bluuebunny/crossref_metadata_2025 huggingface dataset:
Experimental and analytic studies of synchronized responses in the chinchilla auditory nerve
Authors: Relkin E. M. , Harris D. M.
Published: October 2005

The current interest in the coding of sounds by the temporal aspects of spike trains in the auditory nerve suggested a parametric study of synchrony of firing to low frequency sounds and the various indices used to quantify synchronization. Index of synchrony and preferred phase were measured as a function of sound pressure and frequency for single units in the auditory nerve of the chinchilla. Results from several previous studies in other laboratories agree well with the current data which are, however, more complete. It was found that synchrony is a function of frequency and level relative to threshold but otherwise independent of fiber characteristic frequency (CF). Preferred phase, however, depended on the relation between frequency and CF as well as level. The maximum value of synchrony obtained for all fibers formed a low-pass function of frequency with a high frequency rolloff of approximately 6 dB/octave. A simple model consisting of a half-wave rectifier followed by a low-pass filter could predict the period histograms remarkably well. Analysis of the indices of synchrony has shown them to be relatively insensitive to the scatter of firing over one half-cycle, explaining, in part, the steepness of synchrony-versus-level functions. [Supported by NINCDS.]
Measurements on phase in primary units in cat
Authors: Allen Jont B.
Published: October 2005

The following procedure was used to measure the neural phase in cat: first tuning curves were made using the procedures of Kiang, Moxon, and Liberman. Then sweeps of given amplitude and frequency were made through the neural super-threshold response region. At each frequency, a histogram to the pure tone was generated phase locked to the tonal stimulus, and based on either two seconds of stimulus or 200 spikes. The phase was then estimated from the Fourier coefficient of the histogram at the stimulus frequency. Below 2 kHz, good correlation was observed in most units as a function of frequency. After removing the phase shift associated with a fixed delay, which was CF dependent, we found a π phase shift below the CF indicating a spectral zero in the transfer function between neural response and ear canal pressure. The bandwidth and frequency of the zero appeared to be quite signal dependent. In other frequency regions the phase was much less signal level dependent.
Effects of dissipation in piezoelectric materials: Reminiscence
Authors: Martin Gordon E.
Published: October 2005

The subject of dissipation in piezoelectric materials is reviewed largely in a personal experience context. Polarized electrostrictive materials were introduced about 1947 and brought new questions regarding dissipative effects. The author derived [J. Acoust. Soc. Am. 36, 413 (1954)] relations between critical frequencies (such as fm and fn) and loss factors (such as Qm and Qe) and showed an early assumption by Professor Cady to be erroneous in the paper reviewed by Professor Van Dyke. The IRE Standard on Piezoelectricity [Proc. IRE 45, 353 (1957)] used such frequency relations. Complex parameters were used to represent losses [J. Acoust. Soc. Am. 35, 510 (1963)]. The elastic, piezoelectric and dielectric dissipation factors were measured in 1962–63 for “33” and “31” ceramic configurations and reported at a Navy conference in 1964. The effects of cement joints between piezoelectric ceramic units was included in analyses [J. Acoust. Soc. Am. 36, paper D2 (1964)]. In addition to a description of results, an overview is given of measurement techniques and problems including nonlinearities.
Blood pressure measurements with foil electret transducer
Authors: West J. E. , von Seggern H. , Nelson J. R. , Kubi R. A.
Published: October 2005

We have designed a Korotkov (K)-sound Foil Electret transducer for use in an automatic blood pressure monitoring system. Systolic and diastolic pressures, were measured at the normal brachial artery location using a cuff technique. Two modifications of the wide band foil electret microphone, normally used for airborne sounds, were necessary to create the K-sound detector. These were (1) to a thicker diaphragm 50 to 75 μm was used in place of the usual 12 to 25 μm diaphragm, (2) multiple ridges spaced 5 mm apart and with heights comparable to the foil thickness were built into the back plate. These changes plus appropriate filtering result in a transducer with uniform sensitivity over the frequency range from 1–60 Hz. There are two major advantages of this system over previously used transducers: (1) Its sensitivity is higher by a factor of at least 50. (2) Its positioning over the artery is not as critical as other designs because its sensitivity is independent of the diaphragm excitation point. These advantages give better K-sound detection capabilities and more accurate blood pressure determinations.
The tense/lax distinction in vowels as a predictor of voicing cues for final, post-vocalic stops
Authors: Parker Frank , Walsh Thomas
Published: October 2005

Stop release [Wang, J. Speech Hearing Res. (1959)] and the presence/absence of a falling F1 transition [Wolf, J. Phonetics (1978)] have been shown to be effective cues to the voicing distinction in final, post-vocalic stops in English. Neither, however, are necessary. Therefore, preliminary speech production data were gathered to determine the interrelationship of these cues. Spectrographic analysis indicates (1) that release is predicted by the tenseness of the preceding vowel: stops following tense vowels are released; those following lax vowels are not. (2) Besides serving as a cue to [± voice] in stops, release functions as a segment cue. Final stops following tense vowels are not predictable by the phonology of the language, whereas those following lax vowels are. That is, stressed, tense vowels can end a word in English, but stressed, lax vowels cannot. Thus, stop release is redundant after a lax vowel, but it has a high utility after a tense vowel in that it signals the presence of a stop. (3) stop release and a falling F1, both of which can signal [+ voice] in final stops, are in complementary distribution. Voiced stops following tense vowels are released and the vowel lacks a falling F1; voiced stops following lax vowels are unreleased and the vowel exhibits a noticeable falling F1. In sum, stop release (as a segment cue) and both release and a falling F1 (as voicing cues) are predictable on the basis of the tense/lax distinction in the vowel. In this case, then, the appearance of a particular acoustic cue is a function of the abstract phonological system of English.
Risk assessment for noise exposure
Authors: Erdreich Linda S. , Erdreich John
Published: October 2005

In this paper we propose that full use is not made of available noise-exposed threshold data. Specifically, data from noise-exposed populations can be analyzed to assess impact of exposure in two ways: (1) to provide estimates of the number of cases of hearing loss attributable to noise in the population or, (2) to provide estimates of increased risk of hearing loss in an exposed individual relative to the unexposed person. Previous studies have neglected prediction of risk to the individual, emphasizing instead the impact on the population. To more fully understand the effect of noise exposure, the approaches must be combined. We will present two statistics which may be calculated from population threshold data: Percent risk is the statistic commonly computed from hearing conservation data. Percent risk is an estimate of the number or of the proportion of cases of hearing loss attributable to noise exposure. Rate ratio is the second statistic we will discuss. The rate ratio estimates the increased risk of hearing loss to an individual as a consequence of exposure. It is adjusted for the background rate and is, therefore, comparable across studies. Each statistic complements the information provided by the other. We will also discuss how potential risk factors for hearing loss may be evaluated and combined to yield a composite risk estimate.
Auditory sensitivity of northern fur seals (Callorhinus ursinus) and a California sea lion (Zalophus californianus) to airborne sound
Authors: Schusterman R. J. , Moore P. W. B.
Published: October 2005

Aerial audiograms were obtained in a specially constructed acoustic chamber from three otariid pinnipeds—two yearling female Callorhinus and one 2-yr-old female Zalophus. A “go/no-go” response procedure was used and threshold estimates were obtained by a tracking method. Behavioral measurements at 1, 2, 4, 8, 16, 24, and 32 kHz resulted in average thresholds for the two Callorhinus, in the order of frequencies given above, of 29, 9, 22, 13, 7, 22, and 34 dB re0.0002 dyn/cm2, and for the one Zalophus they were, respectively, 41, 19, 26, 16, 28, 37, and 61 dB re 0.0002 dyn/cm2. The thresholds for Callorhinus, although inferior in air compared to in water, show good accommodation for hearing airborne sounds. The present results from Zalophus indicate that a previous study [R. J. Schusterman, J. Acoust. Soc. Am. 56, 1248–1251 (1974)] was noise limited in threshold determinations below 24 kHz. Although only a limited number of species have been looked at, the otariid pinnipeds appear to be more sensitive to airborne sounds than do the phocid pinnipeds. [Work supported by ONR.]
Separation of voiced and unvoiced speech, and silence, using energy and periodicity
Authors: Bancroft John
Published: October 2005

The percentage of voiced speech, unvoiced speech, and silence (VUS) are quantifiable acoustic parameters which may be used to evaluate speech production and intelligibility. It is difficult, however, to separate these parameters using computational methods. A new procedure was therefore developed to assist in the analysis of these parameters by comparing energy and periodicity on a contour map. The contour map is a two-dimensional histogram that is formed when energy and periodicity are measured on small sections (20 ms) of continuous speech. The regions of VUS are then distinguishable and may be separated and quantified by allocating areas on the two-dimensional histogram. This method has been undertaken in an attempt to quantify some of the factors which effect speech intelligibility in hearing-impaired young adults. Examples will be presented, comparing the VUS contours of normal hearing and hearing-impaired speakers. [Work supported by U.S. Department of Education.]
The use of tracking rules in automatic formant analysis of speech
Authors: Dupree B. C.
Published: October 2005

The J.S.R.U. computer-simulated automatic formant analyser uses analysis by synthesis to achieve high reliability and accuracy in the matching of synthetic formants to the peaks of pitch synchronous short term spectra of natural speech [P.M. Seeviour, J. N. Holmes and M. W. Judd, IEEE Int. Conf. Acoust., Speech, Signal Processing, Philadelphia, PA, pp. 609–693 (1976)]. However, analysis by synthesis does not of itself necessarily provide consistent labeling of the peaks of the natural spectrum. The specific case of this problem as it appears in the J.S.R.U. formant analyser will be discussed. The task is to choose, for each 10 ms frame of speech, from among a number of attempts to produce a synthetic match to the natural spectrum. The restrictions on the algorithm, particularly the need to avoid excessive smoothing of formant movement, will be mentioned. Simple heuristic rules currently in use will be described and the performance obtainable with such rules will be illustrated by means of spectrograms and audio tape recordings.
Signal excess measurements using a vertical line array
Authors: Ramsdale Dan J. , Howerton Roger A. , Watkins Cecil L.
Published: October 2005

In August 1978, an experiment was conducted in the Northeast Pacific to determine the signal excess received on a vertical line array placed near the bottom when a cw source (54 Hz) was towed at a nominal depth of 91 m radially away from the array. Beamformed data from the array as well as that from the individual hydrophones were analyzed to determine signal excess in an 0.625-Hz processing band. The beamformed results exhibited not only direct arrivals but also arrivals which had one, two and sometimes three interactions with the bottom. No bottom arrivals were detected at ranges less than 5 km, probably due to the inability of bottom refracted energy to return to the water column at these high grazing angles. The signal excess on the array at these short ranges was markedly better than that on the best single hydrophone. At longer ranges (5–25 km) where the beamformed results exhibited up to four distinct arrivals, an interference pattern was present across the array; the signal excess of many of the beamformed arrivals was only marginally better than that of the best single hydrophone. The interference pattern was dependent on the range to the source so that no one hydrophone exhibited the best signal excess over the entire range interval. [Work supported by NORDA and the Naval Electronic System Command.]
Statistical descriptors of time spread sonar targets
Authors: Berkman E. F. , Mark W. D. , Smith P. W.
Published: October 2005

Sonar design and performance analysis requires a description of the distribution of echo energy in time parameterized by frequency and nominal incidence angle. The variability of echoes from sonar targets motivates a statistical description. The ensemble averaged instantaneous power spectrum of the target echo impulse response is the functional target descriptor which matches these requirements. The relationship of the instantaneous power spectrum to other statistical channel descriptors and conventional measures of target strength is discussed. Further interpretations of and measurement procedures for the instantaneous power spectra of the impulse response of random time spread transmission channels are provided. Higher order statistical descriptors and extension to frequency spread targets are commented on.
Calibration of a force platform instrumentation system using random noise and transfer function measurements
Authors: Huang P. H. , Yaniv S. L.
Published: October 2005

A true dynamic calibration technique for determining an impact force and the mechanical impedance of a force platform is described. The force platform system developed at NBS makes use of a rigid cross-stiffened aluminum plate (23 cm × 38 cm × 10 cm) which is supported by four piezoelectric load washers above a steel-concrete foundation. The calibration technique employs random signals for the excitation of an impedance head which serves as a reference force input. Calibration functions over a continuous frequency range for the impedance head and the force platform are obtained from transfer function measurements which offer significant advantages over traditional calibration methods (R. Bouche, 30th Shock & Vibration Symposium, Detroit, Michigan, October 1961; T. Dimoff et al., National Bureau of Standards Report 9526, June 1967). The coherence function of the measured transfer function indicates that the calibration data are limited by the low-frequency performance of the impedance head below 150 Hz.
Minimizing cable noise
Authors: Edelman S. , Roth S.C. , DeReggi A. S.
Published: October 2005

In acquiring a signal from a vibrating body, commercial low-noise cable generates some noise and it has sufficient stiffness and mass to affect the motion of compliant vibrating bodies such as sheet metal or thin rods. Connectors developed for piezoelectric polymer gages are more satisfactory. These consist of a pair of polymer ribbons typically 25 μm thick and 3 mm wide. Each ribbon has a strip of vapor-deposited metal typically about 1 mm wide and a few tens of nanometers thick down one side and a coating of vapor-deposited metal over the other side. The connector consists of two such ribbons cemented to each other so that the vapor-deposited strips come together. The outer coatings form a shield at ground potential. Such a connector is more flexible and lighter than commercial cable. A polymer vapor-deposited connector and a commercial cable of the same unsupported length were each connected to a dummy sensor on a shake table and the noise generated by each was measured when the shaker was driven over a wide range of frequencies. The results are presented.
The effect of structural inhomogeneities and supports on the frequency response of a vibrator
Authors: Skudrzyk E. J.
Published: October 2005

Suspensions and supports generate forces of the same magnitude as the driver. Structural inhomogeneities also have a considerable effect on the frequency response of a vibrator. They introduce additional resonances and antiresonances and dispersion curves like fluctuations in the frequency response curve. Their effect usually depends on the degree of symmetry of the driver point relative to the vibrator and their distance from the driven point. For instance, a mass at the driver reduces drastically the high frequency response of a vibrator, whereas a mass, a few bending wavelengths away from a driver, has only a relatively small effect on the response curve. In many instances, the presence of inhomogeneities, their magnitude, and position can be analyzed from the frequency response of the vibrator. A spherical shell, when driven at its weld (equator), has a response that is about 10 dB lower than when driven at its pole. A number of typical example curves are computed, or measured, and discussed in this paper. [Work supported by ONR 474.]
Predicting the random acoustic response of a complex structure using FEM techniques
Authors: Voorhees C. R.
Published: October 2005

Spacecraft built at RCA Astro undergo a sequence of environmental tests which include an acoustic test. Its purpose is to demonstrate the ability of the spacecraft and its components to withstand the acoustic vibration environment imposed on it during launch and transonic flight. This paper describes work done to predict the vibration response of the structure and of some critical components to the severe random acoustic excitation. Finite element techniques were used to model two different equipment panels and a typical stowed solar panel of the satellite. Using the STARDYNE structural analysis software package, the structural responses of these models were predicted. The random excitation capabilities of the program were used to excite the models in a way such as to best simulate the physical mechanisms for random acoustic excitation. The corresponding structural responses on the flight spacecraft were then measured during a flight simulation acoustic test in RCA's reverberation test facility. The predicted and measured responses are presented for comparison.
Standards Committee S1, Acoustics
Authors: Flynn D.
Published: October 2005

Working Group chairpersons will report on their progress in the preparation of standards, methods of measurement and testing, and terminology in physical acoustics, electroacoustics, sonics, ultrasonics, and underwater sound. Work in progress includes measurement of noise sources, noise dosimeters, intergrating sound-level meters, and revision and extension of sound level meter specifications. Open discussion of committee reports is encouraged.
Convergence zone positions via ray-mode theory
Authors: Beilis A.
Published: October 2005

A numerical algorithm is developed, based on the ray-mode equivalence of Tindle and Guthrie [C. T. Tindle, L. M. Guthrie, J. Sound Vibration 4, 291–295 (1974)], to computationally predict the positions of convergence zones. The sound speed profile is characteristic of the type found in the western Atlantic. One uses the fact that convergence zones are, in ray theory, groups of rays converging in a region of space. In the ray-mode theory, a ray is shown to be equivalent to a group of modes. Thus, convergence zones are made up of specific groups of adjacent modes. The algorithm numerically determines the dominating groups of modes to be used in the ray-mode path equation. We are presently extending this method to include slow range variations of the sound speed profile within the context of the adiabatic approximation of mode theory.
Acoustic fluctuations of long-range resolved ocean multipaths
Authors: Spiesberger John L. , Worcester Peter F.
Published: October 2005

A phase-coded signal with 64-ms resolution was transmitted at 10-min intervals between an acoustic source moored at 2000-m depth and a bottom mounted receiver at ∼3000-m depth and at ∼900-km range. Time and frequency (cw) domain fluctuations of three resolved multipaths (for a one-month record) are compared to theoretical predictions based on internal wave scattering in the fully saturated regime. Fluctuations due to non-internal wave processes (below one cycle per day) are significant.
Modeling sound propagation through a coastal ocean upwelling
Authors: Malay J. T. , Browning D. G. , Scully-Power P. D.
Published: October 2005

Coastal ocean upwellings occur throughout the world's seas; many are regions of high marine productivity. Two factors associated with these features could significantly effect sound propagation: Abrupt temperature changes and a sloping topography. An analysis is made utilizing a range dependent propagation model and environmental data from upwelling areas off the coast of California and Brazil. The effect on sound propagation is compared with adjacent open ocean conditions for various source and receiver depths over a frequency range of 10–5000 Hz.
Equivalent bottom for use with the split-step parabolic equation sound propagation model
Authors: Bucker Homer P.
Published: October 2005

The split-step parabolic equation propagation model is the standard Navy model for cases where the complex sound field (i.e., phase and amplitude) is required and there are significant horizontal variations. This model is restricted when there is strong bottom interaction. For example, the density discontinuity at the water/sediment interface imposes a small step size and the effects of rigidity in the sediment are not easy to take into account. The equivalent bottom method is a systematic procedure for replacing a realistic bottom model by a set of uniform thickness absorbing liquid layers that have no density changes and controlled sound speed changes between the water and sediment and between adjacent sediment layers. The sound speeds in the layers are adjusted so that the equivalent bottom model has the same bottom loss versus grazing angle curve as is measured or calculated for a realistic geoacoustic bottom model. Application of the method will be illustrated using bottom loss data collected in the Northwest Pacific Ocean.
Very long-range underwater sound propagation between Australia and Bermuda: Theory and experiment
Authors: Northrop John , Hartdegen Carl
Published: October 2005

Effective horizontal sound-speed predictions using the RAYWAVE computer program were made for the 19 763-km great circle path between two deep water sites west of Perth, Australia, and Bermuda. Nine sound-speed profiles and 50 bottom depths were used as input to the program. Results, compared with the R/V VEMA and HMAS DIAMANTINA shots in 1960, are that the predicted sound speeds are slightly lower than observed (1480.9 m/s vice 1481.0 m/s for the VEMA shots and 1478.6 m/s vice 1481.1 m/s for the DIAMANTINAs). Predictions for the DIAMANTINA shots also recorded at Fernando de Noronha gave 1475.12 m/s vice measured sound speeds of 1478.6 m/s to 1480.12 m/s. Ray tracing showed that most of the energy received was in the −8.5° and +11.5° rays. A propagation loss model, previously found applicable to RSR paths in high latitudes, was used to compute the propagation loss in the same frequency band. The model, which assumed spherical spreading out to 10 miles and semispherical (15 Log R) spreading beyond plus attenuation gave a predicted loss of 159 dB vice a measured loss of 162 dB.
Verification of the adiabatic invariant approximation in computing effective horizontal sound speeds in a range dependent environment
Authors: Hansen P. G. , Northrop J.
Published: October 2005

Transmission times were computed for explosive underwater sound signals from West of Perth, Australia, to Bermuda, U.K., using the NOSC raytrace model RAYWAVE II, and the Adiabatic Invariant Approximation [R. C. Shockley, J. Acoust. See. Am. 64, 1171–1177 (1978)]. Nine sound-speed profiles and 50 bottom depths obtained from archival data were used as input to the computer models. The results were consistent, and compared favorably with the experimental data obtained by R/V VEMA and HMAS DIAMANTINA in 1960.
Acoustic reflections from a seamount in the northeast Pacific
Authors: Ebbeson G. R. , Turner R. G.
Published: October 2005

Acoustic propagation experiments at 230 Hz were made over Dickins Seamount to determine the extent of reflections from the seamount and the mechanisms responsible. A receiver was suspended at a depth of 329 m, 60 km from the seamount peak. A source was towed toward the seamount at depths of 18 and 184 m. Doppler-shifted reflections with levels 6 to 12 dB below the direct signal level were measured when the shallow source was between 3 and 5 km from the seamount peak. The dominant arrivals were found to undergo two up-slope and two down-slope reflections before propagating to the receiver via conventional second bottom bounce paths. When the source was within 3 kilometers of the seamount, single down-slope reflections which propagated to the receiver via RSR paths were found to be responsible for an enhancement of the direct signal at the receiver. Reflections were not apparent in the measured results for the deep source because the source energy carried by the reflected arrivals was insignificant when compared to that propagating directly to the receiver along the sound channel axis.
The rubber mandrel as an impedance transformer
Authors: Jameson P. W.
Published: October 2005

In designing a fiber optic hydrophone in which a fiber waveguide is wrapped on a rubber mandrel it is useful to describe the behavior of the rubber in terms of compression and shear waves propagation in the rubber. The rubber may then be viewed as a compression wave antenna or a device for matching the high impedance of the fiber to the lower impedance of the incoming compression wave. We describe the response of an infinitely long fiber-covered mandrel to an incident pressure field at arbitrary angle of incidence. At or near endfire there is a null in the radial response of the mandrel, although not in the axial response. The analysis provides several prescriptions for circumventing the effect of the null in the radial response. We also describe modifications to the analysis for the case of a finite-length mandrel, a much more difficult problem.
Signal and noise sensitivities of mandrel-mounted fiber optic hydrophones
Authors: Africk S. A. , Jameson P. W. , Rudd M. J.
Published: October 2005

The acoustic sensitivity of an optical fiber consists of optical path length shifts resulting from pressure-induced density-mediated variations in index of refraction and from actual path length changes (strains) due to acoustically-induced stress. For free fibers these effects tend to cancel, thereby limiting sensitivity. By wrapping a fiber on a mandrel of a material with dynamic modulus smaller than that of the glass fiber, the pressure-induced strain in the fiber becomes governed by the response of the more compliant mandrel and the acoustic sensitivity of the resulting sensor can be significantly greater than that of a simple fiber. Models for acoustical (signal) sensitivity, response to noise environments typical of sonar spaces, and vibration sensitivity—all as functions of frequency—have been developed. These models include effects of the dynamic properties of the constituent materials, the dimensions of the mandrels, the angle of the wrap, etc. Analytical estimates for preliminary sensor designs and certain confirming laboratory measurements indicate that such sensors may perform as well or better than conventional hydrophones.
Microphones and the recording environment
Authors: Eargle John
Published: October 2005

The largest set of variables in the music recording and reproducing chain occurs at the very beginning and is in the hands of the recording producer. His options range from rational microphone techniques, which conformally map the performance environment into the listening environment, to empirical techniques which have grown out of the last twenty years of popular recording. The presentation will describe various approaches commonly used and explore some of the criteria by which the producer's decisions are made.
Computerized ultrasonic tomography
Authors: Greenleaf J. F.
Published: October 2005

Ultrasonic energy interacts very strongly with biological tissue because the inhomogeneties in the intrinsic properties (density, compressibility, attentuation) have spatial dimensions ranging from much less than a wave length to much greater than a wave length at medical ultrasound frequencies. The magnitude of variations in the intrinsic properties varies from less than 10% to greater than 200%. Thus, the interaction of biological tissue with ultrasonic energy must be considered to include both refractive terms and diffractive terms. Wave equations that characterize tissue/ultrasound interactions can be developed which give insight into various imaging modes currently used in diagnostic instruments. Restraints on transducer and scan geometries created by biological barriers to ultrasonic energy, such as gas and bone, provide specific constraints on methods of solving the wave equations for the intrinsic tissue properties. Computerized ultrasonic tomographic methods useful in diagnosis of breast cancer will be described. Future development of methods for diffraction limited quantitative imaging of tissues will require numerical if not analytical inverse scattering solutions to the appropriate wave equations.
Signal processing in geophysical exploration
Authors: Lines Larry , Treitel Sven
Published: October 2005

Seismic signal processing plays a vital role in modern petroleum exploration. The production of good seismograms generally depends on velocity analysis, deconvolution, and migration. Velocity analysis and the correction for normal moveout is an important tool for enhancing primary reflections and suppressing multiples. Another method of suppressing multiples is that of predictive deconvolution. Multiples, such as water bottom bounces, often occur at predictable well defined intervals. Consequently, a prediction-error filter can be designed to remove the repetitive events from the seismogram. Wavelet deconvolution can be applied to resolve events on the seismogram. Various methods have been devised for estimating the source generated wavelet. The application of spiking filters can remove the smearing effects of such wavelets. These filters are particularly important when attempting to resolve thin beds. In areas where the geology deviates from flat layer geometry, reflection events must be placed in their proper position by migration. A brief review of modern migration techniques is given. In summary, the authors design their own prediction filter to speculate on future developments in seismic processing.
Elastic wave scattering from multiple and compound flaws
Authors: Varadan V. V. , Varadan V. K. , Tittmann B.
Published: October 2005

If the nondestructive evaluation (NDE) of flaws in structural materials it is often important to distinguish between a single flaw or cavity, two cavities separated by a finite distance and a compound flaw when two cavities intersect. Using the T-matrix method, numerical results have been obtained for P waves incident on such geometries as a function of incident wave frequency and the scattering geometry. In each case, results have been obtained using several approaches. Approximate solutions including only single scattering effects are compared with exact results as the distance between the scatterers change. The compound flaw can be treated as a single flaw or as two flaws in contact. The above results will be compared with experimental results showing good agreement.
Inversion of acoustic and elastic scattering data with the time domain ramp function
Authors: Cook Bill D. , McKinney Ronald L. , Wilson Shelford E.
Published: October 2005

Scattering data at low frequencies contains sufficient information upon inversion to yield the basic shape of the scatterer. By shaping the incident wave to have the form of a time domain ramp function, this information can be extracted directly in the time domain. An approximate theory states that the backscattered time signal appropriately corrected for range is proportional to the crossectional area of the scatterer as a function of distance along the line of sight from transducer to target. We are testing the approximate theory with calculations of the exact theory for elastic and acoustic spherical scatterers. The results are encouraging in that in most cases, the size of the scatterer can be inferred.
Anomalies in the spectrum of scattered ultrasonic waves from periodic interfaces
Authors: Jungman A. , Quentin Gerard , Adler Laszlo
Published: October 2005

Scattering of broadband ultrasonic waves from water-solid, solid-air and solid-solid interfaces with periodic structures was investigated by ultrasonic spectroscopy. The interface is a periodic grating (of triangular shape) with periodicity ranged from 100 to 1800 μ. Both infinite and finite surfaces are studied. The spectrum of the scattered wave shows sharp minima (anomalies) at certain frequencies which are characteristic of the materials and parameters of the grating. These “missing” frequencies are identified as diffracted orders along the interface with bulk or surface wave velocities. Studying bondings between two surfaces with the aid of these spectral anomalies is suggested. [This work was supported jointly by the French CNRS and by the Center for Advanced NDE operated by the Science Center, Rockwell International, for ARPA and AFML.]
An interrelationship between self-consistent fields and effective scattering parameters
Authors: Jameson P. W.
Published: October 2005

Various authors [V. Twersky, J. Acoust. Soc. Am. 36, 1314 (1964)], [Keller, Proc. Symp. Appl. Math. 16, 145 (1964)], [Ishmirau, Proc. IEEE 65, 1030 (1977)], [A. J. Devaney, submitted to J. Math. Phys.] have derived a dispersion relation for the propagation of the coherent wave through a medium containing scatterers. The expression involves forward scattering amplitudes. We show how for the case of spherical scatterers this expression can be interpreted in terms of the effective dynamic parameters of a unit spherical cell containing the scatterer. These effective parameters are derived by integrating stresses and strains over the surface of the unit sphere.
Frequency resolution and the growth of masking
Authors: Stout Nigel
Published: October 2005

In psychophysical masking experiments frequency resolution is often estimated by comparing thresholds obtained by varying the frequency relation between signal and masker. It is well established that when signal and masker overlap in both time and frequency domains a 10-dB increment in masker level requires approximately a 10dB increase in signal level for equal detectability. However, when the masker is removed from the signal in either time or frequency the rate of masking growth may change [Widin and Viemeister, J. Acoust. Soc. Am. 66, 388–395 (1979)]. This introduces several theoretical problems in the interpretation of frequency resolution experiments, particularly when we wish to compare the results obtained from simultaneous masking with those from forward masking, in which the masker precedes the signal in time. Such problems are illustrated for the techniques of rippled noise resolution and pure tone masking. An attempt is made to distinguish those features of the masking data that are traceable to the rate of masking growth from those traceable to changes in frequency resolution.
Simulation of auditory adaptive filtering capability through a generalized signal processing procedure
Authors: Adelman Roger , Weiler Ernest
Published: October 2005

Discrimination of spoken language under noise conditions (biological or physical) may be enhanced through signal processing techniques (Flanagan, 1972.) The paper discusses computer models which simulate an adaptive filter mechanism having functional similarity to the so-called “secondary cochlear filtering system.” The functional information for the model is derived from microanatomical characteristics of inner and outer hair cells, and the differences in their afferent and efferent innervation. The feedback system derived places the outer hair cells at the control stage of the loop where they serve as regulators of the tuned response of the inner hair cells. While the model may successfully substitute for the secondary filtering system, the point to point correspondence with the auditory periphery is strongly suggestive, but not conclusive. Recordings of interrupted speech signals readily demonstrate the sensitivity of the process to transient portions of the signal. Examples of physiological and psycho-acoustical procedures which are in consonance with the function of this signal processing will be discussed.
Sound reflection by a coated spherical shell
Authors: Gaunaurd G. C. , Kalnins A.
Published: October 2005

We have developed an analytical model for the transmission (interior fields), and principally for the reflection (exterior fields) of acoustic waves from a coated spherical shell covered with a layer of absorbent material. We consider an elastic spherical shell covered with a viscoelastic layer, immersed in an acoustic medium and containing a different fluid in its interior. The exact equations of elasticity and viscoelasticity govern the behavior of the spherical shell and the coating layer, respectively, and plane compressional incident waves were assumed. This model permits the quantitative study of the way in which absorbing layers of increasingly higher effectiveness alter the sonar cross section of a coated structure, and it also allows the assessment of the relative importance of the individual modal contributions (i.e., partial waves) contained in it. It was possible to isolate resonance features in these modal contributions which originate in the shell and in the coating. The reduction in level of the waves reflected by spherical bilaminar shells is more clearly understood by means of this model, which will also permit resonance-interpretation in the light of shell theories.
Bounds on the reflection loss of a single homogeneous layer
Authors: Africk Steven A. , Sachs David A. , Berkman E. F.
Published: October 2005

This paper develops the optimal choice of material properties and resulting optimal value of a particular measure of echo reduction for a single homogeneous lossy wave-bearing layer backed by a completely reflecting surface. This echo is the superposition of (a) a front-surface reflection and (b) sound which has penetrated into the layer and undergone multiple orders of internal propagation and reflection. To assess designs which are insensitive to variations in parameters (e.g., frequency and temperature and back surface impedance) and to manufacturing tolerances, we characterize the reflection from the layer by the Incoherent Reflection Loss (IRL), a decibel measure of the incoherent sum of the two components (a) and (b) above. To minimize front surface reflection, low material loss factor ana a close match of specific acoustic impedance of the layer (pc) to that of the medium (p0C0) is desired. Conversely, to minimize the contribution due to penetration of sound into the layer, high material loss factor and low internal sound speed are desired. Because minimization of each of these two components puts conflicting demands on layer properties, there is an optimal solution for the incoherent sum. Surprisingly, the optimal layer modulus and loss factor and optimal IRL performance depend only on the layer mass, frequency, and specific acoustic impedance (p0C0) of the medium.
Baffle effectiveness against acoustic and flexural noise
Authors: Martin Nathan C.
Published: October 2005

The relative effectiveness of compliant acoustic baffles against acoustic and flexural noise is analyzed using a simple model composed of a flat base plate of infinite extent, a wave-bearing compliant layer, and an outer cover plate in contact with a fluid half-space. The term, flexural noise, refers to the evanescent nearfield associated with free flexural waves in the base plate. It is shown that the effectiveness of a baffle against flexural noise is less than its normal incidence acoustic transmission loss (TL) at low frequencies and greater than TL at higher frequencies. The crossover occurs at approximately one fifth the critical frequency of the base plate. Computational results are presented to illustrate the relative baffle effectiveness against acoustic and flexural noise for several values of base plate thickness and various baffle parameters.
Analysis of isolated word time warping techniques
Authors: Waibel Alex , Yegnanarayana B.
Published: October 2005

This paper presents experimental results aimed at evaluating various design choices that effect performance of dynamic time warping algorithms. Three major aspects that have been suggested in the literature have been investigated: (1) relaxation of the boundary conditions to allow for inaccurate begin/end-time detection, (2) choice of warping algorithm, e.g., Itakura asymmetric, Sakoe and Chiba asymmetric, Sakoe and Chiba symmetric, and (3) choice of an appropriate warping window to restrict computation to the minimum needed for best recognition results. Recognition results were tested on two vocabularies: the digits and a highly confusable subset of the alphabet (e.g., e, b, d, p, t, g, v, c, z). (1) The relaxation of the boundary conditions degraded the peformance of the confusable subset, while only slightly improving the results for the digits. (2) The asymmetric Itakura warping algorithm yielded better recognition results for the confusables, while we obtained slightly better results for the digits using the symmetric Sakoe and Chiba algorithm. (3) The choice of a 100-ms warping window appears to be optimal for both vocabularies used.
Efficient implementation of a frequency-domain technique for frequency scaling of speech signals
Authors: Malah D. , Flanagan J. L.
Published: October 2005

Frequency scaling of speech signals is a useful technique for reducing bandwidth requirements in analog and digital speech transmission systems. A recently developed frequency domain scaling (FDS) technique is based on scaling the phase of the signal's short-time Fourier transform (STFT), evaluated at a discrete set of equally spaced frequencies. For the particular scale factor of 2, a simple sign tracking algorithm is used which avoids explicit phase computation and unwrapping. A computationally efficient implementation of this technique is presented. It is based on performing the STFT analysis, modification and synthesis by means of the FFT algorithm, using specially chosen analysis and synthesis window functions. In addition, efficient decimation and interpolation of the complex bandpass signals, along with the required change in output sampling rate are embedded in the implementation scheme. Simulation results show that this FDS technique is particularly robust to environmental conditions such as background noise, room reverberation, and multiple speakers. Processed examples will be played along with results obtained with the earlier reported pitch-adaptive, time-domain harmonic scaling (TDHS) technique. A hybrid scheme, which combines the two techniques will also be discussed and demonstrated.
Intelligibility evaluation of a 4.8 kb/s residual-excited linear prediction coder
Authors: Nakatsui M. , Mermelstein P. , Stevenson D. C.
Published: October 2005

Residual-Excited Linear Prediction (RELP) coding offers an alternative to Linear Prediction Coding in environments where background noise and transmission errors are likely to degrade the intelligibility of the received speech. The RELP evaluated here encodes the low-frequency component of the linear-prediction residual into two sub-bands. The component of the residual above 800 Hz is regenerated at the receiver, combined with the received low-frequency residual and used to drive the LPC synthesis filter. Sub-band coding of the low-frequency residual at 2 bits per sample yields quality comparable to coding the same residual with a single quantizer at 3 bits per sample. Evaluation with the aid of the Diagnostic Rhyme Test yields 95% intelligibility, significantly higher than most conventional linear-prediction coders. The intelligibility advantage derives from avoiding the voiced/unvoiced and pitch tracking decisions required for LPC and significantly increasing the time-resolution with which the excitation signal is encoded. Speech quality is midway between 3 and 4 bit log PCM coding. In 10-dB SNR background noise, intelligibility of the coded speech drops to 84% from 89% for the uncoded signal. The coder appears particularly suitable for use in noisy environments such as airplane to satellite communication where conventional LPC coders suffer serious degradations. [Work supported in part by the Communications Research Centre, Canada.
The effect of airport noise upon human prenatal growth
Authors: Schell Lawrence M.
Published: October 2005

The effect of mother's exposure to aircraft noise and social factors on birthweight and length of gestation was studied in a sample of births drawn from a circum-airport community where noise levels had been measured. Mother's exposure was based upon known noise levels near her residence(s) in the community. Data from 116 births were used, these being from mothers whose noise exposure history was complete. Means of high, medium, and low exposure groups differed in birthweight by almost 200 g and in gestation length by nearly 10 days in favor of the low exposure group, but the differences were not statistically significant. Using regression analysis to correct for social factors, the partial correlation coefficient of noise exposure and gestation length was significant in girls (r.. = −0.48, p < 0.001), but not in boys (r.. = −0.18, p = 0.11). Partial correlations of noise exposure and birthweight were negative but not statistically significant in either boys or girls. These results agree best with previous studies which suggest that noise may have a depressing influence upon growth. [Work supported by U.S.P.H.S.]
Noise in the nursery: Background stimulation and infant cognitive development
Authors: Wachs Theodore D.
Published: October 2005

Existing human early experience theories assume a linear model relating amount of stimulation to development: i.e., the more stimulation, the more rapid development. Our research suggests that a linear model is incorrect particularly insofar as background noise in the home is concerned. Our predictor variable is a thirty item inventory of the physical environment. Two items in this inventory define a factor characterizing the amount of noise-confusion in the home. Our results consistently indicate a negative and significant relationship between noise confusion in the home and level of cognitive development between 6 and 30 months of age. Our results also indicate that the relationship of noise confusion to development is mediated by individual difference factors such as infant's sex or temperature. The negative relationship between noise-confusion and development may be due to delays in the development of critical mediators such as exploration or language. In terms of intervention, our infrahuman research indicates that the detrimental effects of noise may be minimized by early handling; at the human level the presence of a “stimulus shelter” in the home may serve a similar function.