide are used to demonstrate the effectiveness of renormalization when applied to both narrow-band and frequency-averaged mode matching.
Source localization: Matched field versus matched mode Synthetic and real data performance analysis
Authors: Jesus Sergio M. , Hamson Rachel M.
Published: October 2005

The present study compares the matched field and the matched mode techniques for passively localizing a narrow-band point source in a shallow-water, range-independent environment. The matched mode technique is fully characterized in terms of sidelobe ambiguity performance and robustness against both system and environment parameter variation and mismatch. Comparative results are also shown for real data detection of a cw source immerged at different depths in a 120-m depth channel using a 62-m aperture vertical array. The results of this study indicate that the matched mode method is much less sensitive to the environmental conditions than the matched field method, and in particular, the result is less degraded by the effects of partial water column sampling (short array). Results obtained on real data showed good agreement with the corresponding tests from simulated data. However, a large sidelobe coverage was found for some situations leading to detection losses. Major causes of performance degradation are the uncertainty on the array sensor position due to array motion and correlated noise due, mainly, to surface-generated noise.
Acoustic wave front distortions at long ranges from internal waves
Authors: Duda Timothy F. , Flatté Stanley M.
Published: October 2005

In the absence of small-scale variations on sound speed in the ocean, a pulsed source delivers a series of smooth wave fronts onto a vertical array at long range from the source (multiray propagation). Small-scale variations such as internal waves induce distortions on the wave fronts with transverse correlation function determined by the phase-structure function, which is itself calculable by integrating appropriate functions along the trajectory of an undistorted ray. Expressions for the phase-structure function at small separations have been previously given in the form of arrival-angle spreads due to internal waves, but these expressions are only for vertical receiver separations up to about 100 m. Evaluations of the phase-structure function for separations up to several kilometers are presented, and particular realizations of wave front distortions that result from these internal-wave effects are shown. [Work supported by ONR, Code 1125OA.]
Preliminary results from the 1988 Monterey Bay acoustic tomography experiment
Authors: Miller James H. , Lynch James F. , Chiu Ching-Sang
Published: October 2005

An ocean acoustic tomography experiment was held 12–16 December 1988 near Monterey, California. The objectives of this experiment were to test a tomographic system to analyze the effects of ocean surface waves, internal waves, and complex three-dimensional bathymetry on long-range acoustic propagation. An acoustic source with a center frequency of 224 Hz and source level of 177 dB was placed on a seamount 30 km off Point Sur. Seven modified sonobuoys (with anchor, bottom-mounted hydrophones, large capacity batteries, and large floats) were placed in Monterey Bay to receive the acoustic signals at ranges of 35 to 60 km from the source. The sonobuoy rf signals were received, demodulated, and the acoustic data were recorded on shore. Oceanographic measurements were taken (for comparison with the acoustically derived results) with a surface wave frequency-directional spectra buoy, surface wave frequency spectra buoys, CTD yo-yo's for internal wave spectra, ADCP, and conventional hydrographic survey for sound-speed profiles. Preliminary analyses and comparison of the acoustic and oceanographic measurements will be presented. An outline of what the complete analysis will entail will also be presented. [Work supported by ONR and Naval Postgraduate School Research Council.]
Warren P. Mason (1900–1986), physicist, engineer, inventor, author, teacher
Authors: Thurston Robert N. , Smith David A. , Baran Jane , Cheung Kwok , Johnson John
Published: October 2005

Warren Perry Mason, a Charter Member, Fellow, President, and Gold Medalist of the Acoustical Society of America, consistently applied his understanding of fundamentals to explain physical processes and create practical devices. As a physicist, he led us to a better understanding of fundamental effects in liquids and solids. He made the first measurement of shear elasticity in liquids and helped establish the type of motion that polymer chains can make. In solids, he contributed to quantitative understandings of phonon drag on charge carriers in semiconductors, fatigue of metals, and damping of acoustic waves in metals, insulators, semiconductors, alloys, and rocks. As an engineer and inventor, he led advances in mufflers and noise control, electromechanical filters for carrier-frequency telephony, piezoelectric crystals and ceramics for electromechanical transducers, and semiconductor strain gauges. With about 200 patents, he is the most prolific inventor in the history of Bell Labs. As an author and teacher, he wrote over 200 papers and 4 reference books that teach fundamental concepts, give complete tensorial descriptions of numerous physical interactions in crystals, describe research results, and guide the reader to the related literature. This talk will sample Mason's contributions to physical acoustics and will give an example of a communications device made at Bellcore, in which Mason would surely have been interested, namely an acoustically tuned optical filter [B. L. Heffner et al., Electron. Lett. 24, 1562–1563 (1988)].
Acoustical oceanography: Child of ocean acoustics
Authors: Medwin Herman
Published: October 2005

Sound propagation is the preeminent technique for sensing, identifying, and communicating under the ocean surface. However, the extraordinary spatial and temporal variability of the ocean has frustrated underwater acousticians for decades. The solution, of course, was to learn more about the causes of this variability. The traditional oceanographic instruments do this rather crudely, and probably the best technique is to use the acoustical behavior to characterize the medium. Several years ago, ocean acoustics gave birth to acoustical oceanography. The acoustical oceanographer inverts the problem; he uses the seemingly capricious nature of sound propagation to learn about the ocean. The many successes of this young science range from the identification and counting of physical and biological inhomogeneities, such as microbubbles, zooplankton, and fish, to the remote sensing of distant rainfall and sea surface roughness, deep sea mountains, rocks and sediments, as well as the shape and strength of immense churning ocean eddies, hundreds of kilometers in extent.
Fan noise control—Challenges and opportunities
Authors: Pei Hsien-sheng (Jason)
Published: October 2005

Fans used in the cooling of mechanical and electrical equipment are often the major sources of noise. This paper reviews the past and current noise control activities in the fan industry and related trade and professional organizations. Technology trends are discussed. Research needs and priorities are identified. The impact of fan design (blade, housing, and motor) and fan manufacturing (materials and processes) on fan noise and system acoustical quality are discussed. This paper illustrates with examples that fan noise control is more a challenge for design, application, and manufacturing than an investigation of fundamental aeroacoustic mechanisms. The approach is multi-disciplinary and comprehensive in its treatment of each integration level—from bearings and windings to motor, housing, and end-use equipment, thereby integrating noise control into fan design and manufacturing to achieve total quality for tomorrow's market.
Using piezoelectric film and ultrasound resonance to determine the complete elastic tensor in one measurement
Authors: Maynard J. D.
Published: October 2005

The ultrasonic and elastic properties of materials is conventionally measured using quartz, lithium niobate, etc., transducers and a pulse-echo technique with the transducer driven at resonance. Some problems include transducer ringing, transducer bonding, parallelism of sample faces, beam diffraction, and the necessity of remounting transducers in order to measure all of the elastic constants. Usually these problems can be minimized, but, with samples that are only a fraction of a millimeter in size, conventional ultrasound measurement becomes difficult, if not impossible. However, nearly all of these problems disappear if a resonance technique is used, and all of the elastic constants may be determined with a single measurement. For the broadband response and minimum loading by the transducer required for a resonance measurement in a small sample, polyvinylidene flouride (PVDF) piezoelectric film as thin as 9 μm is ideally suitable. Small active areas and leads are produced with metalization patterns on each side of the PVDF film. For resonance measurements, electrical cross talk across the small sample is processed by frequency modulating the drive and using phase sensitive detection. Samples with dimensions of only a few hundred microns may be measured with large signal-to-noise ratios. [Work supported by the Office of Naval Research and NSF Grant DMR 8701682.]
Resonance of a parallelepiped to determine single-crystal elastic constants up to 1800 K
Authors: Anderson Orson L.
Published: October 2005

The experimental free oscillation spectrum of a rectangular parallelepiped specimen of a single crystal has been obtained on corundum up to 1800 K, using buffer rods to separate the specimen from transducers. By inversion techniques, a theoretical spectrum can be generated, which depends upon the elastic constants, crystal symmetry, and dimensions. Using computational techniques involving the inversion of large matrices, a set of elastic constants is found that produces a theoretical spectrum matching the experimental spectrum out to about 40 modes. Using this technique, the mineral physics laboratory of UCLA has determined the elastic constants Cij and dCij/dT as functions of T with good precision up to temperatures far in excess of the Debye temperature. Some of the results for MgO include: the accurate determination of the Grüneisen parameter to high T; the conclusion that (∂CV/∂T)P = 0 at high temperature (i.e., there is no perceptible anharmonicity in specific heat); and the finding that the anharmonic parameters δS and δT are independent of T at high T.
Warren P. Mason: Some brief encounters, some long memories
Authors: Testardi Louis R.
Published: October 2005

My introduction to the science of acoustics and my first meeting with Warren Mason were nearly simultaneous if not synonomous events at Bell Labs during the mid 1960s. Living in the house of the giant has left some memories and helped shape some personal views on the nature of human achievement and the diversity of greatness. I'll recount some of the ways our paths crossed over the ensuing 20 years and try to explain how his influence went beyond what I could have imagined at that first meeting.
Time-freqyuency spectra for nonstationary acoustic signals—The Wigner distribution, the evolutionary spectrum, the modified moving window spectrum, and their interrelationships
Authors: Moss Jennie , Lee Jong-Sik , Adamopoulos Panos G. , Hammond Joseph K.
Published: October 2005

The time-frequency description of acoustic signals is a common requirement (applications include speech, sonagrams, frequency tracking, etc.), and this applies to transients and nonstationary signals (both random and nonrandom). Three time-frequency spectra that are used are the Wigner-Ville distribution, Priestley's evolutionary spectral density, and Kodera's modification to the moving spectrogram. This paper will present: a summary of the definitions and computation of these three different time-frequency spectra; the modeling of acoustic signals due to propagating sources in a form allowing prediction of time-frequency spectra; development of the interrelationships between the spectra; and theoretical and simulation results from time-frequency spectra. Examples of the application of all three approaches will be illustrated by using frequency-modulated signals. The signals will model sound as perceived by an observer due to a convecting acoustic source (i.e., including Doppler, range, and directivity effects). The source signal will exhibit both tonal and random components.
Measurement of the equivalent point source position for an arbitrary distribution of sources
Authors: Glegg S. A. L. , Yoon J. R.
Published: October 2005

In many applications, knowledge of the equivalent point source position of a source distribution is required for noise control calculations. For instance, the insertion loss of a noise barrier depends on the height of the noise source above the ground, and some estimate of this must be used at the design stage. The equivalent point source position is defined as the location at which a single source would be placed to give the best representation of the acoustic field over the largest possible solid angle. This paper will describe how the equivalent point source position can be measured using parametric methods. Least-squares estimators using a microphone array are shown to be of limited value, especially at low frequencies, and often the best results are obtained using a two-microphone method. [Work sponsored by the Florida Department of Transportation.]
Nearfield high-resolution source localization
Authors: Bertrand Paul
Published: October 2005

An example of passive acoustic array processing consists of localizing the emanating machinery noise sources along a ship shell. When possible, an efficient way to increase the poor accuracy of this localization is to move the array Fresnel zone up to the sources. A linear beamforming with a spherical wave model then provides source bearing, range, and amplitude estimation. However, even for large antennas, the spatial resolution of this nearfield beamforming is limited to a wavelength fraction. To increase the localization accuracy, especially for low frequencies, “high-resolution” methods should be used. This paper presents, in this case, Capon's method, also called adaptive beamforming. It's performances and resolving power are computed and validated through numerical simulations. They extend to the nearfield case the results usually found in a farfield situation. However, much more drastically than with plane waves, experimental results show that this method breaks down if the source model mismatches the wave fronts actually received. To avoid this strong model sensitivity, a robust approach, extending the spectral density matrix source subspace, is proposed. This method yields correct estimation of the amplitude and position of sources too close to be resolved by linear processing. This enhancement of Capon's method does not require an accurate source model. In addition, it has a low computational cost and is not nearfield dependent. [Work supported by DCN.]
Development of noise-suppression saws
Authors: Liu T. I.
Published: October 2005

Conventional wood-cutting saws have even-spaced teeth and therefore generate pure-tone noise in the cutting process. For the same noise level, the pure-tone noise is more annoying than a combination of frequencies. Thus an optimization technique is used to develop noise-suppression saws that have irregular spacing between adjacent teeth. The Fourier series can be used to express the cutting noise of the circular saws: f(θ) = f(θ)¯ +  ∑ n = 1∞ An sin(nθi + φn), where θi is the angle of the saw teeth, and An is the nth Fourier coefficient, which is a function of θ and represents the amplitude of the cutting noise. A conventional circular saw has a very high noise level when n is equal to integer multiples of the number of teeth. This high pure-tone noise level can be reduced and distributed to other n values if the saw teeth are unevenly spaced. In this research, an optimization algorithm is used so that An is close to being a constant. Producibility of the noise-suppression saws has been considered during design. Experiments have been conducted to verify the new design. This work is very helpful in improving the working environment of the woodworking factory.
Noise measurements on machinery near a reflecting plane
Authors: Moreland J. B. , McKendree F. S.
Published: October 2005

Noise test specifications for large equipment often requires that the measurements be made with the equipment out-of-doors resting on the ground (usually a smooth, paved surface), and with the sound level meter microphone positioned between 1 and 2 m above the ground. Constructive and destructive interference between the sound radiated directly from the equipment and the sound reflected from the ground produces a sound field in which the variation of the A-weighted sound level with distance from the unit does not exactly follow the “inverse square law.” The directional characteristics of the equipment under test further complicate the interpretation of the noise measurements, so that using sound levels at one distance, together with the inverse square law to predict the sound levels at other distances, will not be very precise. This paper discusses the problem of machinery noise measurement made above a reflecting plane and how modern spectral analysis techniques can be used to extract information from the noise data.
Discrimination abilities of impaired listeners compared to the range of variation in performance of normal listeners
Authors: Espinoza-Varas Blas , Watson Charles S. , Patterson Suzanne E.
Published: October 2005

Discrimination abilities of sensorineural hearing-impaired listeners have been compared in most studies to average performance of small samples (n≃10) of selected, highly trained, normal listeners. Such comparisons may be biased because: (a) They do not take into account the broad distribution of the performance of normal listeners, that is, the large intersubject differences exhibited by normal listeners on many auditory tasks; (b) unusually poor performers have been exluded from samples of normal listeners; and (c) normal listeners are often college students, impaired listeners are typically more heterogeneous samples. In the present study discrimination performance of 23 impaired listeners was compared to the range of discrimination performance exhibited by large (n > 100), fairly heterogeneous samples of untrained normal listeners. The tasks included discrimination of frequency, duration, intensity, tonal patterns, and nonsense syllables, and were presented at sufficiently high sensation levels (⩾25–30 dB SL). The range of normal performance was defined as the interval between the average psychometric function of the listeners comprising the best 10% of the normal sample and the function for the worst 10%. Performance of normal listeners was distributed over broad ranges in all tasks. Performance of impaired listeners was clearly bracketed by the range of normal performance in all of the nonspeech discrimination tasks. Variability of impaired listeners' performance in the nonspeech tasks was comparable to that of normal listeners, and their psychometric functions were uniformly distributed over the range of those for normal performance.
A different approach to the noise/interference problem of the hearing impaired
Authors: Villchur Edgar
Published: October 2005

Speech has been called an error-resistant code because it contains many redundant cues to its meaning. These cues provide a reserve against the loss of cues masked by interference and allow normal listeners to understand speech in noisy environments. Evidence from experiments with processed speech suggests that the special vulnerability of the hearing impaired to interference is not an intrinsic feature of the impairment but typically derives from a loss of perception of the redundant cues enjoyed by normal listeners. If such is the case, restoring lost speech cues to the impaired listener is likely to be more helpful against interference than increasing signal-to-interference ratios. The lecture will include a demonstration of speech processed to simulate the effects of recruitment and accentuated high-frequency loss, and the vulnerability of this speech to interference. Lost cues will then be restored by compensatory processing (two-channel amplitude compression plus frequency shaping), which increases the intelligibility of the speech in the presence of interference even though it decreases the signal-to-interference ratio.
A comparison of speech discrimination with cochlear implants and tactile aids
Authors: Carney A. E. , Osberger M. J. , Robbins A. , Renshaw J. , Miyamoto R.
Published: October 2005

Speech discrimination was assessed in three groups of hearing-impaired children fitted with three different sensory aids: (1) single-channel House 3M cochlear implants; (2) 22-channel Nucleus cochlear implants; and (3) Tactaid II devices. Nine subtests of a speech discrimination battery were constructed, which focused both on suprasegmental and segmental speech contrasts. A change/no change task was used to assess discrimination. A change trial consisted of four standard stimuli followed by four contrast stimuli; the no-change trial had eight standard stimuli. Hits and false alarms were tallied, and P(C)max was computed for each subject for each subtest. Results indicated that, in general, subjects with Nucleus devices had the highest mean P(C)max across subtests, followed by subjects with 3M House and Tactaid II devices. However, intersubject variability was high for each device. In addition, differences between devices frequently varied on individual subtests. The ability of this test battery to reveal differences between children's speech perception with different sensory aids will be discussed. [Work supported by NIH (NS24875).]
Two-channel FFT techniques for measuring hearing aids with adaptive signal processing
Authors: Bareham John R.
Published: October 2005

Hearing-aid measurements made in test enclosures have traditionally utilized a sinusoidal method such as that of ANSI Standard S3.22-1987. While this method works well with essentially linear aids, it gives misleading results with automatic gain control (AGC) aids. A new method using shaped noise has been under study in Working Group S3-48. This works well with aids having linear, AGC, or adaptive frequency response characteristics, but does not work with aids having speech versus noise discrimination based on detection of the modulation of the received signal. This paper proposes an extension of the shaped noise method that seems to work well with aids having modulation detection. The proposed method uses a combination of pulsed, shaped noise and continuous bandlimited noise as the test signal. The analysis is by two-channel FFT methods. Examples of appropriate and inappropriate measurement methods will be shown for all types of aids mentioned above.
Recognition of Thai consonants by listeners with normal hearing and noise-induced hearing loss
Authors: Maroonroge Sumalai , Nabelek Igor V. , U-phanich Anothai
Published: October 2005

The purpose of this study was to compare recognition of Thai consonants in listeners with normal and noise-induced hearing loss. The subjects consisted of 36 normal-hearing and 120 hearing-impaired listeners whose hearing loss starting at 3, 4, or 6 kHz. All subjects were native Thais. The test materials were meaningful Thai monosyllabic words recorded by a female speaker. The test was presented at 35 dB SL in quiet and in noise (noise level was 10 dB below signal level). There was a significant difference between scores obtained by listeners with normal hearing and those with noise-induced hearing loss. Consonant recognition scores in quiet and in noise were also different. The overall scores from the three groups of noise-induced listeners were similar but the patterns of consonant recognition were distinct for each group and test condition. Recognition scores for stops, fricatives, and affricates were relatively low. Aspirated consonants showed lower scores than unaspirated sounds. Confusion errors occurred across both place and manner of productions.
Simulation of sensorineural hearing loss
Authors: Duchnowski Paul , Zurek Patrick M.
Published: October 2005

Previous work [Zurek and Delhorne, J. Acoust. Soc. Am. 82, 1548–1559 (1987)] found that noise masking, while degrading the speech reception of normal listeners to resemble closely that of the impaired listeners, could only be used to simulate losses less than 70 dB. With greater losses, the noise is unacceptably intense. In the present simulation, a variation of Villchur's [J. Acoust. Soc. Am. 62, 665–674 (1977)] system, the input signal is divided into 14 1/3-oct bands and each band level (determined over a 20-ms window) is compared to the impaired threshold at the center frequency of the band. For input levels below threshold, the signal is attenuated to essentially zero. Signals in an X-dB range (X≈30 dB) immediately above threshold are attenuated according to an expansive mapping of input level into the range between normal threshold and impaired threshold plus X dB. Signals more than X dB above the impaired threshold are passed unattenuated. The system thus simulates the two primary characteristics of sensorineural hearing loss—elevated threshold and recruitment. In pilot studies, this technique was used successfully to simulate consonant reception of impaired listeners described by Zurek and Delhorne. To evaluate the simulation further, new subjects (both impaired and normal) are being tested with both syllables and sentences with variations in overall level, signal-to-noise ratio, and frequency-gain characteristic. Intelligibility scores and SINFA analyses obtained thus far show this processing to be as good as additive noise in simulating sensorineural hearing loss. [Work supported by NIH.]
Some factors affecting recovery from poststimulatory fatigue. I
Authors: Young M. , Lowry L. D.
Published: October 2005

Recovery from poststimulatory fatigue was measured on subjects with normal hearing and patients with unilateral cochlear and retrocochlear hearing impairments. Ears were stimulated by steady tones at various intensity levels for 3 min. Thresholds during the recovery period were obtained at the stimulated frequencies by using both the steady tone only and the tones alternated between interrupted and steady tones. For all subjects, alternate presentation of tones resulted in more rapid recovery to preexposure level than steady tone presentation. These results seem to indicate that the off-time of interrupted tones permits the fast recovery from fatigue for the auditory nervous system. It is presumed that, by varying the on-off ratio, a variety of recovery patterns may be measured.
Final lengthening: A local tempo change
Authors: Edwards Jan , Beckman Mary E. , Fletcher Janet
Published: October 2005

This paper compares the articulatory correlates of lengthening at phrase boundaries to those of slowing down from normal to slow tempo. The velocity, displacement, and duration of opening and closing mandibular gestures of syllables in intonation-phrase-final and nonfinal position were examined for four speakers. The longer durations of phrase-final syllables are due primarily to slower closing gestures, without any accompanying differences in articulator displacement. Similarly, changing from normal to slow tempo is effected by a decrease in peak velocity for both the opening and closing gestures, without any accompanying differences in articulator displacement. Thus final lengthening can be described as a local slowing down before the phrase boundary. Interestingly, two subjects did not differentiate between the peak velocity of final and nonfinal closing gestures at the slow rate, probably because they had reached a lower limit on peak velocity. However, these subjects did maintain a difference in vowel period duration between nonfinal and final syllables at the slow tempo. This result suggests that final lengthening has a directly temporal target at a more abstract level of specification than the articulatory dynamics. [Work supported by the National Science Foundation under grants IRI-8617873 and IRI-8617852.]
Articulatory consonant and vowel timing in rhythmically produced syllables
Authors: Kühnert Barbara , Pompino-Marschall Bernd , Tillmann Hans G.
Published: October 2005

The results of experiments using simultaneous EMG recordings for consonantal (OOS) and vocalic gestures (ABD) in rhythmically produced speech are discussed in connection with an articulatorily based interpretation of the P-center phenomenon. Three subjects produced sequences of five monosyllabic tokens of the form /Cak/ (with /C/ = /p/, /pf/, /pl/, /pr/, /pfl/) in time to a metronome. In contradiction to the hypothesis of Tuller and Fowler [Percept. Psychophys. 27, 277–283 (1980)], the data in this study suggest that the timing of the articulatory vocalic gesture is not a direct correlate of the P center and, furthermore, not independent of the syllable-initial consonants. In addition, although the acoustic syllable onset shows the well-known systematic deviations from isochrony depending on the length of the initial consonant cluster, the articulatory consonantal gesture clearly does not show a parallel systematic shift to earlier onset. These results support the hypothesis that the articulatory timing pattern of monosyllabic utterances seems to be determined on the basis of the phonetic structure of the entire syllable.
Primary and secondary mechanoreceptors
Authors: Bolanowski S. J.
Published: October 2005

A brief review of the structure and function of primary (i.e., lacking a synapse) and secondary (i.e., having a synapse) mechanoreceptors will be presented. The central theme of this review and the session in general will be “What can be learned about mechanotransduction in general, based on the similarities and differences of the two receptor types?”